From phillclarke at gmail.com Fri Apr 1 02:20:43 2011 From: phillclarke at gmail.com (Phill Clarke) Date: Fri, 1 Apr 2011 01:20:43 +0100 Subject: [FFmpeg-user] Cropping different amounts from top and bottom (or left and right sides) Message-ID: I'm struggling to understand how to crop a different amount from the top and bottom of a video. All the documentation and examples seem to show either cropping the same amount from both top and bottom or cropping from just top or bottom. As an example, I have a PAL MPEG2 file at 720x608. The top 32 lines are VITC. If I want to convert to PAL DV, then I need to crop just those top 32 lines. -vf crop=720x576 is not good as it takes a centre cut, which is 16 lines off both the top and bottom. VITC still visible. -vf crop:720x576:0:32 works well and takes 32 lines off the top. However, if I use this same file and want to convert to NTSC DV I need a resolution of 720x480. That's a reduction of 128 lines from the original. I could use: -vf crop: 720x408 it works kind of OK, because 64 lines are taken off both the top and bottom. However, what I really want it to take 80 lines off the top, to account for the 32 lines of VITC, and 48 lines off the bottom. That's what I can't figure out how to do. The examples in the Libavfilter documentation only show taking the same amounts from both top and bottom: http://www.ffmpeg.org/libavfilter.html#SEC15 Many thanks Phill From dashing.meng at gmail.com Fri Apr 1 10:56:19 2011 From: dashing.meng at gmail.com (littlebat) Date: Fri, 1 Apr 2011 16:56:19 +0800 Subject: [FFmpeg-user] question about "Buffering several frames is not supported. Please consume all available frames before adding a new one." Message-ID: <20110401165619.3320f4a7.dashing.meng@gmail.com> Hi, I tried to get an avi snippet from a rmvb movie using ffmpeg, this is my command line: ffmpeg -i yyzs.rmvb -ss 02:40:00 -t 00:10:00 yyzs17.avi But, it spend about 10 minutes on the part before intended snippet and encode zero frame, the last 1 minutes it worked on that snippet. The key message is: "Buffering several frames is not supported. Please consume all available frames before adding a new one." I tried encoding rmvb to avi, rmvb to mp4, mp4 to mp4, mp4 to avi, got the same result. Is it normal? or need add special parameters to avoid wasteing unnecessary time on unintended part? ffmpeg compiled from newest git repository: FFmpeg version git-N-28755-g7ac1674 Below is the detail(break the long time running and omitted some repeat messages): mdx at u1004b2-desktop:/media/1217e/mdx/share/rwshare/yyzs$ ffmpeg -i yyzs.rmvb -ss 02:40:00 -t 00:10:00 yyzs17.ffmpeg.refresh.avi FFmpeg version git-N-28755-g7ac1674, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 1 2011 15:36:56 with gcc 4.4.3 configuration: libavutil 50. 40. 0 / 50. 40. 0 libavcodec 52.116. 0 / 52.116. 0 libavformat 52.104. 0 / 52.104. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 76. 0 / 1. 76. 0 libswscale 0. 13. 0 / 0. 13. 0 [NULL @ 0xa9f3010] Unsupported video codec [rm @ 0xa9ea600] max_analyze_duration reached Input #0, rm, from 'yyzs.rmvb': Metadata: title : author : copyright : comment : Duration: 02:45:12.13, start: 0.000000, bitrate: 608 kb/s Stream #0.0: Video: rv40, yuv420p, 640x360, 531 kb/s, 12.05 fps, 12 tbr, 1k tbn, 12 tbc Stream #0.1: Audio: cook, 44100 Hz, stereo, s16, 64 kb/s Stream #0.2: Data: [0][0][0][0] / 0x0000 [buffer @ 0xaa76500] w:640 h:360 pixfmt:yuv420p Output #0, avi, to 'yyzs17.ffmpeg.refresh.avi': Metadata: INAM : author : ICOP : ICMT : ISFT : Lavf52.104.0 Stream #0.0: Video: mpeg4, yuv420p, 640x360, q=2-31, 200 kb/s, 12 tbn, 12 tbc Stream #0.1: Audio: mp2, 44100 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding [buffer @ 0xaa76500] Buffering several frames is not supported. Please consume all available frames before adding a new one. frame= 0 fps= 0 q=0.0 size= 10kB time=10000000000.00 bitrate= 0.0kbit frame= 0 fps= 0 q=0.0 size= 10kB time=10000000000.00 bitrate= 0.0kbit frame= 0 fps= 0 q=0.0 size= 10kB time=10000000000.00 bitrate= 0.0kbit frame= 0 fps= 0 q=0.0 size= 10kB time=10000000000.00 bitrate= 0.0kbit frame= 0 fps= 0 q=0.0 size= 10kB time=10000000000.00 bitrate= 0.0kbit ...(omitted) frame= 0 fps= 0 q=0.0 size= 10kB time=10000000000.00 bitrate= 0.0kbit Last message repeated 45778 times Buffering several frames is not supported. Please consume all available frames before adding a new one. [buffer @ 0xaa76500] Buffering several frames is not supported. Please consume all available frames before adding a new one. thanks. littlebat From stefano.sabatini-lala at poste.it Fri Apr 1 11:25:14 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Fri, 1 Apr 2011 11:25:14 +0200 Subject: [FFmpeg-user] question about "Buffering several frames is not supported. Please consume all available frames before adding a new one." In-Reply-To: <20110401165619.3320f4a7.dashing.meng@gmail.com> References: <20110401165619.3320f4a7.dashing.meng@gmail.com> Message-ID: <20110401092514.GA13881@geppetto> On date Friday 2011-04-01 16:56:19 +0800, littlebat encoded: > Hi, > > I tried to get an avi snippet from a rmvb movie using ffmpeg, this is my > command line: ffmpeg -i yyzs.rmvb -ss 02:40:00 -t 00:10:00 yyzs17.avi > But, it spend about 10 minutes on the part before intended snippet and > encode zero frame, the last 1 minutes it worked on that snippet. The > key message is: "Buffering several frames is not supported. Please > consume all available frames before adding a new one." > > I tried encoding rmvb to avi, rmvb to mp4, mp4 to mp4, mp4 to avi, got > the same result. > > Is it normal? or need add special parameters to avoid > wasteing unnecessary time on unintended part? > > ffmpeg compiled from newest git repository: FFmpeg version > git-N-28755-g7ac1674 > > Below is the detail(break the long time running and omitted some > repeat messages): > > mdx at u1004b2-desktop:/media/1217e/mdx/share/rwshare/yyzs$ > ffmpeg -i yyzs.rmvb -ss 02:40:00 -t 00:10:00 yyzs17.ffmpeg.refresh.avi [...] > [buffer @ 0xaa76500] Buffering several frames is not supported. Please > consume all available frames before adding a new one. frame= 0 fps= > 0 q=0.0 size= 10kB time=10000000000.00 bitrate= 0.0kbit > frame= 0 fps= 0 q=0.0 size= 10kB time=10000000000.00 > bitrate= 0.0kbit frame= 0 fps= 0 q=0.0 size= 10kB > time=10000000000.00 bitrate= 0.0kbit frame= 0 fps= 0 q=0.0 > size= 10kB time=10000000000.00 bitrate= 0.0kbit frame= 0 > fps= 0 q=0.0 size= 10kB time=10000000000.00 bitrate= > 0.0kbit ...(omitted) frame= 0 fps= 0 q=0.0 size= 10kB > time=10000000000.00 bitrate= 0.0kbit Last message repeated 45778 > times Buffering several frames is not supported. Please consume all > available frames before adding a new one. [buffer @ 0xaa76500] > Buffering several frames is not supported. Please consume all available > frames before adding a new one. > > thanks. Can you share the sample (and possibly file a bug-report)? That would help. -- ffmpeg-user random tip #13 Have you ever *seen* ffmpeg? find ~/src/ffmpeg/ -type f | xargs cat | ffmpeg -s qcif \ -f rawvideo -i - -y ff.mpeg From thilo.borgmann at googlemail.com Fri Apr 1 12:13:58 2011 From: thilo.borgmann at googlemail.com (Thilo Borgmann) Date: Fri, 01 Apr 2011 12:13:58 +0200 Subject: [FFmpeg-user] Cropping different amounts from top and bottom (or left and right sides) In-Reply-To: References: Message-ID: <4D95A566.1000605@googlemail.com> Am 01.04.11 02:20, schrieb Phill Clarke: > I'm struggling to understand how to crop a different amount from the top and > bottom of a video. All the documentation and examples seem to show either > cropping the same amount from both top and bottom or cropping from just top > or bottom. > > As an example, I have a PAL MPEG2 file at 720x608. The top 32 lines are > VITC. If I want to convert to PAL DV, then I need to crop just those top 32 > lines. > > -vf crop=720x576 is not good as it takes a centre cut, which is 16 lines off > both the top and bottom. VITC still visible. > -vf crop:720x576:0:32 works well and takes 32 lines off the top. Thus you already know the answer to your question. "out_w:out_h:x:y" In the docs you've pointed to: "The x and y parameters specify the expressions for the position of the top-left corner of the output (non-cropped) area." > However, if I use this same file and want to convert to NTSC DV I need a > resolution of 720x480. That's a reduction of 128 lines from the original. I > could use: > > -vf crop: 720x408 it works kind of OK, because 64 lines are taken off both > the top and bottom. > > However, what I really want it to take 80 lines off the top, to account for > the 32 lines of VITC, and 48 lines off the bottom. So your parameters for your example should be: 720:408:0:80 for use with PAL DV or 720:408:0:112 for use with the original I hope that helps. -Thilo From dashing.meng at gmail.com Fri Apr 1 16:27:36 2011 From: dashing.meng at gmail.com (littlebat) Date: Fri, 1 Apr 2011 22:27:36 +0800 Subject: [FFmpeg-user] question about "Buffering several frames is not supported. Please consume all available frames before adding a new one." In-Reply-To: <20110401092514.GA13881@geppetto> References: <20110401165619.3320f4a7.dashing.meng@gmail.com> <20110401092514.GA13881@geppetto> Message-ID: <20110401222736.3013894f.dashing.meng@gmail.com> On Fri, 1 Apr 2011 11:25:14 +0200 Stefano Sabatini wrote: > On date Friday 2011-04-01 16:56:19 +0800, littlebat encoded: > > Hi, > > > > I tried to get an avi snippet from a rmvb movie using ffmpeg, this > > is my command line: ffmpeg -i yyzs.rmvb -ss 02:40:00 -t 00:10:00 > > yyzs17.avi But, it spend about 10 minutes on the part before > > intended snippet and encode zero frame, the last 1 minutes it > > worked on that snippet. The key message is: "Buffering several > > frames is not supported. Please consume all available frames before > > adding a new one." > > > > I tried encoding rmvb to avi, rmvb to mp4, mp4 to mp4, mp4 to avi, > > got the same result. > > > > Is it normal? or need add special parameters to avoid > > wasteing unnecessary time on unintended part? > > > > ffmpeg compiled from newest git repository: FFmpeg version > > git-N-28755-g7ac1674 > > > > Below is the detail(break the long time running and omitted some > > repeat messages): > > > > mdx at u1004b2-desktop:/media/1217e/mdx/share/rwshare/yyzs$ > > ffmpeg -i yyzs.rmvb -ss 02:40:00 -t 00:10:00 > > yyzs17.ffmpeg.refresh.avi > [...] > > [buffer @ 0xaa76500] Buffering several frames is not supported. > > Please consume all available frames before adding a new one. > > frame= 0 fps= 0 q=0.0 size= 10kB time=10000000000.00 > > bitrate= 0.0kbit frame= 0 fps= 0 q=0.0 size= 10kB > > time=10000000000.00 bitrate= 0.0kbit frame= 0 fps= 0 q=0.0 > > size= 10kB time=10000000000.00 bitrate= 0.0kbit frame= 0 > > fps= 0 q=0.0 size= 10kB time=10000000000.00 bitrate= > > 0.0kbit frame= 0 fps= 0 q=0.0 size= 10kB > > time=10000000000.00 bitrate= 0.0kbit ...(omitted) frame= 0 fps= > > 0 q=0.0 size= 10kB time=10000000000.00 bitrate= 0.0kbit > > Last message repeated 45778 times Buffering several frames is not > > supported. Please consume all available frames before adding a new > > one. [buffer @ 0xaa76500] Buffering several frames is not > > supported. Please consume all available frames before adding a new > > one. > > > > thanks. > > Can you share the sample (and possibly file a bug-report)? That would > help. > -- > ffmpeg-user random tip #13 > Have you ever *seen* ffmpeg? > find ~/src/ffmpeg/ -type f | xargs cat | ffmpeg -s qcif \ > -f rawvideo -i - -y ff.mpeg Yes, I reported it as a bug at: https://ffmpeg.org/trac/ffmpeg/ticket/32 and filed a sample file too. littlebat From nicolas.george at normalesup.org Fri Apr 1 17:10:58 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Fri, 1 Apr 2011 17:10:58 +0200 Subject: [FFmpeg-user] question about "Buffering several frames is not supported. Please consume all available frames before adding a new one." In-Reply-To: <20110401092514.GA13881@geppetto> References: <20110401165619.3320f4a7.dashing.meng@gmail.com> <20110401092514.GA13881@geppetto> Message-ID: <20110401151058.GA2915@phare.normalesup.org> Le duodi 12 germinal, an CCXIX, Stefano Sabatini a ?crit?: > Can you share the sample (and possibly file a bug-report)? That would > help. It's just the way output-based seeking (-ss after -i rather than before) is implemented: the frames are decoded but not encoded. Obviously, now that there are filters, they are fed to the input of the filter chain but not taken out. Any sample with video should show the problem. Even -f rawvideo -s 42x42 /dev/zero does, as long as there is a -ss after the -i. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From eye.of.the.8eholder at gmail.com Fri Apr 1 22:55:26 2011 From: eye.of.the.8eholder at gmail.com (Khelben Blackstaff) Date: Fri, 1 Apr 2011 23:55:26 +0300 Subject: [FFmpeg-user] Issues using ffmpeg (as mencoder replacement) Message-ID: Hello. Although ffmpeg and mencoder use the same library, i always used mencoder because i found its syntax easier and because ffmpeg didn't have dvd support (without ripping it to disk i mean). I read for some time now that mencoder has slow development and people should use ffmpeg, but since mencoder worked for me i didn't change. Or so i thought. I noticed some days ago that the videos i converted in the last 2 months were wrong, so i finally did the change and compiled ffmpeg. Now, i have some questions regarding ffmpeg usage. In mencoder i had set up a x264 profile that had ovc=x264=1 vf=harddup x264ecnopts=subq=5:frameref=3:weight_b=1:nr=500:keyint=125:bframes=3: 8x8dct:psnr:threads=auto When i wanted to encode a dvd the command i used would be similar to this: mencoder -profile x264 dvd://1 -oac copy -vf-pre crop=720:432:0:72 -endpos 7500 -o outfile.avi -x264encopts bitrate=2500 -vobsubout outfile -slang el -vobsuboutid el After reading the ffmpeg manpage i tried to convert my command for ffmpeg and came to the following command: ffmpeg -i stream.dump -y -t 02:05:00 -vcodec libx264 -vpre fast -b 2500k -vf crop=720:432:0:72 -acodec copy -threads 2 outfile.avi It works fine (except that for some weird reason it is quite slower than with mencoder) but i didn't find a way to output the subtitle stream. I can use -scodec copy when the container supports subtitle streams (like matroska) but i can't make them output to a different file like the -vobsubout command. Is there any way to do this ? After producing the avi with mencoder i used to mux to matroska it together with the vobsub files using mkvmerge/mmg so -vobsubout isn't so important because with ffmpeg i can output .mkv in one step but it would be nice if it can be done. My primary problem is that i can't get ffmpeg to properly recognise the subs on the ripped file. mplayer dvd://1 gives me: audio stream: 0 format: ac3 (5.1) language: en aid: 128. audio stream: 1 format: dts (5.1) language: en aid: 137. audio stream: 2 format: ac3 (stereo) language: en aid: 130. number of audio channels on disk: 3. subtitle ( sid ): 0 language: en subtitle ( sid ): 1 language: nl subtitle ( sid ): 17 language: en subtitle ( sid ): 18 language: nl number of subtitles on disk: 4 sid 0-1 are the normal subtitles while sid 17-18 are the director's commentary. mplayer stream.dump correctly shows 4 subtitle streams but it doesn't know what language they are ofcourse. When i run ffmpeg or ffprobe i get: [mpeg @ 0x940050] max_analyze_duration reached Input #0, mpeg, from 'stream.dump': Duration: 02:28:34.30, start: 0.287267, bitrate: 3990 kb/s Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 64:45 DAR 16:9], 9800 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.2[0x89]: Audio: dca (DTS), 48000 Hz, 5.1, s16, 768 kb/s Stream #0.3[0x82]: Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s Stream #0.4[0x31]: Subtitle: dvdsub Stream #0.5[0x32]: Subtitle: dvdsub As you see it finds only 2 subtitle streams. I would not mind if they were the correct but they are both the director's commentary (sid 17-18 on mplayer). This happens only on 2 files though. It recognises other dvds fine. Is there any way to force ffmpeg to see/use 0x20, 0x21 Stream ID ? My ffmpeg version is the latest as of now. FFmpeg version git-N-28764-g7b9750e, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 1 2011 21:48:17 with gcc 4.5.2 configuration: --prefix=/usr --libdir=/usr/lib64 --mandir=/usr/man --enable-shared --enable-gpl --disable-ffserver --enable-postproc --enable-libx264 --enable-pic --enable-libmp3lame If i forgot to mention information that is vital, i will be glad to add it. Thank you for your time. From leo.izen at gmail.com Sat Apr 2 00:53:19 2011 From: leo.izen at gmail.com (Leo Izen) Date: Fri, 1 Apr 2011 18:53:19 -0400 Subject: [FFmpeg-user] Annoying Linker error on vdpau even with --disable-vdpau Message-ID: I'm getting this annoying linker error when compiling ffmpeg for windows: LD ffmpeg_g.exe /home/Leo/ffmpeg-w64/ffmpeg/ libavcodec/libavcodec.a(allcodecs.o):allcodecs.c:(.text+0x16): undefined reference to `ff_mpeg1_vdpau_hwaccel' /home/Leo/ffmpeg-w64/ffmpeg/libavcodec/libavcodec.a(allcodecs.o):allcodecs.c:(.text+0x2c): undefined reference to `ff_mpeg2_vdpau_hwaccel' /home/Leo/ffmpeg-w64/ffmpeg/libavcodec/libavcodec.a(mpeg12.o):mpeg12.c:(.text+0x5811): undefined reference to `ff_vdpau_mpeg_picture_complete' collect2: ld returned 1 exit status make: *** [ffmpeg_g.exe] Error 1 I configured ffmpeg with this command: ./configure --enable-static --disable-shared --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libdirac --enable-libfaac --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-zlib --arch=x86_64 --cpu=x86_64 --enable-pic --enable-sram --prefix=/usr/local/x86_64-w64-mingw32 --cross-prefix=/usr/x86_64-w64-mingw32/bin/ --target-os=mingw32 --pkg-config=pkg-config --extra-libs=-lorc-0.4\ -lx264 --enable-stripping --disable-debug --disable-dxva2 --disable-vdpau --enable-w32threads Is there any way to fix this linker error? It didn't happen with a build I made two days ago. From lou at fakeoutdoorsman.com Sat Apr 2 01:47:23 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Fri, 1 Apr 2011 15:47:23 -0800 Subject: [FFmpeg-user] Annoying Linker error on vdpau even with --disable-vdpau In-Reply-To: References: Message-ID: <20110401154723.2f1ae23a@lrcd.com> On Fri, 1 Apr 2011 18:53:19 -0400 Leo Izen wrote: > I'm getting this annoying linker error when compiling ffmpeg for > windows: > > LD ffmpeg_g.exe > /home/Leo/ffmpeg-w64/ffmpeg/ > libavcodec/libavcodec.a(allcodecs.o):allcodecs.c:(.text+0x16): > undefined reference to `ff_mpeg1_vdpau_hwaccel' > /home/Leo/ffmpeg-w64/ffmpeg/libavcodec/libavcodec.a(allcodecs.o):allcodecs.c:(.text+0x2c): > undefined reference to `ff_mpeg2_vdpau_hwaccel' > /home/Leo/ffmpeg-w64/ffmpeg/libavcodec/libavcodec.a(mpeg12.o):mpeg12.c:(.text+0x5811): > undefined reference to `ff_vdpau_mpeg_picture_complete' > collect2: ld returned 1 exit status > make: *** [ffmpeg_g.exe] Error 1 > > I configured ffmpeg with this command: > ./configure --enable-static --disable-shared --enable-gpl > --enable-version3 --enable-nonfree --enable-postproc > --enable-runtime-cpudetect --enable-avisynth --enable-bzlib > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libdirac --enable-libfaac --enable-libmp3lame > --enable-libopenjpeg --enable-libschroedinger --enable-libspeex > --enable-libtheora --enable-libvorbis --enable-libvpx > --enable-libx264 --enable-libxvid --enable-zlib --arch=x86_64 > --cpu=x86_64 --enable-pic --enable-sram > --prefix=/usr/local/x86_64-w64-mingw32 > --cross-prefix=/usr/x86_64-w64-mingw32/bin/ --target-os=mingw32 > --pkg-config=pkg-config --extra-libs=-lorc-0.4\ -lx264 > --enable-stripping --disable-debug --disable-dxva2 --disable-vdpau > --enable-w32threads > > Is there any way to fix this linker error? It didn't happen with a > build I made two days ago. Git pull and try again. Fixed in: From leo.izen at gmail.com Sat Apr 2 02:05:49 2011 From: leo.izen at gmail.com (Leo Izen) Date: Fri, 1 Apr 2011 20:05:49 -0400 Subject: [FFmpeg-user] Annoying Linker error on vdpau even with --disable-vdpau In-Reply-To: <20110401154723.2f1ae23a@lrcd.com> References: <20110401154723.2f1ae23a@lrcd.com> Message-ID: On Fri, Apr 1, 2011 at 7:47 PM, Lou wrote: > > > Git pull and try again. Fixed in: > < > http://git.videolan.org/?p=ffmpeg.git;a=commit;h=ecf94986c8ba7c9fb4f674ee93e3be5f73d8ccdd > > > Thanx it worked. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From brent at thegoodnite.com Sat Apr 2 02:46:32 2011 From: brent at thegoodnite.com (Brent Murata) Date: Fri, 1 Apr 2011 17:46:32 -0700 Subject: [FFmpeg-user] FFmpeg configure script, iOS 4.3, XCode 3.2.6 Message-ID: Hello, I'm having problems with the FFmpeg configure scripts, iOS 4.3, and XCode 3.2.6. I am a novice when it comes to gcc, so please forgive my questions. FYI, it seems that iOS 4.1, XCode 3.2.6 works ok. I've tried the built in configure scripts without success. At the command prompt I enter: ./configure --cc=/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/gcc --as=' gas-preprocessor.pl/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/gcc' --sysroot=/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk --enable-cross-compile --target-os=darwin --cpu=arm1176jzf-s --arch=arm --extra-cflags='-arch armv6' --extra-ldflags='-arch armv6' I have the FFmpeg git snapshot from 3/31/2011. I've placed the gas-preprocessor.pl file in /usr/local/bin/ The error message is: ld: file not found: /usr/lib/system/libcache.dylib for architecture armv6 Config.log is attached. I've read online that when using version 3.2.6 and version 4.0 of XCode we should be using -isysroot instead of --sysroot. I tried this without success. Any advice / help would be much appreciated. Thanks, Brent Murata -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: application/octet-stream Size: 71639 bytes Desc: not available URL: From dashing.meng at gmail.com Sat Apr 2 03:44:33 2011 From: dashing.meng at gmail.com (littlebat) Date: Sat, 2 Apr 2011 09:44:33 +0800 Subject: [FFmpeg-user] question about "Buffering several frames is not supported. Please consume all available frames before adding a new one." In-Reply-To: <20110401151058.GA2915@phare.normalesup.org> References: <20110401165619.3320f4a7.dashing.meng@gmail.com> <20110401092514.GA13881@geppetto> <20110401151058.GA2915@phare.normalesup.org> Message-ID: <20110402094433.5e66e1db.dashing.meng@gmail.com> On Fri, 1 Apr 2011 17:10:58 +0200 Nicolas George wrote: > Le duodi 12 germinal, an CCXIX, Stefano Sabatini a ?crit?: > > Can you share the sample (and possibly file a bug-report)? That > > would help. > > It's just the way output-based seeking (-ss after -i rather than > before) is implemented: the frames are decoded but not encoded. > Obviously, now that there are filters, they are fed to the input of > the filter chain but not taken out. > > Any sample with video should show the problem. Even -f rawvideo -s > 42x42 /dev/zero does, as long as there is a -ss after the -i. > > Regards, > > -- > Nicolas George Put -i after -ss solved the problem. thanks. littlebat From seandarcy2 at gmail.com Sat Apr 2 03:47:16 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Fri, 01 Apr 2011 21:47:16 -0400 Subject: [FFmpeg-user] fps in March 13 git: -r 29.97 = 2997 tbn ??? Message-ID: Converting to an android playable file, I can't seem to set the fps. The input is 29.97. I tried setting -r to 29.97: + ffmpeg -i ffv1.avi -an -pass 1 -r 29.97 -vcodec libx264 -level 30 -vpre slow_firstpass -bufsize 10000000 -maxrate 10000000 -b 10000000 -threads 0 OUTPUT1.m4v FFmpeg version git-0e6a3ee, Copyright (c) 2000-2011 the FFmpeg developers built on Mar 13 2011 12:25:45 with gcc 4.5.1 20100924 (Red Hat 4.5.1-4) ............ Input #0, avi, from 'ffv1.avi': Metadata: encoder : Lavf52.102.0 Duration: 01:02:39.18, start: 0.000000, bitrate: 39880 kb/s Stream #0.0: Video: ffv1, yuv411p, 720x480, PAR 8:9 DAR 4:3, 29.97 tbr, 29.97 tbn, 29.97 tbc Stream #0.1: Audio: pcm_s16le, 32000 Hz, 2 channels, s16, 1024 kb/s ............ Output #0, ipod, to 'OUTPUT1.m4v': Metadata: encoder : Lavf52.102.0 Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 8:9 DAR 4:3], q=0-69, pass 1, 10000 kb/s, 2997 tbn, 29.97 tbc ........ Then I tried setting -r to 30000/1001: ffmpeg -i ffv1.avi -an -pass 1 -r 30000/1001 -vcodec libx264 -level 30 -vpre slow_firstpass -bufsize 10000000 -maxrate 10000000 -b 10000000 -threads 0 OUTPUT1.m4v ................... Output #0, ipod, to 'OUTPUT1.m4v': Metadata: encoder : Lavf52.102.0 Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 8:9 DAR 4:3], q=0-69, pass 1, 10000 kb/s, 30k tbn, 29.97 tbc Which also happens if I don't set the fps: Output #0, ipod, to 'OUTPUT1.m4v': Metadata: encoder : Lavf52.102.0 Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 8:9 DAR 4:3], q=0-69, pass 1, 10000 kb/s, 30k tbn, 29.97 tbc tbn is the framerate for the AVStream of the container, as I gather from google. It seems that -r affects tbn of the output. But really oddly. How do I set fps of the output? Thanks. sean From leo.izen at gmail.com Sat Apr 2 04:11:25 2011 From: leo.izen at gmail.com (Leo Izen) Date: Fri, 1 Apr 2011 22:11:25 -0400 Subject: [FFmpeg-user] question about "Buffering several frames is not supported. Please consume all available frames before adding a new one." In-Reply-To: <20110402094433.5e66e1db.dashing.meng@gmail.com> References: <20110401165619.3320f4a7.dashing.meng@gmail.com> <20110401092514.GA13881@geppetto> <20110401151058.GA2915@phare.normalesup.org> <20110402094433.5e66e1db.dashing.meng@gmail.com> Message-ID: On Fri, Apr 1, 2011 at 9:44 PM, littlebat wrote: > On Fri, 1 Apr 2011 17:10:58 +0200 > Nicolas George wrote: > > > Le duodi 12 germinal, an CCXIX, Stefano Sabatini a ?crit : > > > Can you share the sample (and possibly file a bug-report)? That > > > would help. > > > > It's just the way output-based seeking (-ss after -i rather than > > before) is implemented: the frames are decoded but not encoded. > > Obviously, now that there are filters, they are fed to the input of > > the filter chain but not taken out. > > > > Any sample with video should show the problem. Even -f rawvideo -s > > 42x42 /dev/zero does, as long as there is a -ss after the -i. > > > > Regards, > > > > -- > > Nicolas George > > Put -i after -ss solved the problem. thanks. > I still recommend putting -ss after -i. If you put it before, the audio and video will almost always get misaligned. The error isn't really an error, so it should be fine. > > littlebat > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From dashing.meng at gmail.com Sat Apr 2 09:48:16 2011 From: dashing.meng at gmail.com (littlebat) Date: Sat, 2 Apr 2011 15:48:16 +0800 Subject: [FFmpeg-user] question about "Buffering several frames is not supported. Please consume all available frames before adding a new one." In-Reply-To: References: <20110401165619.3320f4a7.dashing.meng@gmail.com> <20110401092514.GA13881@geppetto> <20110401151058.GA2915@phare.normalesup.org> <20110402094433.5e66e1db.dashing.meng@gmail.com> Message-ID: <20110402154816.2f3cff72.dashing.meng@gmail.com> On Fri, 1 Apr 2011 22:11:25 -0400 Leo Izen wrote: > On Fri, Apr 1, 2011 at 9:44 PM, littlebat > wrote: > > > On Fri, 1 Apr 2011 17:10:58 +0200 > > Nicolas George wrote: > > > > > Le duodi 12 germinal, an CCXIX, Stefano Sabatini a ?crit : > > > > Can you share the sample (and possibly file a bug-report)? That > > > > would help. > > > > > > It's just the way output-based seeking (-ss after -i rather than > > > before) is implemented: the frames are decoded but not encoded. > > > Obviously, now that there are filters, they are fed to the input > > > of the filter chain but not taken out. > > > > > > Any sample with video should show the problem. Even -f rawvideo -s > > > 42x42 /dev/zero does, as long as there is a -ss after the -i. > > > > > > Regards, > > > > > > -- > > > Nicolas George > > > > Put -i after -ss solved the problem. thanks. > > > > I still recommend putting -ss after -i. If you put it before, the > audio and video will almost always get misaligned. The error isn't > really an error, so it should be fine. Add "-async 1" will avoid syncing problem of audio and video in my case, for example: ffmpeg -ss 02:40:00 -t 00:10:00 -i yyzs.rmvb -async 1 yyzs17.mp4 Although I don't understand "-async" parameter clearly. On my Core(TM)2 Duo CPU E7400 machine, if I put -i before -ss I will spend about 11 minutes to finish the job above while only 1 minutes by putting -ss before -i. littlebat From nicolas.george at normalesup.org Sat Apr 2 12:37:32 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Sat, 2 Apr 2011 12:37:32 +0200 Subject: [FFmpeg-user] question about "Buffering several frames is not supported. Please consume all available frames before adding a new one." In-Reply-To: References: <20110401165619.3320f4a7.dashing.meng@gmail.com> <20110401092514.GA13881@geppetto> <20110401151058.GA2915@phare.normalesup.org> <20110402094433.5e66e1db.dashing.meng@gmail.com> Message-ID: <20110402103731.GA24569@phare.normalesup.org> Le duodi 12 germinal, an CCXIX, Leo Izen a ?crit?: > I still recommend putting -ss after -i. It is wrong to recommend one rather than the other: they just do different tasks, and each shall be used whenever it best suits the needs of the current problem. Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From mjs973 at optonline.net Sat Apr 2 13:50:58 2011 From: mjs973 at optonline.net (Mike Scheutzow) Date: Sat, 02 Apr 2011 07:50:58 -0400 Subject: [FFmpeg-user] fps in March 13 git: -r 29.97 = 2997 tbn ??? In-Reply-To: References: Message-ID: <4D970DA2.1070302@optonline.net> sean darcy wrote: > Converting to an android playable file, I can't seem to set the fps. > The input is 29.97. I tried setting -r to 29.97: > > ... > > Output #0, ipod, to 'OUTPUT1.m4v': > Metadata: > encoder : Lavf52.102.0 > Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 8:9 DAR 4:3], > q=0-69, pass 1, 10000 kb/s, 30k tbn, 29.97 tbc > > tbn is the framerate for the AVStream of the container, as I gather > from google. It seems that -r affects tbn of the output. But really > oddly. > > How do I set fps of the output? > You misunderstand the meaning of tbn. tbn is not the "framerate" of the container, it is the timebase. A timebase is like a clock, and indicates ticks per second. Generally, higher values are better. Both values of tbn you saw should work fine for your file. You got different values because 29.97 is not *exactly* equal to 30000/1001. Also, be careful believing the tbn / tbc "Output #0" stats - they have been broken for months. Use 'ffmpeg -i infile' to get accurate values. Mike Scheutzow From h.reindl at thelounge.net Sat Apr 2 14:06:40 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sat, 02 Apr 2011 14:06:40 +0200 Subject: [FFmpeg-user] --enable-filter=drawtext ignored? Message-ID: <4D971150.4020102@thelounge.net> "--enable-filter=drawtext" seems to be ignored or do i understand something wrong? Filters: anull Pass the source unchanged to the output. anullsrc Null audio source, never return audio frames. anullsink Do absolutely nothing with the input audio. blackframe Detect frames that are (almost) black. copy Copy the input video unchanged to the output. crop Crop the input video to width:height:x:y. cropdetect Auto-detect crop size. drawbox Draw a colored box on the input video. fade Fade in/out input video fifo Buffer input images and send them when they are requested. format Convert the input video to one of the specified pixel formats. gradfun Debands video quickly using gradients. hflip Horizontally flip the input video. hqdn3d Apply a High Quality 3D Denoiser. mp libmpcodecs wrapper. noformat Force libavfilter not to use any of the specified pixel formats for the input to the next filter. null Pass the source unchanged to the output. ocv Apply transform using libopencv. overlay Overlay a video source on top of the input. pad Pad input image to width:height[:x:y[:color]] (default x and y: 0, default color: black). pixdesctest Test pixel format definitions. scale Scale the input video to width:height size and/or convert the image format. setdar Set the frame display aspect ratio. setpts Set PTS for the output video frame. setsar Set the pixel sample aspect ratio. settb Set timebase for the output link. slicify Pass the images of input video on to next video filter as multiple slices. transpose Transpose input video. unsharp Sharpen or blur the input video. vflip Flip the input video vertically. yadif Deinterlace the input image buffer Buffer video frames, and make them accessible to the filterchain. color Provide an uniformly colored input, syntax is: [color[:size[:rate]]] movie Read from a movie source. nullsrc Null video source, never return images. nullsink Do absolutely nothing with the input video ______________________ FFmpeg version git-N-28755-g7ac1674, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 2 2011 13:58:11 with gcc 4.5.1 20100924 (Red Hat 4.5.1-4) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O6 -march=core2 -mtune=core2 -mmmx -msse2 -msse3 -msse4.1 -fopenmp -mfpmath=sse -fno-delete-null-pointer-checks -fstack-protector -pipe -D_FORTIFY_SOURCE=2' --extra-version=thelounge.net --enable-nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libvpx --enable-librtmp --enable-vdpau --enable-x11grab --enable-avfilter --enable-postproc --enable-swscale --enable-bzlib --enable-zlib --enable-pthreads --enable-fastdiv --enable-pic --enable-libopencv --enable-bsf=aac_adtstoasc --enable-bsf=chomp --enable-bsf=dump_extradata --enable-bsf=h264_mp4toannexb --enable-bsf=imx_dump_header --enable-bsf=mjpeg2jpeg --enable-bsf=mjpegadump --enable-bsf=mov2textsub --enable-bsf=mp3comp --enable-bsf=mp3decomp --enable-bsf=noise --enable-bsf=remove_extra --enable-bsf=text2movsub --enable-filter=anull --enable-filter=anullsink --enable-filter=anullsrc --enable-filter=blackframe --enable-filter=buffer --enable-filter=color --enable-filter=copy --enable-filter=crop --enable-filter=cropdetect --enable-filter=drawbox --enable-filter=drawtext --enable-filter=fade --enable-filter=fifo --enable-filter=format --enable-filter=gradfun --enable-filter=hflip --enable-filter=hqdn3d --enable-filter=movie --enable-filter=mp --enable-filter=noformat --enable-filter=null --enable-filter=nullsink --enable-filter=nullsrc --enable-filter=ocv --enable-filter=overlay --enable-filter=pad --enable-filter=pixdesctest --enable-filter=scale --enable-filter=setdar --enable-filter=setpts --enable-filter=setsar --enable-filter=settb --enable-filter=slicify --enable-filter=transpose --enable-filter=unsharp --enable-filter=vflip --enable-filter=yadif --enable-shared --disable-static --disable-debug --disable-stripping --shlibdir=/usr/lib64 --cpu=core2 --enable-runtime-cpudetect libavutil 50. 40. 0 / 50. 40. 0 libavcodec 52.116. 0 / 52.116. 0 libavformat 52.104. 0 / 52.104. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 76. 0 / 1. 76. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 ______________________ 2 hours ago Stefano Sabatini drawtext: extend/clarify examples in the filter description master commit | commitdiff | tree | snapshot 2 hours ago Stefano Sabatini drawtext: fix wrong reference in boxcolor option descri... commit | commitdiff | tree | snapshot 2 hours ago Stefano Sabatini drawtext: add documentation for the shadow parameters commit | commitdiff | tree | snapshot 2 hours ago Stefano Sabatini drawtext: allow negative shadow offset values -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From stefano.sabatini-lala at poste.it Sat Apr 2 14:34:53 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sat, 2 Apr 2011 14:34:53 +0200 Subject: [FFmpeg-user] --enable-filter=drawtext ignored? In-Reply-To: <4D971150.4020102@thelounge.net> References: <4D971150.4020102@thelounge.net> Message-ID: <20110402123453.GA29604@geppetto> On date Saturday 2011-04-02 14:06:40 +0200, Reindl Harald encoded: > "--enable-filter=drawtext" seems to be ignored or > do i understand something wrong? This is how the configure system works, all filters are enabled by default, dependency checks are done and since libfreetype is not enabled drawtext is disabled (with no warning). I'm looking at how to improve this behavior. From h.reindl at thelounge.net Sat Apr 2 14:47:40 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sat, 02 Apr 2011 14:47:40 +0200 Subject: [FFmpeg-user] --enable-filter=drawtext ignored? In-Reply-To: <20110402123453.GA29604@geppetto> References: <4D971150.4020102@thelounge.net> <20110402123453.GA29604@geppetto> Message-ID: <4D971AEC.4000004@thelounge.net> thank you! i thought i have everything enabled what did not running away from me, but maybe this is a newer option rebuild is running............. Am 02.04.2011 14:34, schrieb Stefano Sabatini: > On date Saturday 2011-04-02 14:06:40 +0200, Reindl Harald encoded: >> "--enable-filter=drawtext" seems to be ignored or >> do i understand something wrong? > > This is how the configure system works, all filters are enabled by > default, dependency checks are done and since libfreetype is not > enabled drawtext is disabled (with no warning). > > I'm looking at how to improve this behavior. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From leo.izen at gmail.com Sat Apr 2 15:42:27 2011 From: leo.izen at gmail.com (Leo Izen) Date: Sat, 2 Apr 2011 09:42:27 -0400 Subject: [FFmpeg-user] FFmpeg configure script, iOS 4.3, XCode 3.2.6 In-Reply-To: References: Message-ID: On Fri, Apr 1, 2011 at 8:46 PM, Brent Murata wrote: > Hello, > > I'm having problems with the FFmpeg configure scripts, iOS 4.3, and XCode > 3.2.6. I am a novice when it comes to gcc, so please forgive my questions. > FYI, it seems that iOS 4.1, XCode 3.2.6 works ok. > > I've tried the built in configure scripts without success. At the command > prompt I enter: > > ./configure > --cc=/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/gcc --as=' > > gas-preprocessor.pl/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/gcc > ' > > --sysroot=/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk > --enable-cross-compile --target-os=darwin --cpu=arm1176jzf-s --arch=arm > --extra-cflags='-arch armv6' --extra-ldflags='-arch armv6' > > I have the FFmpeg git snapshot from 3/31/2011. > > I've placed the gas-preprocessor.pl file in /usr/local/bin/ > > The error message is: ld: file not found: /usr/lib/system/libcache.dylib > for architecture armv6 > > Config.log is attached. > > I've read online that when using version 3.2.6 and version 4.0 of XCode we > should be using -isysroot instead of --sysroot. I tried this without > success. > 1. Try adding --extra-cflags='--sysroot ...' and --extra-ldflags='-isysroot ...' According to this page: http://www.doitscared.com/?p=521 that's what you should do. 2. Will this new FFmpeg be on Cydia, or is it a private build for an App Store App? > > Any advice / help would be much appreciated. > > Thanks, > Brent Murata > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From leo.izen at gmail.com Sat Apr 2 15:44:35 2011 From: leo.izen at gmail.com (Leo Izen) Date: Sat, 2 Apr 2011 09:44:35 -0400 Subject: [FFmpeg-user] fps in March 13 git: -r 29.97 = 2997 tbn ??? In-Reply-To: <4D970DA2.1070302@optonline.net> References: <4D970DA2.1070302@optonline.net> Message-ID: On Sat, Apr 2, 2011 at 7:50 AM, Mike Scheutzow wrote: > > > > You misunderstand the meaning of tbn. tbn is not the "framerate" of the > container, it is the timebase. > > A timebase is like a clock, and indicates ticks per second. Generally, > higher values are better. > > Both values of tbn you saw should work fine for your file. You got > different values because 29.97 is not *exactly* equal to 30000/1001. > > Also, be careful believing the tbn / tbc "Output #0" stats - they have been > broken for months. Use 'ffmpeg -i infile' to get accurate values. > So tbn is timebase, but what is that exactly? Also, what are tbr and tbc? > > Mike Scheutzow > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From seandarcy2 at gmail.com Sat Apr 2 21:28:36 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Sat, 02 Apr 2011 15:28:36 -0400 Subject: [FFmpeg-user] fps in March 13 git: -r 29.97 = 2997 tbn ??? In-Reply-To: <4D970DA2.1070302@optonline.net> References: <4D970DA2.1070302@optonline.net> Message-ID: On 04/02/2011 07:50 AM, Mike Scheutzow wrote: > sean darcy wrote: >> Converting to an android playable file, I can't seem to set the fps. >> The input is 29.97. I tried setting -r to 29.97: >> >> ... >> >> Output #0, ipod, to 'OUTPUT1.m4v': >> Metadata: >> encoder : Lavf52.102.0 >> Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 8:9 DAR 4:3], >> q=0-69, pass 1, 10000 kb/s, 30k tbn, 29.97 tbc >> >> tbn is the framerate for the AVStream of the container, as I gather >> from google. It seems that -r affects tbn of the output. But really >> oddly. >> >> How do I set fps of the output? >> > > You misunderstand the meaning of tbn. tbn is not the "framerate" of the > container, it is the timebase. > > A timebase is like a clock, and indicates ticks per second. Generally, > higher values are better. > > Both values of tbn you saw should work fine for your file. You got > different values because 29.97 is not *exactly* equal to 30000/1001. > > Also, be careful believing the tbn / tbc "Output #0" stats - they have > been broken for months. Use 'ffmpeg -i infile' to get accurate values. > > Mike Scheutzow Maybe more accurate, but much more confusing: Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'OUTPUT1.m4v': Metadata: major_brand : M4V minor_version : 512 compatible_brands: isomiso2avc1 creation_time : 2011-04-02 03:00:26 encoder : Lavf52.102.0 Duration: 00:05:44.34, start: 0.000000, bitrate: 9920 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x480 [PAR 8:9 DAR 4:3], 9916 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc This is not interlaced, why 59.94? And tbn, whatever it is, is 30,000??? sean From leo.izen at gmail.com Sat Apr 2 21:51:42 2011 From: leo.izen at gmail.com (Leo Izen) Date: Sat, 2 Apr 2011 15:51:42 -0400 Subject: [FFmpeg-user] fps in March 13 git: -r 29.97 = 2997 tbn ??? In-Reply-To: References: <4D970DA2.1070302@optonline.net> Message-ID: On Sat, Apr 2, 2011 at 3:28 PM, sean darcy wrote: > On 04/02/2011 07:50 AM, Mike Scheutzow wrote: > >> sean darcy wrote: >> >>> Converting to an android playable file, I can't seem to set the fps. >>> The input is 29.97. I tried setting -r to 29.97: >>> >>> ... >>> >>> Output #0, ipod, to 'OUTPUT1.m4v': >>> Metadata: >>> encoder : Lavf52.102.0 >>> Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 8:9 DAR 4:3], >>> q=0-69, pass 1, 10000 kb/s, 30k tbn, 29.97 tbc >>> >>> tbn is the framerate for the AVStream of the container, as I gather >>> from google. It seems that -r affects tbn of the output. But really >>> oddly. >>> >>> How do I set fps of the output? >>> >>> >> You misunderstand the meaning of tbn. tbn is not the "framerate" of the >> container, it is the timebase. >> >> A timebase is like a clock, and indicates ticks per second. Generally, >> higher values are better. >> >> Both values of tbn you saw should work fine for your file. You got >> different values because 29.97 is not *exactly* equal to 30000/1001. >> >> Also, be careful believing the tbn / tbc "Output #0" stats - they have >> been broken for months. Use 'ffmpeg -i infile' to get accurate values. >> >> Mike Scheutzow >> > > Maybe more accurate, but much more confusing: > > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'OUTPUT1.m4v': > Metadata: > major_brand : M4V > minor_version : 512 > compatible_brands: isomiso2avc1 > creation_time : 2011-04-02 03:00:26 > encoder : Lavf52.102.0 > Duration: 00:05:44.34, start: 0.000000, bitrate: 9920 kb/s > Stream #0.0(und): Video: h264 (High), yuv420p, 720x480 [PAR 8:9 DAR > 4:3], 9916 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc > > This is not interlaced, why 59.94? And tbn, whatever it is, is 30,000??? > Here's one where none of the numbers match: Input #0, mpeg, from 'IndianaJones3.vob': Duration: 02:06:52.96, start: 1.000000, bitrate: 6245 kb/s Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [PAR 32:27 DAR 16:9], 9800 kb/s, 23.98 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s > > sean > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From seandarcy2 at gmail.com Sun Apr 3 16:36:30 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Sun, 03 Apr 2011 10:36:30 -0400 Subject: [FFmpeg-user] MP4Box can't mux ffmpeg x264 output: "IsoMedia File is truncated" Message-ID: Using yesterday's git, I'm trying to make an android playable mp4. Here's the script full-level3-nero-mp4box: INPUT=total.file ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -vpre slower_firstpass \ -bufsize 10000000 -maxrate 10000000 -b 10000000 -threads 0 -f mp4 -y /dev/null ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -vpre slower -timestamp now \ -bufsize 10000000 -maxrate 10000000 -b 10000000 -threads 0 $1.m4v ffmpeg -i $INPUT -vn -acodec copy -f wav - | neroAacEnc -q 0.3 -ignorelength -if - -of $1.m4a MP4Box -fps 29.97 -add $1.m4v -add $1.m4a -new $1.mp4 I'm using MP4Box from this week: MP4Box -version MP4Box - GPAC version 0.4.6-DEV-rev2812M The command ./full-level3-nero-mp4box full-test runs fine until: + MP4Box -fps 29.97 -add full-test.m4v -add full-test.m4a -new full-test.mp4 [iso file] Incomplete box ..5. [iso file] Incomplete box ..5. MP4Box thinks full-test.m4a is fine but not m4v: MP4Box -info full-test.m4v [iso file] Incomplete box ..5. [iso file] Incomplete box ..5. Truncated file - missing 1219279998 bytes Error opening file full-test.m4v: IsoMedia File is truncated FWIW: 14854937328 Apr 2 16:36 total.file 4962718305 Apr 3 02:25 full-test.m4v ffprobe full-test.m4v FFprobe version git-N-28804-gf38fa75, Copyright (c) 2007-2011 the FFmpeg developers built on Apr 2 2011 23:14:01 with gcc 4.5.1 20100924 (Red Hat 4.5.1-4) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --shlibdir=/usr/lib64 --extra-cflags='-O2 -march=native -mtune=native -fopenmp -fomit-frame-pointer -pipe' --enable-static --enable-shared --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-gray --enable-vaapi --enable-hardcoded-tables --enable-frei0r --enable-libopencv --enable-libdirac --disable-decoder=libdirac --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --disable-encoder=libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-zlib --disable-debug --cpu=amdfam10 --arch=x86_64 --enable-pic libavutil 50. 40. 0 / 50. 40. 0 libavcodec 52.117. 0 / 52.117. 0 libavformat 52.104. 0 / 52.104. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 76. 0 / 1. 76. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'full-test.m4v': Metadata: major_brand : M4V minor_version : 512 compatible_brands: isomiso2avc1 creation_time : 2011-04-03 03:39:34 encoder : Lavf52.104.0 Duration: 01:06:26.64, start: 0.000000, bitrate: 9958 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x480 [PAR 8:9 DAR 4:3], 9953 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 2011-04-03 03:39:34 ffprobe total.file ....... Input #0, avi, from 'total.file': Duration: 00:00:06.60, start: 0.000000, bitrate: -2147483 kb/s Stream #0.0: Video: dvvideo, yuv411p, 720x480 [PAR 8:9 DAR 4:3], 29.97 tbr, 29.97 tbn, 29.97 tbc Stream #0.1: Audio: pcm_s16le, 32000 Hz, 2 channels, s16, 1024 kb/s Is this related to the odd tbn and tbr in full-test.m4v? From an earlier thread, I thought that didn't matter. But if not that, what is causing the failure? sean From michaelni at gmx.at Sun Apr 3 19:13:01 2011 From: michaelni at gmx.at (Michael Niedermayer) Date: Sun, 3 Apr 2011 19:13:01 +0200 Subject: [FFmpeg-user] [RELEASE] ffmpeg-0.6.90-rc0 Message-ID: <20110403171300.GT19660@kiste2> Hi Iam proud to announce that after a way too long time we finally are making a new release. See: http://ffmpeg.org/releases/ Please help test it and report regression & security bugs to ffmpeg.org/trac so we can fix them in rc1 or the final release. (depending on if we will need a rc1) Also please help testing on not so common platforms if you can. (as you know the 3 server admins took all machines they had access to with them for their fork) This release also has fully merged all changes from the following ffmpeg forks: ffmpeg-mt libav And partially merged: ffdshow (i have reviewed all changes but it seems they have already submited back most clean changes, the rest needs (alot) more work to be merged back) -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB In a rich man's house there is no place to spit but his face. -- Diogenes of Sinope -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From seandarcy2 at gmail.com Sun Apr 3 23:20:03 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Sun, 03 Apr 2011 17:20:03 -0400 Subject: [FFmpeg-user] How to create file for ffserver? Index & Id do not match Message-ID: I'd like to have ffserver serve a file. Here's ffserver.conf Port 8090 BindAddress 0.0.0.0 MaxHTTPConnections 2000 MaxClients 1000 MaxBandwidth 1000 # Access log file (uses standard Apache log file format) # '-' is the standard output. CustomLog - # Suppress that if you want to launch ffserver as a daemon. NoDaemon # You could specify # ReadOnlyFile # This marks the file as readonly and it will not be deleted or updated. ReadOnlyFile /opt/Annie-July-Play-2004.asf FileMaxSize 400M # Specify launch in order to start ffmpeg automatically. # First ffmpeg must be defined with an appropriate path if needed, # after that options can follow, but avoid adding the http:// field #Launch ffmpeg # Only allow connections from localhost to the feed. ACL allow 127.0.0.1 Feed Annie.ffm Format asf VideoFrameRate 15 VideoSize 720?480 VideoBitRate 256 VideoBufferSize 40 VideoGopSize 30 AudioBitRate 64 StartSendOnKey I've created the file Annie.ffm: ffmpeg -i total.file -r 15 -vb 256000 -ab 64000 -f asf Annie.ffm FFmpeg version git-N-28804-gf38fa75, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 2 2011 23:14:01 with gcc 4.5.1 20100924 (Red Hat 4.5.1-4) ............. Output #0, asf, to 'Annie.ffm': Metadata: WM/EncodingSettings: Lavf52.104.0 Stream #0.0: Video: msmpeg4, yuv420p, 720x480 [PAR 8:9 DAR 4:3], q=2-31, 256 kb/s, 1k tbn, 15 tbc Stream #0.1: Audio: libmp3lame, 32000 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 But: ffserver ....... Sun Apr 3 17:14:35 2011 Index & Id do not match for stream 0 (/opt/1500gig/MyBook-backup/video-2/plp-2004/play-july25/dv/tmp/Annie.ffm) From is091516 at fhstp.ac.at Mon Apr 4 09:15:20 2011 From: is091516 at fhstp.ac.at (Markus Riegler) Date: Mon, 04 Apr 2011 09:15:20 +0200 Subject: [FFmpeg-user] VP8 Decoding Message-ID: <4D997008.7090304@fhstp.ac.at> Hi, I'm working on my master thesis of the forensic analysis of the VP8 videocodec. I want to extract single Keyframes (Intraframes) of an .webm file and paste the data of the keyframe into a .ivf header to decode the single frame. After that I want to display this single frame. I can do this procedure with the commercial Software CodecVisa. My question about that is, is it possible to decode single vp8 keyframes in an .ivf header and display it with ffmpeg and how? Best regards from Austria -- ______________________________________________ BSc. Riegler Markus Information Security Fachhochschule St.P?lten GmbH Matthias Covernius-Stra?e 15, A-3100 St.P?lten E: is091516 at fhstp.ac.at I: www.fhstp.ac.at From victor.petrescu13 at gmail.com Mon Apr 4 11:50:19 2011 From: victor.petrescu13 at gmail.com (Victor Petrescu) Date: Mon, 4 Apr 2011 12:50:19 +0300 Subject: [FFmpeg-user] How to discard probe frames ? In-Reply-To: <4D934151.7070805@orange-ftgroup.com> References: <4D934151.7070805@orange-ftgroup.com> Message-ID: I would also be interested in an answer to this question. So... as Alex said... any takers? :) 2011/3/30 Alexandre Ferrieux > Hello, > > At the beginning of a stream, av_read_frame_internal is called repeatedly > to probe codec parameters. Once this probing is completed, the frames are > spit back from memory into the processing chain. Now in case of real-time > transcoding this is problematic because it introduces a fixed, possibly > large latency. > > Is there a (command-line) way to simply *discard* these stored frames, > hence keeping up with real time ? > > Thanks in advance, > > -Alex > > PS: I am aware of -probesize to forcibly shorten this loop, but it is not > always desirable, as the needed probe size may depend on stream contents and > position within GOP (or wrt SPS/PPS retransmission in the H264 case). > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From victor.petrescu13 at gmail.com Mon Apr 4 14:03:42 2011 From: victor.petrescu13 at gmail.com (Victor Petrescu) Date: Mon, 4 Apr 2011 15:03:42 +0300 Subject: [FFmpeg-user] Two outputs for ffmpeg Message-ID: Good day to all. I need to stream something from a live cam and to record it in the same time. I use ffsever for streaming and use a pipe to feed ffmpeg. So I have the 2 commands ffmpeg -i pipe.avi -async 1 -r 25 http://localhost:8090/feed1.ffm -> for streaming ffmpeg -i pipe.avi -async 1 -vcodec flv -f flv -r 25 somename.flv -> for recording Both work fine independent. The problem is that when I use them together The recording is decent (not good) but the stream look like hell (random colors everywhere). Can any1 tell me how to fix this? Thank you for your time. Victor From pastas4 at gmail.com Mon Apr 4 19:15:07 2011 From: pastas4 at gmail.com (Dainius (GreatEmerald)) Date: Mon, 4 Apr 2011 20:15:07 +0300 Subject: [FFmpeg-user] VBR 2-pass or CRF when recording with libx264 Message-ID: I am trying to determine the best way to record my screen here using FFmpeg from Linux. I have already tried a lot of options, but I would like to hear the opinion of others as well. The thing that makes recording difficult in my case is that I need to use sox to actually record and mix my microphone and system sounds. If I pipe sox to FFmpeg, the recording performance plummets from constant 25 FPS to constant 5 FPS, so obviously that's not usable. However, if I use sox to output the MP3 file and then use FFmpeg to mix the two, I get a video that is usable. However, there is something I want to find out that I don't really get: FFmpeg can record using two-pass VBR, which is probably the best way to save both quality and disk space. Is it feasible to use it in my case, since I don't have the original recording (it was the screen itself, after all)? If I try to run the first pass, I do get a log file with all of the info in it, so I could use that and use the video output I got from the first pass, but wouldn't that destroy the point of doing a two-pass recording in the first place? And if not, and two-pass VBR is out of the question, then what would be a good way to encode the video for a good quality/size ratio? Single-pass VBR? If so, what settings are the most important for me (since I'm recording a game through x11grab, and in case you want to know, it's Heroes of Might and Magic III :) - it gets a fair share of pretty static images)? For example, how different is changing qscale from qmax/qmin? And should I be concerned with things like qdiff, qblur and qcomp? In fact, does libx264 even acknowledge these settings to begin with? By the way, x264 guides say that the best single-pass option using libx264 is crf, '-crf' in FFmpeg, but the FFmpeg manual does not list that at all. So where should the argument go, and what are valid options for it? From brandono at gmail.com Mon Apr 4 21:14:38 2011 From: brandono at gmail.com (Brandon Ooi) Date: Mon, 4 Apr 2011 12:14:38 -0700 Subject: [FFmpeg-user] Two outputs for ffmpeg In-Reply-To: References: Message-ID: On Mon, Apr 4, 2011 at 5:03 AM, Victor Petrescu wrote: > Good day to all. > > I need to stream something from a live cam and to record it in the same > time. > > I use ffsever for streaming and use a pipe to feed ffmpeg. > > So I have the 2 commands > > ffmpeg -i pipe.avi -async 1 -r 25 http://localhost:8090/feed1.ffm -> for > streaming > ffmpeg -i pipe.avi -async 1 -vcodec flv -f flv -r 25 somename.flv -> for > recording > > Both work fine independent. > > The problem is that when I use them together The recording is decent (not > good) but the stream look like hell (random colors everywhere). > > Can any1 tell me how to fix this? > > A pipe as in a fifo? To use correctly you need to split/copy the fifo into 2 fifos (mkfifo + tee?). Because once one ffmpeg pulls data from a fifo, it's gone. The other ffmpeg gets whatever comes after. From lars.taeuber at gmx.net Mon Apr 4 21:50:09 2011 From: lars.taeuber at gmx.net (Lars =?UTF-8?B?VMOkdWJlcg==?=) Date: Mon, 4 Apr 2011 21:50:09 +0200 Subject: [FFmpeg-user] strange ffmpeg effect on bash Message-ID: <20110404215009.20fe8fff.lars.taeuber@gmx.net> Hi, I've got a problem with ffmpeg in a shell script. It's a bash script to be precise. The script looks like this: =================== #!/bin/bash LOG=${0%.sh}.log ERR=${0%.sh}.err JOBS=jobs.lst >"$LOG" >"$ERR" while read LINE do FILE="" NAME="" eval $LINE test -z "$NAME" -o ! -r "$FILE" && continue ffmpeg -an -sn -i "$FILE" -vcodec ffvhuff -t 30 -f matroska -y "$NAME.mkv" >>"$LOG" 2>>"$ERR" # mencoder -ovc copy -oac pcm -o "$NAME.avi" -endpos 30 "$FILE" >>"$LOG" 2>>"$ERR" echo $? >> "$LOG" done < "$JOBS" echo "Ende" >> "$LOG" =================== With the ffmpeg command the script ends after the first line read from "$JOBS". Somehow bash interpretes its completion like a ?break? command. When you replace the ffmpeg command with the (commented out) mplayer command (or any other programm) all lines from "$JOBS" are read and interpreted as expected. The "$JOBS" file looks like this: =================== FILE=a.avi;NAME=b FILE=c.avi;NAME=d FILE=e.avi;NAME=e =================== How comes? What's the special thing here with ffmpeg? Maybe I should ask this on a bash mailing list? Thanks Lars PS: GNU bash, Version 4.2.8(1)-release (x86_64-pc-linux-gnu) FFmpeg version git-N-28820-g7f125c3 -------------- next part -------------- A non-text attachment was scrubbed... Name: e.sh Type: text/x-sh Size: 433 bytes Desc: not available URL: From cbeerse at gmail.com Mon Apr 4 22:59:57 2011 From: cbeerse at gmail.com (cbeerse at gmail.com) Date: Mon, 04 Apr 2011 22:59:57 +0200 Subject: [FFmpeg-user] strange ffmpeg effect on bash In-Reply-To: <20110404215009.20fe8fff.lars.taeuber@gmx.net> References: <20110404215009.20fe8fff.lars.taeuber@gmx.net> Message-ID: <4D9A314D.5020002@gmail.com> On 4-4-2011 21:50, Lars T?uber wrote: > Hi, > > I've got a problem with ffmpeg in a shell script. > It's a bash script to be precise. > The script looks like this: > > =================== > #!/bin/bash > > LOG=${0%.sh}.log > ERR=${0%.sh}.err > JOBS=jobs.lst >> "$LOG" >> "$ERR" > while read LINE > do > FILE="" > NAME="" > eval $LINE > > test -z "$NAME" -o ! -r "$FILE"&& continue > > ffmpeg -an -sn -i "$FILE" -vcodec ffvhuff -t 30 -f matroska -y "$NAME.mkv">>"$LOG" 2>>"$ERR" > # mencoder -ovc copy -oac pcm -o "$NAME.avi" -endpos 30 "$FILE">>"$LOG" 2>>"$ERR" > echo $?>> "$LOG" > > done< "$JOBS" > echo "Ende">> "$LOG" > =================== > > With the ffmpeg command the script ends after the first line read from "$JOBS". Somehow bash interpretes its completion like a ?break? command. > When you replace the ffmpeg command with the (commented out) mplayer command (or any other programm) all lines from "$JOBS" are read and interpreted as expected. > > The "$JOBS" file looks like this: > =================== > FILE=a.avi;NAME=b > FILE=c.avi;NAME=d > FILE=e.avi;NAME=e > =================== > > How comes? > What's the special thing here with ffmpeg? I think (close to being sure) that ffmpeg does something with stdin and stdout. Hence as you use that too, that can somehow somewhere get mixed up the wrong way. If you rewrite your script to read the file not using the stdin but using one of the other I/Os: while read -fd 3 LINE do # your code here is the same.. done 3< "$JOBS" while read LINE <&3 do # your code here is the same.. done 3< "$JOBS" Or use the for command (might behave different with blanks/spaces in the lines): | |for LINE in `cat "$JOBS"` do # your code here is the same done The above is rough code, not tested. It is about using one of the other file descripters. e.g. advanced scripting. > Maybe I should ask this on a bash mailing list? I bet the bash list will point you to ffmpeg. However, the use of alternate file descripters is definitly using bash. The question there should be: I'd like to use my code but I expect ffmpeg is using the std i/o. Ho can I get around that? CBee From baptiste.coudurier at gmail.com Mon Apr 4 23:56:04 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Mon, 04 Apr 2011 14:56:04 -0700 Subject: [FFmpeg-user] fps in March 13 git: -r 29.97 = 2997 tbn ??? In-Reply-To: <4D970DA2.1070302@optonline.net> References: <4D970DA2.1070302@optonline.net> Message-ID: <4D9A3E74.6040906@gmail.com> Hi, On 04/02/2011 04:50 AM, Mike Scheutzow wrote: > sean darcy wrote: >> Converting to an android playable file, I can't seem to set the fps. >> The input is 29.97. I tried setting -r to 29.97: >> >> ... >> >> Output #0, ipod, to 'OUTPUT1.m4v': >> Metadata: >> encoder : Lavf52.102.0 >> Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 8:9 DAR 4:3], >> q=0-69, pass 1, 10000 kb/s, 30k tbn, 29.97 tbc >> >> tbn is the framerate for the AVStream of the container, as I gather >> from google. It seems that -r affects tbn of the output. But really >> oddly. >> >> How do I set fps of the output? >> > > You misunderstand the meaning of tbn. tbn is not the "framerate" of the > container, it is the timebase. > > A timebase is like a clock, and indicates ticks per second. Generally, > higher values are better. > > Both values of tbn you saw should work fine for your file. You got > different values because 29.97 is not *exactly* equal to 30000/1001. > > Also, be careful believing the tbn / tbc "Output #0" stats - they have > been broken for months. Use 'ffmpeg -i infile' to get accurate values. No, the values are correct and accurate, you can trust them. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From baptiste.coudurier at gmail.com Tue Apr 5 00:00:56 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Mon, 04 Apr 2011 15:00:56 -0700 Subject: [FFmpeg-user] MP4Box can't mux ffmpeg x264 output: "IsoMedia File is truncated" In-Reply-To: References: Message-ID: <4D9A3F98.1050304@gmail.com> Hi, On 04/03/2011 07:36 AM, sean darcy wrote: > Using yesterday's git, I'm trying to make an android playable mp4. > > Here's the script full-level3-nero-mp4box: > > INPUT=total.file > ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -vpre > slower_firstpass \ > -bufsize 10000000 -maxrate 10000000 -b 10000000 -threads 0 -f mp4 > -y /dev/null > ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -vpre slower > -timestamp now \ > -bufsize 10000000 -maxrate 10000000 -b 10000000 -threads 0 $1.m4v > > [...] > > I'm using MP4Box from this week: > > MP4Box -version > MP4Box - GPAC version 0.4.6-DEV-rev2812M > > > The command ./full-level3-nero-mp4box full-test runs fine until: > > + MP4Box -fps 29.97 -add full-test.m4v -add full-test.m4a -new > full-test.mp4 > [iso file] Incomplete box ..5. > [iso file] Incomplete box ..5. > > MP4Box thinks full-test.m4a is fine but not m4v: > > MP4Box -info full-test.m4v > [iso file] Incomplete box ..5. > [iso file] Incomplete box ..5. > Truncated file - missing 1219279998 bytes > Error opening file full-test.m4v: IsoMedia File is truncated Can you upload the m4v somewhere ? -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From chipperb at nc.rr.com Tue Apr 5 01:35:58 2011 From: chipperb at nc.rr.com (Ralph Blach) Date: Mon, 04 Apr 2011 19:35:58 -0400 Subject: [FFmpeg-user] compiling libpostproc as a shared library Message-ID: <4D9A55DE.2010507@nc.rr.com> I am trying to compile libpostproc as a shared library on fedora core 12 I am using the configure line ./configure --enable-shared --enable-version3 --enable-nonfree --prefix=/usr/local but lib postproc get build as a .a static library How do I compile libpostproc as a shared library? Thanks in advance Chip From drurowin at gmail.com Tue Apr 5 05:43:05 2011 From: drurowin at gmail.com (Lucien Pullen) Date: Mon, 04 Apr 2011 21:43:05 -0600 Subject: [FFmpeg-user] Compiling FFMPEG with ffplay support In-Reply-To: References: <4D91A112.70001@bbc.co.uk> Message-ID: <4D9A8FC9.30106@gmail.com> Also sprach Stas Oskin at 3/29/11 7:10 AM: > Hi. > > ffplay needs SDL. Many FFmpeg --enable-* options are silently self-disabled >> if 3rd-party library not available. >> So probably something happens with your SDL installation... >> ----------------------------------------------- >> >> > > Is there a way to see what was disabled and why? You can check config.log to see what was disabled. If you are getting no ffplay, then check the line ``USE_SDL='' to see if that has anything. To make sure SDL gets enabled, you'll want to set the environment variable ``USE_SDL'' to ``/path/to/sdl-config''. Passing the output of that to configure is a nightmare you don't want. I had the entire build process (from git pull to install) in a script and noticed that ffplay hadn't been updated in a month, and that fixed it. From max at quendi.de Tue Apr 5 10:28:42 2011 From: max at quendi.de (Max Horn) Date: Tue, 5 Apr 2011 10:28:42 +0200 Subject: [FFmpeg-user] (x)WMA / WMA Pro 10 support Message-ID: <33157B41-0D51-422E-846A-4740E3B28BB0@quendi.de> Hello everybody, I was wondering if ffmpeg / libavcodec supports decoding both WMA Pro 10 ? On , it lists decoding of "Windows Media Audio Pro". But it doesn't specify if that means both WMA Pro 9 and 10, or just 9. According to some sources (see e.g. here on Wikipedia: ), version 10 files can be decoded by a v9 decoder, but only at reduced quality. I actually tried finding out by looking at the source code (specifically, at wmaprodec.c), but couldn't figure this out there myself. Thus I'd really appreciate a clarification. Maybe this could then also be clarified on the mentioned page. Thanks & Cheers, Max From stas.oskin at gmail.com Tue Apr 5 14:42:12 2011 From: stas.oskin at gmail.com (Stas Oskin) Date: Tue, 5 Apr 2011 15:42:12 +0300 Subject: [FFmpeg-user] Compiling FFMPEG with ffplay support In-Reply-To: <4D9A8FC9.30106@gmail.com> References: <4D91A112.70001@bbc.co.uk> <4D9A8FC9.30106@gmail.com> Message-ID: Hi. You can check config.log to see what was disabled. If you are getting no > ffplay, then check the line ``USE_SDL='' to see if that has anything. > > Originally there was no USE_SDL message at all. > To make sure SDL gets enabled, you'll want to set the environment > variable ``USE_SDL'' to ``/path/to/sdl-config''. Passing the output of > that to configure is a nightmare you don't want. > Did this with no results - ffplay is still not being build. Any idea what else I can check? Thanks. From stas.oskin at gmail.com Tue Apr 5 14:46:59 2011 From: stas.oskin at gmail.com (Stas Oskin) Date: Tue, 5 Apr 2011 15:46:59 +0300 Subject: [FFmpeg-user] Compiling FFMPEG with ffplay support In-Reply-To: References: <4D91A112.70001@bbc.co.uk> <4D9A8FC9.30106@gmail.com> Message-ID: By checking the configure output, I see the following: SDL support no Any way to see why it was disabled? Thanks. From h.reindl at thelounge.net Tue Apr 5 15:23:37 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Tue, 05 Apr 2011 15:23:37 +0200 Subject: [FFmpeg-user] Compiling FFMPEG with ffplay support In-Reply-To: References: <4D91A112.70001@bbc.co.uk> <4D9A8FC9.30106@gmail.com> Message-ID: <4D9B17D9.1080709@thelounge.net> Am 05.04.2011 14:46, schrieb Stas Oskin: > By checking the configure output, I see the following: > > SDL support no > > Any way to see why it was disabled? > > Thanks. Have you installed the devel-package? [builduser at buildserver64:~]$ rpm -qa | grep SDL SDL-devel-1.2.14-10.fc14.x86_64 SDL-1.2.14-10.fc14.x86_64 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From mike.scheutzow at alcatel-lucent.com Tue Apr 5 15:33:03 2011 From: mike.scheutzow at alcatel-lucent.com (Mike Scheutzow) Date: Tue, 05 Apr 2011 09:33:03 -0400 Subject: [FFmpeg-user] fps in March 13 git: -r 29.97 = 2997 tbn ??? In-Reply-To: <4D9A3E74.6040906@gmail.com> References: <4D970DA2.1070302@optonline.net> <4D9A3E74.6040906@gmail.com> Message-ID: <4D9B1A0F.1010203@alcatel-lucent.com> Baptiste Coudurier wrote: >> Also, be careful believing the tbn / tbc "Output #0" stats - they have >> been broken for months. Use 'ffmpeg -i infile' to get accurate values. > > No, the values are correct and accurate, you can trust them. It was still broken in Michael's git as of 2011-mar-16. The test case is very simple: h264 in .ts transcoded to h264 in .ts. During encode run, for "Output #0" ffmpeg prints: Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 10:11 DAR 15:11], q=10-51, 200 kb/s, 90k tbn, 29.97 tbc Yet, ffmpeg -i on the resulting file says: Stream #0.0[0x100]: Video: h264 (High), yuv420p, 720x480 [PAR 10:11 DAR 15:11], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Do the tbc look the same to you? Mike Scheutzow From baptiste.coudurier at gmail.com Tue Apr 5 18:20:51 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Tue, 05 Apr 2011 09:20:51 -0700 Subject: [FFmpeg-user] fps in March 13 git: -r 29.97 = 2997 tbn ??? In-Reply-To: <4D9B1A0F.1010203@alcatel-lucent.com> References: <4D970DA2.1070302@optonline.net> <4D9A3E74.6040906@gmail.com> <4D9B1A0F.1010203@alcatel-lucent.com> Message-ID: <4D9B4163.9030407@gmail.com> On 04/05/2011 06:33 AM, Mike Scheutzow wrote: > Baptiste Coudurier wrote: >>> Also, be careful believing the tbn / tbc "Output #0" stats - they have >>> been broken for months. Use 'ffmpeg -i infile' to get accurate values. >> >> No, the values are correct and accurate, you can trust them. > > It was still broken in Michael's git as of 2011-mar-16. > > The test case is very simple: h264 in .ts transcoded to h264 in .ts. > > During encode run, for "Output #0" ffmpeg prints: > > Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 10:11 DAR 15:11], > q=10-51, 200 kb/s, 90k tbn, 29.97 tbc > > Yet, ffmpeg -i on the resulting file says: > > Stream #0.0[0x100]: Video: h264 (High), yuv420p, 720x480 [PAR 10:11 > DAR 15:11], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc > > Do the tbc look the same to you? And which one do you think are wrong anyway ? Information printed is accurate, the interpretation is different. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From klgiger at gmail.com Tue Apr 5 19:08:34 2011 From: klgiger at gmail.com (kevin giger) Date: Tue, 5 Apr 2011 11:08:34 -0600 Subject: [FFmpeg-user] ffmpeg compile errors undefined reference to x264_param_default Message-ID: I appreciate any help on this issue. Here is my compile string ./configure --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-libdirac --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab root at main [/usr/src/ffmpeg]# And here is the error. LD ffmpeg_g /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `x264_param_default' /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `x264_encoder_headers' /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `x264_encoder_encode' /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `x264_encoder_reconfig' /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `x264_encoder_delayed_frames' /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `x264_encoder_open_114' /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `x264_picture_init' /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `x264_encoder_close' collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1 I have removed and reinstalle x264 with enable share setting. Ran ldconfig. But I still get this error every time. Thanks! From lars.taeuber at gmx.net Tue Apr 5 20:58:47 2011 From: lars.taeuber at gmx.net (Lars =?UTF-8?B?VMOkdWJlcg==?=) Date: Tue, 5 Apr 2011 20:58:47 +0200 Subject: [FFmpeg-user] strange ffmpeg effect on bash In-Reply-To: <4D9A314D.5020002@gmail.com> References: <20110404215009.20fe8fff.lars.taeuber@gmx.net> <4D9A314D.5020002@gmail.com> Message-ID: <20110405205847.d9652a3f.lars.taeuber@gmx.net> Hallo CBee, On Mon, 04 Apr 2011 22:59:57 +0200 "cbeerse at gmail.com" wrote: > On 4-4-2011 21:50, Lars T?uber wrote: > > Hi, > > > > I've got a problem with ffmpeg in a shell script. > > It's a bash script to be precise. [...] > > I think (close to being sure) that ffmpeg does something with stdin and > stdout. Hence as you use that too, that can somehow somewhere get mixed > up the wrong way. you're right. Maybe ffmpeg should test for the shell being interactive or not. > If you rewrite your script to read the file not using the stdin but > using one of the other I/Os: Thanks. I did it like you suggested. It works very well. Lars From leo.izen at gmail.com Tue Apr 5 22:47:43 2011 From: leo.izen at gmail.com (Leo Izen) Date: Tue, 5 Apr 2011 16:47:43 -0400 Subject: [FFmpeg-user] FFmpeg on Windows can't find a file that exists Message-ID: I have a file on windows, and when I try to run ffmpeg on it, it says it doesn't exist, even though it does. Why is this? How can I fix it? Here is my output: C:\Users\todd\bin>dir "..\Boombox (ft. Julian Casablancas).mp4" Volume in drive C is OS Volume Serial Number is B845-7168 Directory of C:\Users\todd 04/03/2011 04:18 PM 70,291,188 Boombox (ft. Julian Casablancas).mp4 1 File(s) 70,291,188 bytes 0 Dir(s) 564,990,820,352 bytes free C:\Users\todd\bin>ffmpeg -i "..\Boombox (ft. Julian Casablancas).mp4" FFmpeg version git-N-28846-g02fd687, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 4 2011 17:29:05 with gcc 4.7.0 20110402 (experimental) configuration: --enable-static --disable-shared --enable-gpl --enable-version3 --disable-nonfree --disable-postproc --enable-runtime-cpudetect --disable-avisynth --enable-bzlib --disable-libopencore-amrnb --disable-libopencore-amrwb --disable-libdirac --disable-libfaac --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --disable-libtheora --disable-libvorbis --disable-libvpx --enable-libx264 --disable-libxvid --enable-zlib --arch=x86_64 --cpu=i7 --enable-pic --enable-sram --prefix=/usr/local/x86_64-w64-mingw32 --cross-prefix=/usr/x86_64-w64-mingw32/bin/ --target-os=mingw32 --pkg-config=pkg-config --extra-libs='-lorc-0.4 -lx264' --enable-stripping --disable-debug --enable-w32threads --disable-encoders --enable-encoder=libx264 --enable-encoder=aac --enable-encoder=pcm_s16le --enable-encoder=alac --disable-muxers --enable-muxer=ipod --enable-muxer=wav --enable-muxer=mp3 --disable-protocols --disable-outdevs --disable-indevs --disable-parsers --disable-bsfs --disable-demuxer=image2 --disable-demuxer=image2pipe --disable-decoder=png --disable-decoder=ppm --disable-decoder=tiff --disable-decoder=bmp --disable-decoder=gif --disable-decoder=targa --enable-encoder=libmp3lame --disable-ffprobe --disable-network libavutil 50. 40. 0 / 50. 40. 0 libavcodec 52.117. 0 / 52.117. 0 libavformat 52.105. 0 / 52.105. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 77. 0 / 1. 77. 0 libswscale 0. 13. 0 / 0. 13. 0 ..\Boombox (ft. Julian Casablancas).mp4: No such file or directory C:\Users\todd\bin> From leo.izen at gmail.com Tue Apr 5 22:56:38 2011 From: leo.izen at gmail.com (Leo Izen) Date: Tue, 5 Apr 2011 16:56:38 -0400 Subject: [FFmpeg-user] How to compile the ffmper full source to take a .dll build. In-Reply-To: References: Message-ID: 2011/3/25 mouli s > hello i tried taking .dll build n succeed in it but whn i tried to add it > in > the project using add reference it shows error....says tat it could not be > added plz make sure tat the file is accessible.....hope the .dll i took is > not working... can any one plz share the working .dll for the fullsource of > ffmpeg??? if so very thank full to them.... thanks for any replies...... > 1. Please use words such as "please" and not "plz" or "that" and not "tat." It is easier to help you if we don't have to parse what you are saying. 2. FFmpeg builds are available at http://hawkeye.arrozcru.org/ 3. Whenever you are asking for help about an error message, post the full un-modified error message, because otherwise we won't have enough information to give a helpful answer. 4. You mention project, but you don't mention what type of project it is, your OS, your compiler, etc. so we don't have enough information to answer your question. Please give as much information as possible, because too much is always better than too little. > > 2011/3/17 V?ctor Paesa > > > Hi, > > > > On Wed, Mar 16, 2011 at 14:32, mouli s wrote: > > > Hello im trying to take a .dll build for the full ffmpeg source.. where > > can > > > i download the full source code??? > > > > http://www.ffmpeg.org/download.html > > > > > n how to take a .dll build for that > > > source??? anyone plz guide me for it as im new to this???? > > > > http://www.ffmpeg.org/general.html#SEC20 > > and/or > > http://ffmpeg.arrozcru.org/wiki/index.php?title=Main_Page > > > > Regards, > > V?ctor > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From james.darnley at gmail.com Tue Apr 5 23:38:13 2011 From: james.darnley at gmail.com (James Darnley) Date: Tue, 5 Apr 2011 23:38:13 +0200 Subject: [FFmpeg-user] FFmpeg on Windows can't find a file that exists In-Reply-To: References: Message-ID: On 05/04/2011, Leo Izen wrote: > --disable-protocols And you wonder why it can't open a file? You need the file protocol for this. From jens.ruda at live.com Wed Apr 6 10:58:30 2011 From: jens.ruda at live.com (Jens Ruda) Date: Wed, 6 Apr 2011 10:58:30 +0200 Subject: [FFmpeg-user] Streaming Message-ID: HiI can use this command and save the output in Ubuntu 10.10. ffmpeg -f alsa -ac 2 -i pulse -f x11grab -r 30 -s 1024x768 -i :0.0 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 output.mkvWhat I need is to stream the output to another machine. I tested all the examples I can find. But I get them not to work with x11grab.I need very low latency. My goal is to stream fullscreen from multiple computers, and mixing them on another computer in real time. Can someone show the way? Do I need to use ffserver? Regards,Jens From victor.petrescu13 at gmail.com Wed Apr 6 11:02:16 2011 From: victor.petrescu13 at gmail.com (Victor Petrescu) Date: Wed, 6 Apr 2011 12:02:16 +0300 Subject: [FFmpeg-user] Streaming In-Reply-To: References: Message-ID: Stream with vlc from the computers. On the server where you mix use ffmpeg+mplayer to mix them (it worked for me). 2011/4/6 Jens Ruda > > HiI can use this command and save the output in Ubuntu 10.10. ffmpeg -f > alsa -ac 2 -i pulse -f x11grab -r 30 -s 1024x768 -i :0.0 -acodec pcm_s16le > -vcodec libx264 -vpre lossless_ultrafast -threads 0 output.mkvWhat I need is > to stream the output to another machine. I tested all the examples I can > find. But I get them not to work with x11grab.I need very low latency. My > goal is to stream fullscreen from multiple computers, and mixing them on > another computer in real time. Can someone show the way? Do I need to use > ffserver? Regards,Jens > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From victor.petrescu13 at gmail.com Wed Apr 6 11:08:13 2011 From: victor.petrescu13 at gmail.com (Victor Petrescu) Date: Wed, 6 Apr 2011 12:08:13 +0300 Subject: [FFmpeg-user] the server crashes when runing shell script that starts ffmpeg Message-ID: Good day to all. I have a virtual machine running centOS 5.5 that I use as a streaming server. Now... if I open 4 terminals and do this: terminal 1: # mkfifo pipe1.avi # mkfifo pipe2.avi # ffserver & # ffmpeg -probesize 164000 -i pipe1.avi -async 1 -r 25 http://localhost:8090/feed1.ffm terminal 2: # ffmpeg -i pipe2.avi -async 1 -r 25 -vcodec flv -f flv somename.flv terminal 3: # mplayer -dumpstream rtp://someIp:somePort -dumpfile pipe1.avi terminal 4: # mplayer -dumpstream rtp://someIp:somePort -dumpfile pipe2.avi Everything is working great. No problems nowhere. Now I'm trying to make a script and run it from a cron. This is the script: ffserver & ffmpeg -probesize 164000 -i pipe1.avi -async 1 -r 25 http://localhost:8090/feed1.ffm & ffmpeg -i pipe2.avi -async 1 -r 25 -vcodec flv -f flv somename.flv & mplayer -dumpstream rtp://someIp:somePort -dumpfile pipe1.avi & mplayer -dumpstream rtp://someIp:somePort -dumpfile pipe2.avi & exit 0 The crontab line is: 10 * * * * script.sh > /dev/null 1>&2 The problem is that when the cron is runned the system crashes (actually I not really crashes the stream (witch I see from another computer) is working perfectly but I can't do anything on the machine. The screen becomes black and I can't do anything except to restart it). I have a script that kills the anterior processes. If I uses it the stream stops, but I still don't regain access to the system (same black screeen whatever I do). Can any1 help me with this? Thank you. From stas.oskin at gmail.com Wed Apr 6 13:49:55 2011 From: stas.oskin at gmail.com (Stas Oskin) Date: Wed, 6 Apr 2011 14:49:55 +0300 Subject: [FFmpeg-user] Compiling FFMPEG with ffplay support In-Reply-To: <4D9B17D9.1080709@thelounge.net> References: <4D91A112.70001@bbc.co.uk> <4D9A8FC9.30106@gmail.com> <4D9B17D9.1080709@thelounge.net> Message-ID: Hi. > Have you installed the devel-package? > > [builduser at buildserver64:~]$ rpm -qa | grep SDL > SDL-devel-1.2.14-10.fc14.x86_64 > SDL-1.2.14-10.fc14.x86_64 > > Yes, I did: [root at nctest ~]# rpm -qa | grep SDL SDL_image-1.2.5-4.el5 SDL-1.2.10-8.el5 SDL-devel-1.2.10-8.el5 Anything else to check? Thanks. From nomiya at galaxy.dti.ne.jp Wed Apr 6 14:08:48 2011 From: nomiya at galaxy.dti.ne.jp (Masaru Nomiya) Date: Wed, 06 Apr 2011 21:08:48 +0900 Subject: [FFmpeg-user] ffmpeg compile errors undefined reference to x264_param_default In-Reply-To: References: Message-ID: <87fwpveg4v.wl%nomiya@galaxy.dti.ne.jp> Hello, In the Message; Subject : [FFmpeg-user] ffmpeg compile errors undefined reference to x264_param_default Message-ID : Date & Time: Tue, 5 Apr 2011 11:08:34 -0600 [KG] == kevin giger has written: [...] KG> And here is the error. KG> LD ffmpeg_g KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to KG> `x264_param_default' KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to KG> `x264_encoder_headers' KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to KG> `x264_encoder_encode' KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to KG> `x264_encoder_reconfig' KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to KG> `x264_encoder_delayed_frames' KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to KG> `x264_encoder_open_114' KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to KG> `x264_picture_init' KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to KG> `x264_encoder_close' KG> collect2: ld returned 1 exit status KG> make: *** [ffmpeg_g] Error 1 KG> I have removed and reinstalle x264 with enable share setting. Ran ldconfig. Have you installed the very latest git of x264? Mine is; $ x264 -V x264 0.114.1924 08d04a4 (libswscale 0.12.0) (libavformat 52.103.0) (ffmpegsource 2.15.0.1) built on Mar 25 2011, gcc: 4.5.0 20100604 [gcc-4_5-branch revision 160292] configuration: --bit-depth=8 Regards, --- ????? Masaru Nomiya mail-to: nomiya @ galaxy.dti.ne.jp ???? ???? "Bill! You married with Computers. Not with Me!" "No..., with money." From victor.petrescu13 at gmail.com Wed Apr 6 15:22:09 2011 From: victor.petrescu13 at gmail.com (Victor Petrescu) Date: Wed, 6 Apr 2011 16:22:09 +0300 Subject: [FFmpeg-user] Seek not available Message-ID: Good day to all. I have a little problem. I am using ffmpeg to record from a live stream. I managed to make everything to work fine but I have no seek on the resulting file. The command I used is this: ffmpeg -i pipe.avi -vcodec flv -f flv -async 1 -r 25 -s 640x480 -sameq somename.flv Now I feed the pipe with mplayer dumpstream rtp://someIp:5004 -dumpfile pipe.avi The resulting file is exactly what I need... but with no seek. Can any1 tell me what I'm doing wrong? Thank you for your time. From jens.ruda at live.com Wed Apr 6 15:22:15 2011 From: jens.ruda at live.com (Jens Ruda) Date: Wed, 6 Apr 2011 15:22:15 +0200 Subject: [FFmpeg-user] Streaming In-Reply-To: References: , Message-ID: > Stream with vlc from the computers. On the server where you mix use > ffmpeg+mplayer to mix them (it worked for me). Thank you. I can not get?VLC?to?work?well.?It?works?only?sporadically?and?withvery?high?latency. Can?someone?show?me?a?foolproof?example?with?ffmpeg,?x11grab and?rtsp?? From prasadn2004 at gmail.com Wed Apr 6 19:24:43 2011 From: prasadn2004 at gmail.com (ram seth) Date: Wed, 6 Apr 2011 22:54:43 +0530 Subject: [FFmpeg-user] Can't encode a rawvideo stream - swScaler: Unknown format is not supported as input pixel format In-Reply-To: <1301409015.10597.26.camel@gundam> References: <1301396828.10597.16.camel@gundam> <1301409015.10597.26.camel@gundam> Message-ID: Looks like your cam is giving YUYV (packed 2 byte per pixel) format video.You need to convert it in Yuv420 planer format then encode it . On Tue, Mar 29, 2011 at 8:00 PM, White_Rabbit wrote: > On Tue, 2011-03-29 at 13:07 +0200, White_R wrote: > > Hi all, > > I couldn't find somebody else with my same issue, so I'm writing here. > > I recorded an audio+video .mkv with > > > > ffmpeg -f video4linux2 -s 352x288 -r 15 -b 3000k -i /dev/video0 -f > > alsa -ac 2 -i hw:0 -vcodec rawvideo part0.mkv > > > > Now I'd like to compress the video stream, [...] > > > > I'm running debian squeeze, with debian-multimedia in apt sources.list > > Thank you for your attention, > > bruno > > Hi again, > I managed to extract the raw video with > ffmpeg -i part0.mkv -an -vcodec copy -f rawvideo part1 > I can reproduce correctly this with > mplayer -demuxer rawvideo -rawvideo fps=15:w=320:h=240:yuy2 part1 > > I still cannot encode/compress it; with > ffmpeg -f rawvideo -s 320x240 -r 15 -i part1 -vcodec mpeg4 > part1_mpeg4.mkv > I get something that doesn't play correctly :( > I uploaded both files > http://tracciabi.li/~bruno/part1 > http://tracciabi.li/~bruno/part1_mpeg4.mkv > > Thanks for your attention > bruno > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From leo.izen at gmail.com Wed Apr 6 23:49:11 2011 From: leo.izen at gmail.com (Leo Izen) Date: Wed, 6 Apr 2011 17:49:11 -0400 Subject: [FFmpeg-user] ffmpeg compile errors undefined reference to x264_param_default In-Reply-To: <87fwpveg4v.wl%nomiya@galaxy.dti.ne.jp> References: <87fwpveg4v.wl%nomiya@galaxy.dti.ne.jp> Message-ID: 2011/4/6 Masaru Nomiya > > KG> And here is the error. > > KG> LD ffmpeg_g > KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to > KG> `x264_param_default' > KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to > KG> `x264_encoder_headers' > KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to > KG> `x264_encoder_encode' > KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to > KG> `x264_encoder_reconfig' > KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to > KG> `x264_encoder_delayed_frames' > KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to > KG> `x264_encoder_open_114' > KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to > KG> `x264_picture_init' > KG> /usr/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to > KG> `x264_encoder_close' > KG> collect2: ld returned 1 exit status > KG> make: *** [ffmpeg_g] Error 1 > > I have found this error too: Try reconfiguring with --extra-libs=-lx264. For some reason FFmpeg doesn't automatically add this. From bryandonnovan at gmail.com Thu Apr 7 02:46:14 2011 From: bryandonnovan at gmail.com (bryandonnovan at gmail.com) Date: Wed, 6 Apr 2011 17:46:14 -0700 Subject: [FFmpeg-user] undefined reference to 'ff_find_pix_fmt' Message-ID: building 0.6.9-rc0 for win32 with a configuration for writing FLV files... When libavformat links, it has an unresolved external 'ff_find_pix_fmt' ff_find_pix_fmt is defined in avcodec/raw.h and avcodec/rawdec.c however it does not appear in avcodec/avcodec-52.def So, my main question is -- what change must I make so that it DOES appear in the DEF file? My second question is the error "./configure: line 761: pkg-config: command not found" -- is this a problem? And if so, what can I do about it? Thanks Ouput from configure and link stages below: ./configure: line 761: pkg-config: command not found install prefix /usr/local source path /C/ffmpeg-0.6.90-rc0 C compiler gcc ARCH x86 (generic) big-endian no runtime cpu detection yes yasm yes MMX enabled yes MMX2 enabled yes 3DNow! enabled yes 3DNow! extended enabled yes SSE enabled yes SSSE3 enabled yes CMOV enabled no CMOV is fast no EBX available yes EBP available no 10 operands supported yes debug symbols no strip symbols yes optimizations yes static yes shared yes postprocessing support no new filter support no network support no threading support w32threads SDL support no Sun medialib support no AVISynth enabled no frei0r enabled no libdc1394 support no libdirac enabled no libfaac enabled no libgsm enabled no libmp3lame enabled yes libnut enabled no libopencore-amrnb support no libopencore-amrwb support no libopencv support no libopenjpeg enabled no librtmp enabled no libschroedinger enabled no libspeex enabled no libtheora enabled no libvorbis enabled no libvpx enabled no libx264 enabled no libxavs enabled no libxvid enabled no zlib enabled no bzlib enabled no Enabled decoders: Enabled encoders: flv h263 libmp3lame Enabled hwaccels: Enabled parsers: Enabled demuxers: Enabled muxers: flv mp3 Enabled protocols: file Enabled filters: Enabled bsfs: mp3_header_compress Enabled indevs: Enabled outdevs: License: LGPL version 2.1 or later Creating config.mak and config.h... libavutil/avconfig.h is unchanged Creating library file: libavutil/libavutil.dll.a lib.exe /machine:i386 /def:libavutil/avutil-50.def /out:libavutil/avutil-50.lib Microsoft (R) Library Manager Version 10.00.30319.01 Copyright (C) Microsoft Corporation. All rights reserved. Creating library libavutil/avutil-50.lib and object libavutil/avutil-50.exp LD libavcodec/avcodec-52.dll Creating library file: libavcodec/libavcodec.dll.a lib.exe /machine:i386 /def:libavcodec/avcodec-52.def /out:libavcodec/avcodec-52.lib Microsoft (R) Library Manager Version 10.00.30319.01 Copyright (C) Microsoft Corporation. All rights reserved. Creating library libavcodec/avcodec-52.lib and object libavcodec/avcodec-52.exp LD libavformat/avformat-52.dll Creating library file: libavformat/libavformat.dll.a libavformat/utils.o:utils.c:(.text+0x9293): undefined reference to `ff_find_pix_fmt' collect2: ld returned 1 exit status make: *** [libavformat/avformat-52.dll] Error 1 From andycivil at gmail.com Thu Apr 7 07:47:26 2011 From: andycivil at gmail.com (Andy Civil) Date: Thu, 07 Apr 2011 01:47:26 -0400 Subject: [FFmpeg-user] FFmpeg documentation confusion Message-ID: <4D9D4FEE.1070002@gmail.com> I think the example given in section 15.15 of the documentation is wrong, unless I've failed to understand it. It says: # Add paddings with color "violet" to the input video. Output video # size is 640x480, the top-left corner of the input video is placed at # row 0, column 40. pad=640:480:0:40:violet but in my experience, this pad value puts the image 40 pixels down, not 40 pixels across. I think it should read: # Add paddings with color "violet" to the input video. Output video # size is 640x480, the top-left corner of the input video is placed at # column 0, row 40. pad=640:480:0:40:violet -- Andy From thilo.borgmann at googlemail.com Thu Apr 7 08:55:32 2011 From: thilo.borgmann at googlemail.com (Thilo Borgmann) Date: Thu, 07 Apr 2011 08:55:32 +0200 Subject: [FFmpeg-user] FFmpeg documentation confusion In-Reply-To: <4D9D4FEE.1070002@gmail.com> References: <4D9D4FEE.1070002@gmail.com> Message-ID: <4D9D5FE4.2090501@googlemail.com> Am 07.04.11 07:47, schrieb Andy Civil: > I think the example given in section 15.15 of the documentation is wrong, unless > I've failed to understand it. It says: > > # Add paddings with color "violet" to the input video. Output video > # size is 640x480, the top-left corner of the input video is placed at > # row 0, column 40. > pad=640:480:0:40:violet > > but in my experience, this pad value puts the image 40 pixels down, not 40 > pixels across. I think it should read: > > # Add paddings with color "violet" to the input video. Output video > # size is 640x480, the top-left corner of the input video is placed at > # column 0, row 40. > pad=640:480:0:40:violet Indeed this is wrong - thanks for reporting! Patch attached. -Thilo -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: doc_pad_filter.patch URL: From krueger at signal7.de Thu Apr 7 09:14:28 2011 From: krueger at signal7.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Thu, 7 Apr 2011 09:14:28 +0200 Subject: [FFmpeg-user] What happens to roundup issues? Message-ID: Hi, I'm not sure whether I have missed an announcement on the list but more or less by accident I noticed that there is a new bug tracker which is probably due to the way the two projects split. This is fine but I would like to know whether bug reporters should resubmit their unresolved roundup bugs to trac or can we rely on the maintainers and other devs to check both bug trackers or are you planning on migrating roundup content to trac? Regards, Robert From stefano.sabatini-lala at poste.it Thu Apr 7 09:52:56 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Thu, 7 Apr 2011 09:52:56 +0200 Subject: [FFmpeg-user] FFmpeg documentation confusion In-Reply-To: <4D9D5FE4.2090501@googlemail.com> References: <4D9D4FEE.1070002@gmail.com> <4D9D5FE4.2090501@googlemail.com> Message-ID: <20110407075256.GA4380@geppetto> On date Thursday 2011-04-07 08:55:32 +0200, Thilo Borgmann encoded: [...] > diff --git a/doc/filters.texi b/doc/filters.texi > index 281fe3d..2547ed5 100644 > --- a/doc/filters.texi > +++ b/doc/filters.texi > @@ -984,7 +984,7 @@ For example: > @example > # Add paddings with color "violet" to the input video. Output video > # size is 640x480, the top-left corner of the input video is placed at > -# row 0, column 40. > +# row 40, column 0. column 0, row 40 (more consistent with the parameters order) > pad=640:480:0:40:violet > @end example Feel free to apply it. -- ffmpeg-user random tip #0 Please try to use a reasonably recent version of ffmpeg before asking questions to ffmpeg-user: older versions aren't supported here! From david at electric-spoon.com Thu Apr 7 13:14:17 2011 From: david at electric-spoon.com (David Pottage) Date: Thu, 07 Apr 2011 12:14:17 +0100 Subject: [FFmpeg-user] Live transcoding to an rtp stream is unreliable. Message-ID: <4D9D9C89.7010702@electric-spoon.com> Hello I am attempting to transcode and re-stream a live TV channel using ffmpeg under linux. My command line is: ffmpeg -loglevel verbose \ -pix_fmt yuv422p -s 720x576 -i http://david-laptop:8000/4228 \ -vn -sn -acodec libfaac -ar 48000 -ab 64000 -ac 2 -async 1 -flags +global_header -f rtp rtp://david-laptop:9022 \ -an -sn -vcodec libx264 -s 320x240 -fpre resources/ffpreset/libx264-hq.ffpreset \ -fpre resources/ffpreset/libx264-ipod320.ffpreset -vglobal 1 -f rtp rtp://david-laptop:9020 -newvideo In the command line above, http://david-laptop:8000/4228 is connected to a DVB capture card, and streams a raw DVB transport stream of a live TV channel. The outgoing stream is sent as two seperate elementary streams (audio and video) over rtp. My problem is that the streaming process is very unreliable. ffmpeg will often hang on start-up, (before it emits the SDP file), or it will start-up normally, but then crash after about a second of transcoding. The reliablity appears to depend on the uptime of the machine, and the number of times it has run before. It usually works if it is the first time that ffmpeg is invoked after a reboot, but becomes less reliable after that. I have tried varying the network port numbers of the outgoing streams in case something is not being cleaned up, but it does not make any difference. Has anyone experienced any symptoms like that. Can anyone suggest what might be going on, or give me any clues for further investigation. Thanks. -- David Pottage. From alexandre.ferrieux at orange-ftgroup.com Thu Apr 7 13:39:05 2011 From: alexandre.ferrieux at orange-ftgroup.com (Alexandre Ferrieux) Date: Thu, 07 Apr 2011 13:39:05 +0200 Subject: [FFmpeg-user] Live transcoding to an rtp stream is unreliable. In-Reply-To: <4D9D9C89.7010702@electric-spoon.com> References: <4D9D9C89.7010702@electric-spoon.com> Message-ID: <4D9DA259.1040701@orange-ftgroup.com> On 07/04/2011 13:14, David Pottage wrote: > Hello > > I am attempting to transcode and re-stream a live TV channel using > ffmpeg under linux. My command line is: > > ffmpeg -loglevel verbose \ > -pix_fmt yuv422p -s 720x576 -i http://david-laptop:8000/4228 \ > -vn -sn -acodec libfaac -ar 48000 -ab 64000 -ac 2 -async 1 -flags > +global_header -f rtp rtp://david-laptop:9022 \ > -an -sn -vcodec libx264 -s 320x240 -fpre > resources/ffpreset/libx264-hq.ffpreset \ > -fpre resources/ffpreset/libx264-ipod320.ffpreset -vglobal 1 -f rtp > rtp://david-laptop:9020 -newvideo > > In the command line above, http://david-laptop:8000/4228 is connected to > a DVB capture card, and streams a raw DVB transport stream of a live TV > channel. > > The outgoing stream is sent as two seperate elementary streams (audio > and video) over rtp. > > My problem is that the streaming process is very unreliable. ffmpeg will > often hang on start-up, (before it emits the SDP file), or it will > start-up normally, but then crash after about a second of transcoding. > > The reliablity appears to depend on the uptime of the machine, and the > number of times it has run before. It usually works if it is the first > time that ffmpeg is invoked after a reboot, but becomes less reliable > after that. > > I have tried varying the network port numbers of the outgoing streams in > case something is not being cleaned up, but it does not make any > difference. > > Has anyone experienced any symptoms like that. Can anyone suggest what > might be going on, or give me any clues for further investigation. First, give the ffmpeg output (giving the version, detected container and codec, stream mapping etc.). Second, better characterize "crashes": open the corefile with gdb and give the stack of every thread. Also, I'd try to decorrelate depayloading/decoding/encoding/repayloading, and see which (if any) crashes. To do so, use several ffmpeg processes connected by pipes, and -[av]codec copy (and explicit -f values). -Alex From david at electric-spoon.com Thu Apr 7 14:41:28 2011 From: david at electric-spoon.com (David Pottage) Date: Thu, 07 Apr 2011 13:41:28 +0100 Subject: [FFmpeg-user] Live transcoding to an rtp stream is unreliable. In-Reply-To: <4D9DA259.1040701@orange-ftgroup.com> References: <4D9D9C89.7010702@electric-spoon.com> <4D9DA259.1040701@orange-ftgroup.com> Message-ID: <4D9DB0F8.4020908@electric-spoon.com> On 07/04/11 12:39, Alexandre Ferrieux wrote: > On 07/04/2011 13:14, David Pottage wrote: >> Hello >> >> I am attempting to transcode and re-stream a live TV channel using >> ffmpeg under linux. My command line is: >> >> ffmpeg -loglevel verbose \ >> -pix_fmt yuv422p -s 720x576 -i http://david-laptop:8000/4228 \ >> -vn -sn -acodec libfaac -ar 48000 -ab 64000 -ac 2 -async 1 -flags >> +global_header -f rtp rtp://david-laptop:9022 \ >> -an -sn -vcodec libx264 -s 320x240 -fpre >> resources/ffpreset/libx264-hq.ffpreset \ >> -fpre resources/ffpreset/libx264-ipod320.ffpreset -vglobal 1 -f rtp >> rtp://david-laptop:9020 -newvideo >> >> In the command line above, http://david-laptop:8000/4228 is connected to >> a DVB capture card, and streams a raw DVB transport stream of a live TV >> channel. >> >> The outgoing stream is sent as two seperate elementary streams (audio >> and video) over rtp. >> >> My problem is that the streaming process is very unreliable. ffmpeg will >> often hang on start-up, (before it emits the SDP file), or it will >> start-up normally, but then crash after about a second of transcoding. >> >> The reliablity appears to depend on the uptime of the machine, and the >> number of times it has run before. It usually works if it is the first >> time that ffmpeg is invoked after a reboot, but becomes less reliable >> after that. >> >> I have tried varying the network port numbers of the outgoing streams in >> case something is not being cleaned up, but it does not make any >> difference. >> >> Has anyone experienced any symptoms like that. Can anyone suggest what >> might be going on, or give me any clues for further investigation. > > First, give the ffmpeg output (giving the version, detected container > and codec, stream mapping etc.). Here: > FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers > built on Apr 4 2011 16:31:39 with gcc 4.4.3 > configuration: --enable-postproc --extra-ldflags=-static > --extra-libs='-lvorbis -logg -lxvidcore -lx264 -lopencore-amrnb > -lopencore-amrwb -lfaad -lfaac -lm -lpthread' --enable-gpl > --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libvorbis --enable-libx264 > --enable-libxvid --enable-nonfree --enable-shared --enable-version3 > --enable-libfaad --enable-small --disable-ffplay --disable-ffserver > --disable-doc > libavutil 50.15. 1 / 50.15. 1 > libavcodec 52.72. 2 / 52.72. 2 > libavformat 52.64. 2 / 52.64. 2 > libavdevice 52. 2. 0 / 52. 2. 0 > libswscale 0.11. 0 / 0.11. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [mpeg2video @ 0x9943990]mpeg_decode_postinit() failure > Last message repeated 13 times > [mpegts @ 0x993dae0]max_analyze_duration reached > [mpegts @ 0x993dae0]Estimating duration from bitrate, this may be > inaccurate > Input #0, mpegts, from 'http://10.10.10.10:8000/4228': > Duration: N/A, start: 55166.391433, bitrate: 15256 kb/s > Program 4228 > Stream #0.0[0x262]: Video: mpeg2video, yuv420p, 720x576 [PAR 64:45 > DAR 16:9], 15000 kb/s, 27.63 fps, 25 tbr, 90k tbn, 50 tbc > Stream #0.1[0x263](eng): Audio: mp2, 48000 Hz, 2 channels, s16, > 256 kb/s > Stream #0.2[0x264](eng): Audio: mp3, 0 channels, s16 > Stream #0.3[0x267](eng): Subtitle: dvbsub > Stream #0.4[0x26b]: Data: 0x0005 > Stream #0.5[0x3f0]: Data: 0x000b > Stream #0.6[0x3f1]: Data: 0x000b > Stream #0.7[0x3f2]: Data: 0x000b > Stream #0.8[0x28a]: Data: 0x000b > Stream #0.9[0x28b]: Data: 0x000b > Stream #0.10[0x28c]: Data: 0x000b > [libx264 @ 0x99fe790]using SAR=4/3 > [libx264 @ 0x99fe790]VBV buffer (3000) > level limit (2000) > [libx264 @ 0x99fe790]using cpu capabilities: MMX2 SSE2 SSE3 Cache64 > [libx264 @ 0x99fe790]profile Constrained Baseline, level 1.3 > [libx264 @ 0x99fe790]264 - core 114 r1913 5fd3dce - H.264/MPEG-4 AVC > codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - > options: cabac=0 ref=4 deblock=1:0:0 analyse=0x1:0x111 me=umh subme=8 > psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 > 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 > threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 > constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 > scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 > bitrate=200 ratetol=20.0 qcomp=0.60 qpmin=10 qpmax=51 qpstep=4 > vbv_maxrate=768 vbv_bufsize=3000 nal_hrd=none ip_ratio=1.41 aq=1:1.00 When it is not working, ffmpeg usualy stops at that point and produces no more output until I kill it with ^C Sometimes it gets as far as emmiting the stream mappings & SDP file. I then get: > Output #0, rtp, to 'rtp://david-laptop:8062': > Metadata: > encoder : Lavf52.64.2 > Stream #0.0(eng): Audio: libfaac, 48000 Hz, 2 channels, s16, 64 kb/s > Output #1, rtp, to 'rtp://david-laptop:8060': > Metadata: > encoder : Lavf52.64.2 > Stream #1.0: Video: libx264, yuv420p, 320x240 [PAR 4:3 DAR 16:9], > q=10-51, 200 kb/s, 90k tbn, 25 tbc > Stream mapping: > Stream #0.1 -> #0.0 > Stream #0.0 -> #1.0 > SDP: > v=0 > o=- 0 0 IN IP4 127.0.0.1 > s=No Name > t=0 0 > a=tool:libavformat 52.64.2 > m=audio 8062 RTP/AVP 97 > c=IN IP4 10.2.2.43 > b=AS:64 > a=rtpmap:97 MPEG4-GENERIC/48000/2 > a=fmtp:97 > profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3; > config=1190 > m=video 8060 RTP/AVP 96 > c=IN IP4 10.2.2.43 > b=AS:200 > a=rtpmap:96 H264/90000 > a=fmtp:96 packetization-mode=1; > sprop-parameter-sets=Z0LADdlBQfsOEAAAAwAQAAADAyDxQqWA,aMkjLIA= > > Press [q] to stop encoding > [mpeg2video @ 0xa193990]warning: first frame is no keyframe > [mpeg2video @ 0xa193990]ac-tex damaged at 11 22=4.52 bitrate= > 73.2kbits/s dup=12 drop=0 > [mpeg2video @ 0xa193990]Warning MVs not available > [mpeg2video @ 0xa193990]concealing 630 DC, 630 AC, 630 MV errors > [mp2 @ 0xa1940f0]incomplete frame > Error while decoding stream #0.1 > frame= 161 fps= 88 q=25.0 Lsize= 43kB time=5.48 bitrate= > 64.3kbits/s dup=12 drop=0 > video:70kB audio:43kB global headers:0kB muxing overhead -61.776752% > [libx264 @ 0xa1e8ac0]frame I:2 Avg QP:35.11 size: 1170 > [libx264 @ 0xa1e8ac0]frame P:159 Avg QP:24.71 size: 430 > [libx264 @ 0xa1e8ac0]mb I I16..4: 81.2% 0.0% 18.8% > [libx264 @ 0xa1e8ac0]mb P I16..4: 0.6% 0.0% 1.3% P16..4: 12.2% > 2.3% 1.0% 0.0% 0.0% skip:82.5% > [libx264 @ 0xa1e8ac0]final ratefactor: 19.82 > [libx264 @ 0xa1e8ac0]coded y,uvDC,uvAC intra: 43.6% 51.2% 36.3% inter: > 3.6% 8.9% 1.7% > [libx264 @ 0xa1e8ac0]i16 v,h,dc,p: 55% 26% 3% 16% > [libx264 @ 0xa1e8ac0]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 21% 15% 6% > 7% 9% 8% 7% 6% > [libx264 @ 0xa1e8ac0]i8c dc,h,v,p: 60% 23% 13% 4% > [libx264 @ 0xa1e8ac0]ref P L0: 79.4% 11.9% 6.7% 2.0% > [libx264 @ 0xa1e8ac0]kb/s:87.93 It only does transcoding for about a second before it stop working, and quits. > Second, better characterize "crashes": open the corefile with gdb and > give the stack of every thread. I should have been more clear. It is not crashing in the sense of a seg fault and core dump. It is just stopping unexpectedly. There is no core dump, it is just not producing anything useful. > Also, I'd try to decorrelate > depayloading/decoding/encoding/repayloading, and see which (if any) > crashes. To do so, use several ffmpeg processes connected by pipes, > and -[av]codec copy (and explicit -f values). OK, I will try that. What container format do you recommend for the pipes? In the past I have tried yuv4mpegpipe, but found it unstable as it only works for video, not audio, so I used matroska. Is that a good choice? Is there another container format that is good for raw streams. -- David Pottage From alexandre.ferrieux at orange-ftgroup.com Thu Apr 7 15:05:18 2011 From: alexandre.ferrieux at orange-ftgroup.com (Alexandre Ferrieux) Date: Thu, 07 Apr 2011 15:05:18 +0200 Subject: [FFmpeg-user] Live transcoding to an rtp stream is unreliable. In-Reply-To: <4D9DB0F8.4020908@electric-spoon.com> References: <4D9D9C89.7010702@electric-spoon.com> <4D9DA259.1040701@orange-ftgroup.com> <4D9DB0F8.4020908@electric-spoon.com> Message-ID: <4D9DB68E.5010008@orange-ftgroup.com> On 07/04/2011 14:41, David Pottage wrote: > On 07/04/11 12:39, Alexandre Ferrieux wrote: >> >> First, give the ffmpeg output (giving the version, detected container >> and codec, stream mapping etc.). > > Here: > >> FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers >> built on Apr 4 2011 16:31:39 with gcc 4.4.3 >> configuration: --enable-postproc --extra-ldflags=-static >> --extra-libs='-lvorbis -logg -lxvidcore -lx264 -lopencore-amrnb >> -lopencore-amrwb -lfaad -lfaac -lm -lpthread' --enable-gpl >> --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb >> --enable-libopencore-amrwb --enable-libvorbis --enable-libx264 >> --enable-libxvid --enable-nonfree --enable-shared --enable-version3 >> --enable-libfaad --enable-small --disable-ffplay --disable-ffserver >> --disable-doc >> libavutil 50.15. 1 / 50.15. 1 >> libavcodec 52.72. 2 / 52.72. 2 >> libavformat 52.64. 2 / 52.64. 2 >> libavdevice 52. 2. 0 / 52. 2. 0 >> libswscale 0.11. 0 / 0.11. 0 >> libpostproc 51. 2. 0 / 51. 2. 0 Thanks. >> [mpeg2video @ 0x9943990]mpeg_decode_postinit() failure >> Last message repeated 13 times >> [mpegts @ 0x993dae0]max_analyze_duration reached >> [mpegts @ 0x993dae0]Estimating duration from bitrate, this may be >> inaccurate >> Input #0, mpegts, from 'http://10.10.10.10:8000/4228': >> Duration: N/A, start: 55166.391433, bitrate: 15256 kb/s >> Program 4228 >> Stream #0.0[0x262]: Video: mpeg2video, yuv420p, 720x576 [PAR 64:45 DAR >> 16:9], 15000 kb/s, 27.63 fps, 25 tbr, 90k tbn, 50 tbc >> Stream #0.1[0x263](eng): Audio: mp2, 48000 Hz, 2 channels, s16, 256 kb/s >> Stream #0.2[0x264](eng): Audio: mp3, 0 channels, s16 >> Stream #0.3[0x267](eng): Subtitle: dvbsub >> Stream #0.4[0x26b]: Data: 0x0005 >> Stream #0.5[0x3f0]: Data: 0x000b >> Stream #0.6[0x3f1]: Data: 0x000b >> Stream #0.7[0x3f2]: Data: 0x000b >> Stream #0.8[0x28a]: Data: 0x000b >> Stream #0.9[0x28b]: Data: 0x000b >> Stream #0.10[0x28c]: Data: 0x000b >> [libx264 @ 0x99fe790]using SAR=4/3 >> [libx264 @ 0x99fe790]VBV buffer (3000) > level limit (2000) >> [libx264 @ 0x99fe790]using cpu capabilities: MMX2 SSE2 SSE3 Cache64 >> [libx264 @ 0x99fe790]profile Constrained Baseline, level 1.3 >> [libx264 @ 0x99fe790]264 - core 114 r1913 5fd3dce - H.264/MPEG-4 AVC >> codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - >> options: cabac=0 ref=4 deblock=1:0:0 analyse=0x1:0x111 me=umh subme=8 >> psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 >> 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 >> threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 >> constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 >> scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 >> bitrate=200 ratetol=20.0 qcomp=0.60 qpmin=10 qpmax=51 qpstep=4 >> vbv_maxrate=768 vbv_bufsize=3000 nal_hrd=none ip_ratio=1.41 aq=1:1.00 > > When it is not working, ffmpeg usualy stops at that point and produces > no more output until I kill it with ^C And no error is printed ? Maybe ffmpeg is not proceeding because it is still expecting a GOP start (a key frame). Do you have an idea of the gop size in your source ? Is it stable ? Also, clearly the reproducibility of a live stream is poor. I'd suggest saving the TS stream to a file for post mortem analysis. > Sometimes it gets as far as emmiting the stream mappings & SDP file. I > then get: > Look at these lines. Aren't they ominous ? >> [mpeg2video @ 0xa193990]ac-tex damaged at 11 22=4.52 bitrate= >> 73.2kbits/s dup=12 drop=0 >> [mp2 @ 0xa1940f0]incomplete frame >> Error while decoding stream #0.1 Clearly this indicates a problem in the input bitstream. Maybe some network latency stalled the TCP (on the http leg) a bit, producing an oveerun at the source, resulting in TS discontinuity ? >> Also, I'd try to decorrelate >> depayloading/decoding/encoding/repayloading, and see which (if any) >> crashes. To do so, use several ffmpeg processes connected by pipes, >> and -[av]codec copy (and explicit -f values). > > OK, I will try that. What container format do you recommend for the > pipes? In the past I have tried yuv4mpegpipe, but found it unstable as > it only works for video, not audio, so I used matroska. Is that a good > choice? Is there another container format that is good for raw streams. Given the above observations, I'd concentrate on the input decoding step, and on the video only. For this, use vocdec copy and the container format that is the simplest for the given codec, here mpeg2video: ffmpeg -loglevel verbose \ -pix_fmt yuv422p -s 720x576 -i http://david-laptop:8000/4228 \ \ -an -sn -vcodec copy -f mpeg2video - > out.m2v With this, you're guaranteed that ffmpeg will not be the limiting factor (unlike a possibly costly x264), since basically it just has to write to disk (you can even use /dev/null instead of out.m2v, just in case). If this setup reproduces the behavior, you have a source overrun or corruption, either due to the network or something inside the source machine itself (CPU bound ?). If not, then ffmpeg's slowness was the reason for the overrun (not swallowing input fast enough), most likely due to the x264 CPU demand; in that case, try with a more modest preset, resampling to smaller size, etc. -Alex From Cecil at decebal.nl Thu Apr 7 15:18:55 2011 From: Cecil at decebal.nl (Cecil Westerhof) Date: Thu, 07 Apr 2011 15:18:55 +0200 Subject: [FFmpeg-user] Adding text to a video Message-ID: <878vvmtd1c.fsf@Compaq.site> I want to make some video's where I explain things. Important things I like to accentuate by integrating text about what is said into the video on the moment it is said. I could not find anything about this. Is this possible? -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof From thilo.borgmann at googlemail.com Thu Apr 7 16:48:04 2011 From: thilo.borgmann at googlemail.com (Thilo Borgmann) Date: Thu, 07 Apr 2011 16:48:04 +0200 Subject: [FFmpeg-user] FFmpeg documentation confusion In-Reply-To: <20110407075256.GA4380@geppetto> References: <4D9D4FEE.1070002@gmail.com> <4D9D5FE4.2090501@googlemail.com> <20110407075256.GA4380@geppetto> Message-ID: <4D9DCEA4.3070900@googlemail.com> Am 07.04.11 09:52, schrieb Stefano Sabatini: > On date Thursday 2011-04-07 08:55:32 +0200, Thilo Borgmann encoded: > [...] >> diff --git a/doc/filters.texi b/doc/filters.texi >> index 281fe3d..2547ed5 100644 >> --- a/doc/filters.texi >> +++ b/doc/filters.texi >> @@ -984,7 +984,7 @@ For example: >> @example >> # Add paddings with color "violet" to the input video. Output video >> # size is 640x480, the top-left corner of the input video is placed at >> -# row 0, column 40. > >> +# row 40, column 0. > > column 0, row 40 (more consistent with the parameters order) My ears are more used to hear it in row-column order. >> pad=640:480:0:40:violet >> @end example > > Feel free to apply it. I don't want to mess with GIT yet, Please apply whaterver variant you like better. Thanks, Thilo From thilo.borgmann at googlemail.com Thu Apr 7 16:57:04 2011 From: thilo.borgmann at googlemail.com (Thilo Borgmann) Date: Thu, 07 Apr 2011 16:57:04 +0200 Subject: [FFmpeg-user] FFmpeg documentation confusion In-Reply-To: <20110407075256.GA4380@geppetto> References: <4D9D4FEE.1070002@gmail.com> <4D9D5FE4.2090501@googlemail.com> <20110407075256.GA4380@geppetto> Message-ID: <4D9DD0C0.2090503@googlemail.com> Am 07.04.11 09:52, schrieb Stefano Sabatini: > On date Thursday 2011-04-07 08:55:32 +0200, Thilo Borgmann encoded: > [...] >> diff --git a/doc/filters.texi b/doc/filters.texi >> index 281fe3d..2547ed5 100644 >> --- a/doc/filters.texi >> +++ b/doc/filters.texi >> @@ -984,7 +984,7 @@ For example: >> @example >> # Add paddings with color "violet" to the input video. Output video >> # size is 640x480, the top-left corner of the input video is placed at >> -# row 0, column 40. > >> +# row 40, column 0. > > column 0, row 40 (more consistent with the parameters order) > >> pad=640:480:0:40:violet >> @end example > > Feel free to apply it. Convenient patch attached. -Thilo -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: doc_pad_filter2.patch URL: From fschwart at ball.com Wed Apr 6 17:30:46 2011 From: fschwart at ball.com (Schwartz, Frank) Date: Wed, 6 Apr 2011 11:30:46 -0400 Subject: [FFmpeg-user] CLI User Documentation Message-ID: <873043AF54FE794B912B616ABA7E9FE508E7E3EE@daytonmsg2k3.AERO.BALL.COM> Hello, I am trying to use FFmpeg and have no prior experience processing video files. To that end, I went to the FFMpeg.org site --> Documentation tab --> Command Line Interface (CLI) and Related User Documentation section --> FFmpeg Documentation link. This document apparently presumes a great deal of background knowledge and used many terms w/o defining them. I have been Google-ing a lot and not getting very far. Can anyone recommend a Glossary or "Video Processing for Dummies"-type site explaining the fundamental concepts and terms? I appreciate the help. Frank Schwartz BATC 937-320-7086 This message and any enclosures are intended only for the addressee. Please notify the sender by email if you are not the intended recipient. If you are not the intended recipient, you may not use, copy, disclose, or distribute this message or its contents or enclosures to any other person and any such actions may be unlawful. Ball reserves the right to monitor and review all messages and enclosures sent to or from this email address. From maxpoulin64 at gmail.com Thu Apr 7 03:13:52 2011 From: maxpoulin64 at gmail.com (Maxime Poulin) Date: Wed, 06 Apr 2011 21:13:52 -0400 Subject: [FFmpeg-user] SIGSEGV when streaming to FFserver Message-ID: <4D9D0FD0.4010605@gmail.com> (This is my first message to any of the ffmpeg's mailing lists, please correct me if I'm doing wrong!) Hi, I compiled FFmpeg a few times this week, trying to get it stream to FFserver, without any success. FFmpeg crashes as soon as it starts sending data to the server (which always receive 4149 bytes of data before the client crashes). I am using the latest Git version as of now (git-N-28846-g02fd687) compiled using these instructions: http://ubuntuforums.org/showthread.php?t=786095 This configuration worked before, but I changed my server and now my computer too, so I had to recompile FFmpeg on both of them. The server looks to work fine, and the local FFmpeg too. I can encode anything to anything, it works. But as soon as I try to send the data to FFserver, either on my local PC or on my server, I get a Segmentation Fault. Here is the command I launch, which worked without problems before: $ ffmpeg -f x11grab -r 60 -s 1920x1080 -i :0 -f alsa -ac 2 -i pulse http://server:8090/video0.ffm As I said, I tried to update it and compile it a few times, but I didn't have any success, so I'm posting here next to the advice of someone on the IRC channel. I attached a gdb log (223 lines), as indicated on the bug report page of the site. Thanks in advance! -------------- next part -------------- A non-text attachment was scrubbed... Name: ffmpeg-gdb.log Type: text/x-log Size: 13734 bytes Desc: not available URL: From stefano.sabatini-lala at poste.it Thu Apr 7 17:49:46 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Thu, 7 Apr 2011 17:49:46 +0200 Subject: [FFmpeg-user] FFmpeg documentation confusion In-Reply-To: <4D9DD0C0.2090503@googlemail.com> References: <4D9D4FEE.1070002@gmail.com> <4D9D5FE4.2090501@googlemail.com> <20110407075256.GA4380@geppetto> <4D9DD0C0.2090503@googlemail.com> Message-ID: <20110407154946.GA7054@geppetto> On date Thursday 2011-04-07 16:57:04 +0200, Thilo Borgmann encoded: > Am 07.04.11 09:52, schrieb Stefano Sabatini: > > On date Thursday 2011-04-07 08:55:32 +0200, Thilo Borgmann encoded: > > [...] > >> diff --git a/doc/filters.texi b/doc/filters.texi > >> index 281fe3d..2547ed5 100644 > >> --- a/doc/filters.texi > >> +++ b/doc/filters.texi > >> @@ -984,7 +984,7 @@ For example: > >> @example > >> # Add paddings with color "violet" to the input video. Output video > >> # size is 640x480, the top-left corner of the input video is placed at > >> -# row 0, column 40. > > > >> +# row 40, column 0. > > > > column 0, row 40 (more consistent with the parameters order) > > > >> pad=640:480:0:40:violet > >> @end example > > > > Feel free to apply it. > > Convenient patch attached. > > -Thilo > diff --git a/doc/filters.texi b/doc/filters.texi > index 281fe3d..32a984d 100644 > --- a/doc/filters.texi > +++ b/doc/filters.texi > @@ -984,7 +984,7 @@ For example: > @example > # Add paddings with color "violet" to the input video. Output video > # size is 640x480, the top-left corner of the input video is placed at > -# row 0, column 40. > +# column 0, row 4. > pad=640:480:0:40:violet > @end example Applied. -- ffmpeg-user random tip #19 X11 session recording with ffmpeg: ffmpeg -f oss -i /dev/audio -f x11grab -s $WIDTHx$HEIGHT -r 5 -i :0.0 \ x11-session.avi Use xdpyinfo to get WIDTH and HEIGHT values. From dave at avpreserve.com Thu Apr 7 17:58:41 2011 From: dave at avpreserve.com (Dave Rice) Date: Thu, 7 Apr 2011 11:58:41 -0400 Subject: [FFmpeg-user] Adding text to a video In-Reply-To: <878vvmtd1c.fsf@Compaq.site> References: <878vvmtd1c.fsf@Compaq.site> Message-ID: On Apr 7, 2011, at 9:18 AM, Cecil Westerhof wrote: > I want to make some video's where I explain things. Important things I > like to accentuate by integrating text about what is said into the > video on the moment it is said. I could not find anything about this. > Is this possible? I suppose the drawtext filter could be used for this, but do you want the text encoded into the video? Depending on how you're presenting it, it may be more efficient and a better UX to use something like popcorn.js. Dave Rice avpreserve.com > -- > Cecil Westerhof > Senior Software Engineer > LinkedIn: http://www.linkedin.com/in/cecilwesterhof > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From thilo.borgmann at googlemail.com Thu Apr 7 18:28:11 2011 From: thilo.borgmann at googlemail.com (Thilo Borgmann) Date: Thu, 07 Apr 2011 18:28:11 +0200 Subject: [FFmpeg-user] FFmpeg documentation confusion In-Reply-To: <20110407154946.GA7054@geppetto> References: <4D9D4FEE.1070002@gmail.com> <4D9D5FE4.2090501@googlemail.com> <20110407075256.GA4380@geppetto> <4D9DD0C0.2090503@googlemail.com> <20110407154946.GA7054@geppetto> Message-ID: <4D9DE61B.2090104@googlemail.com> Am 07.04.11 17:49, schrieb Stefano Sabatini: > On date Thursday 2011-04-07 16:57:04 +0200, Thilo Borgmann encoded: >> Am 07.04.11 09:52, schrieb Stefano Sabatini: >>> On date Thursday 2011-04-07 08:55:32 +0200, Thilo Borgmann encoded: >>> [...] >>>> diff --git a/doc/filters.texi b/doc/filters.texi >>>> index 281fe3d..2547ed5 100644 >>>> --- a/doc/filters.texi >>>> +++ b/doc/filters.texi >>>> @@ -984,7 +984,7 @@ For example: >>>> @example >>>> # Add paddings with color "violet" to the input video. Output video >>>> # size is 640x480, the top-left corner of the input video is placed at >>>> -# row 0, column 40. >>> >>>> +# row 40, column 0. >>> >>> column 0, row 40 (more consistent with the parameters order) >>> >>>> pad=640:480:0:40:violet >>>> @end example >>> >>> Feel free to apply it. >> >> Convenient patch attached. >> >> -Thilo > >> diff --git a/doc/filters.texi b/doc/filters.texi >> index 281fe3d..32a984d 100644 >> --- a/doc/filters.texi >> +++ b/doc/filters.texi >> @@ -984,7 +984,7 @@ For example: >> @example >> # Add paddings with color "violet" to the input video. Output video >> # size is 640x480, the top-left corner of the input video is placed at >> -# row 0, column 40. >> +# column 0, row 4. ^^ should be "40". Sorry, my fault! >> pad=640:480:0:40:violet >> @end example > > Applied. -Thilo From av at bsbc.nb.ca Thu Apr 7 18:29:35 2011 From: av at bsbc.nb.ca (Anthony Brown) Date: Thu, 07 Apr 2011 13:29:35 -0300 Subject: [FFmpeg-user] HD-SDI Message-ID: <4D9DE66F.4040707@bsbc.nb.ca> Hi: Does anyone here successfully use an HD-SDI card with ffmpeg in linux? If so, which one? FYI: I'm currently capturing HDV using firewire on a windows box that is close to my HD camera, piping via http using VLC to a somewhat remote linux box running ffmpeg/ffserver and streaming from there. I'd like to remove the windows box from the equation which I could probably do with a firewire extender and a firewire card in my linux box. OR I could do it the correct way and use HD-SDI instead. Suggestions? -- Anthony Brown Audiovisual coordinator Brunswick Street Baptist Church Telephone: (506)-458-8348 (leave message) Email: av at bsbc.nb.ca -------------- next part -------------- A non-text attachment was scrubbed... Name: av.vcf Type: text/x-vcard Size: 163 bytes Desc: not available URL: From thilo.borgmann at googlemail.com Thu Apr 7 18:29:47 2011 From: thilo.borgmann at googlemail.com (Thilo Borgmann) Date: Thu, 07 Apr 2011 18:29:47 +0200 Subject: [FFmpeg-user] FFmpeg documentation confusion In-Reply-To: <4D9DE61B.2090104@googlemail.com> References: <4D9D4FEE.1070002@gmail.com> <4D9D5FE4.2090501@googlemail.com> <20110407075256.GA4380@geppetto> <4D9DD0C0.2090503@googlemail.com> <20110407154946.GA7054@geppetto> <4D9DE61B.2090104@googlemail.com> Message-ID: <4D9DE67B.7080704@googlemail.com> Am 07.04.11 18:28, schrieb Thilo Borgmann: > Am 07.04.11 17:49, schrieb Stefano Sabatini: >> On date Thursday 2011-04-07 16:57:04 +0200, Thilo Borgmann encoded: >>> Am 07.04.11 09:52, schrieb Stefano Sabatini: >>>> On date Thursday 2011-04-07 08:55:32 +0200, Thilo Borgmann encoded: >>>> [...] >>>>> diff --git a/doc/filters.texi b/doc/filters.texi >>>>> index 281fe3d..2547ed5 100644 >>>>> --- a/doc/filters.texi >>>>> +++ b/doc/filters.texi >>>>> @@ -984,7 +984,7 @@ For example: >>>>> @example >>>>> # Add paddings with color "violet" to the input video. Output video >>>>> # size is 640x480, the top-left corner of the input video is placed at >>>>> -# row 0, column 40. >>>> >>>>> +# row 40, column 0. >>>> >>>> column 0, row 40 (more consistent with the parameters order) >>>> >>>>> pad=640:480:0:40:violet >>>>> @end example >>>> >>>> Feel free to apply it. >>> >>> Convenient patch attached. >>> >>> -Thilo >> >>> diff --git a/doc/filters.texi b/doc/filters.texi >>> index 281fe3d..32a984d 100644 >>> --- a/doc/filters.texi >>> +++ b/doc/filters.texi >>> @@ -984,7 +984,7 @@ For example: >>> @example >>> # Add paddings with color "violet" to the input video. Output video >>> # size is 640x480, the top-left corner of the input video is placed at >>> -# row 0, column 40. >>> +# column 0, row 4. > ^^ > should be "40". Sorry, my fault! Oh and I'm going to check with cvslog before declaring my fault next time ;) -Thilo From renambot at gmail.com Thu Apr 7 19:52:06 2011 From: renambot at gmail.com (luc Renambot) Date: Thu, 7 Apr 2011 12:52:06 -0500 Subject: [FFmpeg-user] HD-SDI In-Reply-To: <4D9DE66F.4040707@bsbc.nb.ca> References: <4D9DE66F.4040707@bsbc.nb.ca> Message-ID: <215F43EE-8555-470C-9F2E-C1CE17A282E6@gmail.com> On Apr 7, 2011, at 11:29 AM, Anthony Brown wrote: > Hi: > > Does anyone here successfully use an HD-SDI card with ffmpeg in linux? If so, which one? > > FYI: I'm currently capturing HDV using firewire on a windows box that is close to my HD camera, piping via http using VLC to a somewhat remote linux box running ffmpeg/ffserver and streaming from there. I'd like to remove the windows box from the equation which I could probably do with a firewire extender and a firewire card in my linux box. OR I could do it the correct way and use HD-SDI instead. > > Suggestions? > > -- > Blackmagic cards have good linux support. Luc From bahamutzero8825 at gmail.com Thu Apr 7 20:37:56 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Thu, 07 Apr 2011 13:37:56 -0500 Subject: [FFmpeg-user] FFmpeg crashes when AC3 audio changes from 5.1ch to 2ch with -f sox Message-ID: <4D9E0484.8030300@gmail.com> I'm trying to convert an AC3 audio stream to AAC and increase the volume along the way. Unfortunately, some bits of the stream are stereo, and when it changes, FFmpeg crashes. This command line: > ffmpeg.exe -i input.ts -f sox - 2>nul | sox.exe -t sox - -t wav - vol > 1.5 2>nul > nul gives this before it crashes: > Seems stream 0 codec frame rate differs from container frame rate: > 59.94 (60000/1001) -> 59.94 (60000/1001) > Input #0, mpegts, from 'p:/input.ts': > Duration: 01:29:55.25, start: 0.166644, bitrate: 14187 kb/s > Program 1 > Stream #0.0[0x1011]: Video: h264 (Main), yuv420p, 1920x1080 [PAR > 1:1 DAR 16:9], 59.96 fps, 59.94 tbr, 90k tbn, 59.94 tbc > Stream #0.1[0x1100](eng): Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s > Incompatible sample format 's16' for codec 'pcm_s32le', auto-selecting > format 's32' > Output #0, sox, to 'pipe:': > Metadata: > encoder : Lavf52.105.0 > Stream #0.0(eng): Audio: pcm_s32le, 48000 Hz, 5.1, s32, 9216 kb/s > Stream mapping: > Stream #0.1 -> #0.0 > Press [q] to stop encoding > Input stream #0.1 frame changed from rate:48000 fmt:s16 ch:6 to > rate:48000 fmt:s16 ch:2 It gets up to about 1241 seconds before it crashes. I've tried adding -ac 6, but that doesn't change anything. I thought that would upmix the audio to 6ch, but I guess it doesn't. It doesn't crash if I change to -f wav, but the audio channels are always mapped wrong if piped into SoX that way, and I really need to increase the volume. From jose.rola at gmail.com Thu Apr 7 21:11:07 2011 From: jose.rola at gmail.com (Jose Rola) Date: Thu, 7 Apr 2011 20:11:07 +0100 Subject: [FFmpeg-user] HD-SDI In-Reply-To: <215F43EE-8555-470C-9F2E-C1CE17A282E6@gmail.com> References: <4D9DE66F.4040707@bsbc.nb.ca> <215F43EE-8555-470C-9F2E-C1CE17A282E6@gmail.com> Message-ID: Hello i agree that black magic have a good unix support and have a good quality/price relation, but i do not believe you will gain anything by using HD-SDI instead of a HDMI extender. I would recomend going HD-SDI if your camera had a HD-SDI output. Regards On Thu, Apr 7, 2011 at 6:52 PM, luc Renambot wrote: > > On Apr 7, 2011, at 11:29 AM, Anthony Brown wrote: > >> Hi: >> >> Does anyone here successfully use an HD-SDI card with ffmpeg in linux? If so, which one? >> >> FYI: I'm currently capturing HDV using firewire on a windows box that is close to my HD camera, piping via http using VLC to a somewhat remote linux box running ffmpeg/ffserver and streaming from there. ?I'd like to remove the windows box from the equation which I could probably do with a firewire extender and a firewire card in my linux box. ?OR I could do it the correct way and use HD-SDI instead. >> >> Suggestions? >> >> -- >> > > Blackmagic cards have good linux support. > > Luc > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From bahamutzero8825 at gmail.com Thu Apr 7 21:24:45 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Thu, 07 Apr 2011 14:24:45 -0500 Subject: [FFmpeg-user] FFmpeg crashes when AC3 audio changes from 5.1ch to 2ch with -f sox In-Reply-To: <4D9E0484.8030300@gmail.com> References: <4D9E0484.8030300@gmail.com> Message-ID: <4D9E0F7D.5060002@gmail.com> Forgot to add that I'm using a very recent git build. > FFmpeg version git-N-28849-ga82cfad-Sherpya From jeff_eckermann at yahoo.com Fri Apr 8 04:27:28 2011 From: jeff_eckermann at yahoo.com (Jeff Eckermann) Date: Thu, 7 Apr 2011 19:27:28 -0700 (PDT) Subject: [FFmpeg-user] CLI User Documentation In-Reply-To: <873043AF54FE794B912B616ABA7E9FE508E7E3EE@daytonmsg2k3.AERO.BALL.COM> References: <873043AF54FE794B912B616ABA7E9FE508E7E3EE@daytonmsg2k3.AERO.BALL.COM> Message-ID: <736963.98642.qm@web33803.mail.mud.yahoo.com> >??? I am trying to use FFmpeg and have no prior experience >processing video files. To that end, I went to the FFMpeg.org site --> >Documentation tab --> Command Line Interface (CLI) and Related User >Documentation section --> FFmpeg Documentation link. This document >apparently presumes a great deal of background knowledge and used many >terms w/o defining them. I have been Google-ing a lot and not getting >very far. Can anyone recommend a Glossary or "Video Processing for >Dummies"-type site explaining the fundamental concepts and terms? >??? I appreciate the help. >Frank Schwartz >BATC >937-320-7086 It would help if you could tell us what you are trying to do. As a general introduction: best to start small and work up.? FFmpeg supports lots of features that most people will rarely if ever use.? A minimal command line is: ffmpeg -i inputfile outputfile FFmpeg will usually do a good job of working out what "inputfile" actually is, and what "outputfile" is intended to be.? If transcoding is required, default values will be used for the optional parameters.? These defaults may be fine for you, or they might not e.g. the default video bitrate is 200kbs, which will give poor quality and is really only suitable for internet streaming.? You can use the "-b" flag to specify the video bitrate, and "-ab" to specify the audio bitrate. You can specify the audio and video format by using "-acodec" and "-vcodec" options, for example: ffmpeg -i dvd_rip.mpg -vcodec mpeg4 -acodec libmp3lame -b 1500k -ab 320k encoded_dvd.mp4 (this assumes that your ffmpeg is compiled with libmp3lame, which it probably is). Try googling "ffmpeg"; you will find plenty of usage examples. From paul at teknique.com Fri Apr 8 04:49:26 2011 From: paul at teknique.com (Paul Wang) Date: Fri, 8 Apr 2011 14:49:26 +1200 Subject: [FFmpeg-user] ref: mov file generated from ffmpeg lib lost audio by players other than VLC Message-ID: <322f448f9e8e095ed5bc3555a3536545@mail.gmail.com> Hi, all, I have a problem using ffmpeg lib APIs. I am using these APIs to generate a MOV container file from a H264 video stream and a AAC audio stream. The audio stream can be save as .aac file, which can be played by QuickTime or MediaPlayer. My problem is after the H264 video and AAC audio stream are saved in a .mov container, the file played by media players other than VLC has no audio. Played by VLC is good and having both video and audio, played by other players, video OK, no audio. No ideas what?s wrong??? This is my first time using ffmpeg, can someone give me a hand?? Thanks. Paul Wang. From mimeini at gmail.com Fri Apr 8 10:04:07 2011 From: mimeini at gmail.com (mikkel meinike) Date: Fri, 8 Apr 2011 10:04:07 +0200 Subject: [FFmpeg-user] CLI User Documentation In-Reply-To: <736963.98642.qm@web33803.mail.mud.yahoo.com> References: <873043AF54FE794B912B616ABA7E9FE508E7E3EE@daytonmsg2k3.AERO.BALL.COM> <736963.98642.qm@web33803.mail.mud.yahoo.com> Message-ID: I'd say go for the faq. http://ffmpeg.org/faq.html even though I have been doing Linux for 12 years now I still don't get the man pages. But the faq is build up on practically explained solution. That works for me :-) Mikkel On Fri, Apr 8, 2011 at 4:27 AM, Jeff Eckermann wrote: >>??? I am trying to use FFmpeg and have no prior experience >>processing video files. To that end, I went to the FFMpeg.org site --> >>Documentation tab --> Command Line Interface (CLI) and Related User >>Documentation section --> FFmpeg Documentation link. This document >>apparently presumes a great deal of background knowledge and used many >>terms w/o defining them. I have been Google-ing a lot and not getting >>very far. Can anyone recommend a Glossary or "Video Processing for >>Dummies"-type site explaining the fundamental concepts and terms? > >>??? I appreciate the help. > >>Frank Schwartz >>BATC >>937-320-7086 > > It would help if you could tell us what you are trying to do. > > As a general introduction: best to start small and work up.? FFmpeg supports > lots of features that most people will rarely if ever use.? A minimal command > line is: > ffmpeg -i inputfile outputfile > > FFmpeg will usually do a good job of working out what "inputfile" actually is, > and what "outputfile" is intended to be.? If transcoding is required, default > values will be used for the optional parameters.? These defaults may be fine for > you, or they might not e.g. the default video bitrate is 200kbs, which will give > poor quality and is really only suitable for internet streaming.? You can use > the "-b" flag to specify the video bitrate, and "-ab" to specify the audio > bitrate. > > You can specify the audio and video format by using "-acodec" and "-vcodec" > options, for example: > > ffmpeg -i dvd_rip.mpg -vcodec mpeg4 -acodec libmp3lame -b 1500k -ab 320k > encoded_dvd.mp4 > > (this assumes that your ffmpeg is compiled with libmp3lame, which it probably > is). > > Try googling "ffmpeg"; you will find plenty of usage examples. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ankuj2004 at gmail.com Fri Apr 8 10:15:01 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Fri, 8 Apr 2011 13:45:01 +0530 Subject: [FFmpeg-user] 4CIF H263 video fails to decode Message-ID: Hi, I am trying to decode a H263 4CIF(704x576) resolution file with ffmpeg, but it is detected as mp3 file and fails to get decoded. I tried with lower resolution H263 files they get decoded easily as well as 16CIF. I am using ffmpeg version 0.6.1 Ankuj Gupta From alabandit at gmail.com Fri Apr 8 12:29:05 2011 From: alabandit at gmail.com (Alastair Mason) Date: Fri, 8 Apr 2011 12:29:05 +0200 Subject: [FFmpeg-user] Building ffmpeg and libx264 on ubuntu Message-ID: Hi I'm trying to build ffmpeg with libx264 on ubuntu 10.04. I have followed this guide http://ubuntuforums.org/showthread.php?t=786095 but i get the following message on both the beta and the latest version of ffmpeg: CC libavutil/x86/cpu.o AR libavutil/libavutil.a CC ffmpeg.o ffmpeg.c: In function ?term_exit?: ffmpeg.c:440: warning: zero-length gnu_printf format string ffmpeg.c: In function ?output_packet?: ffmpeg.c:1057: warning: dereferencing pointer ?picture2? does break strict-aliasing rules ffmpeg.c:1623: note: initialized from here CC cmdutils.o cmdutils.c: In function ?read_file?: cmdutils.c:736: warning: ignoring return value of ?fread?, declared with attribute warn_unused_result LD ffmpeg_g /home/alastair/progs/ffmpeg/ libavcodec/libavcodec.a(libx264.o): In function `X264_frame': /home/alastair/progs/ffmpeg/libavcodec/libx264.c:92: undefined reference to `x264_picture_init' /home/alastair/progs/ffmpeg/libavcodec/libx264.c:121: undefined reference to `x264_encoder_delayed_frames' /home/alastair/progs/ffmpeg/libavcodec/libavcodec.a(libx264.o): In function `X264_init': /home/alastair/progs/ffmpeg/libavcodec/libx264.c:308: undefined reference to `x264_encoder_open_114' collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1 i have used this guide a number of times before successfully. Googled but no luck on an answer. the .configure: ./configure --enable-gpl --enable-version3 --enable-nonfree --enable-postproc \ --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb \ --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis \ --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab Thanks for your help in advance Alastair From james.darnley at gmail.com Fri Apr 8 12:38:55 2011 From: james.darnley at gmail.com (James Darnley) Date: Fri, 8 Apr 2011 12:38:55 +0200 Subject: [FFmpeg-user] Building ffmpeg and libx264 on ubuntu In-Reply-To: References: Message-ID: On 08/04/2011, Alastair Mason wrote: > /home/alastair/progs/ffmpeg/libavcodec/libx264.c:308: undefined > reference to `x264_encoder_open_114' Your x264 header version does not match the library version. Try the guide again, especially part 1. From alabandit at gmail.com Fri Apr 8 15:05:26 2011 From: alabandit at gmail.com (Alastair Mason) Date: Fri, 8 Apr 2011 15:05:26 +0200 Subject: [FFmpeg-user] Building ffmpeg and libx264 on ubuntu In-Reply-To: References: Message-ID: > Your x264 header version does not match the library version. ?Try the > guide again, especially part 1. I have tried that 3 times now... is it perhaps the version of x264? I'm building the main tree git clone git://git.videolan.org/x264 built it again from a make distclean and a clean build after running : sudo apt-get remove ffmpeg x264 libx264-dev tried calling x264 incase i missed a build and there's defiantly nothing there. thanks for the help From alabandit at gmail.com Fri Apr 8 18:06:12 2011 From: alabandit at gmail.com (Alastair Mason) Date: Fri, 8 Apr 2011 18:06:12 +0200 Subject: [FFmpeg-user] Building ffmpeg and libx264 on ubuntu In-Reply-To: References: Message-ID: > You can also try: find / -name '*x264*' after rerunning sudo apt-get remove ffmpeg x264 libx264-dev (just to be safe) i ran running sudo find / -name '*x264*' i found the following: /usr/share/ffmpeg/libx264-max.ffpreset /usr/share/ffmpeg/libx264-lossless_slow.ffpreset /usr/share/ffmpeg/libx264-lossless_ultrafast.ffpreset /usr/share/ffmpeg/libx264-lossless_slower.ffpreset /usr/share/ffmpeg/libx264-lossless_max.ffpreset /usr/share/ffmpeg/libx264-lossless_medium.ffpreset /usr/share/ffmpeg/libx264-ipod640.ffpreset /usr/share/ffmpeg/libx264-hq.ffpreset /usr/share/ffmpeg/libx264-baseline.ffpreset /usr/share/ffmpeg/libx264-main.ffpreset /usr/share/ffmpeg/libx264-fastfirstpass.ffpreset /usr/share/ffmpeg/libx264-lossless_fast.ffpreset /usr/share/ffmpeg/libx264-ipod320.ffpreset /usr/share/ffmpeg/libx264-normal.ffpreset /usr/share/ffmpeg/libx264-default.ffpreset /usr/share/ffmpeg/libx264-slowfirstpass.ffpreset /usr/share/doc/libx264-85 /usr/lib/libx264.so.85 /usr/lib/i686/sse2/libx264.so.85 /usr/lib/vlc/codec/libx264_plugin.so /usr/lib/gstreamer-0.10/libgstx264.so /usr/local/lib/libx264.so /usr/local/lib/libx264.so.67 /var/cache/apt/archives/libx264-dev_2%3a0.85.1448+git1a6d32-4_i386.deb /var/cache/apt/archives/x264_2%3a0.85.1448+git1a6d32-4_i386.deb /var/lib/dpkg/info/libx264-85.list /var/lib/dpkg/info/libx264-85.postinst /var/lib/dpkg/info/libx264-85.shlibs /var/lib/dpkg/info/libx264-85.md5sums /var/lib/dpkg/info/libx264-85.postrm I found the following files that i can't account for. Would it be safe to delete all of them, except: /usr/lib/vlc/codec/libx264_plugin.so /usr/lib/gstreamer-0.10/libgstx264.so or is there a better way way to deal with them? thanks again for the help From curt at curtronics.com Fri Apr 8 18:50:51 2011 From: curt at curtronics.com (Curtis J Blank) Date: Fri, 08 Apr 2011 11:50:51 -0500 Subject: [FFmpeg-user] =?windows-1252?q?Compiling_ffmpeg=3A_implicit_decla?= =?windows-1252?q?ration_of_function_=91lame=5Fset=5FVBR=5Fquality=92?= Message-ID: <4D9F3CEB.4050001@curtronics.com> For the life of me I can't get past this. CC libavcodec/libmp3lame.o libavcodec/libmp3lame.c: In function ?MP3lame_encode_init?: libavcodec/libmp3lame.c:63:9: error: implicit declaration of function ?lame_set_VBR_quality? So I look in lame.h /* VBR quality level. 0=highest 9=lowest, Range [0,...,10[ */ int CDECL lame_set_VBR_quality(lame_global_flags *, float); float CDECL lame_get_VBR_quality(const lame_global_flags *); I've got the lame packages installed lame-3.98.4-5.1.x86_64 libmp3lame-devel-3.98.4-5.1.x86_64 libmp3lame0-3.98.4-5.1.x86_64 libtwolame0-0.3.12-5.1.x86_64 I've tried ffmpeg-0.6.90-rc0.tar.gz ffmpeg-HEAD-9d4cb45.tar.gz ffmpeg-snapshot.tar.gz These are my configure options ./configure --shlibdir=/usr/lib64 --libdir=/usr/lib64 --enable-shared --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libspeex --enable-libxvid --enable-postproc --enable-gpl --enable-x11grab --enable-libgsm --enable-libschroedinger --enable-libdirac --enable-avfilter --enable-libvpx --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libdc1394 --enable-librtmp --enable-libfaac --enable-nonfree --enable-libopencv --enable-runtime-cpudetect But I just can't get it to compile. What am I missing? A compiler flag or something? This is on openSuSE 11.2 From belcampo at zonnet.nl Fri Apr 8 19:17:17 2011 From: belcampo at zonnet.nl (belcampo) Date: Fri, 08 Apr 2011 19:17:17 +0200 Subject: [FFmpeg-user] Building ffmpeg and libx264 on ubuntu In-Reply-To: References: Message-ID: <4D9F431D.1080503@zonnet.nl> On 04/08/11 12:38, James Darnley wrote: > On 08/04/2011, Alastair Mason wrote: >> /home/alastair/progs/ffmpeg/libavcodec/libx264.c:308: undefined >> reference to `x264_encoder_open_114' As stated above, you'll have to get version >= 114, your version is 85. Current version is 114. Downloadable at: ftp://ftp.videolan.org/pub/videolan/x264/snapshots/ You need yasm installed for building and configure with ./configure --prefix=/desired/path --enable-shared > > Your x264 header version does not match the library version. Try the > guide again, especially part 1. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From curt at curtronics.com Fri Apr 8 20:11:59 2011 From: curt at curtronics.com (Curtis J Blank) Date: Fri, 08 Apr 2011 13:11:59 -0500 Subject: [FFmpeg-user] =?windows-1252?q?Compiling_ffmpeg=3A_implicit_decla?= =?windows-1252?q?ration_of_function_=91lame=5Fset=5FVBR=5Fquality=92?= In-Reply-To: <4D9F3CEB.4050001@curtronics.com> References: <4D9F3CEB.4050001@curtronics.com> Message-ID: <4D9F4FEF.4010100@curtronics.com> Ok, it seems I got over that hump. Had an errant copy as in old of lame.h in /usr/local/include/lame/ cleaned that up and got past that. Always seem to find the fix to the problem right after I ask. Now I'm on to a new problem: libavfilter/vf_libopencv.c:29:28: fatal error: opencv/cxtypes.h: No such file or directory And yeah I can't find it. On openSuSE 11.2 opencv-devel-2.2.0-1.pm.12.52.x86_64 libopencv2_2-2.2.0-1.pm.12.52.x86_64 libopencv2-2.1.0-2.37.x86_64 opencv-2.2.0-1.pm.12.52.x86_64 And on openSuSE 11.4 opencv-devel-2.2.0-1.pm.12.34.x86_64 libopencv2_2-2.2.0-1.pm.12.34.x86_64 opencv-2.2.0-1.pm.12.34.x86_64 On 04/08/11 11:50, Curtis J Blank wrote: > For the life of me I can't get past this. > > CC libavcodec/libmp3lame.o > libavcodec/libmp3lame.c: In function ?MP3lame_encode_init?: > libavcodec/libmp3lame.c:63:9: error: implicit declaration of function > ?lame_set_VBR_quality? > > So I look in lame.h > > /* VBR quality level. 0=highest 9=lowest, Range [0,...,10[ */ > int CDECL lame_set_VBR_quality(lame_global_flags *, float); > float CDECL lame_get_VBR_quality(const lame_global_flags *); > > I've got the lame packages installed > > lame-3.98.4-5.1.x86_64 > libmp3lame-devel-3.98.4-5.1.x86_64 > libmp3lame0-3.98.4-5.1.x86_64 > libtwolame0-0.3.12-5.1.x86_64 > > I've tried > > ffmpeg-0.6.90-rc0.tar.gz > ffmpeg-HEAD-9d4cb45.tar.gz > ffmpeg-snapshot.tar.gz > > These are my configure options > > ./configure --shlibdir=/usr/lib64 --libdir=/usr/lib64 --enable-shared > --enable-libmp3lame --enable-libvorbis --enable-libtheora > --enable-libspeex --enable-libxvid --enable-postproc --enable-gpl > --enable-x11grab --enable-libgsm --enable-libschroedinger > --enable-libdirac --enable-avfilter --enable-libvpx --enable-version3 > --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 > --enable-libdc1394 --enable-librtmp --enable-libfaac --enable-nonfree > --enable-libopencv --enable-runtime-cpudetect > > But I just can't get it to compile. What am I missing? A compiler flag > or something? > > This is on openSuSE 11.2 > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From mark at mdsh.com Fri Apr 8 20:45:37 2011 From: mark at mdsh.com (Mark Himsley) Date: Fri, 08 Apr 2011 19:45:37 +0100 Subject: [FFmpeg-user] 4CIF H263 video fails to decode In-Reply-To: References: Message-ID: <4D9F57D1.3020402@mdsh.com> On 08/04/11 09:15, Ankuj Gupta wrote: > Hi, > > I am trying to decode a H263 4CIF(704x576) resolution file with ffmpeg, but > it is detected as mp3 file and fails to get decoded. I tried with lower > resolution H263 files they get decoded easily as well as 16CIF. I am using > ffmpeg version 0.6.1 Please send the command you are using and the untrimmed output of that command. -- Mark From seandarcy2 at gmail.com Sat Apr 9 07:21:51 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Sat, 09 Apr 2011 01:21:51 -0400 Subject: [FFmpeg-user] fps in March 13 git: -r 29.97 = 2997 tbn ??? In-Reply-To: <4D9B4163.9030407@gmail.com> References: <4D970DA2.1070302@optonline.net> <4D9A3E74.6040906@gmail.com> <4D9B1A0F.1010203@alcatel-lucent.com> <4D9B4163.9030407@gmail.com> Message-ID: On 04/05/2011 12:20 PM, Baptiste Coudurier wrote: > On 04/05/2011 06:33 AM, Mike Scheutzow wrote: >> Baptiste Coudurier wrote: >>>> Also, be careful believing the tbn / tbc "Output #0" stats - they have >>>> been broken for months. Use 'ffmpeg -i infile' to get accurate values. >>> >>> No, the values are correct and accurate, you can trust them. >> >> It was still broken in Michael's git as of 2011-mar-16. >> >> The test case is very simple: h264 in .ts transcoded to h264 in .ts. >> >> During encode run, for "Output #0" ffmpeg prints: >> >> Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 10:11 DAR 15:11], >> q=10-51, 200 kb/s, 90k tbn, 29.97 tbc >> >> Yet, ffmpeg -i on the resulting file says: >> >> Stream #0.0[0x100]: Video: h264 (High), yuv420p, 720x480 [PAR 10:11 >> DAR 15:11], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc >> >> Do the tbc look the same to you? > > And which one do you think are wrong anyway ? > > Information printed is accurate, the interpretation is different. > How is the interpretation different? If it were interlaced, I guess one would be fields, the other frames. But it's not interlaced. And which of these fields does "-r" affect? sean From vmrsss at gmail.com Sat Apr 9 13:08:23 2011 From: vmrsss at gmail.com (vmrsss) Date: Sat, 9 Apr 2011 12:08:23 +0100 Subject: [FFmpeg-user] subtitles from DVB streams Message-ID: <1D111E63-4A02-48E7-8FB4-393AAE30143C@gmail.com> Hello there, I have got a stream (a DVB-S2 from EyeTV) which looks like in the output below. Can please anybody explain how I could use ffmpeg extract the subtitle track and dump it either as it is or convert to some meaningful format, eg srt or idx/sub ? thanks. FFmpeg version git-N-28891-gda53f05, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 5 2011 21:35:12 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --arch=x86_64 --cpu=core2 --cc='gcc -arch x86_64 -march=core2 -msse4 -O2 -pipe' --enable-shared --enable-nonfree --disable-ffserver --disable-debug --enable-gpl --enable-pthreads --enable-postproc --enable-libx264 --enable-libxvid --enable-libvorbis --enable-libdirac --enable-libschroedinger --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-runtime-cpudetect --enable-hardcoded-tables libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.117. 0 / 52.117. 0 libavformat 52.105. 0 / 52.105. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 77. 1 / 1. 77. 1 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpegts @ 0x101814e00] max_analyze_duration reached [NULL @ 0x101843e00] start time is not set in av_estimate_timings_from_pts [NULL @ 0x101845c00] start time is not set in av_estimate_timings_from_pts [NULL @ 0x101847a00] start time is not set in av_estimate_timings_from_pts [NULL @ 0x101849800] start time is not set in av_estimate_timings_from_pts [NULL @ 0x10184f200] start time is not set in av_estimate_timings_from_pts [NULL @ 0x101851000] start time is not set in av_estimate_timings_from_pts [NULL @ 0x101852e00] start time is not set in av_estimate_timings_from_pts [NULL @ 0x101854c00] start time is not set in av_estimate_timings_from_pts [NULL @ 0x101856a00] start time is not set in av_estimate_timings_from_pts [NULL @ 0x101858800] start time is not set in av_estimate_timings_from_pts [NULL @ 0x10185a600] start time is not set in av_estimate_timings_from_pts [NULL @ 0x10185c400] start time is not set in av_estimate_timings_from_pts [NULL @ 0x10185e200] start time is not set in av_estimate_timings_from_pts Input #0, mpegts, from '00000000134cfc94.mpg': Duration: 00:59:02.44, start: 57154.402333, bitrate: 10231 kb/s Program 6940 Metadata: service_name : BBC HD service_provider: BSkyB Stream #0.0[0x157c]: Video: h264 (High), yuv420p, 1440x1080 [PAR 4:3 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x157e](NAR): Audio: mp2, 48000 Hz, stereo, s16, 256 kb/s Stream #0.2[0x157f](eng): Subtitle: [6][0][0][0] / 0x0006 Stream #0.3[0xf05]: Data: [11][0][0][0] / 0x000B Stream #0.4[0xf06]: Data: [11][0][0][0] / 0x000B Stream #0.5[0xf07]: Data: [11][0][0][0] / 0x000B Stream #0.6[0xf08]: Data: [11][0][0][0] / 0x000B Stream #0.7[0x1580](eng): Subtitle: dvbsub Stream #0.8[0x157d](eng): Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s Stream #0.9[0x1582]: Data: [5][0][0][0] / 0x0005 Stream #0.10[0xf00]: Data: [5][0][0][0] / 0x0005 Stream #0.11[0xf01]: Data: [5][0][0][0] / 0x0005 Stream #0.12[0xf02]: Data: [5][0][0][0] / 0x0005 Stream #0.13[0xf03]: Data: [5][0][0][0] / 0x0005 Stream #0.14[0xf04]: Data: [5][0][0][0] / 0x0005 Stream #0.15[0x901]: Data: [5][0][0][0] / 0x0005 Stream #0.16[0x902]: Data: [5][0][0][0] / 0x0005 Stream #0.17[0x903]: Data: [5][0][0][0] / 0x0005 Program 6941 Metadata: service_name : BBC One HD service_provider: BSkyB Program 6942 Metadata: service_name : ITV1 HD service_provider: BSkyB Program 6945 Metadata: service_name : 6945 service_provider: BSkyB Program 1035 Metadata: service_name : 6942 service_provider: BSkyB At least one output file must be specified From s.helden at gmx.de Sat Apr 9 21:45:44 2011 From: s.helden at gmx.de (Stephan van Helden) Date: Sat, 09 Apr 2011 21:45:44 +0200 Subject: [FFmpeg-user] multi-angle DVDs Message-ID: <4DA0B768.90006@gmx.de> I converted most of my DVDs to MKV/H.264 files. However, I have a number of DVDs where this fails. It seems that the issue is related to multiple viewing angles. One of the DVDs that I used for deeper investigation is "Exposed" by Mike Oldfield. I concatenated all VOBs into a single file and then tried to encode on Windows with this commandline: ffmpeg -y -i exposed.vob -map 0.0 -map 0.1 -sn -threads 4 -vcodec libx264 -acodec copy -crf 24 -vpre hq -acodec copy exposed.mkv I tried with different ffmpeg versions (including the latest and also very old ones), but always received the error "Application provided invalid, non monotonically increasing dts to muxer". I followed the suggestion that can be found at http://forum.doom9.org/archive/index.php/t-158193.html and elsewhere and compiled my own ffmpeg build that ignores this error. But when I do that, I get an MKV file that seems to contain all of the camera angels multiplexed - few frames one angle, few frames of another angle, and so on. So my basic question is: Is there a way of encoding only the main camera angle? Is there a switch that I could use in the encoding process itself, or is there any tool that can extract the main angle from the VOB file? From coniophora at gmail.com Sun Apr 10 00:34:21 2011 From: coniophora at gmail.com (Jim Worrall) Date: Sat, 9 Apr 2011 16:34:21 -0600 Subject: [FFmpeg-user] C compiler test failure Message-ID: <5FFC2A61-A0B6-4349-90EC-F9E5C4A3B86B@gmail.com> I was trying to compile ffmpeg on MacOS 10.6, and ran into trouble. I am not well versed in unix, so please bear with me. I installed all the libraries and related files using Fink, but the version of ffmpeg that Fink has is not very recent, so I got a snapshot from the repository. I would be very grateful for some pointers, keeping in mind I'm a unix dummy. Perhaps you can tell me if the configure options also make sense. Here is the error and the end of the config.log: ERROR IN TERMINAL: Jims-MacBook-Pro:ffmpeg jim$ ./configure --prefix=${TARGET} --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --arch=x86_64 --enable-runtime-cpudetect && make -j 4 && make install gcc is unable to create an executable file. If gcc is a cross-compiler, use the --enable-cross-compile option. Only do this if you know what cross compiling means. C compiler test failed. If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. FROM CONFIG.LOG: BEGIN /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.LfOoLRwa.c 1 int main(void){ return 0; } END /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.LfOoLRwa.c gcc -c -o /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.s6HaVnj9.o /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.LfOoLRwa.c gcc =L/Volumes/Ramdisk/sw/lib -o /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.CpxM7G19 /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.s6HaVnj9.o i686-apple-darwin10-gcc-4.2.1: =L/Volumes/Ramdisk/sw/lib: No such file or directory C compiler test failed. Thanks, Jim From joey.blow.uk at gmail.com Sun Apr 10 01:16:58 2011 From: joey.blow.uk at gmail.com (Joey Blow) Date: Sun, 10 Apr 2011 00:16:58 +0100 Subject: [FFmpeg-user] Linker error when building on Arm Message-ID: I'm trying to compile ffmpeg on Arm and I've run into exactly the problem described here (I don't how to reply to a message that was sent before I joined the list). I don't really speak C, and I can't quite follow what the fix is. I can see that I need to surround part of the code in an if statement, but I've not had any success in my attempts. Could anyone give me a more detailed pointer as to what exactly I need to put in the code? I'm working on a fresh git clone. Thanks. jb From curt at curtronics.com Sun Apr 10 11:22:56 2011 From: curt at curtronics.com (Curtis J Blank) Date: Sun, 10 Apr 2011 04:22:56 -0500 Subject: [FFmpeg-user] C compiler test failure In-Reply-To: <5FFC2A61-A0B6-4349-90EC-F9E5C4A3B86B@gmail.com> References: <5FFC2A61-A0B6-4349-90EC-F9E5C4A3B86B@gmail.com> Message-ID: <4DA176F0.6040805@curtronics.com> I had that problem "unable to create an executable file" today trying to recompile ZoneMinder after upgrading to openSuSE 11.4. I had this line for ./configure: --with-extralibs="-lfaac -lmp3lame -logg -lvorbis -lamrnb -lamrwb" Once I removed ogg & vorbis it worked. I see you have vorbis, try it without that. If that doesn't work try removing libraries one at a time and see if that helps. On 04/09/11 17:34, Jim Worrall wrote: > I was trying to compile ffmpeg on MacOS 10.6, and ran into trouble. I am not well versed in unix, so please bear with me. I installed all the libraries and related files using Fink, but the version of ffmpeg that Fink has is not very recent, so I got a snapshot from the repository. I would be very grateful for some pointers, keeping in mind I'm a unix dummy. Perhaps you can tell me if the configure options also make sense. Here is the error and the end of the config.log: > > ERROR IN TERMINAL: > Jims-MacBook-Pro:ffmpeg jim$ ./configure --prefix=${TARGET} --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --arch=x86_64 --enable-runtime-cpudetect&& make -j 4&& make install > gcc is unable to create an executable file. > If gcc is a cross-compiler, use the --enable-cross-compile option. > Only do this if you know what cross compiling means. > C compiler test failed. > > If you think configure made a mistake, make sure you are using the latest > version from Git. If the latest version fails, report the problem to the > ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. > Include the log file "config.log" produced by configure as this will help > solving the problem. > > FROM CONFIG.LOG: > BEGIN /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.LfOoLRwa.c > 1 int main(void){ return 0; } > END /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.LfOoLRwa.c > gcc -c -o /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.s6HaVnj9.o /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.LfOoLRwa.c > gcc =L/Volumes/Ramdisk/sw/lib -o /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.CpxM7G19 /var/folders/ns/nsHE53MCGNOl7NqfmlgXX++++TI/-Tmp-//ffconf.s6HaVnj9.o > i686-apple-darwin10-gcc-4.2.1: =L/Volumes/Ramdisk/sw/lib: No such file or directory > C compiler test failed. > > Thanks, > Jim > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From akilan27 at gmail.com Sun Apr 10 13:13:40 2011 From: akilan27 at gmail.com (Akilan R) Date: Sun, 10 Apr 2011 16:43:40 +0530 Subject: [FFmpeg-user] piping images into ffmpeg Message-ID: Hi, I have a C program which generates a series of images and I wanted to make them into a video which should be streamed in real time or stored in a file. While reading up ffmpeg documentation I came across repeatedly that ffmpeg can take input from named pipes. My question is in what format should the files given into the pipe should be and how to input the files into the pipe. I tried this: $ cat hello-*.ppm | ffmpeg -f image2pipe -vcodec ppm -r 25 -i pipe: -r 25 output.mp4 but video was bad. (Here the input comes from already stored files as i was testing; once it works i'll pipe in output from my program like this or using named pipes.) There is no problem with image files as this one worked flawlessly: $ ffmpeg -f image2 -r 25 -i hello-%03d.ppm -r 25 output.mp4 -- ?????? (Akilan R) [ blog.akilan.in ] *I should have no use for a paradise in which I should be deprived of the right to prefer hell.* --Jean Rostand From sarkiaponius at alice.it Sun Apr 10 13:34:47 2011 From: sarkiaponius at alice.it (andrea) Date: Sun, 10 Apr 2011 13:34:47 +0200 Subject: [FFmpeg-user] How to speed up RTSP... Message-ID: <1302435287.19282.1.camel@debian> Hi, I'm using a trivial command line like the following to download old episodes of programmes only available as Real Audio streams: ffmpeg -i rtsp://mm6.rai.it/radiofonia/radio3/napoli/uomini_profeti/2010/uomini_profeti2010_09_19.ra test.ra This works perfectly, but it takes 45 minutes to finish, because this is the duration of the episode. Is there a way to tell ffmpeg to go much faster? I was used to do that with mplayer's option "-bandwidth" set to a very high value, but currently mplayer doesn't work here at home with RTSP, most likely because of stronger limitations on the traffic adopted by my new ISP. Best regards. -- andrea _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From seandarcy2 at gmail.com Sun Apr 10 18:50:07 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Sun, 10 Apr 2011 12:50:07 -0400 Subject: [FFmpeg-user] How to speed up RTSP... In-Reply-To: <1302435287.19282.1.camel@debian> References: <1302435287.19282.1.camel@debian> Message-ID: On 04/10/2011 07:34 AM, andrea wrote: > ffmpeg -i > rtsp://mm6.rai.it/radiofonia/radio3/napoli/uomini_profeti/2010/uomini_profeti2010_09_19.ra test.ra > I'm amazed you got rtsp to work at all. Here's what I get from ffmpeg -protocols: ...... IO. rtmp IO. rtmpt IO. rtmpe IO. rtmpte IO. rtmps IO. rtp IO. tcp IO. udp And that's by building with --enable-librtmp. sean From paul at teknique.com Fri Apr 8 01:30:30 2011 From: paul at teknique.com (Paul Wang) Date: Fri, 8 Apr 2011 11:30:30 +1200 Subject: [FFmpeg-user] ref: mov file generated from ffmpeg lib lost audio by players other than VLC Message-ID: <8b0f60043bd3687fb4ec50de2285b1b3@mail.gmail.com> Hi, all, I have a problem using ffmpeg lib APIs. I am using these APIs to generate a MOV container file from a H264 video stream and a AAC audio stream. The audio stream can be save as .aac file, which can be played by QuickTime or MediaPlayer. My problem is after the H264 video and AAC audio stream are saved in a .mov container, the file played by media players other than VLC has no audio. Played by VLC is good and having both video and audio, played by other players, video OK, no audio. No ideas what?s wrong??? This is my first time using ffmpeg, can someone give me a hand. Thanks. Paul Wang. From marillat at debian-multimedia.org Fri Apr 8 18:20:54 2011 From: marillat at debian-multimedia.org (Christian Marillat) Date: Fri, 08 Apr 2011 18:20:54 +0200 Subject: [FFmpeg-user] Building ffmpeg and libx264 on ubuntu In-Reply-To: (Alastair Mason's message of "Fri, 8 Apr 2011 18:06:12 +0200") References: Message-ID: <877hb4pvdl.fsf@christian.marillat.net> Alastair Mason writes: [...] > /usr/local/lib/libx264.so > /usr/local/lib/libx264.so.67 Remove these files. I think you also have x264 include files in /usr/locale/include (x264.h and x264_config.h) the se files should be removed. Christian From prashant at b-one.net Sun Apr 10 16:02:07 2011 From: prashant at b-one.net (Prashant Rathi) Date: Sun, 10 Apr 2011 18:02:07 +0400 Subject: [FFmpeg-user] Query on audio_decode_example() in api-example.c Message-ID: <000a01cbf787$e2038c00$a60aa400$@b-one.net> Hi FFMPEG Users, I am new to FFMPEG. My requirement is to convert any given input audio file to .wav pcm format by using FFMPEG API's. I tried using api-example.c code given for audio_decode_example() . Input to this api is outfilename and filename. Outfilename: I have given path to some location\test.wav Filename: I have given path to m4a file available on my system. But it always fails at avcodec_decode_audio3() call. This call always returns -1 with following console message for m4a files as input. Channel 0.0 is not allocated I tried to convert mp3 file to wav also using the same approach as above but again it failed at same api call with "Header Missing " message. Please support me with right code to meet my requirements. I want API usage not console command to do this. Thanks in advance! Prashant From sarkiaponius at alice.it Sun Apr 10 23:36:47 2011 From: sarkiaponius at alice.it (andrea) Date: Sun, 10 Apr 2011 23:36:47 +0200 Subject: [FFmpeg-user] How to speed up RTSP... In-Reply-To: References: <1302435287.19282.1.camel@debian> Message-ID: <1302471407.23874.5.camel@debian> On Sun, 2011-04-10 at 12:50 -0400, sean darcy wrote: > On 04/10/2011 07:34 AM, andrea wrote: > > ffmpeg -i > > rtsp://mm6.rai.it/radiofonia/radio3/napoli/uomini_profeti/2010/uomini_profeti2010_09_19.ra test.ra > > > I'm amazed you got rtsp to work at all. Here's what I get from ffmpeg > -protocols: I swear it works, tried right now. I got --enable-librtmp, too. I cannot figure out why it works, but it does. The problem is that it seems there is no option to accelerate the stream someway. With mplayer, as I think I said, the -bandwidth option does the trick, if you set it to a quite high value (say 4000000 or such). But I caould not find something like that for ffmpeg. Best regards. > > ...... > IO. rtmp > IO. rtmpt > IO. rtmpe > IO. rtmpte > IO. rtmps > IO. rtp > IO. tcp > IO. udp > > And that's by building with --enable-librtmp. > > sean > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From seandarcy2 at gmail.com Sun Apr 10 23:56:09 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Sun, 10 Apr 2011 17:56:09 -0400 Subject: [FFmpeg-user] How to speed up RTSP... In-Reply-To: <1302471407.23874.5.camel@debian> References: <1302435287.19282.1.camel@debian> <1302471407.23874.5.camel@debian> Message-ID: On 04/10/2011 05:36 PM, andrea wrote: > On Sun, 2011-04-10 at 12:50 -0400, sean darcy wrote: >> On 04/10/2011 07:34 AM, andrea wrote: >>> ffmpeg -i >>> rtsp://mm6.rai.it/radiofonia/radio3/napoli/uomini_profeti/2010/uomini_profeti2010_09_19.ra test.ra >>> >> I'm amazed you got rtsp to work at all. Here's what I get from ffmpeg >> -protocols: > > I swear it works, tried right now. I got --enable-librtmp, too. I cannot > figure out why it works, but it does. The problem is that it seems there > is no option to accelerate the stream someway. With mplayer, as I think > I said, the -bandwidth option does the trick, if you set it to a quite > high value (say 4000000 or such). But I caould not find something like > that for ffmpeg. > > Best regards. >> >> ...... >> IO. rtmp >> IO. rtmpt >> IO. rtmpe >> IO. rtmpte >> IO. rtmps >> IO. rtp >> IO. tcp >> IO. udp >> >> And that's by building with --enable-librtmp. >> >> sean >> Well glad it works for you. In addition to accelerating the stream, I wish ffmpeg (and mplayer) buffered it. That way you could pause, let the buffer fill, then play without interruptions or stuttering. sean From paul at teknique.com Sun Apr 10 23:56:54 2011 From: paul at teknique.com (Paul Wang) Date: Mon, 11 Apr 2011 09:56:54 +1200 Subject: [FFmpeg-user] ref: My mov file has no audio on players other than VLC Message-ID: Hi, all, I am new to ffmpeg. I am using the ffmpeg to write my h264 and AAC stream into the mov container. My AAC stream is not from ffmpeg, but from a third-party. It works fine, when I write the stream onto a .AAC file without using ffmpeg, it can be played by Quicktime, VLC and media player etc. When I am using ffmpeg to put the video and audio together into a .mov file, the audio is lost on most of players other than VLC. My code is most likely as output_example.c with the ffmpeg, only without calling avcodec_encode_audio(). On VLC, it works fine, but has some warnings. I attached the message of verbosity level 2 for the audio (video disabled) as the following: ?? *mp4** warning: *unknown box type (incompletely loaded) *mp4** debug: *read box: "soun" in stsd channel 2 sample size 16 sample rate 48000.000000 *mp4** debug: *read box: "stsd" entry-count 1 *mp4** debug: *found Box: stts size 24 *mp4** debug: *read box: "stts" entry-count 1 *mp4** debug: *found Box: stsc size 28 *mp4** debug: *read box: "stsc" entry-count 1 *mp4** debug: *found Box: stsz size 1084 *mp4** debug: *read box: "stsz" sample-size 0 sample-count 266 *mp4** debug: *found Box: stco size 1080 *mp4** debug: *read box: "co64" entry-count 266 *mp4** debug: *found Box: udta size 31 *mp4** debug: *found Box: cenc size 23 *mp4** debug: *read box: "cenc" text=`Lavf52.64.2' *mp4** debug: *dumping root Box "root" *mp4** debug: *| + ftyp size 20 *mp4** debug: *| + wide size 8 *mp4** debug: *| + mdat size 90806 *mp4** debug: *| + moov size 2807 *mp4** debug: *| | + mvhd size 108 *mp4** debug: *| | + trak size 2660 *mp4** debug: *| | | + tkhd size 92 *mp4** debug: *| | | + mdia size 2560 *mp4** debug: *| | | | + mdhd size 32 *mp4** debug: *| | | | + hdlr size 45 *mp4** debug: *| | | | + minf size 2475 *mp4** debug: *| | | | | + smhd size 16 *mp4** debug: *| | | | | + hdlr size 44 *mp4** debug: *| | | | | + dinf size 36 *mp4** debug: *| | | | | | + dref size 28 *mp4** debug: *| | | | | | | + url size 12 *mp4** debug: *| | | | | + stbl size 2371 *mp4** debug: *| | | | | | + stsd size 147 *mp4** debug: *| | | | | | | + mp4a size 131 *mp4** debug: *| | | | | | | | + wave size 79 *mp4** debug: *| | | | | | | | | + frma size 12 *mp4** debug: *| | | | | | | | | + mp4a size 12 *qt4** debug: *Updating the geometry *mp4** debug: *| | | | | | | | | + esds size 39 *mp4** debug: *| | | | | | | | | + size 8 *qt4** debug: *Updating the geometry *mp4** debug: *| | | | | | + stts size 24 *mp4** debug: *| | | | | | + stsc size 28 *mp4** debug: *| | | | | | + stsz size 1084 *mp4** debug: *| | | | | | + stco size 1080 *mp4** debug: *| | + udta size 31 *mp4** debug: *| | | + cenc size 23 *mp4** debug: *unrecognized major file specification (qt ). *mp4** debug: *found 1 tracks *mp4** debug: *track[Id 0x1] read 266 chunk *mp4** debug: *track[Id 0x1] read 266 samples length:5s *main** debug: *selecting program id=0 *mp4** debug: *adding track[Id 0x1] audio (enable) language en *main** debug: *using demux module "mp4" *main** debug: *TIMER module_need() : 3.000 ms - Total 3.000 ms / 1 intvls (Avg 3.000 ms) *mp4** warning: *control query 14 unimplemented *mp4** warning: *DEMUX_GET_FPS unimplemented !! *main** debug: *looking for a subtitle file in I:\ *main** debug: *looking for decoder module: 35 candidates *main** debug: *using decoder module "faad" *main** debug: *TIMER module_need() : 0.000 ms - Total 0.000 ms / 1 intvls (Avg 0.000 ms) *main** debug: *thread (decoder) created at priority 2 (../.././src/input/decoder.c:315) *main** debug: *thread started *main** debug: *`I:\AAA.mov' successfully opened ?. When played with mplayer, I had no sound, but the following errors: FAAD: Failed to decode frame: Invalid number of channels FAAD: Failed to decode frame: Invalid number of channels FAAD: Failed to decode frame: Invalid number of channels FAAD: Failed to decode frame: Invalid number of channels FAAD: Failed to decode frame: Invalid number of channels FAAD: Failed to decode frame: Invalid number of channels FAAD: Failed to decode frame: Invalid number of channels FAAD: Failed to decode frame: Invalid number of channels ?. Thanks in advance for any help . Paul Wang. From jblee15 at hotmail.com Mon Apr 11 00:05:06 2011 From: jblee15 at hotmail.com (J K) Date: Sun, 10 Apr 2011 22:05:06 +0000 Subject: [FFmpeg-user] Latest ffmpeg git error Message-ID: Hi All, I am using the latest ffmpeg git and I keep getting the below error.. does anybody know how to fix? make[1]: *** [allformats.o] Error 1make[1]: Leaving directory `/usr/src/ffmpegscript/mplayer/ffmpeg/libavformat'make: *** [ffmpeg/libavformat/libavformat.a] Error 2cp: cannot create regular file `/usr/local/cpffmpeg/etc/mplayer/codecs.conf': No such file or directory Tks. From kshawkeye at gmail.com Mon Apr 11 07:37:17 2011 From: kshawkeye at gmail.com (Kyle) Date: Mon, 11 Apr 2011 01:37:17 -0400 Subject: [FFmpeg-user] ffplay displaying only black Message-ID: <4DA2938D.3010001@gmail.com> I compiled FFmpeg with: --enable-runtime-cpudetect --enable-memalign-hack --enable-avis ynth --cross-prefix=i686-w64-mingw32- --target-os=mingw32 --arch=i686 --extra-cf lags=-I/home/kyle/software/ffmpeg/external-libraries/win32/include --extra-ldfla gs=-L/home/kyle/software/ffmpeg/external-libraries/win32/lib --pkg-config=pkg-co nfig My FFmpeg version is git-N-29032-gab8cfd4 and It was compiled with gcc 4.7.0 20110409 and mingw-w64 I'm getting an error while trying to play back something with ffplay, I used a simple play command: ffplay "infile.mp4" All I see is black, except for (probably 8 pixels total) color in the very top left corner. My SDL configured fine and had no errors, same goes for FFmpeg. I compiled a similar version for 64-bit, same FFmpeg source version, same toolchain (gcc, binutils, mingw-w64 versions as the 32-bit), just changed the --cross-prefix and the --arch and the 64-bit version works perfect. Does anyone have any clues as to what might be happening? If I had to take a guess I would say that it's resizing it to those 8 pixels, since I see them changing colors and stop changing if I pause it. Also, the ffplay output looks good as far as I can tell: FFplay version git-N-29032-gab8cfd4, Copyright (c) 2003-2011 the FFmpeg develope rs built on Apr 11 2011 01:12:08 with gcc 4.7.0 20110409 (experimental) configuration: --enable-runtime-cpudetect --enable-memalign-hack --enable-avis ynth --cross-prefix=i686-w64-mingw32- --target-os=mingw32 --arch=i686 --extra-cf lags=-I/home/kyle/software/ffmpeg/external-libraries/win32/include --extra-ldfla gs=-L/home/kyle/software/ffmpeg/external-libraries/win32/lib --pkg-config=pkg-co nfig libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.117. 1 / 52.117. 1 libavformat 52.106. 0 / 52.106. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 78. 0 / 1. 78. 0 libswscale 0. 13. 0 / 0. 13. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'infile.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2011-03-03 01:26:03 Duration: 00:01:55.93, start: 0.000000, bitrate: 2980 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 2846 kb/s, 29.97 fps, 29.92 tbr, 1k tbn, 59.83 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 128 kb/s Metadata: creation_time : 2011-03-03 01:26:05 Seek to 40% ( 0:00:46) of total duration ( 0:01:55) f=0/0 f=0/0 Seek to 40% ( 0:00:46) of total duration ( 0:01:55) f=0/0 47.04 A-V: -0.072 s:0.0 aq= 320KB vq= 6224KB sq= 0B f=1/1 Finally, my sdl-config is the same for both 32-bit and 64-bit, as goes for my sdl.pc Any ideas would be greatly appreciated, thanks. From tim.nicholson at bbc.co.uk Mon Apr 11 08:56:33 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Mon, 11 Apr 2011 07:56:33 +0100 Subject: [FFmpeg-user] piping images into ffmpeg In-Reply-To: References: Message-ID: <4DA2A621.4090704@bbc.co.uk> On 10/04/11 12:13, Akilan R wrote: > Hi, > > I have a C program which generates a series of images and I wanted to make > them into a video which should be streamed in real time or stored in a file. > While reading up ffmpeg documentation I came across repeatedly that ffmpeg > can take input from named pipes. > > My question is in what format should the files given into the pipe should be > and how to input the files into the pipe. > > I tried this: > > $ cat hello-*.ppm | ffmpeg -f image2pipe -vcodec ppm -r 25 -i pipe: -r 25 > output.mp4 > > but video was bad. (Here the input comes from already stored files as i was > testing; once it works i'll pipe in output from my program like this or > using named pipes.) > There is no problem with image files as this one worked flawlessly: > > $ ffmpeg -f image2 -r 25 -i hello-%03d.ppm -r 25 output.mp4 > > Err, if "-f image2" is correct then why change it to "-f image2pipe" ? What does something like:- $ cat hello-*.ppm | ffmpeg -f image2 -r 25 -i pipe: -r 25 output.mp4 do? I think image2pipe is more for output formatting. yuv4mpegpipe certainly is. -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From akilan27 at gmail.com Mon Apr 11 10:07:26 2011 From: akilan27 at gmail.com (Akilan R) Date: Mon, 11 Apr 2011 13:37:26 +0530 Subject: [FFmpeg-user] piping images into ffmpeg In-Reply-To: <4DA2A621.4090704@bbc.co.uk> References: <4DA2A621.4090704@bbc.co.uk> Message-ID: On Mon, Apr 11, 2011 at 12:26 PM, Tim Nicholson wrote: > Err, if "-f image2" is correct then why change it to "-f image2pipe" ? image2 is correct for getting input from consecutively numbered files. image2pipe is for getting from pipe. > I think image2pipe is more for output formatting. yuv4mpegpipe certainly is. Don't read '2pipe' as 'to pipe'; notice that first option also has 2 in the end. :-) -- ?????? (Akilan R) [ blog.akilan.in ] *I should have no use for a paradise in which I should be deprived of the right to prefer hell.* --Jean Rostand From david at electric-spoon.com Mon Apr 11 14:26:22 2011 From: david at electric-spoon.com (David Pottage) Date: Mon, 11 Apr 2011 13:26:22 +0100 Subject: [FFmpeg-user] Live transcoding to an rtp stream is unreliable. In-Reply-To: <4D9DB68E.5010008@orange-ftgroup.com> References: <4D9D9C89.7010702@electric-spoon.com> <4D9DA259.1040701@orange-ftgroup.com> <4D9DB0F8.4020908@electric-spoon.com> <4D9DB68E.5010008@orange-ftgroup.com> Message-ID: <4DA2F36E.3000808@electric-spoon.com> On 07/04/11 14:05, Alexandre Ferrieux wrote: > Look at these lines. Aren't they ominous ? > >>> [mpeg2video @ 0xa193990]ac-tex damaged at 11 22=4.52 bitrate= >>> 73.2kbits/s dup=12 drop=0 >>> [mp2 @ 0xa1940f0]incomplete frame >>> Error while decoding stream #0.1 > > Clearly this indicates a problem in the input bitstream. Maybe some > network latency stalled the TCP (on the http leg) a bit, producing an > oveerun at the source, resulting in TS discontinuity ? > > Given the above observations, I'd concentrate on the input decoding > step, and on the video only. For this, use vocdec copy and the > container format that is the simplest for the given codec, here > mpeg2video: > > ffmpeg -loglevel verbose \ > -pix_fmt yuv422p -s 720x576 -i http://david-laptop:8000/4228 \ > \ > -an -sn -vcodec copy -f mpeg2video - > out.m2v > > With this, you're guaranteed that ffmpeg will not be the limiting > factor (unlike a possibly costly x264), since basically it just has to > write to disk (you can even use /dev/null instead of out.m2v, just in > case). I tried that with this command line: ffmpeg -loglevel debug -pix_fmt yuv422p -s 720x576 -i http://10.10.10.10:8000/4228 -an -sn -vcodec copy -f mpeg2video - > /dev/null > FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers > built on Apr 4 2011 16:31:39 with gcc 4.4.3 > configuration: --enable-postproc --extra-ldflags=-static > --extra-libs='-lvorbis -logg -lxvidcore -lx264 -lopencore-amrnb > -lopencore-amrwb -lfaad -lfaac -lm -lpthread' --enable-gpl > --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libvorbis --enable-libx264 > --enable-libxvid --enable-nonfree --enable-shared --enable-version3 > --enable-libfaad --enable-small --disable-ffplay --disable-ffserver > --disable-doc > libavutil 50.15. 1 / 50.15. 1 > libavcodec 52.72. 2 / 52.72. 2 > libavformat 52.64. 2 / 52.64. 2 > libavdevice 52. 2. 0 / 52. 2. 0 > libswscale 0.11. 0 / 0.11. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [NULL @ 0xa89cae0]Probed with size=2048 and score=100 > [mpegts @ 0xa89cae0]stream=0 stream_type=2 pid=262 prog_reg_desc= > [mpegts @ 0xa89cae0]stream=1 stream_type=3 pid=263 prog_reg_desc= > [mpegts @ 0xa89cae0]stream=2 stream_type=3 pid=264 prog_reg_desc= > [mpegts @ 0xa89cae0]stream=3 stream_type=6 pid=267 prog_reg_desc= > [mpegts @ 0xa89cae0]stream=4 stream_type=5 pid=26b prog_reg_desc= > [mpegts @ 0xa89cae0]stream=5 stream_type=b pid=3f0 prog_reg_desc= > [mpegts @ 0xa89cae0]stream=6 stream_type=b pid=3f1 prog_reg_desc= > [mpegts @ 0xa89cae0]stream=7 stream_type=b pid=3f2 prog_reg_desc= > [mpegts @ 0xa89cae0]stream=8 stream_type=b pid=28a prog_reg_desc= > [mpegts @ 0xa89cae0]stream=9 stream_type=b pid=28b prog_reg_desc= > [mpegts @ 0xa89cae0]stream=10 stream_type=b pid=28c prog_reg_desc= > [mpeg2video @ 0xa8a2990]mpeg_decode_postinit() failure > Last message repeated 7 times > [mpegts @ 0xa89cae0]max_analyze_duration reached > [mpegts @ 0xa89cae0]Estimating duration from bitrate, this may be > inaccurate > Input #0, mpegts, from 'http://10.10.10.10:8000/4228': > Duration: N/A, start: 33989.713744, bitrate: 15256 kb/s > Program 4228 > Stream #0.0[0x262], 129, 1/90000: Video: mpeg2video, yuv420p, > 720x576 [PAR 64:45 DAR 16:9], 1/50, 15000 kb/s, 26.24 fps, 25 tbr, 90k > tbn, 50 tbc > Stream #0.1[0x263](eng), 211, 1/90000: Audio: mp2, 48000 Hz, 2 > channels, s16, 256 kb/s > Stream #0.2[0x264](eng), 0, 1/90000: Audio: mp3, 0 channels, s16 > Stream #0.3[0x267](eng), 1, 1/90000: Subtitle: dvbsub > Stream #0.4[0x26b], 0, 1/90000: Data: 0x0005 > Stream #0.5[0x3f0], 0, 1/90000: Data: 0x000b > Stream #0.6[0x3f1], 0, 1/90000: Data: 0x000b > Stream #0.7[0x3f2], 0, 1/90000: Data: 0x000b > Stream #0.8[0x28a], 0, 1/90000: Data: 0x000b > Stream #0.9[0x28b], 0, 1/90000: Data: 0x000b > Stream #0.10[0x28c], 0, 1/90000: Data: 0x000b > Output #0, mpeg2video, to 'pipe:': > Metadata: > encoder : Lavf52.64.2 > Stream #0.0, 0, 1/90000: Video: mpeg2video, yuv420p, 720x576 [PAR > 64:45 DAR 16:9], 1/25, q=2-31, 15000 kb/s, 90k tbn, 25 tbc > Stream mapping: > Stream #0.0 -> #0.0 > Press [q] to stop encoding > [mpegts @ 0xa89cae0]Invalid timestamps stream=0, pts=7076, > dts=8589930868, size=23135 > [mpegts @ 0xa89cae0]Invalid timestamps stream=0, pts=3684, > dts=8589927476, size=10251 > [mpegts @ 0xa89cae0]Invalid timestamps stream=0, pts=292, > dts=8589924084, size=8591 > frame=6744789 fps= 25 q=-1.0 Lsize=95343472kB time=269792.20 > bitrate=2895.0kbits/s > video:95343472kB audio:0kB global headers:0kB muxing overhead 0.000000% > Received signal 2: terminating. It ran all weekend until I stopped it myself, so it looks like the input processing is stable. > Maybe ffmpeg is not proceeding because it is still expecting a GOP > start (a key frame). Do you have an idea of the gop size in your > source ? Is it stable ? > > Also, clearly the reproducibility of a live stream is poor. I'd > suggest saving the TS stream to a file for post mortem analysis. The input stream is broadcast TV. From memory there is an I frame about once every 2 seconds, and about 3 P frames per second. I saved a local copy of the input stream with wget: > wget http://10.10.10.10:8000/4228 -O BBC2_4228.dvb I then ran the live transcode command line again replacing the input URL with the filename: ffmpeg -loglevel verbose -pix_fmt yuv422p -s 720x576 -i BBC2_4228.dvb -vn -sn -acodec libfaac -ar 48000 -ab 64000 -ac 2 -async 1 -flags +global_header -f rtp rtp://web-01:9022 -an -sn -vcodec libx264 -s 320x240 -fpre resources/ffpreset/libx264-hq.ffpreset -fpre resources/ffpreset/libx264-ipod320.ffpreset -vglobal 1 -f rtp rtp://web-01:9020 -newvideo This time the transcode ran to completion and did not fall over. > If this setup reproduces the behavior, you have a source overrun or > corruption, either due to the network or something inside the source > machine itself (CPU bound ?). > > If not, then ffmpeg's slowness was the reason for the overrun (not > swallowing input fast enough), most likely due to the x264 CPU demand; > in that case, try with a more modest preset, resampling to smaller > size, etc. Could there be another explanation? I have been doing these tests on a powerful 8 Core machine, that is able to concurrently transcode 5 similar input streams to x264 using similar codec settings but using VLC. These 5 transcode jobs use about 60% of the machine's CPU together. It sounds unlikey that one transcode process with ffmpeg would tax the machine if it can normaly do five such transcode processes using vlc. -- David Pottage From mahakcay at gmail.com Mon Apr 11 17:18:17 2011 From: mahakcay at gmail.com (Mahmut Akcay) Date: Mon, 11 Apr 2011 16:18:17 +0100 Subject: [FFmpeg-user] Compile error when building for iOS Message-ID: I'm having the following problem when cross-compiling for iOS 4.2 : AS libavcodec/arm/dsputil_vfp.o CC libavcodec/arm/fft_fixed_init_arm.o AS libavcodec/arm/fft_fixed_neon.o Too many arguments to macro movrel at /usr/local/bin/gas-preprocessor.plline 255, line 174. make: *** [libavcodec/arm/fft_fixed_neon.o] Error 255 Below is the config params I'm using: ./configure --enable-cross-compile --arch=arm --target-os=darwin --cc='/Developer/Platforms/iPhoneOS.platform/Developer/usr/bin/gcc -arch armv7' --sysroot=/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.2.sdk --cpu=cortex-a8 --enable-pic --disable-shared --enable-static --disable-ffmpeg --disable-ffserver --disable-ffplay --disable-encoders --enable-extra-warnings --disable-debug I'm using the latest version of gas-preprocessor.pl fromhttps://github.com/yuvi/gas-preprocessor. Any help? Thanks in advance. mahmut From sarkiaponius at alice.it Mon Apr 11 20:37:49 2011 From: sarkiaponius at alice.it (andrea) Date: Mon, 11 Apr 2011 20:37:49 +0200 Subject: [FFmpeg-user] How to speed up RTSP... In-Reply-To: References: <1302435287.19282.1.camel@debian> <1302471407.23874.5.camel@debian> Message-ID: <1302547069.4259.2.camel@debian> On Sun, 2011-04-10 at 17:56 -0400, sean darcy wrote: > On 04/10/2011 05:36 PM, andrea wrote: > > On Sun, 2011-04-10 at 12:50 -0400, sean darcy wrote: > >> On 04/10/2011 07:34 AM, andrea wrote: > >>> ffmpeg -i > >>> rtsp://mm6.rai.it/radiofonia/radio3/napoli/uomini_profeti/2010/uomini_profeti2010_09_19.ra test.ra > >>> > >> I'm amazed you got rtsp to work at all. Here's what I get from ffmpeg > >> -protocols: > > > > I swear it works, tried right now. I got --enable-librtmp, too. I cannot > > figure out why it works, but it does. The problem is that it seems there > > is no option to accelerate the stream someway. With mplayer, as I think > > I said, the -bandwidth option does the trick, if you set it to a quite > > high value (say 4000000 or such). But I caould not find something like > > that for ffmpeg. > > > > Best regards. > >> > >> ...... > >> IO. rtmp > >> IO. rtmpt > >> IO. rtmpe > >> IO. rtmpte > >> IO. rtmps > >> IO. rtp > >> IO. tcp > >> IO. udp > >> > >> And that's by building with --enable-librtmp. > >> > >> sean > >> > > Well glad it works for you. In addition to accelerating the stream, I > wish ffmpeg (and mplayer) buffered it. That way you could pause, let the > buffer fill, then play without interruptions or stuttering. Well, actually my goal is not to watch/listen to a stream, but just download it. With mplayer I was able to download a 45 minutes long stream in less than one minute. I think it's strange that such a feature is missing in ffmpeg. Best regards. > > sean > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From joolzg at btinternet.com Mon Apr 11 22:09:59 2011 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Mon, 11 Apr 2011 21:09:59 +0100 (BST) Subject: [FFmpeg-user] DVB Subtitles In-Reply-To: <1302547069.4259.2.camel@debian> Message-ID: <892773.38384.qm@web86402.mail.ird.yahoo.com> Two questions 1. has anybody managed to get DVB Subtitles taken from a MPEG2-TS remuxed into another MPEG2-TS after re-encoding the Audio/Video, if so HOW 2. If the above is not possible, is it possible to have the DVB Subtitles displayed on the uncompressed video so when its encoded they are "burned in" joolz From acmilan_2008 at hotmail.com Tue Apr 12 04:38:51 2011 From: acmilan_2008 at hotmail.com (Erik) Date: Tue, 12 Apr 2011 02:38:51 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Can=27t_locate_entry_av=5Faes=5Fcrpyt_in_?= =?utf-8?q?avutil-49=2Edll?= Message-ID: I just start to write some application using FFMPEG, the application itself is OK but before it start running I got this error message I think I may have a incompatible version of avutil-49.dll...I download FFMPEG dev packet here(it seems doesn't include those DLLs)and I download the avformat-52.dll, avcodec-51.dll and avutil-49.dll. At least it seems no error in avformat-52.dll, but the avutil-49.dll get wrong My avcodec-51.dll: 6,212,096 bytes avformat-52.dll: 713,728 bytes avutil-49.dll: 19,968 bytes I even downloaded a avutil.dll(57,344 bytes)and tried to rename it to avutil- 49.dll but it still error at different entry. How to get the correct DLL? I'm sorry I don't know how to use tools like msys at all so don't just tell me to compile FFMPEG myself thx From acmilan_2008 at hotmail.com Tue Apr 12 11:18:35 2011 From: acmilan_2008 at hotmail.com (Erik) Date: Tue, 12 Apr 2011 09:18:35 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Can=27t_locate_entry_av=5Faes=5Fcrpyt_in_?= =?utf-8?q?avutil-49=2Edll?= References: Message-ID: I just realized what I downloaded are "app" dlls, not dev version. But I downloaded the compiled shared dlls here, I get another problem of the same type It says can't locate entry av_freep in avformat-52.dll I was developing on windows 7 and when I right-click the computer icon, it says 32-bit system. I downloaded 32-bit shared dll too, I don't think I downloaded wrong version of dlls this time From gavr.mail at gmail.com Tue Apr 12 09:26:38 2011 From: gavr.mail at gmail.com (Kirill Gavrilov) Date: Tue, 12 Apr 2011 11:26:38 +0400 Subject: [FFmpeg-user] Can't locate entry av_aes_crpyt in avutil-49.dll In-Reply-To: References: Message-ID: > > I even downloaded a avutil.dll(57,344 bytes)and tried to rename it to > avutil- > 49.dll but it still error at different entry. How to get the correct DLL? > I'm > sorry I don't know how to use tools like msys at all so don't just tell me > to > compile FFMPEG myself thx > Your avutil-49 is too old. The modern version is avutil-50. Also you should not ever try to cross FFmpeg libraries from different sources - they have too much cross dependencies and you'll got a broken setup in most cases. Win binaries you can download here http://win32.libav.org/ and here http://ffmpeg.arrozcru.org/ (I not test them). 2011/4/12 Erik > I just start to write some application using FFMPEG, the application itself > is > OK but before it start running I got this error message > I think I may have a incompatible version of avutil-49.dll...I download > FFMPEG > dev packet here(it seems doesn't include those DLLs)and I download the > avformat-52.dll, avcodec-51.dll and avutil-49.dll. At least it seems no > error > in avformat-52.dll, but the avutil-49.dll get wrong > My avcodec-51.dll: 6,212,096 bytes > avformat-52.dll: 713,728 bytes > avutil-49.dll: 19,968 bytes > I even downloaded a avutil.dll(57,344 bytes)and tried to rename it to > avutil- > 49.dll but it still error at different entry. How to get the correct DLL? > I'm > sorry I don't know how to use tools like msys at all so don't just tell me > to > compile FFMPEG myself thx > _________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user ----------------------------------------------- Kirill Gavrilov, Software designer. From ankuj2004 at gmail.com Wed Apr 13 05:32:25 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Wed, 13 Apr 2011 09:02:25 +0530 Subject: [FFmpeg-user] 4CIF H263 video fails to decode In-Reply-To: <4D9F57D1.3020402@mdsh.com> References: <4D9F57D1.3020402@mdsh.com> Message-ID: Hi Mark The command I am giving is ffmpeg -i h263_704x576_15fps_3370kbps.263 4cif.yuv The untrimmed output of the command is FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers built on Apr 8 2011 14:08:35 with gcc 4.1.2 20080704 (Red Hat 4.1.2-46) configuration: --disable-demuxer=v4l --disable-demuxer=v4l2 --disable-indevs libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.11. 0 / 0.11. 0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f9163c Returning from fn av_probe_input_format2() fmt=0, score_max=70f9163c, scor=0 No. of chunks = 512 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=0, score_max=70f911bc, scor=0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=0, score_max=70f911bc, scor=0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=0, score_max=70f911bc, scor=0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=0, score_max=70f911bc, scor=0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=0, score_max=70f911bc, scor=0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=0, score_max=70f911bc, scor=0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=0, score_max=70f911bc, scor=0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=0, score_max=70f911bc, scor=0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=0, score_max=70f911bc, scor=0 Inside fn av_probe_input_format2(), first_iformat=b61680, score_max=70f911bc Returning from fn av_probe_input_format2() fmt=b60360, score_max=70f911bc, scor=0 [NULL @ 0x5115420]Format detected only with low score of 24, misdetection possible! [mp3 @ 0x51166d0]Header missing Last message repeated 17 times [mp3 @ 0x51166d0]big_values too big [mp3 @ 0x5115420]max_analyze_duration reached [mp3 @ 0x5115420]Estimating duration from bitrate, this may be inaccurate Input #0, mp3, from 'h263_704x576_15fps_3370kbps.263': Duration: 01:45:31.25, start: 0.000000, bitrate: 96 kb/s Stream #0.0: Audio: mp2, 44100 Hz, 2 channels, s16, 96 kb/s Output #0, rawvideo, to '4cif.yuv': Output file #0 does not contain any stream Ankuj On Sat, Apr 9, 2011 at 12:15 AM, Mark Himsley wrote: > On 08/04/11 09:15, Ankuj Gupta wrote: > >> Hi, >> >> I am trying to decode a H263 4CIF(704x576) resolution file with ffmpeg, >> but >> it is detected as mp3 file and fails to get decoded. I tried with lower >> resolution H263 files they get decoded easily as well as 16CIF. I am using >> ffmpeg version 0.6.1 >> > > Please send the command you are using and the untrimmed output of that > command. > > -- > Mark > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From tristan at trutna.net Wed Apr 13 06:43:28 2011 From: tristan at trutna.net (Tristan Trutna) Date: Tue, 12 Apr 2011 21:43:28 -0700 Subject: [FFmpeg-user] Metadata muxer/demuxer documentation? Message-ID: The metadata documentation for ffmetadata ( http://www.ffmpeg.org/ffmpeg.html#TOC130 ) clearly explains the fileformat, but I am unable to find documentation regarding input or output of these files. Specifically, I would like to create an ffmetadata file from an input mp3 file, edit that file, and then transcode the mp3 file into an AAC file using the edited ffmetadata file to provide metadata to the new AAC file rather than using the -map_meta_data argument. What is the argument I should be using to output or input ffmetadata files? Thank you, Tristan From tim.nicholson at bbc.co.uk Wed Apr 13 09:27:00 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Wed, 13 Apr 2011 08:27:00 +0100 Subject: [FFmpeg-user] 4CIF H263 video fails to decode In-Reply-To: References: <4D9F57D1.3020402@mdsh.com> Message-ID: <4DA55044.2060408@bbc.co.uk> On 13/04/11 04:32, Ankuj Gupta wrote: > Hi Mark > > The command I am giving is > > ffmpeg -i h263_704x576_15fps_3370kbps.263 4cif.yuv > > The untrimmed output of the command is > > FFmpeg version 0.6.1, Copyright (c) 2000-2010 the FFmpeg developers > built on Apr 8 2011 14:08:35 with gcc 4.1.2 20080704 (Red Hat 4.1.2-46) > configuration: --disable-demuxer=v4l --disable-demuxer=v4l2 > --disable-indevs > libavutil 50.15. 1 / 50.15. 1 > libavcodec 52.72. 2 / 52.72. 2 > libavformat 52.64. 2 / 52.64. 2 > libavdevice 52. 2. 0 / 52. 2. 0 > libswscale 0.11. 0 / 0.11. 0 > [..] > Input #0, mp3, from 'h263_704x576_15fps_3370kbps.263': > Duration: 01:45:31.25, start: 0.000000, bitrate: 96 kb/s > Stream #0.0: Audio: mp2, 44100 Hz, 2 channels, s16, 96 kb/s > Output #0, rawvideo, to '4cif.yuv': > Output file #0 does not contain any stream So ffmpeg thinks the input file contains only mp2 audio and you are trying to output this to a video only format... So either the auto detection has failed in which case try forcing the input format, or the file doesn't contain what you think it should. -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From zhanwu.xiong at gmail.com Wed Apr 13 21:48:15 2011 From: zhanwu.xiong at gmail.com (Zhanwu Xiong) Date: Wed, 13 Apr 2011 21:48:15 +0200 Subject: [FFmpeg-user] how to specify search folder for external lib while compiling? Message-ID: Hi all, I tried to compile and build ffmpeg in my home folder since I don't have sudo privilege, and I have the following questions: First, I don't know how to specify search directories for external libraries while running *configure*? I try to enable libx264, which is installed in my home folder instead of '/usr/local/', but *./configure --enable-libx264* returns me that libx264 can not be found. I proceeded without libx264, because I noticed that h264 decoder is enabled anyway. Does it mean that ffmpeg have a native h264 decoder? After installation, I tried to extract frames from a h264 encoded video and save it to png files: *ffmpeg -i VIRAT_S_000201_03_000640_000672.mp4 -r 30 -f image2 frame%04d.png * I got the frames, however there are two message that I couldn't understand 1. *Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 30.00 (30/1)* 2. *Incompatible pixel format 'yuv420p' for codec 'png', auto-selecting format 'rgb24'* My guess for the second message is that the original pixel format yuv420p couldn't be used for the output 'png' file, so it is automatically converted into 'rgb24' format, and this is done by calling *swscale*, right? For the first one, the only thing I know is that the frame rate of the input video is 30 frames/sec, which is obtained by viewing the property of it in nautilus, but what is the container frame rate? best regards /zhanwu From bahamutzero8825 at gmail.com Wed Apr 13 21:55:18 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Wed, 13 Apr 2011 14:55:18 -0500 Subject: [FFmpeg-user] how to specify search folder for external lib while compiling? In-Reply-To: References: Message-ID: <4DA5FFA6.6030609@gmail.com> On 2011.04.13 02:48 PM, Zhanwu Xiong wrote: > I proceeded without libx264, because I noticed that h264 decoder is enabled > anyway. Does it mean that ffmpeg have a native h264 decoder? x264 is an AVC/H.264 /encoder/ . libavcodec, which is part of FFmpeg can decode AVC/H.264. From zhanwu.xiong at gmail.com Wed Apr 13 22:01:16 2011 From: zhanwu.xiong at gmail.com (Zhanwu Xiong) Date: Wed, 13 Apr 2011 22:01:16 +0200 Subject: [FFmpeg-user] how to specify search folder for external lib while compiling? In-Reply-To: <4DA5FFA6.6030609@gmail.com> References: <4DA5FFA6.6030609@gmail.com> Message-ID: I thought both are encoder and decoder, now I see. No surprise that my build of ffmpeg support h264 decoding but not encoding. best /zhanwu On Wed, Apr 13, 2011 at 9:55 PM, Andrew Berg wrote: > On 2011.04.13 02:48 PM, Zhanwu Xiong wrote: > > I proceeded without libx264, because I noticed that h264 decoder is > enabled > > anyway. Does it mean that ffmpeg have a native h264 decoder? > x264 is an AVC/H.264 /encoder/ . libavcodec, which is part of FFmpeg can > decode AVC/H.264. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From lukas.oesterreicher at inode.at Wed Apr 13 22:51:27 2011 From: lukas.oesterreicher at inode.at (=?iso-8859-15?Q?Lukas_=D6sterreicher?=) Date: Wed, 13 Apr 2011 22:51:27 +0200 Subject: [FFmpeg-user] multi-threaded video decoding options vs mencoder Message-ID: <85276957.20110413225127@inode.at> Hello. I'm trying to use ffmpeg or mencoder to encode video for the purpose of streaming to a flash player, which is used inside subsonic. The current encoding option I am using is: ffmpeg -ss %o -i %s -async 1 -b %bk -s %wx%h -ar 44100 -ac 2 -v 0 -threads 4 -f flv -vcodec libx264 -fpre /var/subsonic/transcode/presets/libx264-ultrafast.ffpreset - (where o=offset in seconds, s=source file, b=bitrate w=width, h=height). This works ok, but since this is running on a NAS box with only has a dual core atom processor, processing time is limited and it can only do SD video properly, but I want to be able to stream 720p content. I tried around and found that when I use mencoder I can encode much faster. With a 720p h264 video i get 16 fps with ffmpeg and 28(!!) with mencoder. So I would either like to improve ffmpeg speed or make mencoder output flash (preferably h264). Here some sample comments with which i got the mentioned fps speeds: ffmpeg -i "/share/MD0_DATA/Multimedia/video1.mkv" -b 500k -s 800x448 -acodec libmp3lame -ab 96k -ar 44100 -ac 2 -f avi -vcodec libx264 -threads 4 -fpre /var/subsonic/transcode/presets/libx264-ultrafast.ffpreset "/share/MD0_DATA/Multimedia/video1-ffmpeg.avi" mencoder "/share/MD0_DATA/Multimedia/video1.mkv" -lavdopts fast:threads=4 -oac mp3lame -lameopts cbr:br=96:mode=1 -srate 44100 -ovc x264 -x264encopts preset=ultrafast:threads=4:bitrate=500 -vf scale=800:448 -of avi -o "/share/MD0_DATA/Multimedia/video1-mencoder.avi" The big difference between the two is -lavdopts fast:threads=4. Without that, mencoder is about as slow as ffmpeg. I hear that ffmpeg does support multi-threaded decoding, but apparently that is only limited to a small set of video conent. I get this line with ffmpeg [h264 @ 0x8d7c940] Cannot parallelize deblocking type 1, decoding such frames in sequential order However mencoder can somehow decode the same video much better with the provided lavdopts optins. So: is there a way to to make ffmpeg decode it the same fast way? If not, can you help me get mencoder output a flash compatible stream? Apparently mencoder cannot support mp4 properly (is that right?), but it does support flv. However when I try to encode flv like this: mencoder "/share/MD0_DATA/Multimedia/video1.mkv" -of lavf -lavdopts fast:threads=4 -oac faac -faacopts br=96 -channels 2 -srate 44100 -ovc lavc -lavcopts vcodec=flv:vbitrate=500 -vf scale=800:448 -o "/share/MD0_DATA/Multimedia/video1.flv" I get this error: [flv @ 0x89b8c60]dimensions not set Floating point exception This also happens with the build i am using with provided here: http://blog.save-privacy.de/index.php?/archives/10-Flash-Videos-mit-dem-Mencoder-erstellen.html which should work according to the author. I use the latest git source for my compile. Maybe the recent version has a problem that an older versions do not? If so, please tell me which version to use? (Note: I also had to revert to a November 2010 version of x264 since the latest version apparently has problems with encoding and often giving out this error: mencoder: encoder/slicetype.c:1632: x264_rc_analyse_slice: Assertion `cost >= 0' failed. see http://forum.qnap.com/viewtopic.php?f=45&t=43340&p=192041#p192041 for more details) Thanx in advance! Lukas From alexandre.ferrieux at orange-ftgroup.com Thu Apr 14 09:56:35 2011 From: alexandre.ferrieux at orange-ftgroup.com (Alexandre Ferrieux) Date: Thu, 14 Apr 2011 09:56:35 +0200 Subject: [FFmpeg-user] Overlay and varied framerates Message-ID: <4DA6A8B3.10805@orange-ftgroup.com> Hi, I'm using the 'overlay' vfilter, along with the 'movie' source (as a patch), with great success. (Thanks again guys). Now when I do this on live streams (from webcams), I see a differential latency settle: cam A is rendered realtime, camB a constant 2 seconds behind. Permutations show this is not linked to the order in the overlay graph, but to the cameras themselves. I suspect this is due to a slight but systematic difference in the cams' effective framerates: an intrinsic delta between the 2 clocks gives a linear drift, and the delay saturates when overrun occurs (ie all various buffers are filled) Q1: how is time handled in the case of a multiple-input vfilter like overlay ? Assuming the following timeline: A: ....F1..F2.......F3..... B: ..G1.G2.G3.G4.G5........ am I right to think that we get: A+B: .....H1.H2.......H3..... with: H1=F1+G1, H2=F2+G2, H3=F3+G3 ? Q2: If yes, what can I do so that, instead, overlay just "samples" the inputs in a nonblocking manner at regular intervals set by the target output framerate, thus: A+B: ....J1..J2..J3..J4..J5.. with: J1=F1+G1, J2=F2+G3, J3=F2+G4, J4=F2+G5, J5=F3+G5 ? Thanks in advance, -Alex From arvindkumargupta at gmail.com Thu Apr 14 10:05:08 2011 From: arvindkumargupta at gmail.com (Arvind Gupta) Date: Thu, 14 Apr 2011 13:35:08 +0530 Subject: [FFmpeg-user] ffmpeg binary usage in linux Message-ID: Hi I want to use ffmpeg for video conversion in Linux but don't want whole source code to compile and then run, rather than if we can install ffmpeg binary directly and use it, it would be great. Do anybody has idea from where I can get that binary to install and how to use it? -- Arvind From gavr.mail at gmail.com Thu Apr 14 10:15:30 2011 From: gavr.mail at gmail.com (Kirill Gavrilov) Date: Thu, 14 Apr 2011 12:15:30 +0400 Subject: [FFmpeg-user] ffmpeg binary usage in linux In-Reply-To: References: Message-ID: 2011/4/14 Arvind Gupta > Hi > I want to use ffmpeg for video conversion in Linux but don't want whole > source code to compile and then run, rather than if we can install ffmpeg > binary directly and use it, it would be great. > Do anybody has idea from where I can get that binary to install and how to > use it? > You should at least name your Linux and your conversion targets because different distros are not binary compatible and FFmpeg may depends on another libraries (libx264 for example if you want to encode x264 video). ----------------------------------------------- Kirill Gavrilov, Software designer. From dave.bevan at bbc.co.uk Thu Apr 14 10:27:54 2011 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Thu, 14 Apr 2011 09:27:54 +0100 Subject: [FFmpeg-user] ffmpeg binary usage in linux In-Reply-To: References: Message-ID: >Hi >I want to use ffmpeg for video conversion in Linux but don't want whole source code to compile and then run, rather than if we >can install ffmpeg binary directly and use it, it would be great. >Do anybody has idea from where I can get that binary to install and how to use it? > >-- >Arvind If you're on Ubuntu, then "sudo apt-get install ffmpeg". --Dave. http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From arvindkumargupta at gmail.com Thu Apr 14 11:24:52 2011 From: arvindkumargupta at gmail.com (Arvind Gupta) Date: Thu, 14 Apr 2011 14:54:52 +0530 Subject: [FFmpeg-user] ffmpeg binary usage in linux In-Reply-To: References: Message-ID: Hi Kirill I am using RHEL 5.0 64 bit and conversion target is mp4. Thanks Arvind On Thu, Apr 14, 2011 at 1:45 PM, Kirill Gavrilov wrote: > 2011/4/14 Arvind Gupta > > > Hi > > I want to use ffmpeg for video conversion in Linux but don't want whole > > source code to compile and then run, rather than if we can install ffmpeg > > binary directly and use it, it would be great. > > Do anybody has idea from where I can get that binary to install and how > to > > use it? > > > You should at least name your Linux and your conversion targets because > different distros are not binary compatible and FFmpeg may depends on > another libraries (libx264 for example if you want to encode x264 video). > ----------------------------------------------- > Kirill Gavrilov, > Software designer. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Arvind Kumar Gupta From s9chhacc at stud.uni-saarland.de Thu Apr 14 13:40:52 2011 From: s9chhacc at stud.uni-saarland.de (Christopher Haccius) Date: Thu, 14 Apr 2011 13:40:52 +0200 Subject: [FFmpeg-user] Saving Video Frames as Images Message-ID: <4DA6DD44.5050806@stud.uni-saarland.de> I am trying to convert a the frames of a video into an image sequence. I used to accomplish this task by the following line: ffmpeg -i input.mpg -y -f image2 out/output%04d.png Yesterday I downloaded the Snapshot from git://git.videolan.org/ffmpeg.git to try to implement some small changes in the ffmpeg code. When I build this snapshot the code line from above does not seem to work any more. Does anyone know why? How far are the git-repositories different from the releases? Has anything changed in the procedure of splitting videos into frames? Any ideas are very much appreciated! Best Chris The following lines are copied from the terminal after executing the input line mentioned above (first the output how I it used to work, below the output of the new version). --------------------------Old (working) output------------------------- ffmpeg -i input.mpg -y -f image2 out/output%04d.png FFmpeg version 0.6-4:0.6-2ubuntu6.1, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 31 2011 18:43:47 with gcc 4.4.5 configuration: --extra-version=4:0.6-2ubuntu6.1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-vaapi --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg @ 0x1aef420]max_analyze_duration reached Input #0, mpeg, from 'input.mpg': Duration: 00:00:10.76, start: 0.500000, bitrate: 7564 kb/s Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 1004x1004 [PAR 1:1 DAR 1:1], 104857 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 23.98 tbc [swscaler @ 0x1af8550]No accelerated colorspace conversion found from yuv420p to rgb24. Output #0, image2, to 'out/output%04d.png': Metadata: encoder : Lavf52.64.2 Stream #0.0: Video: png, rgb24, 1004x1004 [PAR 1:1 DAR 1:1], q=2-31, 200 kb/s, 90k tbn, 23.98 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 259 fps= 11 q=0.0 Lsize= -0kB time=10.80 bitrate= -0.0kbits/s video:128012kB audio:0kB global headers:0kB muxing overhead -100.000017% --------------New (not working) output--------------------------------------- ./ffmpeg -i input.mpg -y -f image2 out/output%04d.png FFmpeg version UNKNOWN, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 13 2011 18:45:22 with gcc 4.4.5 configuration: libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.119. 0 / 52.119. 0 libavformat 52.106. 0 / 52.106. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 78. 0 / 1. 78. 0 libswscale 0. 13. 0 / 0. 13. 0 [mpeg @ 0x1b656e0] max_analyze_duration reached Input #0, mpeg, from 'input.mpg': Duration: 00:00:10.76, start: 0.500000, bitrate: 7564 kb/s Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 1004x1004 [PAR 1:1 DAR 1:1], 104857 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 23.98 tbc *Output #0, image2, to 'out/output%04d.png': Stream #0.0: Invalid Codec type -1 Could not find input stream matching output stream #0.0* From aviadr1 at gmail.com Thu Apr 14 15:13:45 2011 From: aviadr1 at gmail.com (aviad rozenhek) Date: Thu, 14 Apr 2011 16:13:45 +0300 Subject: [FFmpeg-user] Live transcoding to an rtp stream is unreliable. In-Reply-To: <4D9DB68E.5010008@orange-ftgroup.com> References: <4D9D9C89.7010702@electric-spoon.com> <4D9DA259.1040701@orange-ftgroup.com> <4D9DB0F8.4020908@electric-spoon.com> <4D9DB68E.5010008@orange-ftgroup.com> Message-ID: On Thu, Apr 7, 2011 at 16:05, Alexandre Ferrieux < alexandre.ferrieux at orange-ftgroup.com> wrote: > On 07/04/2011 14:41, David Pottage wrote: > >> On 07/04/11 12:39, Alexandre Ferrieux wrote: >> >>> >>> Look at these lines. Aren't they ominous ? > > [mpeg2video @ 0xa193990]ac-tex damaged at 11 22=4.52 bitrate= >>> 73.2kbits/s dup=12 drop=0 >>> [mp2 @ 0xa1940f0]incomplete frame >>> Error while decoding stream #0.1 >>> >> > Clearly this indicates a problem in the input bitstream. Maybe some network > latency stalled the TCP (on the http leg) a bit, producing an oveerun at the > source, resulting in TS discontinuity ? > > > FYI reading from UDP in ffmpeg is completely broken for any bitrates above 800kbps the loop that reads from UDP also goes through to demuxing, thus a lot of UDP packets are simply dropped. the solution is to implement a tight UDP loop in a custom application, or wait until this is solved in libavformat -- Aviad Rozenhek From s9chhacc at stud.uni-saarland.de Thu Apr 14 17:35:10 2011 From: s9chhacc at stud.uni-saarland.de (Christopher Haccius) Date: Thu, 14 Apr 2011 17:35:10 +0200 Subject: [FFmpeg-user] Saving Video Frames as Images In-Reply-To: <4DA6DD44.5050806@stud.uni-saarland.de> References: <4DA6DD44.5050806@stud.uni-saarland.de> Message-ID: <4DA7142E.9050206@stud.uni-saarland.de> On 04/14/2011 01:40 PM, Christopher Haccius wrote: > I am trying to convert a the frames of a video into an image sequence. > I used to accomplish this task by the following line: > > ffmpeg -i input.mpg -y -f image2 out/output%04d.png > > Yesterday I downloaded the Snapshot from > git://git.videolan.org/ffmpeg.git to try to implement some small > changes in the ffmpeg code. When I build this snapshot the code line > from above does not seem to work any more. Does anyone know why? How > far are the git-repositories different from the releases? Has anything > changed in the procedure of splitting videos into frames? > > Any ideas are very much appreciated! > Best > Chris > > The following lines are copied from the terminal after executing the > input line mentioned above (first the output how I it used to work, > below the output of the new version). > > --------------------------Old (working) output------------------------- > ffmpeg -i input.mpg -y -f image2 out/output%04d.png > FFmpeg version 0.6-4:0.6-2ubuntu6.1, Copyright (c) 2000-2010 the > FFmpeg developers > built on Mar 31 2011 18:43:47 with gcc 4.4.5 > configuration: --extra-version=4:0.6-2ubuntu6.1 --prefix=/usr > --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib > --enable-libgsm --enable-libschroedinger --enable-libspeex > --enable-libtheora --enable-libvorbis --enable-vaapi --enable-pthreads > --enable-zlib --enable-libvpx --disable-stripping > --enable-runtime-cpudetect --enable-gpl --enable-postproc > --enable-x11grab --enable-libdc1394 --enable-shared --disable-static > libavutil 50.15. 1 / 50.15. 1 > libavcodec 52.72. 2 / 52.72. 2 > libavformat 52.64. 2 / 52.64. 2 > libavdevice 52. 2. 0 / 52. 2. 0 > libavfilter 1.19. 0 / 1.19. 0 > libswscale 0.11. 0 / 0.11. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [mpeg @ 0x1aef420]max_analyze_duration reached > Input #0, mpeg, from 'input.mpg': > Duration: 00:00:10.76, start: 0.500000, bitrate: 7564 kb/s > Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 1004x1004 [PAR 1:1 > DAR 1:1], 104857 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 23.98 tbc > [swscaler @ 0x1af8550]No accelerated colorspace conversion found from > yuv420p to rgb24. > Output #0, image2, to 'out/output%04d.png': > Metadata: > encoder : Lavf52.64.2 > Stream #0.0: Video: png, rgb24, 1004x1004 [PAR 1:1 DAR 1:1], > q=2-31, 200 kb/s, 90k tbn, 23.98 tbc > Stream mapping: > Stream #0.0 -> #0.0 > Press [q] to stop encoding > frame= 259 fps= 11 q=0.0 Lsize= -0kB time=10.80 bitrate= > -0.0kbits/s > video:128012kB audio:0kB global headers:0kB muxing overhead -100.000017% > > > --------------New (not working) > output--------------------------------------- > ./ffmpeg -i input.mpg -y -f image2 out/output%04d.png > FFmpeg version UNKNOWN, Copyright (c) 2000-2011 the FFmpeg developers > built on Apr 13 2011 18:45:22 with gcc 4.4.5 > configuration: > libavutil 50. 40. 1 / 50. 40. 1 > libavcodec 52.119. 0 / 52.119. 0 > libavformat 52.106. 0 / 52.106. 0 > libavdevice 52. 4. 0 / 52. 4. 0 > libavfilter 1. 78. 0 / 1. 78. 0 > libswscale 0. 13. 0 / 0. 13. 0 > [mpeg @ 0x1b656e0] max_analyze_duration reached > Input #0, mpeg, from 'input.mpg': > Duration: 00:00:10.76, start: 0.500000, bitrate: 7564 kb/s > Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 1004x1004 [PAR 1:1 > DAR 1:1], 104857 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 23.98 tbc > *Output #0, image2, to 'out/output%04d.png': > Stream #0.0: Invalid Codec type -1 > Could not find input stream matching output stream #0.0* > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user The problem seems to be that I am building ffmpeg without all the configuration options needed to do what I want to get done. Could anyone direct me to a manual or tell me how to set the build-options such that the configurations of my own build are comparable to a Synaptic-build? Appreciating any help! Chris From thilo.borgmann at googlemail.com Thu Apr 14 17:52:37 2011 From: thilo.borgmann at googlemail.com (Thilo Borgmann) Date: Thu, 14 Apr 2011 17:52:37 +0200 Subject: [FFmpeg-user] Saving Video Frames as Images In-Reply-To: <4DA7142E.9050206@stud.uni-saarland.de> References: <4DA6DD44.5050806@stud.uni-saarland.de> <4DA7142E.9050206@stud.uni-saarland.de> Message-ID: <4DA71845.9020404@googlemail.com> Am 14.04.11 17:35, schrieb Christopher Haccius: > On 04/14/2011 01:40 PM, Christopher Haccius wrote: >> I am trying to convert a the frames of a video into an image sequence. I used >> to accomplish this task by the following line: >> >> ffmpeg -i input.mpg -y -f image2 out/output%04d.png >> >> Yesterday ... >> >> --------------------------Old (working) output------------------------- >> ffmpeg -i input.mpg -y -f image2 out/output%04d.png >> FFmpeg version 0.6-4:0.6-2ubuntu6.1, Copyright (c) 2000-2010 the FFmpeg >> developers >> ... >> >> >> --------------New (not working) output--------------------------------------- >> ./ffmpeg -i input.mpg -y -f image2 out/output%04d.png This line works perfectly fine for me with todays git code. >> FFmpeg version UNKNOWN, Copyright (c) 2000-2011 the FFmpeg developers >> built on Apr 13 2011 18:45:22 with gcc 4.4.5 >> configuration: >> ... > > The problem seems to be that I am building ffmpeg without all the configuration > options needed to do what I want to get done. Could anyone direct me to a manual > or tell me how to set the build-options such that the configurations of my own > build are comparable to a Synaptic-build? You should check the output of your configure call. "png" should be listed at enabled encoders. If it is not, you should be able to output your frames into using another image codec/format. Or install a proper png-library and configure and build ffmpeg again. -Thilo From james.darnley at gmail.com Thu Apr 14 18:58:24 2011 From: james.darnley at gmail.com (James Darnley) Date: Thu, 14 Apr 2011 18:58:24 +0200 Subject: [FFmpeg-user] Saving Video Frames as Images In-Reply-To: <4DA6DD44.5050806@stud.uni-saarland.de> References: <4DA6DD44.5050806@stud.uni-saarland.de> Message-ID: On 14/04/2011, Christopher Haccius wrote: > I am trying to convert a the frames of a video into an image sequence. I > used to accomplish this task by the following line: > > ffmpeg -i input.mpg -y -f image2 out/output%04d.png > > Yesterday I downloaded the Snapshot from > git://git.videolan.org/ffmpeg.git to try to implement some small changes > in the ffmpeg code. When I build this snapshot the code line from above > does not seem to work any more. Does anyone know why? How far are the > git-repositories different from the releases? Has anything changed in > the procedure of splitting videos into frames? > > Any ideas are very much appreciated! > Best > Chris > ./ffmpeg -i input.mpg -y -f image2 out/output%04d.png > FFmpeg version UNKNOWN, Copyright (c) 2000-2011 the FFmpeg developers > built on Apr 13 2011 18:45:22 with gcc 4.4.5 > configuration: > libavutil 50. 40. 1 / 50. 40. 1 > libavcodec 52.119. 0 / 52.119. 0 > libavformat 52.106. 0 / 52.106. 0 > libavdevice 52. 4. 0 / 52. 4. 0 > libavfilter 1. 78. 0 / 1. 78. 0 > libswscale 0. 13. 0 / 0. 13. 0 > [mpeg @ 0x1b656e0] max_analyze_duration reached > Input #0, mpeg, from 'input.mpg': > Duration: 00:00:10.76, start: 0.500000, bitrate: 7564 kb/s > Stream #0.0[0x1e0]: Video: mpeg1video, yuv420p, 1004x1004 [PAR 1:1 > DAR 1:1], 104857 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 23.98 tbc > *Output #0, image2, to 'out/output%04d.png': > Stream #0.0: Invalid Codec type -1 > Could not find input stream matching output stream #0.0* You need zlib. Make sure that configure says it is enabled. From benoit at friry.net Thu Apr 14 22:21:17 2011 From: benoit at friry.net (Benoit) Date: Thu, 14 Apr 2011 22:21:17 +0200 Subject: [FFmpeg-user] AVI size limit Message-ID: <20110414202117.GG25387@friry.net> Hello, In order to rip VHS on my rather old box, I purchased a Dazzle USB device. My command is ffmpeg \ -f alsa \ -i plughw:0,0 \ -f video4linux2 -tvstd SECAM \ -i /dev/video0 \ -acodec pcm_s16le \ -ac 2 -ar 44100 \ -vcodec mjpeg \ -s 720x576 -r 25 -sameq \ -f avi \ output.avi I cannot compress in real-time (CPU too weak). So I grab in mjpeg, and will reencode later. It works as expected for short videos. But with a longer video, ffmpeg is killed after 1h35'50". The output is 13836 MB. I don't understand why. File too big? The output file is readable by vlc ou avidemux, but ffplay is wrong about it: Input #0, avi, from 'brut.avi': Metadata: ISFT : Lavf52.64.2 Duration: 00:07:35.12, start: 0.000000, bitrate: 255030 kb/s Stream #0.0: Video: mjpeg, yuvj420p, 720x576, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s Don't understant why it only "see" 7'35". Any idea? Thanks, Benoit From tristan at trutna.net Thu Apr 14 22:29:44 2011 From: tristan at trutna.net (Tristan Trutna) Date: Thu, 14 Apr 2011 16:29:44 -0400 Subject: [FFmpeg-user] Metadata muxer/demuxer documentation? In-Reply-To: References: Message-ID: To answer my own question: Extracting a ffmetadata file goes as follows: $ ffmpeg -i INPUT -f ffmetadata OUTPUT i.e. $ ffmpeg -i test.mp3 -f ffmetadata metadata.txt Reinserting ffmetadata: $ ffmpeg -i INPUT -i INPUT_FFMETADATAFILE -map_metadata 0:1 OUTPUT i.e. $ ffmpeg -i test.mp3 -i metadata.txt -map_metadata 0:1 -acodec libfaac test.m4a From kshawkeye at gmail.com Thu Apr 14 23:16:31 2011 From: kshawkeye at gmail.com (Kyle) Date: Thu, 14 Apr 2011 17:16:31 -0400 Subject: [FFmpeg-user] undefined reference to `_inflate' Message-ID: <4DA7642F.8090608@gmail.com> When trying to compile ffmpeg with rtmp I get this error: i686-w64-mingw32-gcc -L/home/kyle/software/ffmpeg/external-libraries/win32/lib -Wl,--as-needed -I/home/kyle/software/ffmpeg/external-libraries/win32/include -L/home/kyle/software/ffmpeg/external-libraries/win32/lib -o /tmp/ffconf.rzFd0GnR.exe /tmp/ffconf.GEn5rmdC.o -lz -lrtmp -lwinmm -lgdi32 -lpolarssl -lws2_32 -lavifil32 -lm -lpthreadGC2 -lpsapi /home/kyle/software/ffmpeg/external-libraries/win32/lib/librtmp.a(hashswf.o):hashswf.c:(.text+0xb6): undefined reference to `_inflate' /home/kyle/software/ffmpeg/external-libraries/win32/lib/librtmp.a(hashswf.o):hashswf.c:(.text+0xc4d): undefined reference to `_inflateInit_' /home/kyle/software/ffmpeg/external-libraries/win32/lib/librtmp.a(hashswf.o):hashswf.c:(.text+0xc95): undefined reference to `_inflateEnd' You can see that -lz is included, so I'm not sure whats actually going wrong? Here it is with -lz at the end: i686-w64-mingw32-gcc -L/home/kyle/software/ffmpeg/external-libraries/win32/lib -Wl,--as-needed -I/home/kyle/software/ffmpeg/external-libraries/win32/include -L/home/kyle/software/ffmpeg/external-libraries/win32/lib -o /tmp/ffconf.U4sdC0la.exe /tmp/ffconf.2UDSSUVZ.o -lrtmp -lwinmm -lgdi32 -lpolarssl -lws2_32 -lz -lavifil32 -lm -lpthreadGC2 -lpsapi /home/kyle/software/ffmpeg/external-libraries/win32/lib/librtmp.a(hashswf.o):hashswf.c:(.text+0xb6): undefined reference to `_inflate' /home/kyle/software/ffmpeg/external-libraries/win32/lib/librtmp.a(hashswf.o):hashswf.c:(.text+0xc4d): undefined reference to `_inflateInit_' /home/kyle/software/ffmpeg/external-libraries/win32/lib/librtmp.a(hashswf.o):hashswf.c:(.text+0xc95): undefined reference to `_inflateEnd' Any ideas? From jieyunfu at mit.edu Fri Apr 15 08:55:05 2011 From: jieyunfu at mit.edu (Jieyun Fu) Date: Fri, 15 Apr 2011 02:55:05 -0400 Subject: [FFmpeg-user] Piping in ffmpeg? Using Libav to write to RTMP directly? Message-ID: Hi all, One short version of the question: is there anyway to let Libav write to a RTMP server directly? I only know how to write to a file on the disk, from output-example.c. A very slightly longer version of the question: I have a video generator that generates the video using libav, and is able to write the video into a file on the disk. Now, I need to use ffmpeg to send this file to a RTMP server simultaneously, when this video is being generated. One way to solve this problem, is to let this video generator write the video to say, output.flv, and then while this video is still being written, fork a process from the video generator that executes: ffmpeg -re -i output.flv -f flv rtmp://server/output/ ( just an example, skipped many parameters) But is there anyway to use piping to avoid writing to the disk and then call ffmpeg, but let ffmpeg takes my video generator as its input directly? Thank you so much! Jieyun From zhanwu.xiong at gmail.com Fri Apr 15 10:18:16 2011 From: zhanwu.xiong at gmail.com (Zhanwu Xiong) Date: Fri, 15 Apr 2011 10:18:16 +0200 Subject: [FFmpeg-user] How can I build a 32 bit ffmpeg on a 64 bit platform Message-ID: Hi all, I want to build a 32 bit version of ffmpeg on a 64 bit platform, in order to enable * * *gcc -m32 main.c otherlib.o -lavformat -lavcodec* where otherlib.o is an 32 bit compiled object and I don't have source of it. I have tried to specify *'--arch=x86_32'* when calling *configure*, which successfully set arch value to x86_32 in *config.mak*, but the generated library files are still incompatible. Can any one tell me which option I should use when *configure*? best, /zhanwu * * * * From ankuj2004 at gmail.com Fri Apr 15 11:35:47 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Fri, 15 Apr 2011 15:05:47 +0530 Subject: [FFmpeg-user] Possible Issue in ITU.H263 Decoder Code Flow Message-ID: Hi, I was trying to decode a H263 Profile 0 stream with ffmpeg. According to the H263 standard if there is an escape code DCT coefficient of value -128 which is forbidden the decoder should treat this as an error and go for concealment. But in FFmpeg version 0.6.1 file ituh263dec.c in function h263_decode_block following code is present if (level == -128) { if (s->codec_id == CODEC_ID_RV10) ????? {??????? ?/* XXX: should patch encoder too */ ???????? level = get_sbits(&s->gb, 12); ?????? } ??????else ????? { level = get_bits(&s->gb, 5); ???????? level |= get_sbits(&s->gb, 6)<<5; ????? } } where else part (modified quantization decoding Annex T) should not be there for Profile 0. My queries are 1) Is it the correct method ? Am I missing something 2) Or this else part should be present only when Modified Quantization is enabled ? Thanks in advance Ankuj Gupta From ankuj2004 at gmail.com Fri Apr 15 12:00:41 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Fri, 15 Apr 2011 15:30:41 +0530 Subject: [FFmpeg-user] Possible Issue in ITU.H263 Decoder Code Flow In-Reply-To: References: Message-ID: More information : [1] We note that (as per h263 ITU standard) Annex-T is not supported in h263 Profile-0 (baseline profile). [2] Annex-T say -- ------- This annex describes an optional Modified Quantization mode of this Recommendation, which modifies quantizer operation. The use of this mode is indicated in the PLUSPTYPE field of the picture header. ------- [3] And in the below mentioned code: --------- if (level == -128) { if (s->codec_id == CODEC_ID_RV10) { /* XXX: should patch encoder too */ level = get_sbits(&s->gb, 12); } else { level = get_bits(&s->gb, 5); level |= get_sbits(&s->gb, 6)<<5; } } --------- Here in the else part, Annex-T decoding is used without any condition check for PLUSPTYPE. Please let me know if I am missing something. On Fri, Apr 15, 2011 at 3:05 PM, Ankuj Gupta wrote: > Hi, > > I was trying to decode a H263 Profile 0 stream with ffmpeg. According > to the H263 standard if there is an escape code DCT coefficient of > value -128 which is forbidden the decoder should treat this as an > error and go for concealment. > > But in FFmpeg version 0.6.1 file ituh263dec.c in function > h263_decode_block following code is present > > > if (level == -128) > { > ? ? ?if (s->codec_id == CODEC_ID_RV10) > ????? {??????? ?/* XXX: should patch encoder too */ > ???????? ? level = get_sbits(&s->gb, 12); > ?????? } > ??????else > ????? { > ? ? ? ? level = get_bits(&s->gb, 5); > ???????? level |= get_sbits(&s->gb, 6)<<5; > > ????? } > } > > where else part (modified quantization decoding Annex T) should not be > there for Profile 0. > > My queries are > 1) Is it the correct method ? Am I missing something > 2) Or this else part should be present only when Modified Quantization > is enabled ? > > Thanks in advance > > Ankuj Gupta > From zhanwu.xiong at gmail.com Fri Apr 15 14:23:37 2011 From: zhanwu.xiong at gmail.com (Zhanwu Xiong) Date: Fri, 15 Apr 2011 14:23:37 +0200 Subject: [FFmpeg-user] How can I build a 32 bit ffmpeg on a 64 bit platform Message-ID: Hi all, I want to build a 32 bit version of ffmpeg on a 64 bit platform, in order to enable * * *gcc -m32 main.c otherlib.o -lavformat -lavcodec* where otherlib.o is an 32 bit compiled object and I don't have source of it. I have tried to specify *'--arch=x86_32'* when calling *configure*, which successfully set arch value to x86_32 in *config.mak*, but the generated library files are still incompatible. Can any one tell me which option I should use when *configure*? best, /zhanwu From zhanwu.xiong at gmail.com Fri Apr 15 14:25:18 2011 From: zhanwu.xiong at gmail.com (Zhanwu Xiong) Date: Fri, 15 Apr 2011 14:25:18 +0200 Subject: [FFmpeg-user] How can I build a 32 bit ffmpeg on a 64 bit platform Message-ID: Hi all, Sorry for the duplicated mail if any. I want to build a 32 bit version of ffmpeg on a 64 bit platform, in order to enable * * *gcc -m32 main.c otherlib.o -lavformat -lavcodec* where otherlib.o is an 32 bit compiled object and I don't have source of it. I have tried to specify *'--arch=x86_32'* when calling *configure*, which successfully set arch value to x86_32 in *config.mak*, but the generated library files are still incompatible. Can any one tell me which option I should use when *configure*? best, /zhanwu From h.reindl at thelounge.net Fri Apr 15 14:27:38 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Fri, 15 Apr 2011 14:27:38 +0200 Subject: [FFmpeg-user] How can I build a 32 bit ffmpeg on a 64 bit platform In-Reply-To: References: Message-ID: <4DA839BA.3010005@thelounge.net> Am 15.04.2011 14:23, schrieb Zhanwu Xiong: > Hi all, > > I want to build a 32 bit version of ffmpeg on a 64 bit platform, in order to > enable you will not get a quicker answer if you post the same message multiple times > *gcc -m32 main.c otherlib.o -lavformat -lavcodec* > where otherlib.o is an 32 bit compiled object and I don't have source of it. how should anybody understand this? WHAT library? how should this work? > I have tried to specify *'--arch=x86_32'* when calling *configure*, > which successfully set arch > value to x86_32 in *config.mak*, but the generated library files are still > incompatible. > > Can any one tell me which option I should use when *configure*? i would generally use a virtual machine with the target os und a build-environment and building distribution-like packages instead of manual compile -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From zhanwu.xiong at gmail.com Fri Apr 15 14:36:44 2011 From: zhanwu.xiong at gmail.com (Zhanwu Xiong) Date: Fri, 15 Apr 2011 14:36:44 +0200 Subject: [FFmpeg-user] How can I build a 32 bit ffmpeg on a 64 bit platform In-Reply-To: <4DA839BA.3010005@thelounge.net> References: <4DA839BA.3010005@thelounge.net> Message-ID: I didn't intended to post it multiple time, really sorry for that. On Fri, Apr 15, 2011 at 2:27 PM, Reindl Harald wrote: > > Am 15.04.2011 14:23, schrieb Zhanwu Xiong: > > Hi all, > > > > I want to build a 32 bit version of ffmpeg on a 64 bit platform, in order > to > > enable > > you will not get a quicker answer if you post the same > message multiple times > > > *gcc -m32 main.c otherlib.o -lavformat -lavcodec* > > where otherlib.o is an 32 bit compiled object and I don't have source of > it. > > how should anybody understand this? > WHAT library? > how should this work? > > otherlib.o is an object file which implemented some function I need, and it is compiled in 32 bit machine. My working machine is 64 bit, so when I compile my program with that object file I tried to force the the compiler to work in 32 bit mode, that is what '-m32' for I guess. In this case, I also need my libavcodec and libavformat to be 32 bit. And my problem is that I dont know how to generate 32 bit libavcodec and libavformat library on a 64 bit platform. asking question is really an art that I dont master ;) > > I have tried to specify *'--arch=x86_32'* when calling *configure*, > > which successfully set arch > > value to x86_32 in *config.mak*, but the generated library files are > still > > incompatible. > > > > Can any one tell me which option I should use when *configure*? > > i would generally use a virtual machine with the target os und > a build-environment and building distribution-like packages > instead of manual compile > I am working on a unix server, so this is not applicable for me > > > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From alexandre.ferrieux at orange-ftgroup.com Fri Apr 15 15:28:11 2011 From: alexandre.ferrieux at orange-ftgroup.com (Alexandre Ferrieux) Date: Fri, 15 Apr 2011 15:28:11 +0200 Subject: [FFmpeg-user] Roundup down ? Message-ID: <4DA847EB.9040505@orange-ftgroup.com> Hi, What's the current status of the bug tracker ? The link given on http://ffmpeg.org is https://ffmpeg.org/trac/ffmpeg , which currently is the generic 'trac' page. -Alex From jeisom at gmail.com Fri Apr 15 15:36:00 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Fri, 15 Apr 2011 08:36:00 -0500 Subject: [FFmpeg-user] Roundup down ? In-Reply-To: <4DA847EB.9040505@orange-ftgroup.com> References: <4DA847EB.9040505@orange-ftgroup.com> Message-ID: On Fri, Apr 15, 2011 at 8:28 AM, Alexandre Ferrieux < alexandre.ferrieux at orange-ftgroup.com> wrote: > Hi, > > What's the current status of the bug tracker ? > > The link given on http://ffmpeg.org is https://ffmpeg.org/trac/ffmpeg , > which currently is the generic 'trac' page. > > Seems to be working. There need to be some personalization for ffmpeg though. there is about 40-50 bugs listed. didn't log in though. Jonathan > -Alex > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From alexandre.ferrieux at orange-ftgroup.com Fri Apr 15 15:43:19 2011 From: alexandre.ferrieux at orange-ftgroup.com (Alexandre Ferrieux) Date: Fri, 15 Apr 2011 15:43:19 +0200 Subject: [FFmpeg-user] Roundup down ? In-Reply-To: References: <4DA847EB.9040505@orange-ftgroup.com> Message-ID: <4DA84B77.9000709@orange-ftgroup.com> On 15/04/2011 15:36, Jonathan Isom wrote: > > > On Fri, Apr 15, 2011 at 8:28 AM, Alexandre Ferrieux > > wrote: > > Hi, > > What's the current status of the bug tracker ? > > The link given on http://ffmpeg.org is > https://ffmpeg.org/trac/ffmpeg , which currently is the generic > 'trac' page. > > > Seems to be working. There need to be some personalization for ffmpeg > though. there is about 40-50 bugs listed. > didn't log in though. Oh, thanks ! Indeed. There "needs" to be something mentioning the project's name. What's the point of being so hidden ? -Alex From bmurphy1976 at gmail.com Sat Apr 16 03:39:47 2011 From: bmurphy1976 at gmail.com (Bryan Murphy) Date: Fri, 15 Apr 2011 20:39:47 -0500 Subject: [FFmpeg-user] Problem converting MPEGTS file to QuickTime compatible MP4 Message-ID: I'm having difficulty converting an h.264+aac MPEGTS file into a QuickTime compatible MP4. If I use ffmpeg to convert a source file direct to MP4, everything works. If I convert to an intermediate MPEGTS file (I have multiple reasons for wanting to do this), the second conversion from MPEGTS to MP4 produces a file that QuickTime refuses to play. I've tried multiple versions of ffmpeg and they all seem to exhibit the same (or very similar) behavior. Here is a shell script that reproduces this behavior. It creates two files, output-good.mp4 which plays in QuickTime, and output-bad.mp4 which does not. #!/bin/bash -e wget http://www.2atoms.com/video/haha/smelly_monkey4.wmv # create file that works ffmpeg -y -i smelly_monkey4.wmv \ -vcodec libx264 -vpre baseline -vpre medium -vpre ipod320 -s 320x240 -r 29.97 -b 256k -bt 256k \ -acodec libfaac -ab 64k -ar 22050 -ac 1 output-good.mp4 # transcode audio seperately ffmpeg -y -i smelly_monkey4.wmv -f wav -acodec pcm_s16le -sameq .temp.audio1.wav sox .temp.audio1.wav .temp.audio2.wav channels 1 rate 22050 pad 0 30 faac .temp.audio2.wav -b 64 -c 22050 -o .temp.audio.aac # transcode video seperately ffmpeg -y -i smelly_monkey4.wmv -vcodec libx264 -vpre baseline -vpre medium -vpre ipod320 -s 320x240 -r 29.97 -b 256k -bt 256k -an .temp.video.h264 # store output as a .ts file ffmpeg -y -shortest -async 1 -i .temp.audio.aac -i .temp.video.h264 -acodec copy -vcodec copy .temp.transcoded.ts # repackage output into various different containers as needed, including QuickTime compatible MP4 ffmpeg -y -i .temp.transcoded.ts -acodec copy -vcodec copy output-bad.mp4 From lukas.oesterreicher at inode.at Sat Apr 16 16:43:34 2011 From: lukas.oesterreicher at inode.at (=?iso-8859-15?Q?Lukas_=D6sterreicher?=) Date: Sat, 16 Apr 2011 16:43:34 +0200 Subject: [FFmpeg-user] Use advanced multi-threaded decoding ? In-Reply-To: References: <4DA847EB.9040505@orange-ftgroup.com> Message-ID: <16210727214.20110416164334@inode.at> Hello. FFmpeg (now) supports multithreaded decoding. However it appears to me it does not support it as well as mencoder or mplayer does. When I use -lavdopts fast:threads=4 in mencoder, decoding of h264 content is much faster than compared to ffmpeg, at least as far as I managed (with ffmpeg I can do 16fps, in mencoder 28! because of this). Is there a way to tell ffmpeg to decode the video with like (or simlarly) mencoder would if passing the "-lavdopts fast:threads=4" option? Please answer to tell me if its possible, and if it is the case how. Thanx in Advance! Lukas From richard at richsim900.plus.com Sat Apr 16 21:07:15 2011 From: richard at richsim900.plus.com (richard) Date: Sat, 16 Apr 2011 20:07:15 +0100 Subject: [FFmpeg-user] FFmpeg produces m4a files that won't play on Marantz CD6003 Message-ID: <1302980835.6732.6.camel@base-desktop> The Marantz CD6003 will play m4a files via a USB device or iPod connected to a USB port, but m4a files created using ffmpeg will not play. Testing has shown that the problem appears to be caused because the average bit rate (avgBitrate ) in the DecoderConfigDescriptor within the esds atom is set to zero. Where the avgBitrate is a non zero value, the m4a files play OK. I'm using Ubuntu 8.04 (Hardy). FFmpeg version git-N-28888-g6f73d5e built on Apr 5 2011 with gcc 4.2.4 If I convert an mp3 to m4a using libfaac: ffmpeg -i test2.mp3 -acodec libfaac test2.m4a the m4a file won't play. Check the metadata using: mp4dump test2.m4a and the avgBitrate in the decConfigDescr in the esds atom = 0. If I convert the same mp3 to m4a using faac: ffmpeg -i test4.mp3 -f wav - | faac -o test4.m4a -b 152 - the m4a file plays OK, but the the avgBitrate in the decConfigDescr in the esds atom = 151859 An m4a file that does not play (avgBitrate = 0) can be made playable by changing a tag (any tag) using EasyTag or mp4tags. When a tag is changed, EasyTag or mp4tags automatically changes the avgBitrate to the correct (or estimated) value (e.g. avgBitrate =128002). The problem with m4a files also occurs when I use get_iplayer. The latest patched version of get_iplayer does this: rtmpdump downloads flv ffmpeg remuxes flv to aac using: ffmpeg -i -vn -acodec copy -y ffmpeg remuxes aac to m4a and removes ADTS using: fmpeg -i -vn -acodec copy -absf aac_adtstoasc -y atomicparsley tags the m4a Again the resulting m4a file will not play, and the avgBitrate in the esds atom is zero. The m4a file can be made playable using mp4tags to alter a tag. This automatically changes the avgBitrate to the correct (or estimated) value, but the file only plays after a long delay. The -optimize option of mp4creator can be used to remove free atoms and the resulting m4a files plays OK without a long delay. Could this problem of setting the AvgBitrate to zero be resolved? ffmpeg does know the average bit rate, but it doesn't store it in the m4a file, and so a zero value results. From lou at fakeoutdoorsman.com Sat Apr 16 21:22:01 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Sat, 16 Apr 2011 11:22:01 -0800 Subject: [FFmpeg-user] Problem converting MPEGTS file to QuickTime compatible MP4 In-Reply-To: References: Message-ID: <20110416112201.73a286d2@lrcd.com> On Fri, 15 Apr 2011 20:39:47 -0500 Bryan Murphy wrote: > I'm having difficulty converting an h.264+aac MPEGTS file into a > QuickTime compatible MP4. If I use ffmpeg to convert a source file > direct to MP4, everything works. If I convert to an intermediate > MPEGTS file (I have multiple reasons for wanting to do this), the > second conversion from MPEGTS to MP4 produces a file that QuickTime > refuses to play. > > I've tried multiple versions of ffmpeg and they all seem to exhibit > the same (or very similar) behavior. > > Here is a shell script that reproduces this behavior. It creates two > files, output-good.mp4 which plays in QuickTime, and output-bad.mp4 > which does not. > > #!/bin/bash -e > > wget http://www.2atoms.com/video/haha/smelly_monkey4.wmv > > # create file that works > ffmpeg -y -i smelly_monkey4.wmv \ > -vcodec libx264 -vpre baseline -vpre medium -vpre ipod320 -s My comments probably won't help your issue, but your commands can be improved. Constraining presets, such as baseline, go after your encoding preset. Also, you don't need to include baseline when using ipod320 as the last preset because ipod320 is baseline with a few additional options and will overwrite whatever baseline sets: -vpre medium -vpre ipod320 > 320x240 -r 29.97 -b 256k -bt 256k \ > -acodec libfaac -ab 64k -ar 22050 -ac 1 output-good.mp4 -bt 256k is being ignored so you can remove it. Why are you not using '-threads 0'? Use CRF if you're doing one pass. See: FFmpeg x264 encoding guide > # transcode audio seperately > ffmpeg -y -i smelly_monkey4.wmv -f wav -acodec pcm_s16le > -sameq .temp.audio1.wav sox .temp.audio1.wav .temp.audio2.wav > channels 1 rate 22050 pad 0 30 faac .temp.audio2.wav -b 64 -c 22050 > -o .temp.audio.aac You don't need -sameq here. > # transcode video seperately > ffmpeg -y -i smelly_monkey4.wmv -vcodec libx264 -vpre baseline -vpre > medium -vpre ipod320 -s 320x240 -r 29.97 -b 256k -bt 256k -an > .temp.video.h264 > > # store output as a .ts file > ffmpeg -y -shortest -async 1 -i .temp.audio.aac -i .temp.video.h264 > -acodec copy -vcodec copy .temp.transcoded.ts > > # repackage output into various di.temp.transcoded.tsfferent > containers as needed,including QuickTime compatible MP4 > ffmpeg -y -i .temp.transcoded.ts -acodec copy -vcodec copy > output-bad.mp4 Does it work with Quicktime if you use .temp.video.h264 and temp.audio.aac as your sources instead of .temp.transcoded.ts? From baptiste.coudurier at gmail.com Sun Apr 17 01:20:06 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Sat, 16 Apr 2011 16:20:06 -0700 Subject: [FFmpeg-user] FFmpeg produces m4a files that won't play on Marantz CD6003 In-Reply-To: <1302980835.6732.6.camel@base-desktop> References: <1302980835.6732.6.camel@base-desktop> Message-ID: <4DAA2426.6020000@gmail.com> Hi, On 4/16/11 12:07 PM, richard wrote: > The Marantz CD6003 will play m4a files via a USB device or iPod > connected to a USB port, but m4a files created using ffmpeg will not > play. Testing has shown that the problem appears to be caused because > the average bit rate (avgBitrate ) in the DecoderConfigDescriptor within > the esds atom is set to zero. Where the avgBitrate is a non zero value, > the m4a files play OK. > > I'm using Ubuntu 8.04 (Hardy). FFmpeg version git-N-28888-g6f73d5e built > on Apr 5 2011 with gcc 4.2.4 > > If I convert an mp3 to m4a using libfaac: > > ffmpeg -i test2.mp3 -acodec libfaac test2.m4a > > the m4a file won't play. Check the metadata using: > > mp4dump test2.m4a > > and the avgBitrate in the decConfigDescr in the esds atom = 0. > > > If I convert the same mp3 to m4a using faac: > > ffmpeg -i test4.mp3 -f wav - | faac -o test4.m4a -b 152 - > > the m4a file plays OK, but the the avgBitrate in the decConfigDescr in > the esds atom = 151859 > > An m4a file that does not play (avgBitrate = 0) can be made playable by > changing a tag (any tag) using EasyTag or mp4tags. When a tag is > changed, EasyTag or mp4tags automatically changes the avgBitrate to the > correct (or estimated) value (e.g. avgBitrate =128002). > > > The problem with m4a files also occurs when I use get_iplayer. The > latest patched version of get_iplayer does this: > > rtmpdump downloads flv > ffmpeg remuxes flv to aac using: ffmpeg -i -vn -acodec copy -y > ffmpeg remuxes aac to m4a and removes ADTS using: fmpeg -i -vn -acodec > copy -absf aac_adtstoasc -y > atomicparsley tags the m4a > > Again the resulting m4a file will not play, and the avgBitrate in the > esds atom is zero. The m4a file can be made playable using mp4tags to > alter a tag. This automatically changes the avgBitrate to the correct > (or estimated) value, but the file only plays after a long delay. The > -optimize option of mp4creator can be used to remove free atoms and the > resulting m4a files plays OK without a long delay. > > Could this problem of setting the AvgBitrate to zero be resolved? > > ffmpeg does know the average bit rate, but it doesn't store it in the > m4a file, and so a zero value results. Should be fixed in latest git. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From baptiste.coudurier at gmail.com Sun Apr 17 02:10:11 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Sat, 16 Apr 2011 17:10:11 -0700 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding Message-ID: <4DAA2FE3.5040708@gmail.com> Hi everyone, Latest git has now added 2 new options for libx264 encoding: -preset -tune These options are directly mapped to internal tune and presets within libx264. The old presets files have been removed, please use the new options. Best regards. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From seandarcy2 at gmail.com Sun Apr 17 03:41:36 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Sat, 16 Apr 2011 21:41:36 -0400 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DAA2FE3.5040708@gmail.com> References: <4DAA2FE3.5040708@gmail.com> Message-ID: On 04/16/2011 08:10 PM, Baptiste Coudurier wrote: > Hi everyone, > > Latest git has now added 2 new options for libx264 encoding: > -preset > -tune > > These options are directly mapped to internal tune and presets within > libx264. The old presets files have been removed, please use the new > options. > > Best regards. > What are the choices/options for -preset and -tune? sean From tariq.connect at gmail.com Sun Apr 17 04:55:00 2011 From: tariq.connect at gmail.com (Md. Tariq Anjum) Date: Sun, 17 Apr 2011 08:25:00 +0530 Subject: [FFmpeg-user] avi to flv conversion and conversion progress tracking system Message-ID: Hello All I have 2 queries. Please help me sort them. 1. I need to convert an avi file to flv during the upload process of the avi 2. I need to be informed about the progress of conversion process done on a regular interval from ffmpeg I use php and my coding language. Please help me.... Thanks Tariq From lou at fakeoutdoorsman.com Sun Apr 17 04:55:12 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Sat, 16 Apr 2011 18:55:12 -0800 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: References: <4DAA2FE3.5040708@gmail.com> Message-ID: <20110416185512.261090a7@lrcd.com> On Sat, 16 Apr 2011 21:41:36 -0400 sean darcy wrote: > What are the choices/options for -preset and -tune? > > sean The same presets and tunes listed in "x264 --help". From lou at fakeoutdoorsman.com Sun Apr 17 05:08:04 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Sat, 16 Apr 2011 19:08:04 -0800 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DAA2FE3.5040708@gmail.com> References: <4DAA2FE3.5040708@gmail.com> Message-ID: <20110416190804.77de7d9e@lrcd.com> On Sat, 16 Apr 2011 17:10:11 -0700 Baptiste Coudurier wrote: > Hi everyone, > > Latest git has now added 2 new options for libx264 encoding: > -preset > -tune Also: -profile Such as: -profile baseline See "x264 --help" to see available presets, tunes, and profiles. > These options are directly mapped to internal tune and presets within > libx264. The old presets files have been removed, please use the new > options. > > Best regards. > > -- > Baptiste COUDURIER > Key fingerprint > 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg > maintainer http://www.ffmpeg.org From david.ariens at gmail.com Sun Apr 17 07:59:40 2011 From: david.ariens at gmail.com (David) Date: Sun, 17 Apr 2011 05:59:40 +0000 (UTC) Subject: [FFmpeg-user] Issue with watermarking a video -> result is garbled (pink and green lines) Message-ID: Hi, I have an issue with watermarking a video using ffmpeg and libavfilter. This is my testvideo: www.coachteam.com/testvideo/paul.mov This is the watermark: www.coachteam.com/testvideo/coachteamlogo.png And this is the result: www.coachteam.com/testvideo/paul.mp4 As you can see the logo is wrongly added (pink, just parts are visible). The command I use is this one: "c:\ffmpeg\bin\ffmpeg.exe" -i %1 -pass 1 -passlogfile log_file -b 500k -vcodec libx264 -vpre slow -vpre baseline -g 30 -y -vf "movie=coachteamlogo.png [logo]; [in][logo] overlay=10:main_h-overlay_h-10 [out]" "%~d1%~p1%~n1.mp4" The complete output of ffmpeg (including release) is pasted at the bottom. I tried changing formats (other video inputs, other images,...). I tried just adding the watermark without further compression of the video. But all without success... Could anyone give me a hint what I am doing wrong? Thanks! David C:\Users\David\Desktop\New folder>"c:\ffmpeg\bin\ffmpeg.exe" -i "C:\Users\David\ Desktop\New folder\paul.mov" -pass 1 -passlogfile log_file -b 500k -vcodec libx2 64 -vpre slow -vpre baseline -g 30 -y -vf "movie=coachteamlogo.png [logo]; [in][ logo] overlay=10:main_h-overlay_h-10 [out]" "C:\Users\David\Desktop\New folder\p aul.mp4" FFmpeg version git-N-28948-g9d4cb45, Copyright (c) 2000-2011 the FFmpeg develope rs built on Apr 8 2011 16:57:22 with gcc 4.5.2 configuration: --enable-gpl --enable-version3 --enable-runtime-cpudetect --ena ble-memalign-hack --arch=i686 --target-os=mingw32 --cross-prefix=i686-w64-mingw3 2- --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb - -enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3la me --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libsp eex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --ena ble-libxavs --enable-libxvid --enable-zlib --extra-cflags=-I/home/kyle/software/ ffmpeg/external-libs/32-bit/include --extra-ldflags=-L/home/kyle/software/ffmpeg /external-libs/32-bit/lib --pkg-config=pkg-config libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.117. 0 / 52.117. 0 libavformat 52.106. 0 / 52.106. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 77. 1 / 1. 77. 1 libswscale 0. 13. 0 / 0. 13. 0 Seems stream 1 codec frame rate differs from container frame rate: 1200.00 (1200 /1) -> 30.00 (30/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\Users\David\Desktop\New folder\paul. mov': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2011-04-09 16:14:17 encoder : 4.3.1 encoder-dut : 4.3.1 date : 2011-04-09T18:13:43+0200 date-dut : 2011-04-09T18:13:43+0200 Duration: 00:00:11.63, start: 0.000000, bitrate: 755 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, mono, s16, 63 kb/s Metadata: creation_time : 2011-04-09 16:14:17 Stream #0.1(und): Video: h264 (Main), yuv420p, 480x272, 688 kb/s, 30 fps, 30 tbr, 600 tbn, 1200 tbc Metadata: creation_time : 2011-04-09 16:14:17 [buffer @ 0199FD70] w:480 h:272 pixfmt:yuv420p [movie @ 01966FD0] seek_point:0 format_name:(null) file_name:coachteamlogo.png s tream_index:0 [overlay @ 01967050] auto-inserting filter 'auto-inserted scaler 0' between the filter 'Parsed filter 0 movie' and the filter 'Parsed filter 1 overlay' [scale @ 01967010] w:100 h:60 fmt:bgra -> w:100 h:60 fmt:yuva420p flags:0x4 [overlay @ 01967050] main w:480 h:272 fmt:yuv420p overlay x:10 y:202 w:100 h:60 fmt:yuva420p [overlay @ 01967050] main_tb:1/1000000 overlay_tb:1/25 -> tb:1/1000000 exact:1 [libx264 @ 01997520] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE 4.1 Cache64 [libx264 @ 01997520] profile Constrained Baseline, level 2.1 [libx264 @ 01997520] 264 - core 114 r1924 08d04a4 - H.264/MPEG-4 AVC codec - Cop yleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 ref=5 deb lock=1:0:0 analyse=0x1:0x111 me=umh subme=8 psy=1 psy_rd=1.00:0.00 mixed_ref=1 m e_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chro ma_qp_offset=-2 threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 constrai ned_intra=0 bframes=0 weightp=0 keyint=30 keyint_min=16 scenecut=40 intra_refres h=0 rc_lookahead=30 rc=abr mbtree=1 bitrate=500 ratetol=1.0 qcomp=0.60 qpmin=0 q pmax=69 qpstep=4 ip_ratio=1.41 aq=1:1.00 Output #0, mp4, to 'C:\Users\David\Desktop\New folder\paul.mp4': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2011-04-09 16:14:17 date-dut : 2011-04-09T18:13:43+0200 encoder-dut : 4.3.1 date : 2011-04-09T18:13:43+0200 encoder : Lavf52.106.0 Stream #0.0(und): Video: libx264, yuv420p, 480x272, q=0-69, pass 1, 500 kb/s , 30 tbn, 30 tbc Metadata: creation_time : 2011-04-09 16:14:17 Stream #0.1(und): Audio: aac, 44100 Hz, mono, s16, 64 kb/s Metadata: creation_time : 2011-04-09 16:14:17 Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 349 fps= 25 q=-1.0 Lsize= 820kB time=11.57 bitrate= 580.8kbits/s video:724kB audio:88kB global headers:0kB muxing overhead 0.976190% frame I:12 Avg QP:21.75 size: 10397 [libx264 @ 01997520] frame P:337 Avg QP:24.11 size: 1827 [libx264 @ 01997520] mb I I16..4: 29.3% 0.0% 70.7% [libx264 @ 01997520] mb P I16..4: 3.7% 0.0% 1.4% P16..4: 55.9% 11.9% 3.4% 0.0% 0.0% skip:23.7% [libx264 @ 01997520] final ratefactor: 22.49 [libx264 @ 01997520] coded y,uvDC,uvAC intra: 45.7% 81.2% 23.4% inter: 11.6% 38. 5% 0.8% [libx264 @ 01997520] i16 v,h,dc,p: 26% 22% 14% 38% [libx264 @ 01997520] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 13% 19% 14% 9% 11% 10% 11% 7% 6% [libx264 @ 01997520] i8c dc,h,v,p: 47% 24% 18% 11% [libx264 @ 01997520] ref P L0: 91.3% 5.4% 2.0% 0.7% 0.6% [libx264 @ 01997520] kb/s:509.22 From roger.pau at entel.upc.edu Sun Apr 17 20:38:43 2011 From: roger.pau at entel.upc.edu (=?UTF-8?Q?Roger_Pau_Monn=C3=A9?=) Date: Sun, 17 Apr 2011 20:38:43 +0200 Subject: [FFmpeg-user] Filter video using libavfilter is longer than input video Message-ID: Hello, I'm trying to develop a new filter using the avfilter API, but it seems to add some frames at the beginning of the video, so input and output videos don't have the same duration (for example if I use an input video of duration 00:01:41.60 the output video has a duration of 00:01:41.64). I don't know if there is a problem with my filter, I don't think I'm adding any frames, but it's the first time I use it, so I'm attaching the filter I'm working on to see if anyone knows what the problem might be. Thanks, Roger. -------------- next part -------------- A non-text attachment was scrubbed... Name: vf_watermark.c Type: text/x-csrc Size: 8687 bytes Desc: not available URL: From stefano.sabatini-lala at poste.it Sun Apr 17 21:06:06 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sun, 17 Apr 2011 21:06:06 +0200 Subject: [FFmpeg-user] Filter video using libavfilter is longer than input video In-Reply-To: References: Message-ID: <20110417190606.GA4688@geppetto> On date Sunday 2011-04-17 20:38:43 +0200, Roger Pau Monn? encoded: > Hello, > > I'm trying to develop a new filter using the avfilter API, but it > seems to add some frames at the beginning of the video, so input and > output videos don't have the same duration (for example if I use an > input video of duration 00:01:41.60 the output video has a duration of > 00:01:41.64). I don't know if there is a problem with my filter, I > don't think I'm adding any frames, but it's the first time I use it, > so I'm attaching the filter I'm working on to see if anyone knows what > the problem might be. > > Thanks, Roger. Hi, from some quick tests it looks a problem related to transcoding, so not necessarily libavfilter-specific. BTW if you want to integrate the filter into FFmpeg you're welcome to post it on ffmpeg-devel, I see there are some simplifications which can be done. -- ffmpeg-user random tip #11 One minute of video silence with ffmpeg: ffmpeg -t 60 -s qcif -f rawvideo -pix_fmt rgb24 -r 25 -i /dev/zero \ -y silence.mpeg From roger.pau at entel.upc.edu Sun Apr 17 21:24:18 2011 From: roger.pau at entel.upc.edu (=?UTF-8?Q?Roger_Pau_Monn=C3=A9?=) Date: Sun, 17 Apr 2011 21:24:18 +0200 Subject: [FFmpeg-user] Filter video using libavfilter is longer than input video In-Reply-To: <20110417190606.GA4688@geppetto> References: <20110417190606.GA4688@geppetto> Message-ID: Hello, Thanks for your quick reply, I think I'm doing something wrong in the filter, because if I execute ffmpeg -i INPUT -vf null OUTPUT the duration of the video is the same, but if I insert my filter, the duration of the video is increased. I would like to add the filter to FFmpeg, but I'm still working on the decoder to extract the mark. Since input and output videos are not synchronized (marked video seems to start later) I cannot recover the mark correctly because there is a difference between input videos (I've based my recover filter on the overlay filter, I will try to clean the code and send it tomorrow). 2011/4/17 Stefano Sabatini : > On date Sunday 2011-04-17 20:38:43 +0200, Roger Pau Monn? encoded: >> Hello, >> >> I'm trying to develop a new filter using the avfilter API, but it >> seems to add some frames at the beginning of the video, so input and >> output videos don't have the same duration (for example if I use an >> input video of duration 00:01:41.60 the output video has a duration of >> 00:01:41.64). I don't know if there is a problem with my filter, I >> don't think I'm adding any frames, but it's the first time I use it, >> so I'm attaching the filter I'm working on to see if anyone knows what >> the problem might be. >> >> Thanks, Roger. > > Hi, > from some quick tests it looks a problem related to transcoding, so > not necessarily libavfilter-specific. > > BTW if you want to integrate the filter into FFmpeg you're welcome to > post it on ffmpeg-devel, I see there are some simplifications which > can be done. > -- > ffmpeg-user random tip #11 > One minute of video silence with ffmpeg: > ffmpeg -t 60 -s qcif -f rawvideo -pix_fmt rgb24 -r 25 -i /dev/zero \ > ? ?-y silence.mpeg > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From acmilan_2008 at hotmail.com Mon Apr 18 03:25:34 2011 From: acmilan_2008 at hotmail.com (Erik) Date: Mon, 18 Apr 2011 01:25:34 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Can=27t_locate_entry_av=5Faes=5Fcrpyt_in_?= =?utf-8?q?avutil-49=2Edll?= References: Message-ID: > > Your avutil-49 is too old. The modern version is avutil-50. Also you should > not ever > try to cross FFmpeg libraries from different sources - they have too much > cross dependencies > and you'll got a broken setup in most cases. > > Win binaries you can download here http://win32.libav.org/ and here I tried this version and the problem changed to can't locate entry of av_alloc_format_context at avformat-52.dll. Seems the main problem is in this file? I noticed in some of the includes, the include path has an additional directory name(e.g. #include "libavutil/avconfig.h" in a include already in the libavutil directory), which makes gcc fail but ok in VC. Is the entry error a incompatible issue with VC? If so, if i want to build it in gcc, how to solve the directory problem, can i simply remove the directory names in includes? From acmilan_2008 at hotmail.com Mon Apr 18 03:35:42 2011 From: acmilan_2008 at hotmail.com (Erik) Date: Mon, 18 Apr 2011 01:35:42 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Can=27t_locate_entry_av=5Faes=5Fcrpyt_in_?= =?utf-8?q?avutil-49=2Edll?= References: Message-ID: Erik hotmail.com> writes: > I tried this version and the problem changed to can't locate entry of > av_alloc_format_context at avformat-52.dll. Seems the main problem is in this > file? I noticed in some of the includes, the include path has an additional > directory name(e.g. #include "libavutil/avconfig.h" in a include already in > the libavutil directory), which makes gcc fail but ok in VC. > Is the entry error a incompatible issue with VC? If so, if i want to build it > in gcc, how to solve the directory problem, can i simply remove the directory > names in includes? > Tried another version of binaries and still the same problem, it's avcodec- 52.dll sry i post that carelessly wrong. In VC I already remember to use extern "C" and when I try to use gcc I removed this direction and changed the file extension to .c. Are there any additional C/C++ problems? From weikun0905 at gmail.com Mon Apr 18 04:25:15 2011 From: weikun0905 at gmail.com (Vincent,Wei) Date: Mon, 18 Apr 2011 10:25:15 +0800 Subject: [FFmpeg-user] About the output_example.c ,help Message-ID: Hi,all I am a new user of ffmpeg, I need to change the RTP packet to the avi file,used the H264 encoder, so I read the code of output_example.c. And I had tried to genrerate a test.avi file just like the example. There are some question about the result: (1) When I use the .avi as the extensions, it uesd the CODEC_ID_MPEG4 as default , I had found that it is defined at the avienc.c line:562 AVOutputFormat avi_muxer = { ..... CODEC_ID_MP2, CODEC_ID_MPEG4, ..... }; I had try to change it to the CODEC_ID_H264, but when I try to genrerat the test.avi again, I found that it can't work rightly,error as follow: error, non monotone timestamps -368934881474191031 >= -368934881474191031 Error while writing video frame . Why? (2) When I edit the output_example.c line:445 fmt = guess_format("h264", filename, NULL); I also set the .avi as the extensions, but it can't generate the avi format file(alougth the file look like .avi extension name,but the format do not fit the avi format standard). It generate the file which fit the .264 file format ,so it is turly encode as the h264, but I need the avi file. (3) The example is worked as the YUV420 as input data, but my data source had already been encoded as H264(RTP packet payload), I thought that I may not need to change the data to YUV first, it's not efficient. I means that if the data is already the encoded ,the avcodec_encode_video() may not be called? If it is right,av_write_frame() need what data as the input data ? raw h264 data ?or some others?? How can I use the ffmpeg to help me do the work? That is all, so many questions for all of you, please help me, thanks From help at nagashare.com Mon Apr 18 07:18:43 2011 From: help at nagashare.com (Bourne) Date: Mon, 18 Apr 2011 13:18:43 +0800 Subject: [FFmpeg-user] About mms streaming recording Message-ID: <000001cbfd88$175cd310$46167930$@com> Dear All, I want use ffmpeg to recording mms streaming to wmv files without encoding, it's possible? Thanks. Bourne From martin.nohr at rimage.com Wed Apr 13 21:04:29 2011 From: martin.nohr at rimage.com (Martin Nohr) Date: Wed, 13 Apr 2011 14:04:29 -0500 Subject: [FFmpeg-user] DV reading problem Message-ID: I found a problem transcoding an interlaced DV SD 4:3 video to DVD mpeg2. The mpg video looks like it has been shifted to the right about a third of the screen. It looks like a black bar in the middle, part of the video on the left, and the rest of the video on the right. I tried the September 2008 ffmpeg and it works fine. The failure is only on the current version. Has anybody seen anything like that? I have tried -top/-deinterlace/ilme and nothing makes it look correct. Martin Nohr From argisshenweiwei at gmail.com Thu Apr 14 12:04:15 2011 From: argisshenweiwei at gmail.com (=?GB2312?B?yern4g==?=) Date: Thu, 14 Apr 2011 18:04:15 +0800 Subject: [FFmpeg-user] make error Message-ID: Dear sir: I have download the following file: FFmpeg 0.6.90-rc0 "Love and Peace"--------------Download bzip2 tarball And I am using WINDOWS XP and MinGW 5.1.6. I copied MinGW 32-make into the file downloaded path and input "mingw32-make" at the downloaded path . But it says "config.mak not foune"-------------------------------So I change the "configure" file's name into "config.mak".But console says "config.mak17:missing operator,stop" -------------------Then I add the 17's line of the config.mak a [tab]-----------------Then it says"config.mak17 : commands commence before first target" . Can you tell me how to make it? Thanks so much! Wei Shen From argisshenweiwei at gmail.com Thu Apr 14 12:05:21 2011 From: argisshenweiwei at gmail.com (=?GB2312?B?yern4g==?=) Date: Thu, 14 Apr 2011 18:05:21 +0800 Subject: [FFmpeg-user] make error Message-ID: Dear sir: I have download the following file: FFmpeg 0.6.90-rc0 "Love and Peace"--------------Download bzip2 tarball And I am using WINDOWS XP and MinGW 5.1.6. I copied MinGW 32-make into the file downloaded path and input "mingw32-make" at the downloaded path . But it says "config.mak not foune"-------------------------------So I change the "configure" file's name into "config.mak".But console says "config.mak17:missing operator,stop" -------------------Then I add the 17's line of the config.mak a [tab]-----------------Then it says"config.mak17 : commands commence before first target" . Can you tell me how to make it? Thanks so much! Wei Shen From bouzidiamine at hotmail.com Thu Apr 14 14:30:01 2011 From: bouzidiamine at hotmail.com (amine bouzidi) Date: Thu, 14 Apr 2011 12:30:01 +0000 Subject: [FFmpeg-user] Problems while configuring the ffmpeg Message-ID: Hi, can anyone please help to configure the ffmpeg as I am encountering problems while configuring it with this command: ./configure --prefix=/usr --target-os=linux --arch=arm --cpu=cortex-a8 --enable-cross-compile --cross-prefix=arm-none-linux-gnueabi- --cc=arm-none-linux-gnueabi-gcc --enable-pic --enable-shared --disable-debug --disable-everything --enable-hardcoded-tables --enable-fft --enable-dct --disable-stripping --extra-cflags="-Wall -pipe -mcpu=cortex-a8 -mfpu=neon -mfloat-abi=softfp -fno-math-errno -fno-signed-zeros -fno-tree-vectorize" the output on terminal widow looks as follow: arm-none-linux-gnueabi-gcc is unable to create an executable file. C compiler test failed. If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. the version that I using is ffmpeg-HEAD-b8a4444.tar.gz downloaded from this link http://ffmpeg.org/download.html -------------- next part -------------- A non-text attachment was scrubbed... Name: config.log Type: text/x-log Size: 58896 bytes Desc: not available URL: From james.darnley at gmail.com Mon Apr 18 14:18:36 2011 From: james.darnley at gmail.com (James Darnley) Date: Mon, 18 Apr 2011 14:18:36 +0200 Subject: [FFmpeg-user] make error In-Reply-To: References: Message-ID: On 14/04/2011, ?? wrote: > Dear sir: > > I have download the following file: > FFmpeg 0.6.90-rc0 "Love and Peace"--------------Download bzip2 > tarball > > > And I am using WINDOWS XP and MinGW 5.1.6. > I copied MinGW 32-make into the file downloaded path and input > "mingw32-make" at the downloaded path . > But it says "config.mak not foune"-------------------------------So I change > the "configure" file's name into "config.mak".But console says > "config.mak17:missing operator,stop" -------------------Then I add the 17's > line of the config.mak a [tab]-----------------Then it says"config.mak17 : > commands commence before first target" . > > Can you tell me how to make it? > > Thanks so much! You need a unix-like shell (msys or cygwin) to run configure first From bmurphy1976 at gmail.com Mon Apr 18 15:52:36 2011 From: bmurphy1976 at gmail.com (Bryan Murphy) Date: Mon, 18 Apr 2011 08:52:36 -0500 Subject: [FFmpeg-user] Problem converting MPEGTS file to QuickTime compatible MP4 In-Reply-To: <20110416112201.73a286d2@lrcd.com> References: <20110416112201.73a286d2@lrcd.com> Message-ID: On Sat, Apr 16, 2011 at 2:22 PM, Lou wrote: > Does it work with Quicktime if you use .temp.video.h264 and > temp.audio.aac as your sources instead of .temp.transcoded.ts? No. There is something about ffmpeg merging the pre-transcoded streams that QuickTime does not like. It's only when it performs the transcode inline that it is happy. ffmpeg -y -shortest -async 1 -i .temp.audio.aac -i .temp.video.h264 -acodec copy -vcodec copy output-bad2.mp4 result: --------------------------- Error --------------------------- Error -2041: an invalid sample description was found in the movie (output-bad2.mp4) --------------------------- OK --------------------------- Here are a few more pieces to the puzzle. Using mp4info, we can see it complaining that the "bad" file's audio stream is missing "GAConfig". And with MP4Box -info, we can see that the stream for the bad file is labeled as having a "Corrupted AAC Config". mp4info output-bad.mp4 Track Type Info 1 video H264 Baseline at 1.3, 9.142 secs, 262 kbps, 320x240 @ 29.971560 fps 2 audio MPEG-4 (no GAConfig), 9.148 secs, 0 kbps, 22050 Hz Metadata Tool: Lavf52.92.0 mp4info output-good.mp4 Track Type Info 1 video H264 Baseline at 1.3, 9.142 secs, 262 kbps, 320x240 @ 29.971560 fps 2 audio MPEG-4 AAC LC, 9.287 secs, 0 kbps, 22050 Hz MP4Box -info output-bad.mp4 * Movie Info * Timescale 1000 - Duration 00:00:09.149 Fragmented File no - 2 track(s) File Brand isom - version 512 Created: GMT Thu Jan 01 00:00:00 1970 File has no MPEG4 IOD/OD iTunes Info: Encoder Software: Lavf52.92.0 Track # 1 Info - TrackID 1 - TimeScale 2997 - Duration 00:00:09.142 Media Info: Language "Undetermined" - Type "vide:avc1" - 274 samples Visual Track layout: x=0 y=0 width=320 height=240 MPEG-4 Config: Visual Stream - ObjectTypeIndication 0x21 AVC/H264 Video - Visual Size 320 x 240 AVC Info: 1 SPS - 1 PPS - Profile Baseline @ Level 1.3 NAL Unit length bits: 32 Self-synchronized Track # 2 Info - TrackID 2 - TimeScale 22050 - Duration 00:00:09.148 Media Info: Language "Undetermined" - Type "soun:mp4a" - 197 samples MPEG-4 Config: Audio Stream - ObjectTypeIndication 0x40 Corrupted AAC Config Synchronized on stream 1 MP4Box -info output-good.mp4 * Movie Info * Timescale 1000 - Duration 00:00:09.288 Fragmented File no - 2 track(s) File Brand isom - version 512 Created: GMT Thu Jan 01 00:00:00 1970 File has no MPEG4 IOD/OD Chapters: Chapter #1 - 00:00:00.000 - "Clip 1" iTunes Info: Name: Smelly Monkey! Artist: 2atoms.com Comment: Hey monkey - don't be doing that! Encoder Software: Lavf52.92.0 Track # 1 Info - TrackID 1 - TimeScale 2997 - Duration 00:00:09.142 Media Info: Language "English" - Type "vide:avc1" - 274 samples Visual Track layout: x=0 y=0 width=320 height=240 MPEG-4 Config: Visual Stream - ObjectTypeIndication 0x21 AVC/H264 Video - Visual Size 320 x 240 AVC Info: 1 SPS - 1 PPS - Profile Baseline @ Level 1.3 NAL Unit length bits: 32 Self-synchronized Track # 2 Info - TrackID 2 - TimeScale 22050 - Duration 00:00:09.287 Media Info: Language "English" - Type "soun:mp4a" - 200 samples MPEG-4 Config: Audio Stream - ObjectTypeIndication 0x40 MPEG-4 Audio AAC LC - 1 Channel(s) - SampleRate 22050 Synchronized on stream 1 From bmurphy1976 at gmail.com Mon Apr 18 15:55:05 2011 From: bmurphy1976 at gmail.com (Bryan Murphy) Date: Mon, 18 Apr 2011 08:55:05 -0500 Subject: [FFmpeg-user] Problem converting MPEGTS file to QuickTime compatible MP4 In-Reply-To: References: <20110416112201.73a286d2@lrcd.com> Message-ID: On Mon, Apr 18, 2011 at 8:52 AM, Bryan Murphy wrote: > No. ?There is something about ffmpeg merging the pre-transcoded > streams that QuickTime does not like. ?It's only when it performs the > transcode inline that it is happy. Also, if I skip the audio stream entirely, QuickTime will play back the video without audio. Everything seems to point to some sort of corruption in the AAC audio stream. Bryan From alexandre.ferrieux at orange-ftgroup.com Mon Apr 18 16:46:57 2011 From: alexandre.ferrieux at orange-ftgroup.com (Alexandre Ferrieux) Date: Mon, 18 Apr 2011 16:46:57 +0200 Subject: [FFmpeg-user] Asynchronous overlay In-Reply-To: <4DA6A8B3.10805@orange-ftgroup.com> References: <4DA6A8B3.10805@orange-ftgroup.com> Message-ID: <4DAC4EE1.7060002@orange-ftgroup.com> Hello, I'd like to use the overlay vfilter to lay out side-by-side several video streams; however they have wildly different framerates, some of them varying. What I'd like to do is a kind of async overlay: that each source be decoded into the overlay buffer at its own rate, and that a separate thread encodes 'snapshots' of this overlay buffer at a regular, independent rate. Is there a way to tell ffmpeg to do this ? -Alex From cholet at logilune.com Mon Apr 18 16:49:47 2011 From: cholet at logilune.com (Eric Cholet) Date: Mon, 18 Apr 2011 14:49:47 +0000 (UTC) Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> Message-ID: > > Latest git has now added 2 new options for libx264 encoding: > > -preset > > -tune > > Also: > -profile > > Such as: > -profile baseline > With freshly downloaded ffmpeg from git, I'm getting an error when trying to use the -profile option: % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b 2500k -r 25 -preset medium -profile main foo.mp4 What am I doing wrong? From cholet at logilune.com Mon Apr 18 17:03:49 2011 From: cholet at logilune.com (=?iso-8859-1?Q?=C9ric_Cholet?=) Date: Mon, 18 Apr 2011 17:03:49 +0200 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> Message-ID: <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> Le 18 avr. 2011 ? 16:49, Eric Cholet a ?crit : >>> Latest git has now added 2 new options for libx264 encoding: >>> -preset >>> -tune >> >> Also: >> -profile >> >> Such as: >> -profile baseline >> > > With freshly downloaded ffmpeg from git, I'm getting an error when trying > to use the -profile option: > > > % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b 2500k -r 25 > -preset medium -profile main foo.mp4 Oops forgot to post the error message: [NULL @ 0x80240b200] [Eval @ 0x7fffffffd110] Undefined constant or missing '(' in 'main' [NULL @ 0x80240b200] Unable to parse option value "main" Invalid value 'main' for option 'profile' > > What am I doing wrong? -- ?ric Cholet From gunvant.sahu at eworks.in Mon Apr 18 17:25:33 2011 From: gunvant.sahu at eworks.in (Gunvant Sahu) Date: Mon, 18 Apr 2011 20:55:33 +0530 Subject: [FFmpeg-user] how to combine videos side by side Message-ID: Hello All I am looking for a solution. Please have a look into this site http://www.shervinemami.co.cc/openCV.html%22 I want to combine videos like this. Is this possible with ffmpeg? Thanks in advance Gunvant From list at samalyse.com Mon Apr 18 21:42:21 2011 From: list at samalyse.com (Olivier Guilyardi) Date: Mon, 18 Apr 2011 21:42:21 +0200 Subject: [FFmpeg-user] MPEG-4/WAV container for quick A/V sync tweaking Message-ID: <4DAC941D.7030103@samalyse.com> Hello list, Which container formats support an MPEG-4 video with a WAV audio stream? I'm looking for a format which is supported by most players. Is there a such format which would allow to adjust the A/V sync offset very quickly, by modifying a field in the header, without the need to copy or transcode the whole file? TIA -- Olivier From bahamutzero8825 at gmail.com Mon Apr 18 22:02:52 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Mon, 18 Apr 2011 15:02:52 -0500 Subject: [FFmpeg-user] MPEG-4/WAV container for quick A/V sync tweaking In-Reply-To: <4DAC941D.7030103@samalyse.com> References: <4DAC941D.7030103@samalyse.com> Message-ID: <4DAC98EC.8050605@gmail.com> On 2011.04.18 02:42 PM, Olivier Guilyardi wrote: > Which container formats support an MPEG-4 video with a WAV audio stream? I'm > looking for a format which is supported by most players. AVI and Matroska. If by MPEG-4 video, you mean AVC/H.264, then I wouldn't recommend AVI. Decent software players should have no problems with this, but very few hardware players would support this combination, even if the codecs and container are supported in other combinations. > Is there a such format which would allow to adjust the A/V sync offset very > quickly, by modifying a field in the header, without the need to copy or > transcode the whole file? Not sure if Matroska can specify an offset, but I know MP4 and MPEG-TS can. From lou at fakeoutdoorsman.com Mon Apr 18 22:51:28 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Mon, 18 Apr 2011 12:51:28 -0800 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> Message-ID: <20110418125128.53550483@lrcd.com> On Mon, 18 Apr 2011 17:03:49 +0200 ?ric Cholet wrote: > Le 18 avr. 2011 ? 16:49, Eric Cholet a ?crit : > > With freshly downloaded ffmpeg from git, I'm getting an error when > > trying to use the -profile option: > > > > > > % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b 2500k > > -r 25 -preset medium -profile main foo.mp4 > > Oops forgot to post the error message: > > [NULL @ 0x80240b200] [Eval @ 0x7fffffffd110] Undefined constant or > missing '(' in 'main' [NULL @ 0x80240b200] Unable to parse option > value "main" Invalid value 'main' for option 'profile' > > > > > > What am I doing wrong? > -- > ?ric Cholet You are doing nothing wrong. It was my mistake to mention -profile. It is listed in 'ffmpeg -h', but is not available yet. I like to verify my suggestions, but this time I didn't. This is what happens when I rely on memory as I previously tested the new options a month ago but apparently forgot that -profile wasn't usable yet. From list at samalyse.com Mon Apr 18 22:56:14 2011 From: list at samalyse.com (Olivier Guilyardi) Date: Mon, 18 Apr 2011 22:56:14 +0200 Subject: [FFmpeg-user] MPEG-4/WAV container for quick A/V sync tweaking In-Reply-To: <4DAC98EC.8050605@gmail.com> References: <4DAC941D.7030103@samalyse.com> <4DAC98EC.8050605@gmail.com> Message-ID: <4DACA56E.3010102@samalyse.com> On 04/18/2011 10:02 PM, Andrew Berg wrote: > On 2011.04.18 02:42 PM, Olivier Guilyardi wrote: >> Which container formats support an MPEG-4 video with a WAV audio stream? I'm >> looking for a format which is supported by most players. > AVI and Matroska. If by MPEG-4 video, you mean AVC/H.264, then I > wouldn't recommend AVI. Okay, yes H.264, not sure about AVC. Should be. Actually, the original data is in a 3GP container, which FFmpeg reports as: Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/tmp/VID_20110418_110132.3gp': Metadata: major_brand : 3gp4 minor_version : 768 compatible_brands: 3gp43gp6 Duration: 00:00:07.47, start: 0.000000, bitrate: 2033 kb/s Stream #0.0(eng), 1, 1/24000: Video: mpeg4, yuv420p, 720x480 [PAR 1:1 DAR 3:2], 1/24, 2030 kb/s, 23.94 fps, 24 tbr, 24k tbn, 24 tbc > Decent software players should have no problems with this, but very few > hardware players would support this combination, even if the codecs and > container are supported in other combinations. If I understand correctly, it is pretty uncommon to have a compressed video stream together with uncompressed audio. >> Is there a such format which would allow to adjust the A/V sync offset very >> quickly, by modifying a field in the header, without the need to copy or >> transcode the whole file? > Not sure if Matroska can specify an offset, but I know MP4 and MPEG-TS can. Hmm, MP4 sounds good. But no WAV according to Wikipedia. MP3 is a bit too much patent encumbered, but MP2 could do. Can I use FFmpeg from the command line to: 1. mux the MPEG-4 data from a 3GP file together with a WAV file, with MP2 encoding, into an MP4 container? I would need pure video copy, no decoding/encoding. FFmpeg would be compiled without H.264 support. 2. after 1. is done, adjust the sync offset in the MP4 container if I need to? Or would I need to code my own header modification routine? Remember, I don't want to copy the whole file, I want very quick A/V sync offset tweaking I can't do step 2. during step 1. for UI reasons. Thanks -- Olivier From baptiste.coudurier at gmail.com Mon Apr 18 22:57:12 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Mon, 18 Apr 2011 13:57:12 -0700 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <20110418125128.53550483@lrcd.com> References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> <20110418125128.53550483@lrcd.com> Message-ID: <4DACA5A8.8060106@gmail.com> On 04/18/2011 01:51 PM, Lou wrote: > On Mon, 18 Apr 2011 17:03:49 +0200 > ?ric Cholet wrote: > >> Le 18 avr. 2011 ? 16:49, Eric Cholet a ?crit : >>> With freshly downloaded ffmpeg from git, I'm getting an error when >>> trying to use the -profile option: >>> >>> >>> % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b 2500k >>> -r 25 -preset medium -profile main foo.mp4 >> >> Oops forgot to post the error message: >> >> [NULL @ 0x80240b200] [Eval @ 0x7fffffffd110] Undefined constant or >> missing '(' in 'main' [NULL @ 0x80240b200] Unable to parse option >> value "main" Invalid value 'main' for option 'profile' >> >> >>> >>> What am I doing wrong? >> -- >> ?ric Cholet > > You are doing nothing wrong. It was my mistake to mention -profile. It > is listed in 'ffmpeg -h', but is not available yet. > > I like to verify my suggestions, but this time I didn't. This is what > happens when I rely on memory as I previously tested the new options a > month ago but apparently forgot that -profile wasn't usable yet. I'm on it, don't worry :) -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From bahamutzero8825 at gmail.com Mon Apr 18 23:15:02 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Mon, 18 Apr 2011 16:15:02 -0500 Subject: [FFmpeg-user] MPEG-4/WAV container for quick A/V sync tweaking In-Reply-To: <4DACA56E.3010102@samalyse.com> References: <4DAC941D.7030103@samalyse.com> <4DAC98EC.8050605@gmail.com> <4DACA56E.3010102@samalyse.com> Message-ID: <4DACA9D6.9020509@gmail.com> On 2011.04.18 03:56 PM, Olivier Guilyardi wrote: > Okay, yes H.264, not sure about AVC. Should be. H.264 and AVC (and MPEG-4 Part 10) are all names for the same video compression standard. > Actually, the original data is > in a 3GP container, which FFmpeg reports as: > > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/tmp/VID_20110418_110132.3gp': > Metadata: > major_brand : 3gp4 > minor_version : 768 > compatible_brands: 3gp43gp6 > Duration: 00:00:07.47, start: 0.000000, bitrate: 2033 kb/s > Stream #0.0(eng), 1, 1/24000: Video: mpeg4, yuv420p, 720x480 [PAR 1:1 DAR > 3:2], 1/24, 2030 kb/s, 23.94 fps, 24 tbr, 24k tbn, 24 tbc That's ambiguous, but I think FFmpeg reports AVC/H.264 as h264 instead of mpeg4 (which means your original is MPEG-4 Part 2). I'm pretty sure 3GP supports MPEG-4 Part 2. You can always use MediaInfo to find out exactly what you have. > Can I use FFmpeg from the command line to: I recommend using MP4Box for this. Of course, if you don't specify an offset (or specify the wrong offset) the first time you mux, you will need to rewrite the entire file. MP4Box is great for producing compliant containers, but it requires rewriting or remuxing for damn near everything. In any case, Matroska will work if it can specify an offset. From baptiste.coudurier at gmail.com Mon Apr 18 23:22:10 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Mon, 18 Apr 2011 14:22:10 -0700 Subject: [FFmpeg-user] Problem converting MPEGTS file to QuickTime compatible MP4 In-Reply-To: References: <20110416112201.73a286d2@lrcd.com> Message-ID: <4DACAB82.3010906@gmail.com> On 04/18/2011 06:55 AM, Bryan Murphy wrote: > On Mon, Apr 18, 2011 at 8:52 AM, Bryan Murphy wrote: >> No. There is something about ffmpeg merging the pre-transcoded >> streams that QuickTime does not like. It's only when it performs the >> transcode inline that it is happy. > > Also, if I skip the audio stream entirely, QuickTime will play back > the video without audio. Everything seems to point to some sort of > corruption in the AAC audio stream. Use -absf aac_adtstoasc -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From list at samalyse.com Mon Apr 18 23:41:01 2011 From: list at samalyse.com (Olivier Guilyardi) Date: Mon, 18 Apr 2011 23:41:01 +0200 Subject: [FFmpeg-user] MPEG-4/WAV container for quick A/V sync tweaking In-Reply-To: <4DACA9D6.9020509@gmail.com> References: <4DAC941D.7030103@samalyse.com> <4DAC98EC.8050605@gmail.com> <4DACA56E.3010102@samalyse.com> <4DACA9D6.9020509@gmail.com> Message-ID: <4DACAFED.8000904@samalyse.com> On 04/18/2011 11:15 PM, Andrew Berg wrote: > That's ambiguous, but I think FFmpeg reports AVC/H.264 as h264 instead > of mpeg4 (which means your original is MPEG-4 Part 2). I'm pretty sure > 3GP supports MPEG-4 Part 2. You can always use MediaInfo to find out > exactly what you have. mediainfo says: General Complete name : /tmp/VID_20110418_110132.3gp Format : MPEG-4 Format profile : 3GPP Media Release 4 Codec ID : 3gp4 File size : 1.81 MiB Duration : 7s 437ms Overall bit rate : 2 045 Kbps Video ID : 1 Format : MPEG-4 Visual Format settings, BVOP : Yes Format settings, QPel : No Format settings, GMC : No warppoints Format settings, Matrix : Default (H.263) Codec ID : 20 Duration : 7s 437ms Bit rate mode : Variable Bit rate : 2 042 Kbps Nominal bit rate : 2 169 Kbps Maximum bit rate : 2 359 Kbps Width : 720 pixels Height : 480 pixels Display aspect ratio : 3:2 Frame rate mode : Variable Frame rate : 24.069 fps Minimum frame rate : 22.727 fps Maximum frame rate : 24.390 fps Standard : NTSC Color space : YUV Bit depth : 8 bits Scan type : Progressive Compression mode : Lossy Bits/(Pixel*Frame) : 0.245 Stream size : 1.81 MiB (100%) Language : English So it seems liks it's H.263. Actually, I can also get H.264. I need to be able to handle both. >> Can I use FFmpeg from the command line to: > I recommend using MP4Box for this. Of course, if you don't specify an > offset (or specify the wrong offset) the first time you mux, you will > need to rewrite the entire file. MP4Box is great for producing compliant > containers, but it requires rewriting or remuxing for damn near everything. Okay, remuxing is not an option. But since I would obtain the file from mp4box, it's not as if I was working on some random mp4 file with weird headers and such. So maybe that I could code something from scratch to tweak the sync offset quickly without using mp4box again after initial muxing. > In any case, Matroska will work if it can specify an offset. Matroska looks more and more attractive to me. It would require quite a lot of extra work in my very case, but it seems like a great format. I especially love the fact that it's designed to ease editing, which could be useful for me in the future. But, I've google all combinations of matroska, AV, sync, offset, timecode etc.. And I couldn't find anything. Any clue? -- Olivier From toronto.ffmpeg at gmail.com Mon Apr 18 23:59:13 2011 From: toronto.ffmpeg at gmail.com (Dennis P) Date: Mon, 18 Apr 2011 17:59:13 -0400 Subject: [FFmpeg-user] Job posting - encoding specialist in Toronto, Canada. Message-ID: Need an encoding specialist in Toronto (full time). For 100+ people Canadian company, young and growing. Must have skills: FFmpeg; x264; Linux; basic C & Java; basic knowledge of delivery methods and encapsulation: RTSP, HTTP, FLASH, SilverLight, iOS-HTTP. Nice to have skills: video quality analysis tools; MPEG2-TS/PS multiplexing; AC3/AAC audio compression and encapsulation; live feed formats. Please send your resume to: *toronto.ffmpeg at gmail.com* Thank you. Dennis. From cholet at logilune.com Tue Apr 19 00:22:41 2011 From: cholet at logilune.com (=?iso-8859-1?Q?=C9ric_Cholet?=) Date: Tue, 19 Apr 2011 00:22:41 +0200 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <20110418125128.53550483@lrcd.com> References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> <20110418125128.53550483@lrcd.com> Message-ID: <36F62FED-88B3-4706-A6FB-461B96929FF1@logilune.com> Le 18 avr. 2011 ? 22:51, Lou a ?crit : > On Mon, 18 Apr 2011 17:03:49 +0200 > ?ric Cholet wrote: > >> Le 18 avr. 2011 ? 16:49, Eric Cholet a ?crit : >>> With freshly downloaded ffmpeg from git, I'm getting an error when >>> trying to use the -profile option: >>> >>> >>> % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b 2500k >>> -r 25 -preset medium -profile main foo.mp4 >> >> Oops forgot to post the error message: >> >> [NULL @ 0x80240b200] [Eval @ 0x7fffffffd110] Undefined constant or >> missing '(' in 'main' [NULL @ 0x80240b200] Unable to parse option >> value "main" Invalid value 'main' for option 'profile' >> >> >>> >>> What am I doing wrong? >> -- >> ?ric Cholet > > You are doing nothing wrong. It was my mistake to mention -profile. It > is listed in 'ffmpeg -h', but is not available yet. > > I like to verify my suggestions, but this time I didn't. This is what > happens when I rely on memory as I previously tested the new options a > month ago but apparently forgot that -profile wasn't usable yet. Thanks for the reply. So how does one go about specifying an x264 profile now that those ffmpeg presets are gone? Should I pull an ealier version and go back to using -vpre main? -- ?ric Cholet From lou at fakeoutdoorsman.com Tue Apr 19 00:41:07 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Mon, 18 Apr 2011 14:41:07 -0800 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <36F62FED-88B3-4706-A6FB-461B96929FF1@logilune.com> References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> <20110418125128.53550483@lrcd.com> <36F62FED-88B3-4706-A6FB-461B96929FF1@logilune.com> Message-ID: <20110418144107.69e91b42@lrcd.com> On Tue, 19 Apr 2011 00:22:41 +0200 ?ric Cholet wrote: > Le 18 avr. 2011 ? 22:51, Lou a ?crit : > > > On Mon, 18 Apr 2011 17:03:49 +0200 > > ?ric Cholet wrote: > > > >> Le 18 avr. 2011 ? 16:49, Eric Cholet a ?crit : > >>> With freshly downloaded ffmpeg from git, I'm getting an error when > >>> trying to use the -profile option: > >>> > >>> > >>> % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b > >>> 2500k -r 25 -preset medium -profile main foo.mp4 > >> > >> Oops forgot to post the error message: > >> > >> [NULL @ 0x80240b200] [Eval @ 0x7fffffffd110] Undefined constant or > >> missing '(' in 'main' [NULL @ 0x80240b200] Unable to parse option > >> value "main" Invalid value 'main' for option 'profile' > >> > >> > >>> > >>> What am I doing wrong? > >> -- > >> ?ric Cholet > > > > You are doing nothing wrong. It was my mistake to mention -profile. > > It is listed in 'ffmpeg -h', but is not available yet. > > > > I like to verify my suggestions, but this time I didn't. This is > > what happens when I rely on memory as I previously tested the new > > options a month ago but apparently forgot that -profile wasn't > > usable yet. > > Thanks for the reply. So how does one go about specifying an x264 > profile now that those ffmpeg presets are gone? Should I pull an > ealier version and go back to using -vpre main? You can pull an earlier version if you need to encode immediately: cd ~/ffmpeg make distclean git checkout abf8342aa94bdf06bb324f6723a6743dd628d5c6 ./configure, make, install, etc make distclean git checkout master git pull (I didn't test this) or you could wait a little while. I would guess no more than a day or two for profile to be added. From bmurphy1976 at gmail.com Tue Apr 19 02:51:59 2011 From: bmurphy1976 at gmail.com (Bryan Murphy) Date: Mon, 18 Apr 2011 19:51:59 -0500 Subject: [FFmpeg-user] Problem converting MPEGTS file to QuickTime compatible MP4 In-Reply-To: <4DACAB82.3010906@gmail.com> References: <20110416112201.73a286d2@lrcd.com> <4DACAB82.3010906@gmail.com> Message-ID: On Mon, Apr 18, 2011 at 4:22 PM, Baptiste Coudurier wrote: >> Also, if I skip the audio stream entirely, QuickTime will play back >> the video without audio. ?Everything seems to point to some sort of >> corruption in the AAC audio stream. > > Use -absf aac_adtstoasc THANK YOU! I've been searching so long for a solution to this problem! I'll have to do some more thorough tests when I get back to work tomorrow, but so far this appears to be working beautifully. Bryan From dusa.adrian at gmail.com Tue Apr 19 10:05:19 2011 From: dusa.adrian at gmail.com (Adrian Dusa) Date: Tue, 19 Apr 2011 11:05:19 +0300 Subject: [FFmpeg-user] video stream out of sync Message-ID: Dear ffmpeg users, I'm a newbie, so trying to be both short and clear. Working under Kubuntu 10.1 Maverick, I have an X11 grab and an external mp3 audio file, the task being to merge both into an .avi file of reasonable dimensions (say 2MB per minute). Steps done: -- grab the video stream: ffmpeg -f x11grab -s 816x816 -i :0.0+407x162 -r 15 -vcodec libx264 -vpre lossless_ultrafast -threads 0 blahvideo.mkv -- cut the video stream from moment X till moment Y (in order to match the audio stream length): ffmpeg -i blavideo.mkv -ss 00:00:XX -t 00:0Y:YY.YY -r 15 -vcodec libx264 -vpre lossless_ultrafast -threads 0 blahcut.mkv -- merge them together: ffmpeg -i blahcut.mkv -i blahaudio.mp3 -acodec copy -vcodec libxvid -qscale 5 -b 2k -r 15 -f avi bla.avi The final result is quite satisfactory in terms of video quality, but towards the end the video stream gets desynced behind the audio stream with about 1 second every 5 minutes. Could you please advice what I could do in order to avoid this? Thanks very much, this is a great software, Adrian From gavr.mail at gmail.com Tue Apr 19 10:11:32 2011 From: gavr.mail at gmail.com (Kirill Gavrilov) Date: Tue, 19 Apr 2011 12:11:32 +0400 Subject: [FFmpeg-user] video stream out of sync In-Reply-To: References: Message-ID: > > Working under Kubuntu 10.1 Maverick, I have an X11 grab and an > external mp3 audio file, the task being to merge both into an .avi > I'm not expert here. But does your MP3 file has VBR or constant bit-rate? Show your output ffmpeg -i myAudio.mp3 if you doesn't understand. ----------------------------------------------- Kirill Gavrilov, Software designer. From dusa.adrian at gmail.com Tue Apr 19 10:25:50 2011 From: dusa.adrian at gmail.com (Adrian Dusa) Date: Tue, 19 Apr 2011 11:25:50 +0300 Subject: [FFmpeg-user] video stream out of sync In-Reply-To: References: Message-ID: On Tue, Apr 19, 2011 at 11:11, Kirill Gavrilov wrote: >> [...] > I'm not expert here. But does your MP3 file has VBR or constant bit-rate? > Show your output ffmpeg -i myAudio.mp3 if you doesn't understand. I don't know whether it is constant or VBR, but I believe this is the chunk of interest from the output of ffmpeg -i $ ffmpeg -i blahaudio.mp3 libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mp3 @ 0x914fdb0]Header missing [mp3 @ 0x914ebc0]max_analyze_duration reached [mp3 @ 0x914ebc0]Estimating duration from bitrate, this may be inaccurate Input #0, mp3, from 'blahaudio.mp3': Duration: 00:05:27.86, start: 0.000000, bitrate: 127 kb/s Stream #0.0: Audio: mp3, 22050 Hz, 1 channels, s16, 128 kb/s Many thanks, Adrian -- Adrian Dusa Romanian Social Data Archive 1, Schitu Magureanu Bd. 050025 Bucharest sector 5 Romania Tel.:+40 21 3126618 \ ? ? ? ?+40 21 3120210 / int.101 Fax: +40 21 3158391 From acgacosta at gmail.com Tue Apr 19 12:39:58 2011 From: acgacosta at gmail.com (Antony Acosta) Date: Tue, 19 Apr 2011 18:39:58 +0800 Subject: [FFmpeg-user] FFMPEG Update Message-ID: Hi, I would like to ask if updating my ffmpeg to the newest version would affect how my current settings are right now. For example I am using certain commands and codes for ffmpeg to convert videos, now if I update my ffmpeg to it's latest version, would it affect how it works? -- Antony [Programmer, Allied Web Design] ?[C][G]? "THere is no cHaLLenge THaT can'T Be FinisHeD!" ? ?'?'\????\?=(???)=?/????/'?'? ? From h.reindl at thelounge.net Tue Apr 19 12:47:06 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Tue, 19 Apr 2011 12:47:06 +0200 Subject: [FFmpeg-user] FFMPEG Update In-Reply-To: References: Message-ID: <4DAD682A.9060903@thelounge.net> Am 19.04.2011 12:39, schrieb Antony Acosta: > Hi, > > I would like to ask if updating my ffmpeg to the newest version would affect > how my current settings are right now. For example I am using certain > commands and codes for ffmpeg to convert videos, now if I update my ffmpeg > to it's latest version, would it affect how it works? You should tell us waht version you are using! pad/crop params are removed and replaced by filters is one thing i needed to change in our php-wrapper some months ago -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From acgacosta at gmail.com Tue Apr 19 12:50:10 2011 From: acgacosta at gmail.com (Antony Acosta) Date: Tue, 19 Apr 2011 18:50:10 +0800 Subject: [FFmpeg-user] FFMPEG Update In-Reply-To: <4DAD682A.9060903@thelounge.net> References: <4DAD682A.9060903@thelounge.net> Message-ID: I am currently using FFmpeg version SVN-r20220 -- Antony [Programmer, Allied Web Design] ?[C][G]? "THere is no cHaLLenge THaT can'T Be FinisHeD!" ? ?'?'\????\?=(???)=?/????/'?'? ? On Tue, Apr 19, 2011 at 6:47 PM, Reindl Harald wrote: > > Am 19.04.2011 12:39, schrieb Antony Acosta: > > Hi, > > > > I would like to ask if updating my ffmpeg to the newest version would > affect > > how my current settings are right now. For example I am using certain > > commands and codes for ffmpeg to convert videos, now if I update my > ffmpeg > > to it's latest version, would it affect how it works? > > You should tell us waht version you are using! > > pad/crop params are removed and replaced by filters is one thing > i needed to change in our php-wrapper some months ago > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From yadnesh at gmail.com Tue Apr 19 13:04:34 2011 From: yadnesh at gmail.com (Yadnesh Phadke) Date: Tue, 19 Apr 2011 16:34:34 +0530 Subject: [FFmpeg-user] Issues in Creating SDP for a H264/AAC file Message-ID: Hi All, I have recorded a video from my Samsung Galaxy S (Android 2.2) phone which I am trying to stream using the following ffmpeg command. ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- ffmpeg -i ~/Media/video-2011-04-13-15-13-09.mp4 -an -vcodec copy -f rtp rtp://192.168.1.122:11111 -i ~/Media/video-2011-04-13-15-13-09.mp4 -vn -strict experimental -acodec copy -flags +global_header -f rtp rtp:// 192.168.1.122:22222 -newaudio ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- The created SDP seems to be incomplete. The output shows no value for " sprop-parameter-sets" Earlier, I have successfully streamed and played the stream for other MP4 files having H264 and AAC encoded media. Facing issue with files created on Android. Is this issue related to any specific encoding level of H264? Here is the complete output of that streaming command: ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- FFmpeg version 0.6, Copyright (c) 2000-2010 the FFmpeg developers built on Sep 20 2010 18:45:33 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) configuration: --prefix=/home/yadnesh --enable-shared --disable-static --disable-optimizations --disable-mmx --disable-stripping --enable-debug --enable-version3 --enable-pthreads --enable-gpl --enable-libfaac --enable-libx264 --enable-nonfree libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.11. 0 / 0.11. 0 Start time = 0 Seems stream 1 codec frame rate differs from container frame rate: 60000.00 (60000/1) -> 1000.00 (1000/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/home/yadnesh/Media/video-2011-04-13-15-13-09.mp4': Metadata: major_brand : 3gp4 minor_version : 768 compatible_brands: 3gp43gp6 Duration: 00:00:09.72, start: 0.000000, bitrate: 3457 kb/s Stream #0.0(eng): Audio: aac, 16000 Hz, mono, s16, 56 kb/s Stream #0.1(eng): Video: h264, yuv420p, 720x480, 3435 kb/s, 29.85 fps, 1k tbr, 30k tbn, 60k tbc Start time = 0 Seems stream 1 codec frame rate differs from container frame rate: 60000.00 (60000/1) -> 1000.00 (1000/1) Input #1, mov,mp4,m4a,3gp,3g2,mj2, from '/home/yadnesh/Media/video-2011-04-13-15-13-09.mp4': Metadata: major_brand : 3gp4 minor_version : 768 compatible_brands: 3gp43gp6 Duration: 00:00:09.72, start: 0.000000, bitrate: 3457 kb/s Stream #1.0(eng): Audio: aac, 16000 Hz, mono, s16, 56 kb/s Stream #1.1(eng): Video: h264, yuv420p, 720x480, 3435 kb/s, 29.85 fps, 1k tbr, 30k tbn, 60k tbc Output #0, rtp, to 'rtp://192.168.1.122:11111': Metadata: encoder : Lavf52.64.2 Stream #0.0(eng): Video: avc1 / 0x31637661, yuv420p, 720x480, q=2-31, 3435 kb/s, 90k tbn, 30k tbc Output #1, rtp, to 'rtp://192.168.1.122:22222': Metadata: encoder : Lavf52.64.2 Stream #1.0(eng): Audio: aac, 16000 Hz, mono, 56 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #1.0 SDP: v=0 o=- 0 0 IN IP4 192.168.1.122 s=No Name t=0 0 a=tool:libavformat 52.64.2 m=video 0 RTP/AVP 96 c=IN IP4 192.168.1.122 b=AS:3435 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1; sprop-parameter-sets= a=control:streamid=0 m=audio 0 RTP/AVP 97 c=IN IP4 192.168.1.122 b=AS:56 a=rtpmap:97 MPEG4-GENERIC/16000/1 a=fmtp:97 profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3; config=1408 a=control:streamid=1 Press [q] to stop encoding frame= 287 fps= 0 q=-1.0 Lsize= 0kB time=9.58 bitrate= 0.0kbits/s video:4032kB audio:68kB global headers:0kB muxing overhead -99.999333% ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- On debugging into the FFMpeg code, I found that while creating sdp it does not go in to the loop (of function static char *extradata2psets(AVCodecContext *c)) while (r < c->extradata + c->extradata_size) { } Can you please help me with the causes of such wrong SDP? Could the cause be the file format (3GP) or the codec (H264)? Regards, -- Yadnesh ---------------------------------------------------------------------- Christopher Morley - There is only one success - to be able to spend your life in your own way. From h.reindl at thelounge.net Tue Apr 19 13:21:16 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Tue, 19 Apr 2011 13:21:16 +0200 Subject: [FFmpeg-user] FFMPEG Update In-Reply-To: References: <4DAD682A.9060903@thelounge.net> Message-ID: <4DAD702C.1010701@thelounge.net> Am 19.04.2011 12:50, schrieb Antony Acosta: > I am currently using FFmpeg version SVN-r20220 sounds quite old but i do not know every svn-release by it's name :-) why do you not post from begin the first output of "ffmpeg" so that anybody has useful informations? [harry at nb-rhsoft:~]$ ffmpeg FFmpeg version git-N-28742-g1caa412, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 13 2011 13:00:04 with gcc 4.5.1 20100924 (Red Hat 4.5.1-4) -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From yadnesh at gmail.com Tue Apr 19 13:30:29 2011 From: yadnesh at gmail.com (Yadnesh Phadke) Date: Tue, 19 Apr 2011 17:00:29 +0530 Subject: [FFmpeg-user] Issues in Creating SDP for a H264/AAC file In-Reply-To: References: Message-ID: The video plays well on Android handset and on VLC player. The issue is only with streaming using FFMpeg. I can upload the file for reference, if required. Regards, Yadnesh On Tue, Apr 19, 2011 at 4:34 PM, Yadnesh Phadke wrote: > Hi All, > > > > I have recorded a video from my Samsung Galaxy S (Android 2.2) phone which > I am trying to stream using the following ffmpeg command. > > > ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- > > ffmpeg -i ~/Media/video-2011-04-13-15-13-09.mp4 -an -vcodec copy -f rtp > rtp://192.168.1.122:11111 -i ~/Media/video-2011-04-13-15-13-09.mp4 -vn > -strict experimental -acodec copy -flags +global_header -f rtp rtp:// > 192.168.1.122:22222 -newaudio > > > ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- > > The created SDP seems to be incomplete. The output shows no value for " > sprop-parameter-sets" > > Earlier, I have successfully streamed and played the stream for other MP4 > files having H264 and AAC encoded media. Facing issue with files created on > Android. Is this issue related to any specific encoding level of H264? > > Here is the complete output of that streaming command: > > > ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- > > FFmpeg version 0.6, Copyright (c) 2000-2010 the FFmpeg developers > > built on Sep 20 2010 18:45:33 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) > > configuration: --prefix=/home/yadnesh --enable-shared --disable-static > --disable-optimizations --disable-mmx --disable-stripping --enable-debug > --enable-version3 --enable-pthreads --enable-gpl --enable-libfaac > --enable-libx264 --enable-nonfree > > libavutil 50.15. 1 / 50.15. 1 > > libavcodec 52.72. 2 / 52.72. 2 > > libavformat 52.64. 2 / 52.64. 2 > > libavdevice 52. 2. 0 / 52. 2. 0 > > libswscale 0.11. 0 / 0.11. 0 > > > Start time = 0 > > > Seems stream 1 codec frame rate differs from container frame rate: 60000.00 > (60000/1) -> 1000.00 (1000/1) > > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from > '/home/yadnesh/Media/video-2011-04-13-15-13-09.mp4': > > Metadata: > > major_brand : 3gp4 > > minor_version : 768 > > compatible_brands: 3gp43gp6 > > Duration: 00:00:09.72, start: 0.000000, bitrate: 3457 kb/s > > Stream #0.0(eng): Audio: aac, 16000 Hz, mono, s16, 56 kb/s > > Stream #0.1(eng): Video: h264, yuv420p, 720x480, 3435 kb/s, 29.85 fps, > 1k tbr, 30k tbn, 60k tbc > > > Start time = 0 > > > Seems stream 1 codec frame rate differs from container frame rate: 60000.00 > (60000/1) -> 1000.00 (1000/1) > > Input #1, mov,mp4,m4a,3gp,3g2,mj2, from > '/home/yadnesh/Media/video-2011-04-13-15-13-09.mp4': > > Metadata: > > major_brand : 3gp4 > > minor_version : 768 > > compatible_brands: 3gp43gp6 > > Duration: 00:00:09.72, start: 0.000000, bitrate: 3457 kb/s > > Stream #1.0(eng): Audio: aac, 16000 Hz, mono, s16, 56 kb/s > > Stream #1.1(eng): Video: h264, yuv420p, 720x480, 3435 kb/s, 29.85 fps, > 1k tbr, 30k tbn, 60k tbc > > Output #0, rtp, to 'rtp://192.168.1.122:11111': > > Metadata: > > encoder : Lavf52.64.2 > > Stream #0.0(eng): Video: avc1 / 0x31637661, yuv420p, 720x480, q=2-31, > 3435 kb/s, 90k tbn, 30k tbc > > Output #1, rtp, to 'rtp://192.168.1.122:22222': > > Metadata: > > encoder : Lavf52.64.2 > > Stream #1.0(eng): Audio: aac, 16000 Hz, mono, 56 kb/s > > Stream mapping: > > Stream #0.1 -> #0.0 > > Stream #0.0 -> #1.0 > > SDP: > > v=0 > > o=- 0 0 IN IP4 192.168.1.122 > > s=No Name > > t=0 0 > > a=tool:libavformat 52.64.2 > > m=video 0 RTP/AVP 96 > > c=IN IP4 192.168.1.122 > > b=AS:3435 > > a=rtpmap:96 H264/90000 > > a=fmtp:96 packetization-mode=1; sprop-parameter-sets= > > a=control:streamid=0 > > m=audio 0 RTP/AVP 97 > > c=IN IP4 192.168.1.122 > > b=AS:56 > > a=rtpmap:97 MPEG4-GENERIC/16000/1 > > a=fmtp:97 > profile-level-id=1;mode=AAC-hbr;sizelength=13;indexlength=3;indexdeltalength=3; > config=1408 > > a=control:streamid=1 > > > Press [q] to stop encoding > > frame= 287 fps= 0 q=-1.0 Lsize= 0kB time=9.58 bitrate= 0.0kbits/s > > video:4032kB audio:68kB global headers:0kB muxing overhead -99.999333% > > > ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- > > > > On debugging into the FFMpeg code, I found that while creating sdp it does > not go in to the loop (of function static char > *extradata2psets(AVCodecContext *c)) > > while (r < c->extradata + c->extradata_size) { > > } > > > > Can you please help me with the causes of such wrong SDP? Could the cause > be the file format (3GP) or the codec (H264)? > > > Regards, > > -- Yadnesh > > ---------------------------------------------------------------------- > Christopher Morley - There is only one success - to be able to spend your > life in your own way. > -- ---------------------------------------------------------------------- Christopher Morley - There is only one success - to be able to spend your life in your own way. From acgacosta at gmail.com Tue Apr 19 13:31:08 2011 From: acgacosta at gmail.com (Antony Acosta) Date: Tue, 19 Apr 2011 19:31:08 +0800 Subject: [FFmpeg-user] FFMPEG Update In-Reply-To: <4DAD702C.1010701@thelounge.net> References: <4DAD682A.9060903@thelounge.net> <4DAD702C.1010701@thelounge.net> Message-ID: This are basically the codes that will be most affected if ever we update the FFmpeg version to its latest: - ffmpeg -i input.flv -ss 00:00:01 -s 640x380 output.jpg - ffmpeg -i input.flv -ar 44100 -f flv -b 1000kb output.flv Will the update affected those codes somehow? Because I also tried other stuffs like converting to mp4 videos such as: - ffmpeg -i input.flv input -acodec aac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +4mv+trell -aic 2 -cmp 2 -subcmp 2 -s 320x180 -title X output.mp4 - ffmpeg -y -i input.flv -b 768 -s 320x240 -vcodec xvid -ab 128 -acodec aac -ac 2 -ab 64 -f mp4 output.mp4 However this didn't work. I'm guessing I would need codecs installed to have this working and to update my FFmpeg to it's latest version? -- Antony [Programmer, Allied Web Design] ?[C][G]? "THere is no cHaLLenge THaT can'T Be FinisHeD!" ? ?'?'\????\?=(???)=?/????/'?'? ? On Tue, Apr 19, 2011 at 7:21 PM, Reindl Harald wrote: > > Am 19.04.2011 12:50, schrieb Antony Acosta: > > I am currently using FFmpeg version SVN-r20220 > > sounds quite old but i do not know every svn-release by it's name :-) > why do you not post from begin the first output of "ffmpeg" so that > anybody has useful informations? > > [harry at nb-rhsoft:~]$ ffmpeg > FFmpeg version git-N-28742-g1caa412, Copyright (c) 2000-2011 the FFmpeg > developers > built on Apr 13 2011 13:00:04 with gcc 4.5.1 20100924 (Red Hat 4.5.1-4) > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From david.ariens at gmail.com Tue Apr 19 13:41:26 2011 From: david.ariens at gmail.com (David) Date: Tue, 19 Apr 2011 11:41:26 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Issue_with_watermarking_a_video_-=3E_resu?= =?utf-8?q?lt_is_garbled=09=28pink_and_green_lines=29?= References: Message-ID: Anyone an idea? David gmail.com> writes: > .... From dusa.adrian at gmail.com Tue Apr 19 13:43:32 2011 From: dusa.adrian at gmail.com (Adrian Dusa) Date: Tue, 19 Apr 2011 14:43:32 +0300 Subject: [FFmpeg-user] video stream out of sync In-Reply-To: References: Message-ID: On Tue, Apr 19, 2011 at 11:25, Adrian Dusa wrote: > On Tue, Apr 19, 2011 at 11:11, Kirill Gavrilov wrote: >>> [...] >> I'm not expert here. But does your MP3 file has VBR or constant bit-rate? >> Show your output ffmpeg -i myAudio.mp3 if you doesn't understand. > > I don't know whether it is constant or VBR, but I believe this is the > chunk of interest from the output of ffmpeg -i Oh, I just edited the mp3 file with Audacity, saved it as constant bit-rate and got the very same output from ffmpeg -i. So the short answer is yes, the file is constant bit-rate. Would this explain the video desync? Best wishes, and thanks again in advance for your time, Adrian From av at bsbc.nb.ca Tue Apr 19 13:49:45 2011 From: av at bsbc.nb.ca (Anthony Brown) Date: Tue, 19 Apr 2011 08:49:45 -0300 Subject: [FFmpeg-user] video stream out of sync In-Reply-To: References: Message-ID: <4DAD76D9.1050005@bsbc.nb.ca> On 11-04-19 08:43 AM, Adrian Dusa wrote: > > Oh, I just edited the mp3 file with Audacity, saved it as constant > bit-rate and got the very same output from ffmpeg -i. > So the short answer is yes, the file is constant bit-rate. Would this > explain the video desync? Maybe you need to experiment with the -async and/or -vsync settings? I've find conversions get totally out of whack without async. -- Anthony Brown Audiovisual coordinator Brunswick Street Baptist Church Telephone: (506)-458-8348 (leave message) Email: av at bsbc.nb.ca -------------- next part -------------- A non-text attachment was scrubbed... Name: av.vcf Type: text/x-vcard Size: 163 bytes Desc: not available URL: From h.reindl at thelounge.net Tue Apr 19 14:34:08 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Tue, 19 Apr 2011 14:34:08 +0200 Subject: [FFmpeg-user] Issue with watermarking a video -> result is garbled (pink and green lines) In-Reply-To: References: Message-ID: <4DAD8140.4080206@thelounge.net> Am 19.04.2011 13:41, schrieb David: > result is garbled (pink and green lines) > Anyone an idea? not really BUT i have this in all videos with the latest snaphots without using watermarks the last one with perfect results is 20110331/git-g1caa412 for me and maybe few days later because i was sometimes too busy for daily snapshot i guessed this is because i made a "unclean"-change in x264 to get "so.102" as binary instead the real used "so.115" because i can would not be able to resolve dependencies on Fedora 14 without and mplayer/vlc/kaffeine are playing h264-files well so this is fine for me as long this is a private build :-) since your posting i do ot longer believe that it has to do with my H264 change and i hope someday the builds are working for mee too again.... -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From dusa.adrian at gmail.com Tue Apr 19 15:29:32 2011 From: dusa.adrian at gmail.com (Adrian Dusa) Date: Tue, 19 Apr 2011 16:29:32 +0300 Subject: [FFmpeg-user] video stream out of sync In-Reply-To: <4DAD76D9.1050005@bsbc.nb.ca> References: <4DAD76D9.1050005@bsbc.nb.ca> Message-ID: On Tue, Apr 19, 2011 at 14:49, Anthony Brown wrote: > On 11-04-19 08:43 AM, Adrian Dusa wrote: >> >> Oh, I just edited the mp3 file with Audacity, saved it as constant >> bit-rate and got the very same output from ffmpeg -i. >> So the short answer is yes, the file is constant bit-rate. Would this >> explain the video desync? > > Maybe you need to experiment with the -async and/or -vsync settings? I've > find conversions get totally out of whack without async. Oh, that might be one solution, only from the man of ffmpeg I am unable to figure out what values of the async stretch or squeeze. Alternatively, I could try to grab both audio and video in the mkv file. I tried: ffmpeg -f alsa -ac 2 -i pulse This records the input from the microphone, and I need to grab what's being presented on the screen (along with the video file). I do have the original audio (a .flv file), but since I bump into these syncing problems I might try to grab them both. So basically now I have two questions: - what are the values for the async for stretching/squeezing of the audio stream? - how to grab the "desktop" sound instead of the "mic" sound? Thanks again, Adrian From av at bsbc.nb.ca Tue Apr 19 15:50:02 2011 From: av at bsbc.nb.ca (Anthony Brown) Date: Tue, 19 Apr 2011 10:50:02 -0300 Subject: [FFmpeg-user] video stream out of sync In-Reply-To: References: <4DAD76D9.1050005@bsbc.nb.ca> Message-ID: <4DAD930A.5070908@bsbc.nb.ca> On 11-04-19 10:29 AM, Adrian Dusa wrote: > Oh, that might be one solution, only from the man of ffmpeg I am > unable to figure out what values of the async stretch or squeeze. The documentation is a bit sketchy on that (and many other points), but them as complain can feel free to rewrite it :). AFAIK, -async X will resample the audio track to speed up or slow down by as much as X samples per second as required to match the video timestamps. You probably don't need to make big adjustments, you're talking 1 second in 5 minutes, less than 1%. If you sample audio at 48000 an -async 480 would be more than enough to compensate. > - how to grab the "desktop" sound instead of the "mic" sound? Can't help you there. -- Anthony Brown Audiovisual coordinator Brunswick Street Baptist Church Telephone: (506)-458-8348 (leave message) Email: av at bsbc.nb.ca -------------- next part -------------- A non-text attachment was scrubbed... Name: av.vcf Type: text/x-vcard Size: 163 bytes Desc: not available URL: From dusa.adrian at gmail.com Tue Apr 19 16:04:04 2011 From: dusa.adrian at gmail.com (Adrian Dusa) Date: Tue, 19 Apr 2011 17:04:04 +0300 Subject: [FFmpeg-user] video stream out of sync In-Reply-To: <4DAD930A.5070908@bsbc.nb.ca> References: <4DAD76D9.1050005@bsbc.nb.ca> <4DAD930A.5070908@bsbc.nb.ca> Message-ID: On Tue, Apr 19, 2011 at 16:50, Anthony Brown wrote: > The documentation is a bit sketchy on that (and many other points), but them > as complain can feel free to rewrite it :). ?AFAIK, -async X will resample > the audio track to speed up or slow down by as much as X samples per second > as required to match the video timestamps. ?You probably don't need to make > big adjustments, you're talking 1 second in 5 minutes, less than 1%. ?If you > sample audio at 48000 an -async 480 would be more than enough to compensate. Thanks very much, no complaining here, I find ffmpeg as a great piece of software :) Will go for trial and error then, and hopefully find the right solution. >> - how to grab the "desktop" sound instead of the "mic" sound? > Can't help you there. Hmm, there must be a way to intercept the sound that is "sent" to the speakers. If only I could find the answer for this one... Best wishes, Adrian From gunvantsahu at gmail.com Mon Apr 18 17:24:00 2011 From: gunvantsahu at gmail.com (gunvant sahu) Date: Mon, 18 Apr 2011 20:54:00 +0530 Subject: [FFmpeg-user] how to combine videos side by side Message-ID: Hello All I am looking for a solution. Please have a look into this site http://www.shervinemami.co.cc/openCV.html%22 I want to combine videos like this. Is this possible with ffmpeg? Thanks in advance Gunvant From sdbhabal at gmail.com Tue Apr 19 15:19:16 2011 From: sdbhabal at gmail.com (santosh bhabal) Date: Tue, 19 Apr 2011 18:49:16 +0530 Subject: [FFmpeg-user] using ffserver Message-ID: Hi All, I m new to using ffserver & ffmpeg. I wanted to stream live containt over internet. I have my "set top box" connected to my system using "osprey capture card". I have googled a lot & found that "ffserver & ffmpeg" can fulfill my requirement. Any help will be apperitiated. Thanks in advance. Regards Santosh From toronto.ffmpeg at gmail.com Tue Apr 19 17:26:31 2011 From: toronto.ffmpeg at gmail.com (Dennis P) Date: Tue, 19 Apr 2011 11:26:31 -0400 Subject: [FFmpeg-user] Job posting - encoding specialist in Toronto, Canada. In-Reply-To: References: Message-ID: Need an encoding specialist in Toronto (full time). For 100+ people Canadian company, young and growing. Must have skills: FFmpeg; x264; Linux; basic C & Java; basic knowledge of delivery methods and encapsulation: RTSP, HTTP, FLASH, SilverLight, iOS-HTTP. Nice to have skills: video quality analysis tools; MPEG2-TS/PS multiplexing; AC3/AAC audio compression and encapsulation; live feed formats. Please send your resume to: *toronto.ffmpeg at gmail.com* Thank you. Dennis. From thilo.borgmann at googlemail.com Tue Apr 19 17:26:42 2011 From: thilo.borgmann at googlemail.com (Thilo Borgmann) Date: Tue, 19 Apr 2011 17:26:42 +0200 Subject: [FFmpeg-user] FFMPEG Update In-Reply-To: References: <4DAD682A.9060903@thelounge.net> <4DAD702C.1010701@thelounge.net> Message-ID: <4DADA9B2.6000907@googlemail.com> Am 19.04.11 13:31, schrieb Antony Acosta: > This are basically the codes that will be most affected if ever we update > the FFmpeg version to its latest: > > > - ffmpeg -i input.flv -ss 00:00:01 -s 640x380 output.jpg > - ffmpeg -i input.flv -ar 44100 -f flv -b 1000kb output.flv > > > Will the update affected those codes somehow? I don't think anyone can say that for sure. You should download the current source from git and compile it to have a current version of ffmpeg next to your existing one. You can test if the current behaviour of ffmpeg still meets your requirements and do that without breaking your current toolchain. > ...Because I also tried other > stuffs like converting to mp4 videos such as: > > > - ffmpeg -i input.flv input -acodec aac -ab 128kb -vcodec mpeg4 -b 1200kb > -mbd 2 -flags +4mv+trell -aic 2 -cmp 2 -subcmp 2 -s 320x180 -title X > output.mp4 > - ffmpeg -y -i input.flv -b 768 -s 320x240 -vcodec xvid -ab 128 -acodec > aac -ac 2 -ab 64 -f mp4 output.mp4 > > However this didn't work. I'm guessing I would need codecs installed to have > this working and to update my FFmpeg to it's latest version? When configuring you will see what video codecs will be available in your new build of FFmpeg and can install necessary missing codec libraries before actually building ffmpeg. -Thilo From alexandre.ferrieux at orange-ftgroup.com Tue Apr 19 17:30:47 2011 From: alexandre.ferrieux at orange-ftgroup.com (Alexandre Ferrieux) Date: Tue, 19 Apr 2011 17:30:47 +0200 Subject: [FFmpeg-user] how to combine videos side by side In-Reply-To: References: Message-ID: <4DADAAA7.6070601@orange-ftgroup.com> On 18/04/2011 17:24, gunvant sahu wrote: > Hello All > > I am looking for a solution. Please have a look into this site > > http://www.shervinemami.co.cc/openCV.html%22 > > I want to combine videos like this. Is this possible with ffmpeg? > > Thanks in advance Look at the 'overlay' and 'movie' vfilters. Also, look for these keywords on the list's archive. -Alex From seandarcy2 at gmail.com Tue Apr 19 17:56:40 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Tue, 19 Apr 2011 11:56:40 -0400 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DAA2FE3.5040708@gmail.com> References: <4DAA2FE3.5040708@gmail.com> Message-ID: On 04/16/2011 08:10 PM, Baptiste Coudurier wrote: > Hi everyone, > > Latest git has now added 2 new options for libx264 encoding: > -preset > -tune > > These options are directly mapped to internal tune and presets within > libx264. The old presets files have been removed, please use the new > options. > > Best regards. > How do these new presets work with 2 passes? For instance, I have a script that used: -pass 1 -vpre slow_firstpass -pass 2 -vpre slow I've changed it to: -pass 1 -preset slow -pass 2 -preset slow I used to get 100 fps on the 1st pass, now about 70. Is this to be expected? Or do I need to do something else on the 1st pass? Here's the full script: ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -preset slow \ -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 -f mp4 -y /dev/null ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -preset slow -timestamp now \ -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 $1.m4v Thanks, sean From seandarcy2 at gmail.com Tue Apr 19 18:17:36 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Tue, 19 Apr 2011 12:17:36 -0400 Subject: [FFmpeg-user] fix for building ffmpeg with opencv-2.2 Message-ID: ffmpeg won't build with --enable-libopencv for opencv-2.2 because of the changed opencv structure. The problem is the second include in libavfilter/vf_libopencv.c : #include just delete it. Don't know if ffmpeg would still build with opencv < 2.2. sean From tim.nicholson at bbc.co.uk Tue Apr 19 18:33:24 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 19 Apr 2011 17:33:24 +0100 Subject: [FFmpeg-user] Issue with watermarking a video -> result is garbled (pink and green lines) In-Reply-To: References: Message-ID: <4DADB954.7020108@bbc.co.uk> On 17/04/11 06:59, David wrote: > Hi, > > I have an issue with watermarking a video using ffmpeg and libavfilter. > > This is my testvideo: www.coachteam.com/testvideo/paul.mov > This is the watermark: www.coachteam.com/testvideo/coachteamlogo.png > And this is the result: www.coachteam.com/testvideo/paul.mp4 I have just done a test using your .png, but with a gash file of my own and the results look correct. ffmpeg -i NeighEC001.mpg -threads 0 \ -vf "movie=coachteamlogo.png [logo]; [in][logo] overlay=10:main_h-overlay_h-10 [out]" \ -vcodec dvvideo -s 720x576 -an -y out.mov -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From tim.nicholson at bbc.co.uk Tue Apr 19 18:36:02 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 19 Apr 2011 17:36:02 +0100 Subject: [FFmpeg-user] Issue with watermarking a video -> result is garbled (pink and green lines) In-Reply-To: <4DADB954.7020108@bbc.co.uk> References: <4DADB954.7020108@bbc.co.uk> Message-ID: <4DADB9F2.9020007@bbc.co.uk> On 19/04/11 17:33, Tim Nicholson wrote: > On 17/04/11 06:59, David wrote: >> Hi, >> >> I have an issue with watermarking a video using ffmpeg and libavfilter. >> >> This is my testvideo: www.coachteam.com/testvideo/paul.mov >> This is the watermark: www.coachteam.com/testvideo/coachteamlogo.png >> And this is the result: www.coachteam.com/testvideo/paul.mp4 > > I have just done a test using your .png, but with a gash file of my own > and the results look correct. > > ffmpeg -i NeighEC001.mpg -threads 0 \ > -vf "movie=coachteamlogo.png [logo]; [in][logo] > overlay=10:main_h-overlay_h-10 [out]" \ > -vcodec dvvideo -s 720x576 -an -y out.mov > > And now with your file:- ffmpeg -i paul.mov -threads 0 \ -vf "movie=coachteamlogo.png [logo]; [in][logo] overlay=10:main_h-overlay_h-10 [out]" \ -vcodec dvvideo -s 720x576 -an -y out-paul.mov and it still looks good..... -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From david.ariens at gmail.com Tue Apr 19 19:24:45 2011 From: david.ariens at gmail.com (David) Date: Tue, 19 Apr 2011 17:24:45 +0000 (UTC) Subject: [FFmpeg-user] Issue with watermarking a video -> result is garbled (pink and green lines) References: <4DADB954.7020108@bbc.co.uk> <4DADB9F2.9020007@bbc.co.uk> Message-ID: Tim Nicholson bbc.co.uk> writes: > > > And now with your file:- > > ffmpeg -i paul.mov -threads 0 \ > -vf "movie=coachteamlogo.png [logo]; [in][logo] > overlay=10:main_h-overlay_h-10 [out]" \ > -vcodec dvvideo -s 720x576 -an -y out-paul.mov > > and it still looks good..... > Tim, When running your code I still got the same crappy output. Which build are you running? I will try testing with another build. From baptiste.coudurier at gmail.com Tue Apr 19 21:27:33 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Tue, 19 Apr 2011 12:27:33 -0700 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: References: <4DAA2FE3.5040708@gmail.com> Message-ID: <4DADE225.6070907@gmail.com> Hi, On 04/19/2011 08:56 AM, sean darcy wrote: > On 04/16/2011 08:10 PM, Baptiste Coudurier wrote: >> Hi everyone, >> >> Latest git has now added 2 new options for libx264 encoding: >> -preset >> -tune >> >> These options are directly mapped to internal tune and presets within >> libx264. The old presets files have been removed, please use the new >> options. >> >> Best regards. >> > How do these new presets work with 2 passes? > > For instance, I have a script that used: > > -pass 1 -vpre slow_firstpass > > -pass 2 -vpre slow > > I've changed it to: > > -pass 1 -preset slow > > -pass 2 -preset slow > > I used to get 100 fps on the 1st pass, now about 70. Is this to be > expected? Or do I need to do something else on the 1st pass? > > Here's the full script: > > ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -preset slow \ > -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 -f mp4 -y > /dev/null > ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -preset slow > -timestamp now \ > -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 $1.m4v fastfirstpass should be enabled by default. Can you please double check that the parameters printed are the same ? -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From baptiste.coudurier at gmail.com Tue Apr 19 22:32:45 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Tue, 19 Apr 2011 13:32:45 -0700 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DACA5A8.8060106@gmail.com> References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> <20110418125128.53550483@lrcd.com> <4DACA5A8.8060106@gmail.com> Message-ID: <4DADF16D.4050003@gmail.com> On 04/18/2011 01:57 PM, Baptiste Coudurier wrote: > On 04/18/2011 01:51 PM, Lou wrote: >> On Mon, 18 Apr 2011 17:03:49 +0200 >> ?ric Cholet wrote: >> >>> Le 18 avr. 2011 ? 16:49, Eric Cholet a ?crit : >>>> With freshly downloaded ffmpeg from git, I'm getting an error when >>>> trying to use the -profile option: >>>> >>>> >>>> % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b 2500k >>>> -r 25 -preset medium -profile main foo.mp4 >>> >>> Oops forgot to post the error message: >>> >>> [NULL @ 0x80240b200] [Eval @ 0x7fffffffd110] Undefined constant or >>> missing '(' in 'main' [NULL @ 0x80240b200] Unable to parse option >>> value "main" Invalid value 'main' for option 'profile' >>> >>> >>>> >>>> What am I doing wrong? >>> -- >>> ?ric Cholet >> >> You are doing nothing wrong. It was my mistake to mention -profile. It >> is listed in 'ffmpeg -h', but is not available yet. >> >> I like to verify my suggestions, but this time I didn't. This is what >> happens when I rely on memory as I previously tested the new options a >> month ago but apparently forgot that -profile wasn't usable yet. > > I'm on it, don't worry :) > All right, -profile should work now. Please report any problem. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From jonathan.michael.henson at gmail.com Tue Apr 19 23:48:39 2011 From: jonathan.michael.henson at gmail.com (Jonathan Michael Henson) Date: Tue, 19 Apr 2011 16:48:39 -0500 Subject: [FFmpeg-user] rtp payloader for h261 Message-ID: <1303249719.3406.4.camel@Linus> I am using gstreamer to create an external rtp channel for a H323/SIP server. My problem is that I cannot find a rtp payloader for ffenc_h261. I know this has to be simple, could anyone please help me out? Thanks, From seandarcy2 at gmail.com Wed Apr 20 00:05:07 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Tue, 19 Apr 2011 18:05:07 -0400 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DADE225.6070907@gmail.com> References: <4DAA2FE3.5040708@gmail.com> <4DADE225.6070907@gmail.com> Message-ID: On 04/19/2011 03:27 PM, Baptiste Coudurier wrote: > Hi, > > On 04/19/2011 08:56 AM, sean darcy wrote: >> On 04/16/2011 08:10 PM, Baptiste Coudurier wrote: >>> Hi everyone, >>> >>> Latest git has now added 2 new options for libx264 encoding: >>> -preset >>> -tune >>> >>> These options are directly mapped to internal tune and presets within >>> libx264. The old presets files have been removed, please use the new >>> options. >>> >>> Best regards. >>> >> How do these new presets work with 2 passes? >> >> For instance, I have a script that used: >> >> -pass 1 -vpre slow_firstpass >> >> -pass 2 -vpre slow >> >> I've changed it to: >> >> -pass 1 -preset slow >> >> -pass 2 -preset slow >> >> I used to get 100 fps on the 1st pass, now about 70. Is this to be >> expected? Or do I need to do something else on the 1st pass? >> >> Here's the full script: >> >> ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -preset slow \ >> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 -f mp4 -y >> /dev/null >> ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -preset slow >> -timestamp now \ >> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 $1.m4v > > fastfirstpass should be enabled by default. > Can you please double check that the parameters printed are the same ? > I've changed the script a little: slow -> slower, and added -tune film. But on pass 1 there's no fastfirstpass. BTW, notice "-level 30" on ffmpeg command line becomes Level 3.1 in x264. ffmpeg -i total.play.2005 -an -pass 1 -vcodec libx264 -level 30 -preset slower -tune film -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 -f mp4 -y /dev/null FFmpeg version git-N-29196-ge61b83d, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 19 2011 11:08:14 with gcc 4.6.0 20110413 (Red Hat 4.6.0-4) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --shlibdir=/usr/lib64 --extra-cflags='-O2 -march=native -mtune=native -fopenmp -fomit-frame-pointer -pipe' --enable-static --enable-shared --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-gray --enable-vaapi --enable-hardcoded-tables --enable-frei0r --enable-libdirac --disable-decoder=libdirac --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --disable-encoder=libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-zlib --disable-debug --cpu=amdfam10 --arch=x86_64 --enable-libopencv --enable-pic libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.120. 0 / 52.120. 0 libavformat 52.108. 0 / 52.108. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 79. 1 / 1. 79. 1 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 [dv @ 0x102d870] Estimating duration from bitrate, this may be inaccurate Input #0, dv, from 'total.play.2005': Duration: 00:51:08.43, start: 0.000000, bitrate: 28771 kb/s Stream #0.0: Video: dvvideo, yuv411p, 720x480, 28771 kb/s, PAR 8:9 DAR 4:3, 29.97 tbr, 29.97 tbn, 29.97 tbc Stream #0.1: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s Incompatible pixel format 'yuv411p' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 0x1034a90] w:720 h:480 pixfmt:yuv411p [ffsink @ 0x1034d40] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x1035050] w:720 h:480 fmt:yuv411p -> w:720 h:480 fmt:yuv420p flags:0xa0000004 [libx264 @ 0x10340b0] using SAR=8/9 [libx264 @ 0x10340b0] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 0x10340b0] profile High, level 3.1 [libx264 @ 0x10340b0] 264 - core 115 r1937+22 337ca0a - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=8 deblock=1:-1:-1 analyse=0x3:0x133 me=umh subme=9 psy=1 psy_rd=1.00:0.15 mixed_ref=1 me_range=16 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-3 threads=9 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=abr mbtree=1 bitrate=1200 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=10000 vbv_bufsize=10000 nal_hrd=none ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to '/dev/null': Metadata: encoder : Lavf52.108.0 Stream #0.0: Video: libx264, yuv420p, 720x480 [PAR 8:9 DAR 4:3], q=2-31, pass 1, 1200 kb/s, 30k tbn, 29.97 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame=91961 fps= 37 q=-1.0 Lsize= 0kB time=3068.37 bitrate= 0.0kbits/s /s video:451863kB audio:0kB global headers:0kB muxing overhead -100.000000% frame I:397 Avg QP:21.13 size: 49992 [libx264 @ 0x10340b0] frame P:24406 Avg QP:23.88 size: 12822 [libx264 @ 0x10340b0] frame B:67158 Avg QP:28.25 size: 1935 [libx264 @ 0x10340b0] consecutive B-frames: 0.8% 1.6% 11.3% 86.3% [libx264 @ 0x10340b0] mb I I16..4: 7.7% 76.8% 15.5% [libx264 @ 0x10340b0] mb P I16..4: 0.7% 2.2% 0.5% P16..4: 47.7% 15.8% 8.5% 0.7% 0.7% skip:23.1% [libx264 @ 0x10340b0] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 34.4% 3.1% 0.9% direct: 3.0% skip:58.4% L0:45.6% L1:44.3% BI:10.1% [libx264 @ 0x10340b0] final ratefactor: 22.71 [libx264 @ 0x10340b0] 8x8 transform intra:65.4% inter:53.8% [libx264 @ 0x10340b0] direct mvs spatial:100.0% temporal:0.0% [libx264 @ 0x10340b0] coded y,uvDC,uvAC intra: 73.6% 65.0% 34.7% inter: 14.3% 12.3% 0.6% [libx264 @ 0x10340b0] i16 v,h,dc,p: 37% 37% 13% 13% [libx264 @ 0x10340b0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 17% 14% 14% 6% 9% 10% 9% 9% 12% [libx264 @ 0x10340b0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 7% 46% 4% 5% 7% 7% 8% 6% 10% [libx264 @ 0x10340b0] i8c dc,h,v,p: 39% 29% 20% 12% [libx264 @ 0x10340b0] Weighted P-Frames: Y:8.7% UV:3.2% [libx264 @ 0x10340b0] ref P L0: 37.6% 11.7% 24.4% 5.5% 7.8% 3.8% 6.2% 2.3% 0.6% 0.0% [libx264 @ 0x10340b0] ref B L0: 64.7% 18.1% 7.3% 4.1% 2.6% 2.3% 0.9% [libx264 @ 0x10340b0] ref B L1: 93.3% 6.7% [libx264 @ 0x10340b0] kb/s:1206.37 sean From baptiste.coudurier at gmail.com Wed Apr 20 00:09:29 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Tue, 19 Apr 2011 15:09:29 -0700 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: References: <4DAA2FE3.5040708@gmail.com> <4DADE225.6070907@gmail.com> Message-ID: <4DAE0819.4040904@gmail.com> Hi Sean, On 04/19/2011 03:05 PM, sean darcy wrote: > On 04/19/2011 03:27 PM, Baptiste Coudurier wrote: >> Hi, >> >> On 04/19/2011 08:56 AM, sean darcy wrote: >>> On 04/16/2011 08:10 PM, Baptiste Coudurier wrote: >>>> Hi everyone, >>>> >>>> Latest git has now added 2 new options for libx264 encoding: >>>> -preset >>>> -tune >>>> >>>> These options are directly mapped to internal tune and presets within >>>> libx264. The old presets files have been removed, please use the new >>>> options. >>>> >>>> Best regards. >>>> >>> How do these new presets work with 2 passes? >>> >>> For instance, I have a script that used: >>> >>> -pass 1 -vpre slow_firstpass >>> >>> -pass 2 -vpre slow >>> >>> I've changed it to: >>> >>> -pass 1 -preset slow >>> >>> -pass 2 -preset slow >>> >>> I used to get 100 fps on the 1st pass, now about 70. Is this to be >>> expected? Or do I need to do something else on the 1st pass? >>> >>> Here's the full script: >>> >>> ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -preset >>> slow \ >>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 -f >>> mp4 -y >>> /dev/null >>> ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -preset slow >>> -timestamp now \ >>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 $1.m4v >> >> fastfirstpass should be enabled by default. >> Can you please double check that the parameters printed are the same ? >> > > I've changed the script a little: slow -> slower, and added -tune film. > But on pass 1 there's no fastfirstpass. I don't understand what you mean. fastfirstpass is activated by default like x264, if you want to disable it use: -fastfirstpass 0. > BTW, notice "-level 30" on ffmpeg command line becomes Level 3.1 in x264. -level has not effect when -preset is used. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From baptiste.coudurier at gmail.com Wed Apr 20 00:26:24 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Tue, 19 Apr 2011 15:26:24 -0700 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DAE0819.4040904@gmail.com> References: <4DAA2FE3.5040708@gmail.com> <4DADE225.6070907@gmail.com> <4DAE0819.4040904@gmail.com> Message-ID: <4DAE0C10.3070305@gmail.com> On 04/19/2011 03:09 PM, Baptiste Coudurier wrote: > Hi Sean, > > On 04/19/2011 03:05 PM, sean darcy wrote: >> On 04/19/2011 03:27 PM, Baptiste Coudurier wrote: >>> Hi, >>> >>> On 04/19/2011 08:56 AM, sean darcy wrote: >>>> On 04/16/2011 08:10 PM, Baptiste Coudurier wrote: >>>>> Hi everyone, >>>>> >>>>> Latest git has now added 2 new options for libx264 encoding: >>>>> -preset >>>>> -tune >>>>> >>>>> These options are directly mapped to internal tune and presets within >>>>> libx264. The old presets files have been removed, please use the new >>>>> options. >>>>> >>>>> Best regards. >>>>> >>>> How do these new presets work with 2 passes? >>>> >>>> For instance, I have a script that used: >>>> >>>> -pass 1 -vpre slow_firstpass >>>> >>>> -pass 2 -vpre slow >>>> >>>> I've changed it to: >>>> >>>> -pass 1 -preset slow >>>> >>>> -pass 2 -preset slow >>>> >>>> I used to get 100 fps on the 1st pass, now about 70. Is this to be >>>> expected? Or do I need to do something else on the 1st pass? >>>> >>>> Here's the full script: >>>> >>>> ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -preset >>>> slow \ >>>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 -f >>>> mp4 -y >>>> /dev/null >>>> ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -preset slow >>>> -timestamp now \ >>>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 $1.m4v >>> >>> fastfirstpass should be enabled by default. >>> Can you please double check that the parameters printed are the same ? >>> Fixed fastfirstpass in git, please update. It seems testing was bumpy here .... sorry about that. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From seandarcy2 at gmail.com Wed Apr 20 00:51:26 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Tue, 19 Apr 2011 18:51:26 -0400 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DAE0819.4040904@gmail.com> References: <4DAA2FE3.5040708@gmail.com> <4DADE225.6070907@gmail.com> <4DAE0819.4040904@gmail.com> Message-ID: Hi, On 04/19/2011 06:09 PM, Baptiste Coudurier wrote: > Hi Sean, > > On 04/19/2011 03:05 PM, sean darcy wrote: >> On 04/19/2011 03:27 PM, Baptiste Coudurier wrote: >>> Hi, >>> >>> On 04/19/2011 08:56 AM, sean darcy wrote: >>>> On 04/16/2011 08:10 PM, Baptiste Coudurier wrote: >>>>> Hi everyone, >>>>> >>>>> Latest git has now added 2 new options for libx264 encoding: >>>>> -preset >>>>> -tune >>>>> >>>>> These options are directly mapped to internal tune and presets within >>>>> libx264. The old presets files have been removed, please use the new >>>>> options. >>>>> >>>>> Best regards. >>>>> >>>> How do these new presets work with 2 passes? >>>> >>>> For instance, I have a script that used: >>>> >>>> -pass 1 -vpre slow_firstpass >>>> >>>> -pass 2 -vpre slow >>>> >>>> I've changed it to: >>>> >>>> -pass 1 -preset slow >>>> >>>> -pass 2 -preset slow >>>> >>>> I used to get 100 fps on the 1st pass, now about 70. Is this to be >>>> expected? Or do I need to do something else on the 1st pass? >>>> >>>> Here's the full script: >>>> >>>> ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -preset >>>> slow \ >>>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 -f >>>> mp4 -y >>>> /dev/null >>>> ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -preset slow >>>> -timestamp now \ >>>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 $1.m4v >>> >>> fastfirstpass should be enabled by default. >>> Can you please double check that the parameters printed are the same ? >>> >> >> I've changed the script a little: slow -> slower, and added -tune film. >> But on pass 1 there's no fastfirstpass. > > I don't understand what you mean. fastfirstpass is activated by default > like x264, if you want to disable it use: -fastfirstpass 0. > I misunderstood you. I thought you meant check ffmpeg parameters v. x264 parameters. I now understand you meant to check if the parameters for ffmpeg were the same with -vpre and -preset. And the answer is yes - they were the same: I was using the identical script. I'm still puzzled by the new behavior. I'm not sure fastfirstpass is enabled, or maybe something else is slowing the first pass. With ffmpeg and the old presets, as I remember it, pass 1 was 80-100 fps. pass 2 was 30-40 fps. So pass 1 was about twice as fast as pass 2. With the new settings (bearing in mind I've gone from slow -> slower) pass 1 is 2 fps _slower_ than pass 2! Pass 1: frame=91961 fps= 37 q=-1.0 Lsize= 0kB time=3068.37 bitrate= 0.0kbits/s /s Pass 2: frame=91961 fps= 39 q=-1.0 Lsize= 450992kB time=3068.37 bitrate=1204.1kbits/s >> BTW, notice "-level 30" on ffmpeg command line becomes Level 3.1 in x264. > > -level has not effect when -preset is used. > I'm trying to encode so that it can be played on android/iphone. Some models require 3.0 (others are ok with 3.1). I assume some of this will be fixed with -profile, but even within a profile, the levels can be important. But I guess this is an x264 issue, not ffmpeg. sean From jgarcia at vicomtech.org Tue Apr 19 17:51:14 2011 From: jgarcia at vicomtech.org (Julen Garcia) Date: Tue, 19 Apr 2011 17:51:14 +0200 Subject: [FFmpeg-user] sws_scale problem from BGRA to YUV420P Message-ID: I get bad results transcoding from a 32bit BGRA image to libx264's YUV420P format using this command: ffmpeg -vstats -benchmark -f image2 -i "input0.png" -vf "format=bgra" -vcodec libx264 -vpre lossless_ultrafast -crf 22 -b 10000k -threads 0 OUTPUT.mp4 However if I use a 24bit BGR image the result is ok ffmpeg -vstats -benchmark -f image2 -i "input1.png" -vf "format=bgr24" -vcodec libx264 -vpre lossless_ultrafast -crf 22 -b 10000k -threads 0 OUTPUT.mp4 I'm using one of the latest git version, exactly "ffmpeg-git-a304071-win64-shared" from here: http://hawkeye.arrozcru.org/ In an older version, "r22941" both commands uset to work fine. Thank you! From baptiste.coudurier at gmail.com Wed Apr 20 03:58:15 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Tue, 19 Apr 2011 18:58:15 -0700 Subject: [FFmpeg-user] sws_scale problem from BGRA to YUV420P In-Reply-To: References: Message-ID: <4DAE3DB7.1000509@gmail.com> Hi, On 04/19/2011 08:51 AM, Julen Garcia wrote: > I get bad results transcoding from a 32bit BGRA image to libx264's > YUV420P format using this command: > > ffmpeg -vstats -benchmark -f image2 -i "input0.png" -vf "format=bgra" > -vcodec libx264 -vpre lossless_ultrafast -crf 22 -b 10000k -threads 0 > OUTPUT.mp4 > > However if I use a 24bit BGR image the result is ok > > ffmpeg -vstats -benchmark -f image2 -i "input1.png" -vf "format=bgr24" > -vcodec libx264 -vpre lossless_ultrafast -crf 22 -b 10000k -threads 0 > OUTPUT.mp4 > > I'm using one of the latest git version, exactly > "ffmpeg-git-a304071-win64-shared" from here: > http://hawkeye.arrozcru.org/ > > In an older version, "r22941" both commands uset to work fine. Can you please upload your png file somewhere ? -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From baptiste.coudurier at gmail.com Wed Apr 20 04:05:51 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Tue, 19 Apr 2011 19:05:51 -0700 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: References: <4DAA2FE3.5040708@gmail.com> <4DADE225.6070907@gmail.com> <4DAE0819.4040904@gmail.com> Message-ID: <4DAE3F7F.10304@gmail.com> On 04/19/2011 03:51 PM, sean darcy wrote: > Hi, > > On 04/19/2011 06:09 PM, Baptiste Coudurier wrote: >> Hi Sean, >> >> On 04/19/2011 03:05 PM, sean darcy wrote: >>> On 04/19/2011 03:27 PM, Baptiste Coudurier wrote: >>>> Hi, >>>> >>>> On 04/19/2011 08:56 AM, sean darcy wrote: >>>>> On 04/16/2011 08:10 PM, Baptiste Coudurier wrote: >>>>>> Hi everyone, >>>>>> >>>>>> Latest git has now added 2 new options for libx264 encoding: >>>>>> -preset >>>>>> -tune >>>>>> >>>>>> These options are directly mapped to internal tune and presets within >>>>>> libx264. The old presets files have been removed, please use the new >>>>>> options. >>>>>> >>>>>> Best regards. >>>>>> >>>>> How do these new presets work with 2 passes? >>>>> >>>>> For instance, I have a script that used: >>>>> >>>>> -pass 1 -vpre slow_firstpass >>>>> >>>>> -pass 2 -vpre slow >>>>> >>>>> I've changed it to: >>>>> >>>>> -pass 1 -preset slow >>>>> >>>>> -pass 2 -preset slow >>>>> >>>>> I used to get 100 fps on the 1st pass, now about 70. Is this to be >>>>> expected? Or do I need to do something else on the 1st pass? >>>>> >>>>> Here's the full script: >>>>> >>>>> ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -preset >>>>> slow \ >>>>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 -f >>>>> mp4 -y >>>>> /dev/null >>>>> ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -preset slow >>>>> -timestamp now \ >>>>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 >>>>> $1.m4v >>>> >>>> fastfirstpass should be enabled by default. >>>> Can you please double check that the parameters printed are the same ? >>>> >>> >>> I've changed the script a little: slow -> slower, and added -tune film. >>> But on pass 1 there's no fastfirstpass. >> >> I don't understand what you mean. fastfirstpass is activated by default >> like x264, if you want to disable it use: -fastfirstpass 0. >> > > > I misunderstood you. I thought you meant check ffmpeg parameters v. x264 > parameters. I now understand you meant to check if the parameters for > ffmpeg were the same with -vpre and -preset. And the answer is yes - > they were the same: I was using the identical script. You have to compare this line: [libx264 @ 0x10340b0] 264 - core 115 r1937+22 337ca0a - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=8 deblock=1:-1:-1 analyse=0x3:0x133 me=umh subme=9 psy=1 psy_rd=1.00:0.15 mixed_ref=1 me_range=16 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-3 threads=9 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=abr mbtree=1 bitrate=1200 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=10000 vbv_bufsize=10000 nal_hrd=none ip_ratio=1.40 aq=1:1.00 between the 2 to make sure they match. > I'm still puzzled by the new behavior. I'm not sure fastfirstpass is > enabled, or maybe something else is slowing the first pass. With ffmpeg > and the old presets, as I remember it, pass 1 was 80-100 fps. pass 2 was > 30-40 fps. So pass 1 was about twice as fast as pass 2. > > With the new settings (bearing in mind I've gone from slow -> slower) > pass 1 is 2 fps _slower_ than pass 2! Can you please recheck with latest git ? > frame=91961 fps= 37 q=-1.0 Lsize= 0kB time=3068.37 bitrate= > 0.0kbits/s /s > > Pass 2: > > frame=91961 fps= 39 q=-1.0 Lsize= 450992kB time=3068.37 > bitrate=1204.1kbits/s > >>> BTW, notice "-level 30" on ffmpeg command line becomes Level 3.1 in > x264. >> >> -level has not effect when -preset is used. >> > > I'm trying to encode so that it can be played on android/iphone. Some > models require 3.0 (others are ok with 3.1). I assume some of this will > be fixed with -profile, but even within a profile, the levels can be > important. But I guess this is an x264 issue, not ffmpeg. Of course it is important. -level should now work, enjoy. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From tim.nicholson at bbc.co.uk Wed Apr 20 09:14:29 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Wed, 20 Apr 2011 08:14:29 +0100 Subject: [FFmpeg-user] Issue with watermarking a video -> result is garbled (pink and green lines) In-Reply-To: References: <4DADB954.7020108@bbc.co.uk> <4DADB9F2.9020007@bbc.co.uk> Message-ID: <4DAE87D5.6070600@bbc.co.uk> On 19/04/11 18:24, David wrote: > Tim Nicholson bbc.co.uk> writes: >> And now with your file:- >> >> ffmpeg -i paul.mov -threads 0 \ >> -vf "movie=coachteamlogo.png [logo]; [in][logo] >> overlay=10:main_h-overlay_h-10 [out]" \ >> -vcodec dvvideo -s 720x576 -an -y out-paul.mov >> >> and it still looks good..... >> > > > Tim, > > When running your code I still got the same crappy output. Which build are you > running? I will try testing with another build. > FFmpeg version git-N-29169-g7e2eb4b, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 18 2011 09:59:45 with gcc 4.5.0 20100604 [gcc-4_5-branch revision 160292] -- Tim http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. From jgarcia at vicomtech.org Wed Apr 20 10:30:02 2011 From: jgarcia at vicomtech.org (Julen Garcia) Date: Wed, 20 Apr 2011 10:30:02 +0200 Subject: [FFmpeg-user] sws_scale problem from BGRA to YUV420P Message-ID: Hi, Here are the results I get after different test sets. if you need more info or logs, just ask ;) Thank you in advance. ___________ ffmpeg -vstats -benchmark -f image2 -i "input24.png" -vf "format=bgr24" -vcodec libx264 -vpre lossless_ultrafast -crf 22 -b 10000k -threads 0 OUTPUT.mp4 [scale @ 0000000001918BF0] w:1280 h:720 fmt:rgb24 -> w:1280 h:720 fmt:bgr24 flags:0x4 [scale @ 0000000001918D30] w:1280 h:720 fmt:bgr24 -> w:1280 h:720 fmt:yuv420p flags:0x4 result: Ok http://dl.dropbox.com/u/17827310/ffmpeg/input24.png Image internal format = rgb24, forced to bgr24 http://dl.dropbox.com/u/17827310/ffmpeg/output24_bgr24.png ___________ ffmpeg -vstats -benchmark -f image2 -i "input24.png" -vcodec libx264 -vpre lossless_ultrafast -crf 22 -b 10000k -threads 0 OUTPUT.mp4 [scale @ 0000000000342090] w:1280 h:720 fmt:rgb24 -> w:1280 h:720 fmt:yuv420p flags:0x4 result: Bad http://dl.dropbox.com/u/17827310/ffmpeg/input24.png Image internal format = rgb24 http://dl.dropbox.com/u/17827310/ffmpeg/output24_rgb24.png ___________ ffmpeg -vstats -benchmark -f image2 -i "input32.png" -vf "format=bgra" -vcodec libx264 -vpre lossless_ultrafast -crf 22 -b 10000k -threads 0 OUTPUT.mp4 or ffmpeg -vstats -benchmark -f image2 -i "input32.png" -vcodec libx264 -vpre lossless_ultrafast -crf 22 -b 10000k -threads 0 OUTPUT.mp4 [scale @ 0000000001952090] w:1280 h:720 fmt:bgra -> w:1280 h:720 fmt:yuv420p flags:0x4 result: Bad http://dl.dropbox.com/u/17827310/ffmpeg/input32.png Image internal format = bgra http://dl.dropbox.com/u/17827310/ffmpeg/output32.png From jieyunfu at gmail.com Thu Apr 21 00:35:28 2011 From: jieyunfu at gmail.com (Jieyun Fu) Date: Wed, 20 Apr 2011 18:35:28 -0400 Subject: [FFmpeg-user] using libav to resample, alias effects Message-ID: Hi all, I realized that when I am writing my own code to upsample a piece of audio from 8000Hz to 22050Hz, I always get some alias effects: the sound got buzzy (with some zee, zee in the background) and looking at the spectrogram we can see the frequency mirrors around 8K Hz, which is a classical alias effects. If I use ffmpeg binaries, I don't have this problem. But I do need to use Libav to achieve that. You guys can download the original raw waveform and the converted raw waveform from ( http://dl.dropbox.com/u/4363247/ffmpeg/before.raw and http://dl.dropbox.com/u/4363247/ffmpeg/after.raw) There are two APIs to resample the audio but both produce these alias effects. Please let me know if I am missing anything. newsize = (long) size * 22050 / 8000.0; ReSampleContext* context = av_audio_resample_init(1, 1, // out channels, in channels 22050, 8000, SAMPLE_FMT_S16, SAMPLE_FMT_S16, 16, 0, 0, 0.8); int audio_resample_rc = audio_resample(context, (short*)converted_samples, (short*)samples, size / 2); if (context) audio_resample_close(context); and int out_rate = 22050; audio_cntx = av_resample_init( out_rate, // out rate RATE, // in rate 16, // filter length 0, // phase count 0, // linear FIR filter 0.8 ); // cutoff frequency assert( audio_cntx && "Failed to create resampling context!" ); int samples_consumed = 0; int samples_output = av_resample( audio_cntx, (short*)out_buffer, (short*)in_buffer, &samples_consumed, bytes_read / 2, 40000 * 4 / 2, 0 ); av_resample_close( audio_cntx ); Jieyun From seandarcy2 at gmail.com Thu Apr 21 05:31:23 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Wed, 20 Apr 2011 23:31:23 -0400 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DAE3F7F.10304@gmail.com> References: <4DAA2FE3.5040708@gmail.com> <4DADE225.6070907@gmail.com> <4DAE0819.4040904@gmail.com> <4DAE3F7F.10304@gmail.com> Message-ID: On 04/19/2011 10:05 PM, Baptiste Coudurier wrote: > On 04/19/2011 03:51 PM, sean darcy wrote: >> Hi, >> >> On 04/19/2011 06:09 PM, Baptiste Coudurier wrote: >>> Hi Sean, >>> >>> On 04/19/2011 03:05 PM, sean darcy wrote: >>>> On 04/19/2011 03:27 PM, Baptiste Coudurier wrote: >>>>> Hi, >>>>> >>>>> On 04/19/2011 08:56 AM, sean darcy wrote: >>>>>> On 04/16/2011 08:10 PM, Baptiste Coudurier wrote: >>>>>>> Hi everyone, >>>>>>> >>>>>>> Latest git has now added 2 new options for libx264 encoding: >>>>>>> -preset >>>>>>> -tune >>>>>>> >>>>>>> These options are directly mapped to internal tune and presets within >>>>>>> libx264. The old presets files have been removed, please use the new >>>>>>> options. >>>>>>> >>>>>>> Best regards. >>>>>>> >>>>>> How do these new presets work with 2 passes? >>>>>> >>>>>> For instance, I have a script that used: >>>>>> >>>>>> -pass 1 -vpre slow_firstpass >>>>>> >>>>>> -pass 2 -vpre slow >>>>>> >>>>>> I've changed it to: >>>>>> >>>>>> -pass 1 -preset slow >>>>>> >>>>>> -pass 2 -preset slow >>>>>> >>>>>> I used to get 100 fps on the 1st pass, now about 70. Is this to be >>>>>> expected? Or do I need to do something else on the 1st pass? >>>>>> >>>>>> Here's the full script: >>>>>> >>>>>> ffmpeg -i $INPUT -an -pass 1 -vcodec libx264 -level 30 -preset >>>>>> slow \ >>>>>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 -f >>>>>> mp4 -y >>>>>> /dev/null >>>>>> ffmpeg -i $INPUT -an -pass 2 -vcodec libx264 -level 30 -preset slow >>>>>> -timestamp now \ >>>>>> -bufsize 10000000 -maxrate 10000000 -b 1200k -threads 0 >>>>>> $1.m4v >>>>> >>>>> fastfirstpass should be enabled by default. >>>>> Can you please double check that the parameters printed are the same ? >>>>> >>>> >>>> I've changed the script a little: slow -> slower, and added -tune film. >>>> But on pass 1 there's no fastfirstpass. >>> >>> I don't understand what you mean. fastfirstpass is activated by default >>> like x264, if you want to disable it use: -fastfirstpass 0. >>> >> >> >> I misunderstood you. I thought you meant check ffmpeg parameters v. x264 >> parameters. I now understand you meant to check if the parameters for >> ffmpeg were the same with -vpre and -preset. And the answer is yes - >> they were the same: I was using the identical script. > > You have to compare this line: > [libx264 @ 0x10340b0] 264 - core 115 r1937+22 337ca0a - H.264/MPEG-4 AVC > codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - > options: cabac=1 ref=8 deblock=1:-1:-1 analyse=0x3:0x133 me=umh subme=9 > psy=1 psy_rd=1.00:0.15 mixed_ref=1 me_range=16 chroma_me=1 trellis=2 > 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-3 threads=9 > sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 > constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 > weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 > intra_refresh=0 rc_lookahead=60 rc=abr mbtree=1 bitrate=1200 ratetol=1.0 > qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=10000 vbv_bufsize=10000 > nal_hrd=none ip_ratio=1.40 aq=1:1.00 > > between the 2 to make sure they match. > >> I'm still puzzled by the new behavior. I'm not sure fastfirstpass is >> enabled, or maybe something else is slowing the first pass. With ffmpeg >> and the old presets, as I remember it, pass 1 was 80-100 fps. pass 2 was >> 30-40 fps. So pass 1 was about twice as fast as pass 2. >> >> With the new settings (bearing in mind I've gone from slow -> slower) >> pass 1 is 2 fps _slower_ than pass 2! > > Can you please recheck with latest git ? > >> frame=91961 fps= 37 q=-1.0 Lsize= 0kB time=3068.37 bitrate= >> 0.0kbits/s /s >> >> Pass 2: >> >> frame=91961 fps= 39 q=-1.0 Lsize= 450992kB time=3068.37 >> bitrate=1204.1kbits/s >> >>>> BTW, notice "-level 30" on ffmpeg command line becomes Level 3.1 in >> x264. >>> >>> -level has not effect when -preset is used. >>> >> >> I'm trying to encode so that it can be played on android/iphone. Some >> models require 3.0 (others are ok with 3.1). I assume some of this will >> be fixed with -profile, but even within a profile, the levels can be >> important. But I guess this is an x264 issue, not ffmpeg. > > Of course it is important. > -level should now work, enjoy. > Most excellent! pass 1: frame=91961 fps=105 q=-1.0 Lsize= 0kB time=3068.37 bitrate= 0.0kbits/s s and it is level 3.0 Thanks enormously for being so responsive. sean From cholet at logilune.com Thu Apr 21 08:12:03 2011 From: cholet at logilune.com (=?iso-8859-1?Q?=C9ric_Cholet?=) Date: Thu, 21 Apr 2011 08:12:03 +0200 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DADF16D.4050003@gmail.com> References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> <20110418125128.53550483@lrcd.com> <4DACA5A8.8060106@gmail.com> <4DADF16D.4050003@gmail.com> Message-ID: <50C150F4-5EAD-4169-86DD-573B562EA0CE@logilune.com> Le 19 avr. 2011 ? 22:32, Baptiste Coudurier a ?crit : > On 04/18/2011 01:57 PM, Baptiste Coudurier wrote: >> On 04/18/2011 01:51 PM, Lou wrote: >>> On Mon, 18 Apr 2011 17:03:49 +0200 >>> ?ric Cholet wrote: >>> >>>> Le 18 avr. 2011 ? 16:49, Eric Cholet a ?crit : >>>>> With freshly downloaded ffmpeg from git, I'm getting an error when >>>>> trying to use the -profile option: >>>>> >>>>> >>>>> % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b 2500k >>>>> -r 25 -preset medium -profile main foo.mp4 >>>> >>>> Oops forgot to post the error message: >>>> >>>> [NULL @ 0x80240b200] [Eval @ 0x7fffffffd110] Undefined constant or >>>> missing '(' in 'main' [NULL @ 0x80240b200] Unable to parse option >>>> value "main" Invalid value 'main' for option 'profile' >>>> >>>> >>>>> >>>>> What am I doing wrong? >>>> -- >>>> ?ric Cholet >>> >>> You are doing nothing wrong. It was my mistake to mention -profile. It >>> is listed in 'ffmpeg -h', but is not available yet. >>> >>> I like to verify my suggestions, but this time I didn't. This is what >>> happens when I rely on memory as I previously tested the new options a >>> month ago but apparently forgot that -profile wasn't usable yet. >> >> I'm on it, don't worry :) >> > > All right, -profile should work now. > Please report any problem. Indeed works like a charm, thanks a lot. -- ?ric Cholet From nitin.bodke at gmail.com Thu Apr 21 16:48:01 2011 From: nitin.bodke at gmail.com (Nitin Bodke) Date: Thu, 21 Apr 2011 20:18:01 +0530 Subject: [FFmpeg-user] Is there any way to hide the configure command options? Message-ID: Hi All, When we execute ffmpeg, ffmpeg displays configuration options. e.g configuration: --prefix=/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/ffmpeg-0.5.1/release --extra-cflags=-I/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/faac-1.28/release/include --extra-cflags=-I/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/faad2-2.7/release/include --extra-cflags=-I/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/x264-20100425-2245/release/include --extra-cflags=-I/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/opencore-amr-0.1.2/release/include --extra-cflags=-I/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/libgsm-1.0.13/inc/ --extra-ldflags=-L/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/faac-1.28/release/lib --extra-ldflags=-L/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/faad2-2.7/release/lib --extra-ldflags=-L/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/x264-20100425-2245/release/lib --extra-ldflags=-L/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/opencore-amr-0.1.2/release/lib --extra-ldflags=-L/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/libgsm-1.0.13/lib --enable-gpl --enable-pthreads --enable-version3 --enable-nonfree --enable-static --enable-shared --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libfaad --enable-libx264 --enable-libgsm --enable-libmp3lame --extra-cflags=-I/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/lame.3.98.4/release/include --extra-ldflags=-L/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/lame.3.98.4/release/lib Is there any way to hide this? Thanks, Nitin. From stefano.sabatini-lala at poste.it Thu Apr 21 17:12:22 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Thu, 21 Apr 2011 17:12:22 +0200 Subject: [FFmpeg-user] Is there any way to hide the configure command options? In-Reply-To: References: Message-ID: <20110421151222.GB31144@geppetto> On date Thursday 2011-04-21 20:18:01 +0530, Nitin Bodke encoded: > Hi All, > > When we execute ffmpeg, ffmpeg displays configuration options. > e.g > > configuration: [...] > Is there any way to hide this? No, please add a feature request on the issue-tracker: https://ffmpeg.org/trac/ffmpeg/report since this is becoming a common complain. -- ffmpeg-user random tip #11 One minute of video silence with ffmpeg: ffmpeg -t 60 -s qcif -f rawvideo -pix_fmt rgb24 -r 25 -i /dev/zero \ -y silence.mpeg From h.reindl at thelounge.net Thu Apr 21 17:12:53 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 21 Apr 2011 17:12:53 +0200 Subject: [FFmpeg-user] Is there any way to hide the configure command options? In-Reply-To: References: Message-ID: <4DB04975.9060501@thelounge.net> Am 21.04.2011 16:48, schrieb Nitin Bodke: > Hi All, > > When we execute ffmpeg, ffmpeg displays configuration options. > e.g > > Is there any way to hide this? AFAIK no, but i would like this too if there would be a otpion like "ffmpeg -configure" to not get lost this information if i have to work on a foreign server and nobody knows how ffmpeg was built -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From developer at noknok.net Thu Apr 21 18:17:25 2011 From: developer at noknok.net (NokNok Developer) Date: Thu, 21 Apr 2011 12:17:25 -0400 Subject: [FFmpeg-user] BOUNTY OFFERED: AAC 5.1 -> AAC Stereo Message-ID: <4DB05895.6010600@noknok.net> Over the years we have seen various patches by folks to enable various forms of 6 Channel to Stereo downsampling, non of which ever were implemented in the core code base for AAC (internal) processing. While the solution to this problem can range from a simple mixing of channels, adjusting volumes, etc to more advanced, our goal is to find a solution acceptable to the ffmpeg team in any form at this point. Many devices support AAC but in a stereo mode only vs 6 channel, etc. We would like to offer such a bounty to get such a solution in the code base which will enable downsampling of AAC 5.1 to AAC stereo when encoding. An example of two previous variations: http://muzso.hu/2009/02/25/downsampling-multichannel-audio-5.1-into-stereo-2-channels-with-ffmpeg Its important that the solution be acceptable in a form which it can be included in the standard code base for ffmpeg so we are not having to patch, etc. Using the internal AAC encoder (-strict experimental) Depending on the solution we would place the bounty anywhere from $50-$250. If you are interested in doing this, or do this and looking for payment, please email ffmpegbounty [at] bitsling.com Not sure where best in placing this bounty, so if not on the proper list, please let us know as well. Thank You From blacktrash at gmx.net Thu Apr 21 19:02:33 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Thu, 21 Apr 2011 18:02:33 +0100 Subject: [FFmpeg-user] Is there any way to hide the configure command options? In-Reply-To: References: Message-ID: <20110421170233.GV777@krille.blacktrash.org> * Nitin Bodke on Thursday, April 21, 2011 at 20:18:01 +0530 > When we execute ffmpeg, ffmpeg displays configuration options. > e.g > > configuration: > --prefix=/sandbox/mmsc/schundi/workspace/CA/msg-platform/transcoder/alembik/codecs/ffmpeg-0.5.1/release > Is there any way to hide this? Redirecting stderr to /dev/null seems to do what you want. ffmpeg [... whatever ...] 2>/dev/null Not entirely certain whether this could have any adverse side-effects. c -- \black\trash movie _SAME TIME SAME PLACE_ New York, in the summer of 2001 --->> http://www.blacktrash.org/underdogma/stsp.php From nicolas.george at normalesup.org Thu Apr 21 19:20:42 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Thu, 21 Apr 2011 19:20:42 +0200 Subject: [FFmpeg-user] Is there any way to hide the configure command options? In-Reply-To: <20110421170233.GV777@krille.blacktrash.org> References: <20110421170233.GV777@krille.blacktrash.org> Message-ID: <20110421172042.GA7060@phare.normalesup.org> Le duodi 2 flor?al, an CCXIX, Christian Ebert a ?crit?: > Redirecting stderr to /dev/null seems to do what you want. > > ffmpeg [... whatever ...] 2>/dev/null > > Not entirely certain whether this could have any adverse > side-effects. Like... discarding error messages and other potentially helpful diagnostics, for example? Regards, -- Nicolas George -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: Digital signature URL: From benoit at friry.net Thu Apr 21 19:57:56 2011 From: benoit at friry.net (Benoit) Date: Thu, 21 Apr 2011 19:57:56 +0200 Subject: [FFmpeg-user] AVI size limit In-Reply-To: <20110414201622.GF25387@friry.net> References: <20110414201622.GF25387@friry.net> Message-ID: <20110421175756.GA19281@friry.net> Hi all, > In order to rip VHS on my rather old box, I purchased a Dazzle > USB device. > My command is > ffmpeg \ > -f alsa \ > -i plughw:0,0 \ > -f video4linux2 -tvstd SECAM \ > -i /dev/video0 \ > -acodec pcm_s16le \ > -ac 2 -ar 44100 \ > -vcodec mjpeg \ > -s 720x576 -r 25 -sameq \ > -f avi \ > output.avi > I cannot compress in real-time (CPU too weak). So I grab in mjpeg, > and will reencode later. > It works as expected for short videos. But with a longer video, > ffmpeg is killed after 1h35'50". The output is 13836 MB. > I don't understand why. File too big? > The output file is readable by vlc ou avidemux, but ffplay is > wrong about it: > Input #0, avi, from 'brut.avi': > Metadata: > ISFT : Lavf52.64.2 > Duration: 00:07:35.12, start: 0.000000, bitrate: 255030 kb/s > Stream #0.0: Video: mjpeg, yuvj420p, 720x576, 25 tbr, 25 tbn, 25 tbc > Stream #0.1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s > Don't understant why it only "see" 7'35". Nobody? Any hint? Anything to look for? How to investigate? I have no idea... :( - Filesystem is ext3 with 4k block size (max file size is 2TB !) - I did not find default value for "-fs". Thanks, Benoit From blacktrash at gmx.net Thu Apr 21 20:11:52 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Thu, 21 Apr 2011 19:11:52 +0100 Subject: [FFmpeg-user] Is there any way to hide the configure command options? In-Reply-To: <20110421172042.GA7060@phare.normalesup.org> References: <20110421170233.GV777@krille.blacktrash.org> <20110421172042.GA7060@phare.normalesup.org> Message-ID: <20110421181152.GW777@krille.blacktrash.org> * Nicolas George on Thursday, April 21, 2011 at 19:20:42 +0200 > Le duodi 2 flor?al, an CCXIX, Christian Ebert a ?crit?: >> Redirecting stderr to /dev/null seems to do what you want. >> >> ffmpeg [... whatever ...] 2>/dev/null >> >> Not entirely certain whether this could have any adverse >> side-effects. > > Like... discarding error messages and other potentially helpful diagnostics, > for example? heh one could not without reason classify compile time options under "potentially helpful diagnostics" too ;-) c -- \black\trash movie _COWBOY CANOE COMA_ Ein deutscher Western/A German Western --->> http://www.blacktrash.org/underdogma/ccc.php From allenskd at ymail.com Thu Apr 21 20:40:28 2011 From: allenskd at ymail.com (David Gonzalez) Date: Thu, 21 Apr 2011 11:40:28 -0700 (PDT) Subject: [FFmpeg-user] Is it possible to do this with avfilter? Message-ID: <268173.7861.qm@web120518.mail.ne1.yahoo.com> I'm trying to do this with ffmpeg where I put a banner on top of the video http://img269.imageshack.us/i/exampleyl.jpg/ like in this example problem is that I tried to do some chaining (scale then overlay) but it didn't go through, if i do it with just "overlay" the image gets in the way so I'm not sure if I'm getting the the chain right or not. Thanks, From stefano.sabatini-lala at poste.it Fri Apr 22 01:03:00 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Fri, 22 Apr 2011 01:03:00 +0200 Subject: [FFmpeg-user] Is it possible to do this with avfilter? In-Reply-To: <268173.7861.qm@web120518.mail.ne1.yahoo.com> References: <268173.7861.qm@web120518.mail.ne1.yahoo.com> Message-ID: <20110421230300.GD1368@geppetto> On date Thursday 2011-04-21 11:40:28 -0700, David Gonzalez encoded: > I'm trying to do this with ffmpeg where I put a banner on top of the video > > http://img269.imageshack.us/i/exampleyl.jpg/ like in this example Why not? > problem is that I tried to do some chaining (scale then overlay) but it didn't > go through, if i do it with just "overlay" the image gets in the way so I'm not > sure if I'm getting the the chain right or not. I don't know how to interpret "didn't go through" and "gets in the way", show us the command you're using. -- ffmpeg-user random tip #7 If you're asking about problems with an ffmpeg tool, *always* post the command you're using and the complete output. From rcunning at gmail.com Tue Apr 19 20:56:14 2011 From: rcunning at gmail.com (rcunning) Date: Tue, 19 Apr 2011 11:56:14 -0700 (PDT) Subject: [FFmpeg-user] Compatibility with ffmpeg RTMP? In-Reply-To: References: Message-ID: <1303239374352-3461243.post@n4.nabble.com> You probably need to put some valid timestamp info into the stream. Try: ./ffmpeg -loglevel debug -timestamp now -i ~/Downloads/20051210-w50s.flv -re -ac 1 -acodec copy -vcodec flv -f copy rtmp://localhost:1936/ You would also need to add timestamp even if you push through ffserver. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Compatibility-with-ffmpeg-RTMP-tp3265373p3461243.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From bhavesh.dhameliya at matrixcomsec.com Thu Apr 21 12:25:22 2011 From: bhavesh.dhameliya at matrixcomsec.com (bhavesh) Date: Thu, 21 Apr 2011 03:25:22 -0700 (PDT) Subject: [FFmpeg-user] Regarding AVI file with variable frame rate Message-ID: <1303381522687-3465567.post@n4.nabble.com> Dear ffmpeg-user, I am new with ffmpeg and i want to convert my h.264 file which has varaible frame rate into avi file. The file converted by ffmpeg was Ok but when i play this file it looks like fast forward view. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Regarding-AVI-file-with-variable-frame-rate-tp3465567p3465567.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From singh.gurpreet452 at gmail.com Wed Apr 20 08:58:09 2011 From: singh.gurpreet452 at gmail.com (singh.gurpreet452 at gmail.com) Date: Tue, 19 Apr 2011 23:58:09 -0700 Subject: [FFmpeg-user] MPEG frame sizes Message-ID: <28458190.1281.1303282689033.JavaMail.nabble@joe.nabble.com> Hello Please tell me how to find I,P,B frame size of mpeg video file . and also tell me how to find size of slice of frames. Number of gop in one mpeg video file number of pack header in one mpeg video file From bhavesh.dhameliya at matrixcomsec.com Thu Apr 21 12:25:16 2011 From: bhavesh.dhameliya at matrixcomsec.com (bhavesh) Date: Thu, 21 Apr 2011 03:25:16 -0700 (PDT) Subject: [FFmpeg-user] Regarding AVI file with variable frame rate Message-ID: <1303381515962-3465566.post@n4.nabble.com> Dear ffmpeg-user, I am new with ffmpeg and i want to convert my h.264 file which has varaible frame rate into avi file. The file converted by ffmpeg was Ok but when i play this file it looks like fast forward view. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Regarding-AVI-file-with-variable-frame-rate-tp3465566p3465566.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From i-love-spam at yandex.ru Fri Apr 22 04:12:42 2011 From: i-love-spam at yandex.ru (i-love-spam) Date: Fri, 22 Apr 2011 06:12:42 +0400 Subject: [FFmpeg-user] Apple live HTTP streaming: video slows down if processed with ffmpeg Message-ID: <59371303438362@web81.yandex.ru> I'm trying to make live http streaming working with and iPhone 4. For live http streaming I have to use mpegts segments that contain h264 and mp3/aac for audio. I use 10 sec segments and I use opensource segmenter.c that uses libavformat/avcodec to segment videos into mpegts chunks. What I noticed, is that many of my segmented videos didn't play properly: after playing for 30 seconds or so video starts to play twice slower while sound plays at normal speed. So I tried to investigate what's wrong and I came to conclusion that chunks generated from flv/mp4 etc always have this problem and segments generated from other ts files do not have the problem. I tried to different tests, and it looks that ffmpeg somehow modifies bitstream so that live http streaming does not work on iPhone. I have an original family guy mpegts video clip. If I use this clip for segmenting then live http streaming works perfectly. If I convert this file with ffmpeg -i famili_guy.ts -acodec copy -vcodec copy family_guy.flv (or mp4) then if I try to segment that resulting file then iPhone will have video slowdown problem (at the beginning it plays it normally, but after 30 seconds or so video starts to play approx twice slower). What I also observed, is that if I take that family_guy.flv or mp4 and convert it back to family_guy2.ts then this new famili_guy2.ts will also have the same problem as mp4 or flv file. I used binary compare tool to check and it looks like after mpegts -> mp4 -> mpegts h264 bitstream becomes somewhat different and doesn't play normally on iPhone 4. Did anybody had the same problem, any idea what I could do to "fix" some of these video to make them segmentable with ffmpeg and segmenter? thank you. PS. just to clarify, I do use -vbsf h264_mp4toannexb to convert from mp4 back to ts file, I think the problem is somehow related to that bitstream re-format. From lou at fakeoutdoorsman.com Fri Apr 22 06:47:11 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Thu, 21 Apr 2011 20:47:11 -0800 Subject: [FFmpeg-user] FFMPEG Update In-Reply-To: References: <4DAD682A.9060903@thelounge.net> <4DAD702C.1010701@thelounge.net> Message-ID: <20110421204711.1012532c@lrcd.com> On Tue, 19 Apr 2011 19:31:08 +0800 Antony Acosta wrote: > This are basically the codes that will be most affected if ever we > update the FFmpeg version to its latest: As with any update there is a chance that your command could become outdated. > - ffmpeg -i input.flv -ss 00:00:01 -s 640x380 output.jpg > - ffmpeg -i input.flv -ar 44100 -f flv -b 1000kb output.flv Both of these commands should still work although flv (the encoder) won't give as good results as libx264 for flv (the container) output. > Will the update affected those codes somehow? Because I also tried > other stuffs like converting to mp4 videos such as: > > > - ffmpeg -i input.flv input -acodec aac -ab 128kb -vcodec mpeg4 -b > 1200kb -mbd 2 -flags +4mv+trell -aic 2 -cmp 2 -subcmp 2 -s 320x180 > -title X output.mp4 Many of your options are outdated, but if you wanted to use this particular command an updated version would be something like: ffmpeg -i input.flv -acodec aac -strict experimental -ab 128k -vcodec mpeg4 -b 1200k -mbd 2 -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title="X" output.mp4 "ffmpeg -h" will give you a massive list of the option names. > - ffmpeg -y -i input.flv -b 768 -s 320x240 -vcodec xvid -ab 128 > -acodec aac -ac 2 -ab 64 -f mp4 output.mp4 ffmpeg -i input.flv -b 768k -s 320x240 -vcodec libxvid -ab 128k -acodec aac -strict experimental -ac 2 -ab 64k output.mp4 Alternatively you could replace "-acodec aac -strict experimental" with: * "-acodec libfaac" or... * "-acodec libvo_aacenc" if you compile FFmpeg to support these third-party libraries > However this didn't work. I'm guessing I would need codecs installed > to have this working and to update my FFmpeg to it's latest version? Error messages are always more useful than, "this didn't work". > Antony > [Programmer, Allied Web Design] > > ?[C][G]? "THere is no cHaLLenge THaT can'T Be FinisHeD!" ? > ?'?'\????\?=(???)=?/????/'?'? ? > > > On Tue, Apr 19, 2011 at 7:21 PM, Reindl Harald > wrote: > > > > > Am 19.04.2011 12:50, schrieb Antony Acosta: > > > I am currently using FFmpeg version SVN-r20220 > > > > sounds quite old but i do not know every svn-release by it's > > name :-) why do you not post from begin the first output of > > "ffmpeg" so that anybody has useful informations? > > > > [harry at nb-rhsoft:~]$ ffmpeg > > FFmpeg version git-N-28742-g1caa412, Copyright (c) 2000-2011 the > > FFmpeg developers > > built on Apr 13 2011 13:00:04 with gcc 4.5.1 20100924 (Red Hat > > 4.5.1-4) From paul__tapper at hotmail.com Fri Apr 22 09:26:12 2011 From: paul__tapper at hotmail.com (Paul Tapper) Date: Fri, 22 Apr 2011 07:26:12 +0000 Subject: [FFmpeg-user] demuxing and re-muxing of mxf video files Message-ID: Hi, I hope this is the right place to post this question. Sorry if it isn't. I'm trying to use FFmpeg to take an .mxf file and:1. demux the audio out of it to a .wav (and then process the .wav myself), then2. mux together the video from the original .mxf with the .wav that I have now processed, to get back to an .mxf file, as similar as possible to the original .mxf, but now with my processed audio, instead of the original audio. I've got step 1. working with a very simple: ffmpeg.exe -i "test.mxf" "test.mxf.wav" but I'm a bit stuck on step 2. I've tried the following: ffmpeg.exe -an -i "test.mxf" -i "test.mxf.wav" -vcodec copy "test.mxf.remux.mxf" but I get an error "[mxf @ 01EF8610] track 0: could not find essence container ul, codec not currently supported in container" Please could anyone suggest what I should do here? Is this even a sensible thing to attempt with FFmpeg? Thanks Paul From ranjiniraguu at gmail.com Fri Apr 22 11:30:59 2011 From: ranjiniraguu at gmail.com (ranjini raguu) Date: Fri, 22 Apr 2011 15:00:59 +0530 Subject: [FFmpeg-user] Reg:conversion of video file Message-ID: Im using ffmpeg for converting audio n video.... the conversion is working fine... but whn i add some presets values the conversion is missing somewhere i.e after conversion the length of the video is not displayed so tat the video cannot be seeked... how to fix this so tat the length should be identifies after conversion..... wat can we do for it? From ranjiniraguu at gmail.com Fri Apr 22 11:34:37 2011 From: ranjiniraguu at gmail.com (ranjini raguu) Date: Fri, 22 Apr 2011 15:04:37 +0530 Subject: [FFmpeg-user] Reg:conversion of video file In-Reply-To: References: Message-ID: Im using ffmpeg for converting audio n video.... the conversion is working fine... but whn i add some presets values the conversion is missing somewhere i.e after conversion the length of the video is not displayed so tat the video cannot be seeked... how to fix this so tat the length should be identifies after conversion..... wat can we do for it? From rodney.baker at iinet.net.au Fri Apr 22 13:42:01 2011 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Fri, 22 Apr 2011 21:12:01 +0930 Subject: [FFmpeg-user] demuxing and re-muxing of mxf video files In-Reply-To: References: Message-ID: <201104222112.01831.rodney.baker@iinet.net.au> On Fri, 22 Apr 2011 16:56:12 Paul Tapper wrote: > Hi, > I hope this is the right place to post this question. Sorry if it isn't. > I'm trying to use FFmpeg to take an .mxf file and:1. demux the audio out of > it to a .wav (and then process the .wav myself), then2. mux together the > video from the original .mxf with the .wav that I have now processed, to > get back to an .mxf file, as similar as possible to the original .mxf, but > now with my processed audio, instead of the original audio. I've got step > 1. working with a very simple: > ffmpeg.exe -i "test.mxf" "test.mxf.wav" > but I'm a bit stuck on step 2. > I've tried the following: > ffmpeg.exe -an -i "test.mxf" -i "test.mxf.wav" -vcodec copy > "test.mxf.remux.mxf" but I get an error "[mxf @ 01EF8610] track 0: could > not find essence container ul, codec not currently supported in container" > Please could anyone suggest what I should do here? Is this even a > sensible thing to attempt with FFmpeg? Thanks > Paul First of all, you're telling ffmpeg that you want no audio (the -an) switch, then trying to add audio. You need to specify the 2 input files as you have done, but then map the streams from each input file to the output file. If the video in test.mxf is stream 0 then you would use -map 0:0 -map 1:0. You also need the "newaudio" switch. Your final command would look something like: ffmpeg -i test.mxf -i test.mxf.wav -acodec copy -vcodec copy -map 0:0 -map 1:0 output.mxf -newaudio You might have to play around with the order of the options (order is important). Others who have done this before may have better suggestions. YMMV. -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From butter1093 at gmail.com Fri Apr 22 12:09:36 2011 From: butter1093 at gmail.com (butter1093) Date: Fri, 22 Apr 2011 03:09:36 -0700 (PDT) Subject: [FFmpeg-user] m2ts to DNxHD convert In-Reply-To: References: Message-ID: <1303466976228-3467729.post@n4.nabble.com> For mts to dhxhd conversion, you can use Aunsoft MTS/M2TS Converter for mac to help you make a converison with high quality. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/m2ts-to-DNxHD-convert-tp939577p3467729.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From kushal.patel at ennovatetech.com Fri Apr 22 12:21:55 2011 From: kushal.patel at ennovatetech.com (kushal) Date: Fri, 22 Apr 2011 03:21:55 -0700 (PDT) Subject: [FFmpeg-user] FFmpeg configure script, iOS 4.3, XCode 3.2.6 In-Reply-To: References: Message-ID: <1303467715039-3467739.post@n4.nabble.com> Hi, In ffmpeg configuration script by default Library path set to "/usr/lib/system" so it can not find library libcache.dylib,so we have to set latest sdk path. Please try by adding extra ldflags as shown below with your ./configure script. --extra-ldflags=-L/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS4.3.sdk/usr/lib/system I have just add this flag and compiled ffmpeg for armv7 and iPhoneOS4.3 sdk with success. Thanks, Kushal Patel -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFmpeg-configure-script-iOS-4-3-XCode-3-2-6-tp3422006p3467739.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mike.scheutzow at alcatel-lucent.com Fri Apr 22 15:25:35 2011 From: mike.scheutzow at alcatel-lucent.com (Mike Scheutzow) Date: Fri, 22 Apr 2011 09:25:35 -0400 Subject: [FFmpeg-user] Regarding AVI file with variable frame rate In-Reply-To: <1303381522687-3465567.post@n4.nabble.com> References: <1303381522687-3465567.post@n4.nabble.com> Message-ID: <4DB181CF.50608@alcatel-lucent.com> bhavesh wrote: > Dear ffmpeg-user, > > I am new with ffmpeg and i want to convert my h.264 file which has varaible > frame rate into avi file. > > The file converted by ffmpeg was Ok but when i play this file it looks like > fast forward view. Did you specify the output frame rate you want using -r ? Please post your ffmpeg command line, and *all* the output produced by ffmpeg. Mike Scheutzow From paul__tapper at hotmail.com Fri Apr 22 15:58:41 2011 From: paul__tapper at hotmail.com (Paul Tapper) Date: Fri, 22 Apr 2011 13:58:41 +0000 Subject: [FFmpeg-user] demuxing and re-muxing of mxf video files In-Reply-To: <201104222112.01831.rodney.baker@iinet.net.au> References: , <201104222112.01831.rodney.baker@iinet.net.au> Message-ID: Hi Rodney, Thanks very much for the help. I tried your suggestion and FFmpeg complained about the number of mappings in the output file, which I think might have been due to the -newaudio switch, so I tried removing that, and now it looks like its trying to do the re-mux, but is failing. I get the following output... Input #0, mxf, from 'C:\ffmpegTest\test.mxf': Duration: 00:00:03.40, start: 0.000000, bitrate: 136319 kb/s Stream #0.0: Video: libopenjpeg, rgb24, 1920x1080, 25 fps, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/sInput #1, wav, from 'C:\ffmpegTest\test.mxf.wav': Duration: 00:00:03.40, bitrate: 1536 kb/s Stream #1.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s[mxf @ 0097C990] track 0: could not find essence container ul, codec not currently supported in containerOutput #0, mxf, to 'C:\ffmpegTest\test.mxf.remux.mxf': Metadata: encoder : Lavf52.108.0 Stream #0.0: Video: [0][0][0][0] / 0x0000, rgb24, 1920x1080, q=2-31, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 48000 Hz, 2 channels, 1536 kb/sStream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1Could not write header for output file #0 (incorrect codec parameters ?) I don't really understand any of this, but the "[mxf @ 0097C990] track 0: could not find essence container ul, codec not currently supported in container" makes me wonder whether there is some problem with the mux-ing of mxf files. Please, do you have any further suggestions? Thanks Paul > From: rodney.baker at iinet.net.au > To: ffmpeg-user at ffmpeg.org > Date: Fri, 22 Apr 2011 21:12:01 +0930 > Subject: Re: [FFmpeg-user] demuxing and re-muxing of mxf video files > > On Fri, 22 Apr 2011 16:56:12 Paul Tapper wrote: > > Hi, > > I hope this is the right place to post this question. Sorry if it isn't. > > I'm trying to use FFmpeg to take an .mxf file and:1. demux the audio out of > > it to a .wav (and then process the .wav myself), then2. mux together the > > video from the original .mxf with the .wav that I have now processed, to > > get back to an .mxf file, as similar as possible to the original .mxf, but > > now with my processed audio, instead of the original audio. I've got step > > 1. working with a very simple: > > ffmpeg.exe -i "test.mxf" "test.mxf.wav" > > but I'm a bit stuck on step 2. > > I've tried the following: > > ffmpeg.exe -an -i "test.mxf" -i "test.mxf.wav" -vcodec copy > > "test.mxf.remux.mxf" but I get an error "[mxf @ 01EF8610] track 0: could > > not find essence container ul, codec not currently supported in container" > > Please could anyone suggest what I should do here? Is this even a > > sensible thing to attempt with FFmpeg? Thanks > > Paul > > First of all, you're telling ffmpeg that you want no audio (the -an) switch, > then trying to add audio. > > You need to specify the 2 input files as you have done, but then map the > streams from each input file to the output file. If the video in test.mxf is > stream 0 then you would use -map 0:0 -map 1:0. > > You also need the "newaudio" switch. > > Your final command would look something like: > > ffmpeg -i test.mxf -i test.mxf.wav -acodec copy -vcodec copy -map 0:0 -map 1:0 > output.mxf -newaudio > > You might have to play around with the order of the options (order is > important). Others who have done this before may have better suggestions. > YMMV. > > -- > =================================================== > Rodney Baker VK5ZTV > rodney.baker at iinet.net.au > =================================================== > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From lou at fakeoutdoorsman.com Fri Apr 22 19:11:54 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Fri, 22 Apr 2011 09:11:54 -0800 Subject: [FFmpeg-user] Reg:conversion of video file In-Reply-To: References: Message-ID: <20110422091154.14da5ccd@lrcd.com> On Fri, 22 Apr 2011 15:00:59 +0530 ranjini raguu wrote: > Im using ffmpeg for converting audio n video.... the conversion is > working fine... but whn i add some presets values the conversion is What preset values? > missing somewhere i.e after conversion the length of the video is not > displayed so tat the video cannot be seeked... how to fix this so tat The length of the video is not displayed properly in ______? FFmpeg, a web flash player, Quicktime, WMP, mediainfo? Seeked with what? > the length should be identifies after conversion..... > wat can we do for it? First of all, you could show us your FFmpeg command and the complete console output. Help us help you by providing any useful and necessary information when asking questions. You should provide enough detailed information that anyone else could attempt to duplicate any issues that you are encountering. From me at shanemcgovern.com Fri Apr 22 19:20:16 2011 From: me at shanemcgovern.com (Shane) Date: Fri, 22 Apr 2011 10:20:16 -0700 Subject: [FFmpeg-user] Basic - Windows Binary AMR to MP3? Message-ID: <4DB1B8D0.4010501@shanemcgovern.com> Hi I'm using the Windows binary of FFmpeg (which is awesome) and have no problems at all converting, clipping and general accomplishing everything I need with video. I cannot seem to figure out the correct command line combination for converting AMR (from my smart phone) files to MP3 (or anything audio). FFmpeg's output cannot be played or is really short and sounds like static. Can someone post a working command line example? From baptiste.coudurier at gmail.com Fri Apr 22 21:55:18 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Fri, 22 Apr 2011 12:55:18 -0700 Subject: [FFmpeg-user] Apple live HTTP streaming: video slows down if processed with ffmpeg In-Reply-To: <59371303438362@web81.yandex.ru> References: <59371303438362@web81.yandex.ru> Message-ID: <4DB1DD26.1030409@gmail.com> Hi, On 04/21/2011 07:12 PM, i-love-spam wrote: > > I'm trying to make live http streaming working with and iPhone 4. For > live http streaming I have to use mpegts segments that contain h264 > and mp3/aac for audio. I use 10 sec segments and I use opensource > segmenter.c that uses libavformat/avcodec to segment videos into > mpegts chunks. > > What I noticed, is that many of my segmented videos didn't play > properly: after playing for 30 seconds or so video starts to play > twice slower while sound plays at normal speed. So I tried to > investigate what's wrong and I came to conclusion that chunks > generated from flv/mp4 etc always have this problem and segments > generated from other ts files do not have the problem. I tried to > different tests, and it looks that ffmpeg somehow modifies bitstream > so that live http streaming does not work on iPhone. > > I have an original family guy mpegts video clip. If I use this clip > for segmenting then live http streaming works perfectly. If I convert > this file with ffmpeg -i famili_guy.ts -acodec copy -vcodec copy > family_guy.flv (or mp4) then if I try to segment that resulting file > then iPhone will have video slowdown problem (at the beginning it > plays it normally, but after 30 seconds or so video starts to play > approx twice slower). What I also observed, is that if I take that > family_guy.flv or mp4 and convert it back to family_guy2.ts then this > new famili_guy2.ts will also have the same problem as mp4 or flv > file. I used binary compare tool to check and it looks like after > mpegts -> mp4 -> mpegts h264 bitstream becomes somewhat different and > doesn't play normally on iPhone 4. > > Did anybody had the same problem, any idea what I could do to "fix" > some of these video to make them segmentable with ffmpeg and > segmenter? > > > thank you. > > > PS. just to clarify, I do use -vbsf h264_mp4toannexb to convert from > mp4 back to ts file, I think the problem is somehow related to that > bitstream re-format. You need to use -absf aac_adtstoasc when going from ts to mp4/flv. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From baptiste.coudurier at gmail.com Fri Apr 22 21:56:27 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Fri, 22 Apr 2011 12:56:27 -0700 Subject: [FFmpeg-user] m2ts to DNxHD convert In-Reply-To: <1303466976228-3467729.post@n4.nabble.com> References: <1303466976228-3467729.post@n4.nabble.com> Message-ID: <4DB1DD6B.5000202@gmail.com> On 04/22/2011 03:09 AM, butter1093 wrote: > For mts to dhxhd conversion, you can use Aunsoft MTS/M2TS Converter for mac > to help you make a converison with high quality. Please do not suggest nor advertise any non open source product/software on this mailing list. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From i-love-spam at yandex.ru Sat Apr 23 00:22:02 2011 From: i-love-spam at yandex.ru (pps) Date: Fri, 22 Apr 2011 18:22:02 -0400 Subject: [FFmpeg-user] Apple live HTTP streaming: video slows down if processed with ffmpeg In-Reply-To: <4DB1DD26.1030409@gmail.com> References: <59371303438362@web81.yandex.ru> <4DB1DD26.1030409@gmail.com> Message-ID: <4DB1FF8A.9070607@yandex.ru> On 22-Apr-11 15:55, Baptiste Coudurier wrote: > Hi, > You need to use -absf aac_adtstoasc when going from ts to mp4/flv. > Actually, I used clips with mp3 and didn't have to do anything. The video source is 24fps, but after it goes to mp4 and back to ts segments video frame timestamps are different a little bit: in original movie all video frame timestamps increase monotonically (by 3715 if I'm correct), but after conversion to mp4 they increase by 3730 3700 3730 3700, so that overall timing is the same, but difference between frames is not constant. In that case live http stream breaks after a couple of minutes of playback: video becomes very slow while audio plays normal. From allenskd at ymail.com Sat Apr 23 01:16:02 2011 From: allenskd at ymail.com (David Guevara) Date: Fri, 22 Apr 2011 16:16:02 -0700 (PDT) Subject: [FFmpeg-user] Is it possible to do this with avfilter? In-Reply-To: <20110421230300.GD1368@geppetto> References: <268173.7861.qm@web120518.mail.ne1.yahoo.com> <20110421230300.GD1368@geppetto> Message-ID: <992525.11955.qm@web120509.mail.ne1.yahoo.com> Thanks for responding, I sat down today to RTFM of libavfilter and its filterchain so I got it working. ________________________________ From: Stefano Sabatini To: FFmpeg user questions and RTFMs Sent: Thu, April 21, 2011 7:03:00 PM Subject: Re: [FFmpeg-user] Is it possible to do this with avfilter? On date Thursday 2011-04-21 11:40:28 -0700, David Gonzalez encoded: > I'm trying to do this with ffmpeg where I put a banner on top of the video > > http://img269.imageshack.us/i/exampleyl.jpg/ like in this example Why not? > problem is that I tried to do some chaining (scale then overlay) but it didn't > go through, if i do it with just "overlay" the image gets in the way so I'm not > > sure if I'm getting the the chain right or not. I don't know how to interpret "didn't go through" and "gets in the way", show us the command you're using. -- ffmpeg-user random tip #7 If you're asking about problems with an ffmpeg tool, *always* post the command you're using and the complete output. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From jeisom at gmail.com Sat Apr 23 11:47:10 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Sat, 23 Apr 2011 04:47:10 -0500 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <4DADF16D.4050003@gmail.com> References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> <20110418125128.53550483@lrcd.com> <4DACA5A8.8060106@gmail.com> <4DADF16D.4050003@gmail.com> Message-ID: On Tue, Apr 19, 2011 at 3:32 PM, Baptiste Coudurier wrote: > On 04/18/2011 01:57 PM, Baptiste Coudurier wrote: >> On 04/18/2011 01:51 PM, Lou wrote: >>> On Mon, 18 Apr 2011 17:03:49 +0200 >>> ?ric Cholet wrote: >>> >>>> Le 18 avr. 2011 ? 16:49, Eric Cholet a ?crit : >>>>> With freshly downloaded ffmpeg from git, I'm getting an error when >>>>> trying to use the -profile option: >>>>> >>>>> >>>>> % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b 2500k >>>>> -r 25 -preset medium -profile main foo.mp4 >>>> >>>> Oops forgot to post the error message: >>>> >>>> [NULL @ 0x80240b200] [Eval @ 0x7fffffffd110] Undefined constant or >>>> missing '(' in 'main' [NULL @ 0x80240b200] Unable to parse option >>>> value "main" Invalid value 'main' for option 'profile' >>>> >>>> >>>>> >>>>> What am I doing wrong? >>>> -- >>>> ?ric Cholet >>> >>> You are doing nothing wrong. It was my mistake to mention -profile. It >>> is listed in 'ffmpeg -h', but is not available yet. >>> >>> I like to verify my suggestions, but this time I didn't. This is what >>> happens when I rely on memory as I previously tested the new options a >>> month ago but apparently forgot that -profile wasn't usable yet. >> >> I'm on it, don't worry :) >> > > All right, -profile should work now. > Please report any problem. As of commit 3d462373f8e8fa58b697a9b2d93a5edf70841937 I get FFmpeg version git-N-29302-g3d46237, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 23 2011 04:42:40 with gcc 4.5.2 configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --cc=x86_64-pc-linux-gnu-gcc --disable-static --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --disable-stripping --disable-debug --disable-doc --enable-libmp3lame --enable-libvorbis --enable-libx264 --enable-libfaac --enable-nonfree --disable-indev=v4l --disable-indev=oss --disable-indev=jack --enable-x11grab --disable-outdev=oss --enable-pthreads --disable-ssse3 --disable-altivec --cpu=amdfam10 --enable-hardcoded-tables libavutil 51. 0. 0 / 51. 0. 0 libavcodec 53. 1. 0 / 53. 1. 0 libavformat 53. 0. 1 / 53. 0. 1 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 0. 0 / 2. 0. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg @ 0x1158ae0] max_analyze_duration reached Input #0, mpeg, from '/tmp//Morning Show.tmp': Duration: 00:45:02.61, start: 0.360000, bitrate: 3747 kb/s Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 [PAR 8:9 DAR 4:3], 15000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s Unrecognized option '-profile' Later Jonathan > -- > Baptiste COUDURIER > Key fingerprint ? ? ? ? ? ? ? ? 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA > FFmpeg maintainer ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?http://www.ffmpeg.org > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From jeisom at gmail.com Sat Apr 23 11:48:27 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Sat, 23 Apr 2011 04:48:27 -0500 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: References: <4DAA2FE3.5040708@gmail.com> <20110416190804.77de7d9e@lrcd.com> <3751C01E-EF65-42BC-A6B1-A3D215941EC0@logilune.com> <20110418125128.53550483@lrcd.com> <4DACA5A8.8060106@gmail.com> <4DADF16D.4050003@gmail.com> Message-ID: On Sat, Apr 23, 2011 at 4:47 AM, Jonathan Isom wrote: > On Tue, Apr 19, 2011 at 3:32 PM, Baptiste Coudurier > wrote: >> On 04/18/2011 01:57 PM, Baptiste Coudurier wrote: >>> On 04/18/2011 01:51 PM, Lou wrote: >>>> On Mon, 18 Apr 2011 17:03:49 +0200 >>>> ?ric Cholet wrote: >>>> >>>>> Le 18 avr. 2011 ? 16:49, Eric Cholet a ?crit : >>>>>> With freshly downloaded ffmpeg from git, I'm getting an error when >>>>>> trying to use the -profile option: >>>>>> >>>>>> >>>>>> % ffmpeg -i foo.mov -f mp4 -vcodec libx264 -ar 22050 -ac 2 -b 2500k >>>>>> -r 25 -preset medium -profile main foo.mp4 >>>>> >>>>> Oops forgot to post the error message: >>>>> >>>>> [NULL @ 0x80240b200] [Eval @ 0x7fffffffd110] Undefined constant or >>>>> missing '(' in 'main' [NULL @ 0x80240b200] Unable to parse option >>>>> value "main" Invalid value 'main' for option 'profile' >>>>> >>>>> >>>>>> >>>>>> What am I doing wrong? >>>>> -- >>>>> ?ric Cholet >>>> >>>> You are doing nothing wrong. It was my mistake to mention -profile. It >>>> is listed in 'ffmpeg -h', but is not available yet. >>>> >>>> I like to verify my suggestions, but this time I didn't. This is what >>>> happens when I rely on memory as I previously tested the new options a >>>> month ago but apparently forgot that -profile wasn't usable yet. >>> >>> I'm on it, don't worry :) >>> >> >> All right, -profile should work now. >> Please report any problem. > > As of commit 3d462373f8e8fa58b697a9b2d93a5edf70841937 I get > > > FFmpeg version git-N-29302-g3d46237, Copyright (c) 2000-2011 the > FFmpeg developers > ?built on Apr 23 2011 04:42:40 with gcc 4.5.2 > ?configuration: --prefix=/usr --libdir=/usr/lib64 > --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared > --cc=x86_64-pc-linux-gnu-gcc --disable-static --enable-gpl > --enable-version3 --enable-postproc --enable-avfilter > --disable-stripping --disable-debug --disable-doc --enable-libmp3lame > --enable-libvorbis --enable-libx264 --enable-libfaac --enable-nonfree > --disable-indev=v4l --disable-indev=oss --disable-indev=jack > --enable-x11grab --disable-outdev=oss --enable-pthreads > --disable-ssse3 --disable-altivec --cpu=amdfam10 > --enable-hardcoded-tables > ?libavutil ? ?51. ?0. 0 / 51. ?0. 0 > ?libavcodec ? 53. ?1. 0 / 53. ?1. 0 > ?libavformat ?53. ?0. 1 / 53. ?0. 1 > ?libavdevice ?53. ?0. 0 / 53. ?0. 0 > ?libavfilter ? 2. ?0. 0 / ?2. ?0. 0 > ?libswscale ? ?0. 13. 0 / ?0. 13. 0 > ?libpostproc ?51. ?2. 0 / 51. ?2. 0 > [mpeg @ 0x1158ae0] max_analyze_duration reached > Input #0, mpeg, from '/tmp//Morning Show.tmp': > ?Duration: 00:45:02.61, start: 0.360000, bitrate: 3747 kb/s > ? ?Stream #0.0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x480 > [PAR 8:9 DAR 4:3], 15000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 > tbc > ? ?Stream #0.1[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 192 kb/s > Unrecognized option '-profile' Sorry User Error. > Later > > Jonathan > >> -- >> Baptiste COUDURIER >> Key fingerprint ? ? ? ? ? ? ? ? 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA >> FFmpeg maintainer ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?http://www.ffmpeg.org >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > From coniophora at gmail.com Sat Apr 23 21:24:56 2011 From: coniophora at gmail.com (Jim Worrall) Date: Sat, 23 Apr 2011 13:24:56 -0600 Subject: [FFmpeg-user] .flv conversion to .avi Message-ID: <0614E95A-48AF-448E-BCFD-F432B568835A@gmail.com> I'm trying to convert a .flv file (H.264/MPEG-4 AVC and AAC - MPEG-4 audio) to something that Windows Media Player can play on a normal Windows installation without installing additional codecs. This could be in a .wmv, .avi, .mpg, and a few others I guess. At first, recoding the video with -sameq gave 10x the file size of the original. Then I realized I could keep the video intact with "-vcodec copy" and still make an avi. ffmpeg -i INPUT.flv -vcodec copy -acodec libmp3lame OUTPUT.avi This played great with VLC on my Mac, but when I tried playing it on a Windows machine, only the audio would play. I would be grateful if someone could recommend an efficient method of transcoding files like this, that would play with codecs available on a basic Windows install. Here is the ffmpeg setup on my Mac. It is built from a git snapshot just two days ago: Jims-MacBook-Pro:~ jim$ ffmpeg FFmpeg version UNKNOWN, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 21 2011 18:00:17 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --arch=x86_64 --enable-runtime-cpudetect libavutil 51. 0. 0 / 51. 0. 0 libavcodec 53. 0. 0 / 53. 0. 0 libavformat 53. 0. 0 / 53. 0. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 0. 0 / 2. 0. 0 libswscale 0. 13. 0 / 0. 13. 0 Thanks, Jim From h.reindl at thelounge.net Sat Apr 23 22:36:22 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sat, 23 Apr 2011 22:36:22 +0200 Subject: [FFmpeg-user] .flv conversion to .avi In-Reply-To: <0614E95A-48AF-448E-BCFD-F432B568835A@gmail.com> References: <0614E95A-48AF-448E-BCFD-F432B568835A@gmail.com> Message-ID: <4DB33846.4090309@thelounge.net> Am 23.04.2011 21:24, schrieb Jim Worrall: > I'm trying to convert a .flv file (H.264/MPEG-4 AVC and AAC - MPEG-4 audio) to something that Windows Media Player can play on a normal Windows installation without installing additional codecs. This could be in a .wmv, .avi, .mpg, and a few others I guess. > At first, recoding the video with -sameq gave 10x the file size of the original. Then I realized I could keep the video intact with "-vcodec copy" and still make an avi. > > ffmpeg -i INPUT.flv -vcodec copy -acodec libmp3lame OUTPUT.avi why in the world "-vcodec copy"? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From david.ariens at gmail.com Sat Apr 23 22:37:41 2011 From: david.ariens at gmail.com (David) Date: Sat, 23 Apr 2011 20:37:41 +0000 (UTC) Subject: [FFmpeg-user] =?utf-8?q?Issue_with_watermarking_a_video_-=3E_resu?= =?utf-8?q?lt_is=09garbled_=28pink_and_green_lines=29?= References: <4DADB954.7020108@bbc.co.uk> <4DADB9F2.9020007@bbc.co.uk> <4DAE87D5.6070600@bbc.co.uk> Message-ID: Tim Nicholson bbc.co.uk> writes: > FFmpeg version git-N-29169-g7e2eb4b, Copyright (c) 2000-2011 the FFmpeg > developers > built on Apr 18 2011 09:59:45 with gcc 4.5.0 20100604 [gcc-4_5-branch > revision 160292] > Hi Tim, Thanks for the reply. I downloaded a new release and the problem was gone. david From bahamutzero8825 at gmail.com Sat Apr 23 23:00:41 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Sat, 23 Apr 2011 16:00:41 -0500 Subject: [FFmpeg-user] .flv conversion to .avi In-Reply-To: <0614E95A-48AF-448E-BCFD-F432B568835A@gmail.com> References: <0614E95A-48AF-448E-BCFD-F432B568835A@gmail.com> Message-ID: <4DB33DF9.9000901@gmail.com> On 2011.04.23 02:24 PM, Jim Worrall wrote: > Then I realized I could keep the video intact with "-vcodec copy" and still make an avi. AVC in AVI is a hack at best. Either WMP supports AVC out-of-the-box or it doesn't. The container the video stream is in is irrelevant. If WMP supports AVC, but not MP4 or Matroska, MPEG-TS is an appropriate container. If it doesn't support AVC, -vcodec copy will give you unusable output 100% of the time. From coniophora at gmail.com Sun Apr 24 00:22:06 2011 From: coniophora at gmail.com (Jim Worrall) Date: Sat, 23 Apr 2011 16:22:06 -0600 Subject: [FFmpeg-user] .flv conversion to .avi In-Reply-To: <4DB33DF9.9000901@gmail.com> References: <0614E95A-48AF-448E-BCFD-F432B568835A@gmail.com> <4DB33DF9.9000901@gmail.com> Message-ID: Thanks, it sounds like there is no reason to think it should work, as indeed it didn't. So my question remains, can someone "recommend an efficient method [that is, one that doesn't increase the file size an order of magnitude, while maintaining quality] of transcoding files like this [./flv with H.264/MPEG-4 AVC and AAC - MPEG-4 audio], that would play with codecs available on a basic Windows install. On Apr 23, 2011, at 3:00 PM, Andrew Berg wrote: On 2011.04.23 02:24 PM, Jim Worrall wrote: > Then I realized I could keep the video intact with "-vcodec copy" and still make an avi. AVC in AVI is a hack at best. Either WMP supports AVC out-of-the-box or it doesn't. The container the video stream is in is irrelevant. If WMP supports AVC, but not MP4 or Matroska, MPEG-TS is an appropriate container. If it doesn't support AVC, -vcodec copy will give you unusable output 100% of the time. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From klaus.kudielka at gmx.net Sun Apr 24 11:39:45 2011 From: klaus.kudielka at gmx.net (Klaus Kudielka) Date: Sun, 24 Apr 2011 11:39:45 +0200 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding Message-ID: <20110424093945.GA31290@mars> Hi, even though in principle I like the new concept of directly taking over the x264 presets in the ffmpeg libx264 wrapper, there is one big difference that makes ffmpeg less usable for x264 encoding: (a) x264 & ffmpeg -vpre => user options override presets (b) ffmpeg -preset => presets override user options In my particular example of encoding HD video: I used to be able to specify the "slower" preset, together with -refs 4 resulting in a high profile, level 4.0 compliant output. This is not possible anymore, since -preset slower now persistently switches to refs=8, and I can't do anything about it :( Personally I would STRONGLY vote that ffmpeg -preset also conforms to behaviour (a). BUT: as I understand libx264.c, AVCodecContext doesn't tell X264_init whether the user has specified -refs or not. So, no trivial way to change the current behaviour. What's the opinion of the maintainers? Regards, Klaus From pyprog05 at gmail.com Sun Apr 24 12:55:17 2011 From: pyprog05 at gmail.com (PyProg PyProg) Date: Sun, 24 Apr 2011 12:55:17 +0200 Subject: [FFmpeg-user] m2ts to DNxHD convert In-Reply-To: References: Message-ID: 2009/2/14 luismanolo : > Hello. > Sorry about my english. > > I`m a new user of ffmpeg, and I have a sony hd camera. And I want to convert > the MTS files of my camera to the codec of Apple Pro Res called dnxhd. I > know that this codecs are supported for ffmpeg but I don`t know command to > convert the video. > I`m a ubuntu linux user. And I use cinelerra to edit the files. > > Many Thanks. Hi, You can use EKD for that. Install the SVN version for this tool. To install this SVN version: http://ekd.tuxfamily.org/index.php/Developpement/SVNEnglish Read this (sorry only in french): http://ekdm.wordpress.com/2010/08/17/arrivee-de-lavid-dnxhd-pour-la-gestion-de-lavchd/ EKD is a GNU/GPL v3 software: http://ekd.tuxfamily.org/index.php/Accueil/DroitsLogsLibresEnglish EKD uses (among others) FFmpeg. a+ -- http://ekd.tuxfamily.org http://ekdm.wordpress.com http://lcs.dunois.clg14.ac-caen.fr/~alama/blog http://lprod.org/wiki/doku.php/video:encodage:avchd_converter From rodney.baker at iinet.net.au Sun Apr 24 14:11:38 2011 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Sun, 24 Apr 2011 21:41:38 +0930 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <20110424093945.GA31290@mars> References: <20110424093945.GA31290@mars> Message-ID: <201104242141.38571.rodney.baker@iinet.net.au> On Sun, 24 Apr 2011 19:09:45 Klaus Kudielka wrote: > Hi, > > even though in principle I like the new concept of directly taking over the > x264 presets in the ffmpeg libx264 wrapper, there is one big difference > that makes ffmpeg less usable for x264 encoding: > > (a) x264 & ffmpeg -vpre => user options override presets > (b) ffmpeg -preset => presets override user options > > In my particular example of encoding HD video: > I used to be able to specify the "slower" preset, together with -refs 4 > resulting in a high profile, level 4.0 compliant output. > This is not possible anymore, since -preset slower now persistently > switches to refs=8, and I can't do anything about it :( > Use -profile high -level 4.0 and it should set the parameters correctly. -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From rickcorteza at gmail.com Sun Apr 24 15:00:59 2011 From: rickcorteza at gmail.com (Rick C.) Date: Sun, 24 Apr 2011 21:00:59 +0800 Subject: [FFmpeg-user] rmvb files Message-ID: Hello, I just wanted to double-check about converting a .rmvb file to let's say .avi. The results I'm getting is choppy video and I noticed this: [NULL @ 0x101829a00] Unsupported video codec [rm @ 0x101813000] max_analyze_duration reached If I just do ffmpeg -i file.rmvb this is printed out. As far as I can tell rv40 is a supported video codec correct? Anyways any input on .rmvb files and/or if I need to do something special in converting them would be helpful. Thanks! rc From klaus.kudielka at gmx.net Sun Apr 24 15:10:09 2011 From: klaus.kudielka at gmx.net (Klaus Kudielka) Date: Sun, 24 Apr 2011 15:10:09 +0200 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <201104242141.38571.rodney.baker@iinet.net.au> References: <20110424093945.GA31290@mars> <201104242141.38571.rodney.baker@iinet.net.au> Message-ID: <20110424131009.GA32022@mars> On Sun, Apr 24, 2011 at 09:41:38PM +0930, Rodney Baker wrote: > > Use -profile high -level 4.0 and it should set the parameters correctly. Unfortunately not. Look at the libx264 output below. It complains that DPB size > level limit, but does not correct the number of reference frames (ref=8). Looking at x264/encoder/set.c, this is the behaviour of x264_validate_levels, as it is. Specifying -level only validates the level, but does not enforce it. /opt/ffmpeg/bin/ffmpeg -i in.m2ts -threads 0 -map 0.0 -vcodec libx264 -b 8000k -maxrate 12000k -bufsize 25000k -preset slower -tune film -profile high -level 4.1 -pass 2 -map 0.1 -acodec ac3 -ac 2 -ab 192k -alang eng -sn -f matroska -y out.mkv -map 0.5 -acodec ac3 -ac 2 -ab 192k -alang ger -newaudio FFmpeg version UNKNOWN, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 23 2011 14:07:36 with gcc 4.4.3 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-pthreads --prefix=/opt/ffmpeg --extra-cflags=-I/opt/x264/include --extra-ldflags=-L/opt/x264/lib libavutil 51. 0. 0 / 51. 0. 0 libavcodec 53. 1. 0 / 53. 1. 0 libavformat 53. 0. 1 / 53. 0. 1 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 0. 0 / 2. 0. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mpegts, from 'in.m2ts': Duration: 02:28:51.67, start: 11.650667, bitrate: 22336 kb/s Program 1 Stream #0.0[0x1011]: Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 90k tbn, 23.98 tbc [[...audio & subtitle streams...]] [buffer @ 0x2ee3f20] w:1920 h:1080 pixfmt:yuv420p [libx264 @ 0x2ec40a0] using SAR=1/1 [libx264 @ 0x2ec40a0] DPB size (8 frames, 25067520 bytes) > level limit (4 frames, 12582912 bytes) [libx264 @ 0x2ec40a0] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 0x2ec40a0] profile High, level 4.1 [libx264 @ 0x2ec40a0] 264 - core 114 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=8 deblock=1:-1:-1 analyse=0x3:0x133 me=umh subme=9 psy=1 psy_rd=1.00:0.15 mixed_ref=1 me_range=16 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-3 threads=6 sliced_threads=0 nr=0 decimate=1 interlaced=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=23 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=2pass mbtree=1 bitrate=8000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 cplxblur=20.0 qblur=0.5 vbv_maxrate=12000 vbv_bufsize=25000 nal_hrd=none ip_ratio=1.40 aq=1:1.00 From rodney.baker at iinet.net.au Sun Apr 24 15:35:05 2011 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Sun, 24 Apr 2011 23:05:05 +0930 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <20110424131009.GA32022@mars> References: <20110424093945.GA31290@mars> <201104242141.38571.rodney.baker@iinet.net.au> <20110424131009.GA32022@mars> Message-ID: <201104242305.05605.rodney.baker@iinet.net.au> On Sun, 24 Apr 2011 22:40:09 Klaus Kudielka wrote: > On Sun, Apr 24, 2011 at 09:41:38PM +0930, Rodney Baker wrote: > > Use -profile high -level 4.0 and it should set the parameters correctly. > > Unfortunately not. Look at the libx264 output below. > It complains that DPB size > level limit, but does not correct the number > of reference frames (ref=8). > > Looking at x264/encoder/set.c, this is the behaviour of > x264_validate_levels, as it is. Specifying -level only validates the > level, but does not enforce it. So if ffmpeg is passing the -level parameter to x264, but x264 is not enforcing it, then that would seem like an x264 bug/feature, not an ffmpeg one. You might like to raise this on the x264-devel list (or log a bug report/feature request for x264) or, even better, submit a patch. :-) -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From klaus.kudielka at gmx.net Sun Apr 24 16:53:15 2011 From: klaus.kudielka at gmx.net (Klaus Kudielka) Date: Sun, 24 Apr 2011 16:53:15 +0200 Subject: [FFmpeg-user] New options -preset and -tune for x264 encoding In-Reply-To: <201104242305.05605.rodney.baker@iinet.net.au> References: <20110424093945.GA31290@mars> <201104242141.38571.rodney.baker@iinet.net.au> <20110424131009.GA32022@mars> <201104242305.05605.rodney.baker@iinet.net.au> Message-ID: <20110424145315.GA32298@mars> On Sun, Apr 24, 2011 at 11:05:05PM +0930, Rodney Baker wrote: > > So if ffmpeg is passing the -level parameter to x264, but x264 is not > enforcing it, then that would seem like an x264 bug/feature, not an ffmpeg > one. You might like to raise this on the x264-devel list (or log a bug > report/feature request for x264) or, even better, submit a patch. :-) > That would certainly be one possibility to solve my current problem. Coming back to my original question, one could also argue that "ffmpeg -preset" currently works more like a reset (overriding user options), than a *p*reset. Well, I'll probably hardcode all options, without any preset. Seems to be the safest approach right now. Regards, Klaus From ankuj2004 at gmail.com Mon Apr 25 07:39:09 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Mon, 25 Apr 2011 11:09:09 +0530 Subject: [FFmpeg-user] How to demux avi file Message-ID: Hi, I am having an avi file which has and H263 video stream. I want to extract video elementary stream from it. I tried looking at the documentation but didnt find anything relevant. Is it possible to do it ? Regards Ankuj From ranjiniraguu at gmail.com Mon Apr 25 07:50:50 2011 From: ranjiniraguu at gmail.com (ranjini raguu) Date: Mon, 25 Apr 2011 11:20:50 +0530 Subject: [FFmpeg-user] Reg:conversion of video file In-Reply-To: <20110422091154.14da5ccd@lrcd.com> References: <20110422091154.14da5ccd@lrcd.com> Message-ID: im sorry for not not providing the usefull information... actually wat im trying to do is trying to convert the video file to flv.... these r the parameters im passing to convert the file to flv for *Flash: Video (flv) for Web use Widescreen *-vcodec flv -f flv -r 29.97 -s 320x180 -aspect 16:9 -b 300kb -g 160 -cmp dct -subcmp dct -mbd 2 -flags +aic+cbp+mv0+mv4 -trellis 1 -ac 1 -ar 22050 -ab 56kb whn i convert it the video is converted to flv format but the length of the video is not shown.. i.e the total time of the video is not shown its showing 00.00..... video can be played but it cannot be forwarded it. if we try to do it then the video starts playing from the first.... this is the prob.... how to solve it??? if any details required im ready to share... thanks for any replies...... On Fri, Apr 22, 2011 at 10:41 PM, Lou wrote: > On Fri, 22 Apr 2011 15:00:59 +0530 > ranjini raguu wrote: > > > Im using ffmpeg for converting audio n video.... the conversion is > > working fine... but whn i add some presets values the conversion is > > What preset values? > > > missing somewhere i.e after conversion the length of the video is not > > displayed so tat the video cannot be seeked... how to fix this so tat > > The length of the video is not displayed properly in ______? FFmpeg, a > web flash player, Quicktime, WMP, mediainfo? > > Seeked with what? > > > the length should be identifies after conversion..... > > wat can we do for it? > > First of all, you could show us your FFmpeg command and the complete > console output. Help us help you by providing any useful and necessary > information when asking questions. You should provide enough detailed > information that anyone else could attempt to duplicate any issues > that you are encountering. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From benoit at friry.net Mon Apr 25 08:23:28 2011 From: benoit at friry.net (Benoit) Date: Mon, 25 Apr 2011 08:23:28 +0200 Subject: [FFmpeg-user] How to demux avi file In-Reply-To: References: Message-ID: <20110425062328.GB4391@friry.net> Hello, Ankuj Gupta a ?crit : > I am having an avi file which has and H263 video stream. I want to > extract video elementary stream from it. I tried looking at the > documentation but didnt find anything relevant. > Is it possible to do it ? Is it correct to say you want to remove audio streams? If so, look at "-vcodec copy" and "-an" options. benoit From ankuj2004 at gmail.com Mon Apr 25 08:37:16 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Mon, 25 Apr 2011 12:07:16 +0530 Subject: [FFmpeg-user] How to demux avi file In-Reply-To: <20110425062328.GB4391@friry.net> References: <20110425062328.GB4391@friry.net> Message-ID: Thanks Benoit. That worked as a charm :) Ankuj On Mon, Apr 25, 2011 at 11:53 AM, Benoit wrote: > Hello, > > Ankuj Gupta a ?crit : >> I am having an avi file which has and H263 video stream. I want to >> extract video elementary stream from it. I tried looking at the >> documentation but didnt find anything relevant. >> Is it possible to do it ?? > > Is it correct to say you want to remove audio streams? > If so, look at "-vcodec copy" and "-an" options. > > benoit > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ankuj2004 at gmail.com Mon Apr 25 10:06:11 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Mon, 25 Apr 2011 13:36:11 +0530 Subject: [FFmpeg-user] How to demux avi file In-Reply-To: References: <20110425062328.GB4391@friry.net> Message-ID: I was able to de-mux some avi files using the following command ffmpeg -i test042.avi -vcodec copy -an test042.263 But for some streams it is returning Unable to find a suitable output format for 'test042.263' I guess the avi file is not in correct format or is there issue with the command that I am using Ankuj On Mon, Apr 25, 2011 at 12:07 PM, Ankuj Gupta wrote: > Thanks Benoit. That worked as a charm :) > > Ankuj > > On Mon, Apr 25, 2011 at 11:53 AM, Benoit wrote: >> Hello, >> >> Ankuj Gupta a ?crit : >>> I am having an avi file which has and H263 video stream. I want to >>> extract video elementary stream from it. I tried looking at the >>> documentation but didnt find anything relevant. >>> Is it possible to do it ?? >> >> Is it correct to say you want to remove audio streams? >> If so, look at "-vcodec copy" and "-an" options. >> >> benoit >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > From benoit at friry.net Mon Apr 25 11:11:03 2011 From: benoit at friry.net (Benoit) Date: Mon, 25 Apr 2011 11:11:03 +0200 Subject: [FFmpeg-user] How to demux avi file In-Reply-To: References: <20110425062328.GB4391@friry.net> Message-ID: <20110425091103.GA3945@friry.net> Ankuj Gupta a ?crit : > I was able to de-mux some avi files using the following command > ffmpeg -i test042.avi -vcodec copy -an test042.263 > But for some streams it is returning > Unable to find a suitable output format for 'test042.263' > I guess the avi file is not in correct format or is there issue with > the command that I am using Try another container! I didn't find a definitive guide to "what can contain what". You should try ".mov" (QuickTime). benoit From Dave.Pope at myxer.com Mon Apr 25 15:21:22 2011 From: Dave.Pope at myxer.com (Dave Pope) Date: Mon, 25 Apr 2011 09:21:22 -0400 Subject: [FFmpeg-user] How to demux avi file In-Reply-To: <20110425091103.GA3945@friry.net> References: <20110425062328.GB4391@friry.net> <20110425091103.GA3945@friry.net> Message-ID: I have found this link to be useful for "what can contain what": http://en.wikipedia.org/wiki/Comparison_of_container_formats -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Benoit Sent: Monday, April 25, 2011 5:11 AM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] How to demux avi file Ankuj Gupta a ?crit : > I was able to de-mux some avi files using the following command > ffmpeg -i test042.avi -vcodec copy -an test042.263 > But for some streams it is returning > Unable to find a suitable output format for 'test042.263' > I guess the avi file is not in correct format or is there issue with > the command that I am using Try another container! I didn't find a definitive guide to "what can contain what". You should try ".mov" (QuickTime). benoit _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From lou at fakeoutdoorsman.com Mon Apr 25 20:07:21 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Mon, 25 Apr 2011 10:07:21 -0800 Subject: [FFmpeg-user] Reg:conversion of video file In-Reply-To: References: <20110422091154.14da5ccd@lrcd.com> Message-ID: <20110425100721.0a1ea283@lrcd.com> On Mon, 25 Apr 2011 11:20:50 +0530 ranjini raguu wrote: > im sorry for not not providing the usefull information... actually > wat im trying to do is trying to convert the video file to flv.... > > these r the parameters im passing to convert the file to flv for > *Flash: Video (flv) for Web use Widescreen > > *-vcodec flv -f flv -r 29.97 -s 320x180 -aspect 16:9 -b 300kb -g 160 > -cmp dct -subcmp dct -mbd 2 -flags +aic+cbp+mv0+mv4 -trellis 1 -ac 1 > -ar 22050 -ab 56kb I would personally use libx264 if I wanted FLV output. > whn i convert it the video is converted to flv format but the length > of the video is not shown.. i.e the total time of the video is not > shown its showing 00.00..... video can be played but it cannot be > forwarded it. if we try to do it then the video starts playing from > the first.... this is the prob.... how to solve it??? if any details > required im ready to share... thanks for any replies...... The length of the video is not displayed properly in what? You didn't answer this last time, so I'll assume you are refering to a flash player such as JW Player, flowplayer, etc. Try using flvtool++, flvmeta, flvtool2, or an equivalent tool on your FFmpeg output: flvtool++ input.flv output.flv > On Fri, Apr 22, 2011 at 10:41 PM, Lou wrote: > > > On Fri, 22 Apr 2011 15:00:59 +0530 > > ranjini raguu wrote: > > > > > Im using ffmpeg for converting audio n video.... the conversion is > > > working fine... but whn i add some presets values the conversion > > > is > > > > What preset values? > > > > > missing somewhere i.e after conversion the length of the video is > > > not displayed so tat the video cannot be seeked... how to fix > > > this so tat > > > > The length of the video is not displayed properly in ______? > > FFmpeg, a web flash player, Quicktime, WMP, mediainfo? > > > > Seeked with what? > > > > > the length should be identifies after conversion..... > > > wat can we do for it? > > > > First of all, you could show us your FFmpeg command and the complete > > console output. Help us help you by providing any useful and > > necessary information when asking questions. You should provide > > enough detailed information that anyone else could attempt to > > duplicate any issues that you are encountering. From gaurav1068 at gmail.com Mon Apr 25 11:44:33 2011 From: gaurav1068 at gmail.com (gaurav1068 at gmail.com) Date: Mon, 25 Apr 2011 02:44:33 -0700 Subject: [FFmpeg-user] ffplay compiling option Message-ID: <30514420.2256.1303724673707.JavaMail.nabble@joe.nabble.com> hey were you able to compile ffplay please let me know the exact text you typed to compile the ffplay From thuyps at yahoo.com Mon Apr 25 18:39:50 2011 From: thuyps at yahoo.com (PhamSonThuy) Date: Mon, 25 Apr 2011 09:39:50 -0700 (PDT) Subject: [FFmpeg-user] [ffmpeg for iphone] img_convert Message-ID: <450482.53650.qm@web30702.mail.mud.yahoo.com> Hi team, My name ThuyPham, from Vietnam. I have a question for you, please help me: Q: I have successfully using the latest ffmpeg, but I found that sws_scale() ate too much CPU. I knew that now img_convert was deprecated and the replacement is sws_scale. But my app does not need any scale, just img_convert is enough. So the question is how can I improve sws_scale() or how to make img_convert() visible? I downloaded the older version that supports img_convert but I could not compile it by using gas_preprocessor (seems gas_preprocessor only for the new versions). I'm a newbie on MacOS, so compiling ffmpeg is really a big challenge to me :( Regards & Thank you! - ThuyPham From lou at fakeoutdoorsman.com Tue Apr 26 05:54:02 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Mon, 25 Apr 2011 19:54:02 -0800 Subject: [FFmpeg-user] ffplay compiling option In-Reply-To: <30514420.2256.1303724673707.JavaMail.nabble@joe.nabble.com> References: <30514420.2256.1303724673707.JavaMail.nabble@joe.nabble.com> Message-ID: <20110425195402.3358428c@lrcd.com> On Mon, 25 Apr 2011 02:44:33 -0700 gaurav1068 at gmail.com wrote: > hey were you able to compile ffplay please let me know the exact text > you typed to compile the ffplay If ffplay isn't compiling for you then you are probably missing SDL. Because you did not mention your platform/OS/distro, or any other useful information, I can't provide very much information to help you. From ranjiniraguu at gmail.com Tue Apr 26 08:22:34 2011 From: ranjiniraguu at gmail.com (ranjini raguu) Date: Tue, 26 Apr 2011 11:52:34 +0530 Subject: [FFmpeg-user] Reg:conversion of video file In-Reply-To: <20110425100721.0a1ea283@lrcd.com> References: <20110422091154.14da5ccd@lrcd.com> <20110425100721.0a1ea283@lrcd.com> Message-ID: hello lou, *The length of the video is not displayed properly in what? * i meant tat the file after conversion does not contain the total length of the video.. for example if the length of the file is 3min 22 sec after conversion it shows 00.00 tats the problem im facing... On Mon, Apr 25, 2011 at 11:37 PM, Lou wrote: > On Mon, 25 Apr 2011 11:20:50 +0530 > ranjini raguu wrote: > > > im sorry for not not providing the usefull information... actually > > wat im trying to do is trying to convert the video file to flv.... > > > > these r the parameters im passing to convert the file to flv for > > *Flash: Video (flv) for Web use Widescreen > > > > *-vcodec flv -f flv -r 29.97 -s 320x180 -aspect 16:9 -b 300kb -g 160 > > -cmp dct -subcmp dct -mbd 2 -flags +aic+cbp+mv0+mv4 -trellis 1 -ac 1 > > -ar 22050 -ab 56kb > > I would personally use libx264 if I wanted FLV output. > > > whn i convert it the video is converted to flv format but the length > > of the video is not shown.. i.e the total time of the video is not > > shown its showing 00.00..... video can be played but it cannot be > > forwarded it. if we try to do it then the video starts playing from > > the first.... this is the prob.... how to solve it??? if any details > > required im ready to share... thanks for any replies...... > > The length of the video is not displayed properly in what? You didn't > answer this last time, so I'll assume you are refering to a flash > player such as JW Player, flowplayer, etc. > > Try using flvtool++, flvmeta, flvtool2, or an equivalent tool on your > FFmpeg output: > > flvtool++ input.flv output.flv > > > On Fri, Apr 22, 2011 at 10:41 PM, Lou wrote: > > > > > On Fri, 22 Apr 2011 15:00:59 +0530 > > > ranjini raguu wrote: > > > > > > > Im using ffmpeg for converting audio n video.... the conversion is > > > > working fine... but whn i add some presets values the conversion > > > > is > > > > > > What preset values? > > > > > > > missing somewhere i.e after conversion the length of the video is > > > > not displayed so tat the video cannot be seeked... how to fix > > > > this so tat > > > > > > The length of the video is not displayed properly in ______? > > > FFmpeg, a web flash player, Quicktime, WMP, mediainfo? > > > > > > Seeked with what? > > > > > > > the length should be identifies after conversion..... > > > > wat can we do for it? > > > > > > First of all, you could show us your FFmpeg command and the complete > > > console output. Help us help you by providing any useful and > > > necessary information when asking questions. You should provide > > > enough detailed information that anyone else could attempt to > > > duplicate any issues that you are encountering. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ranjiniraguu at gmail.com Tue Apr 26 09:49:35 2011 From: ranjiniraguu at gmail.com (ranjini raguu) Date: Tue, 26 Apr 2011 13:19:35 +0530 Subject: [FFmpeg-user] Reg:conversion of video file In-Reply-To: References: <20110422091154.14da5ccd@lrcd.com> <20110425100721.0a1ea283@lrcd.com> Message-ID: tat prob arises only when im converting wmv file to flv file. On Tue, Apr 26, 2011 at 11:52 AM, ranjini raguu wrote: > hello lou, > > > *The length of the video is not displayed properly in what? * > > i meant tat the file after conversion does not contain the total length of > the video.. for example if the length of the file is 3min 22 sec after > conversion it shows 00.00 tats the problem im facing... > > > On Mon, Apr 25, 2011 at 11:37 PM, Lou wrote: > >> On Mon, 25 Apr 2011 11:20:50 +0530 >> ranjini raguu wrote: >> >> > im sorry for not not providing the usefull information... actually >> > wat im trying to do is trying to convert the video file to flv.... >> > >> > these r the parameters im passing to convert the file to flv for >> > *Flash: Video (flv) for Web use Widescreen >> > >> > *-vcodec flv -f flv -r 29.97 -s 320x180 -aspect 16:9 -b 300kb -g 160 >> > -cmp dct -subcmp dct -mbd 2 -flags +aic+cbp+mv0+mv4 -trellis 1 -ac 1 >> > -ar 22050 -ab 56kb >> >> I would personally use libx264 if I wanted FLV output. >> >> > whn i convert it the video is converted to flv format but the length >> > of the video is not shown.. i.e the total time of the video is not >> > shown its showing 00.00..... video can be played but it cannot be >> > forwarded it. if we try to do it then the video starts playing from >> > the first.... this is the prob.... how to solve it??? if any details >> > required im ready to share... thanks for any replies...... >> >> The length of the video is not displayed properly in what? You didn't >> answer this last time, so I'll assume you are refering to a flash >> player such as JW Player, flowplayer, etc. >> >> Try using flvtool++, flvmeta, flvtool2, or an equivalent tool on your >> FFmpeg output: >> >> flvtool++ input.flv output.flv >> >> > On Fri, Apr 22, 2011 at 10:41 PM, Lou wrote: >> > >> > > On Fri, 22 Apr 2011 15:00:59 +0530 >> > > ranjini raguu wrote: >> > > >> > > > Im using ffmpeg for converting audio n video.... the conversion is >> > > > working fine... but whn i add some presets values the conversion >> > > > is >> > > >> > > What preset values? >> > > >> > > > missing somewhere i.e after conversion the length of the video is >> > > > not displayed so tat the video cannot be seeked... how to fix >> > > > this so tat >> > > >> > > The length of the video is not displayed properly in ______? >> > > FFmpeg, a web flash player, Quicktime, WMP, mediainfo? >> > > >> > > Seeked with what? >> > > >> > > > the length should be identifies after conversion..... >> > > > wat can we do for it? >> > > >> > > First of all, you could show us your FFmpeg command and the complete >> > > console output. Help us help you by providing any useful and >> > > necessary information when asking questions. You should provide >> > > enough detailed information that anyone else could attempt to >> > > duplicate any issues that you are encountering. >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From smouli60 at gmail.com Tue Apr 26 13:39:29 2011 From: smouli60 at gmail.com (mouli s) Date: Tue, 26 Apr 2011 17:09:29 +0530 Subject: [FFmpeg-user] conversion from wmv to avi is not working Message-ID: hello im trying to convert .wmv to .avi im using the following parameters -f avi -r 29.97 -vcodec libxvid -vtag XVID -s 640x480 -aspect 4:3 -maxrate 1800kb -b 1500kb -qmin 3 -qmax 5 -bufsize 4096 -mbd 2 -bf 2 -flags +4mv -trellis -aic -cmp 2 -subcmp 2 -g 300 -acodec libmp3lame -ar 48000 -ab 128kb -ac 2 but the conversion is not proper it chops, video is not clear how to solve this problem????? FYI - this conversion is for *XviD FullScreen* but when i pass this ordinary parameter alone -ab 128 -b 1200 it works fine conversion is good wats wrong with the first parameter i mentioned.... thanks for any replies. From smouli60 at gmail.com Tue Apr 26 13:39:29 2011 From: smouli60 at gmail.com (mouli s) Date: Tue, 26 Apr 2011 17:09:29 +0530 Subject: [FFmpeg-user] conversion from wmv to avi is not working Message-ID: hello im trying to convert .wmv to .avi im using the following parameters -f avi -r 29.97 -vcodec libxvid -vtag XVID -s 640x480 -aspect 4:3 -maxrate 1800kb -b 1500kb -qmin 3 -qmax 5 -bufsize 4096 -mbd 2 -bf 2 -flags +4mv -trellis -aic -cmp 2 -subcmp 2 -g 300 -acodec libmp3lame -ar 48000 -ab 128kb -ac 2 but the conversion is not proper it chops, video is not clear how to solve this problem????? FYI - this conversion is for *XviD FullScreen* but when i pass this ordinary parameter alone -ab 128 -b 1200 it works fine conversion is good wats wrong with the first parameter i mentioned.... thanks for any replies. From jogga at bitfield.se Tue Apr 26 13:47:44 2011 From: jogga at bitfield.se (=?iso-8859-1?Q?Isaksson_J=F6rgen?=) Date: Tue, 26 Apr 2011 13:47:44 +0200 Subject: [FFmpeg-user] Problems converting image only movies Message-ID: <91475F42-3002-4AE0-AA18-C463E72CA669@bitfield.se> Hi List, I'm trying to overlay a QuickTime movie that consists of PNG images on top of another movie using FFMEG. I have noticed that FFMPEG doesn't seem to handle such an image only movie correctly. To illustrate the problem I have made a simple example movie that has one image sample for each second with a timecode written to it. When I try to convert this movie and change the framerate FFMPEG seems misinterpret samples randomly. Download the sample movie from http://dl.dropbox.com/u/659266/timecodes.mov and try with the following simple command line: ffmpeg -y -i timecodes.mov -r 25 timecodes_copy.mov You will see that FFMPEG will shorten the first sample to 1 frame instead of 25? The sample that should appear on frame 26 will instead appear on frame 2? What could be wrong? Did I find a bug? Looking forward to a reply / Joergen From news at mschlosser.de Tue Apr 26 12:59:52 2011 From: news at mschlosser.de (news.gmane.org) Date: Tue, 26 Apr 2011 12:59:52 +0200 Subject: [FFmpeg-user] Use libavfilter to automatically create cutpoint files (e.g. for ProjectX) Message-ID: <20110426125952.00003df6@unknown> Hi, I have lots of recordings of TV series and I'm too lazy to cut them all manually in ProjectX. Since the first and the last frame (intro and credits) are always the same and there are no commercial breaks, I had the idea of writing a filter plugin which basically takes one or more picture files and compares them to each frame in a video file. The position of the best matching video frame(s) is written to a (text) file. This way I want to automatically create the cutpoint files for ProjectX. The plugin already works and reliably detects the start and the end of my series but neither the pts (presentation time stamp) nor the frame number given by the AVFilterBufferRef structure produce valid cutpoints in ProjectX. I'm new to most of the video internals and I unfortunately have no idea how to solve this. It would be great if someone could take a look at the code and point me to the right direction or tell me how to improve things. -- Many thanks in advance! Martin Schlosser -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: vf_detectpicture.c URL: From radioconfusion at gmail.com Tue Apr 26 14:32:46 2011 From: radioconfusion at gmail.com (Konfuusio Tonttu) Date: Tue, 26 Apr 2011 15:32:46 +0300 Subject: [FFmpeg-user] Remuxed TS-file doesn't have any frame rate Message-ID: Hi I have a problem when remuxing AVC+AAC in MP4 to MPEG-TS. The TS-file doesn't have any frame rate. I have tested with "-r" option but no luck. VLC plays broken file fine, but MPlayer doesn't show video without "-fps" option. Mediainfo doesn't show any frame rate to TS-file. How I can solve this problem? Thanks. Best Regards, Jussi $ ffmpeg -i test1.mp4 -acodec copy -vcodec copy -vbsf h264_mp4toannexb -f mpegts test1.ts ffmpeg version git-N-29386-g3788a3c, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 26 2011 14:50:11 with gcc 4.4.3 configuration: --prefix=/usr --enable-avfilter --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static --enable-libmp3lame libavutil 51. 0. 0 / 51. 0. 0 libavcodec 53. 1. 0 / 53. 1. 0 libavformat 53. 0. 3 / 53. 0. 3 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 0. 0 / 2. 0. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 1 codec frame rate differs from container frame rate: 1200.00 (1200/1) -> 25.00 (25/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test1.mp4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2008-10-08 18:07:47 Duration: 00:02:40.89, start: 0.000000, bitrate: 6434 kb/s Stream #0.0(eng): Audio: aac, 32000 Hz, stereo, s16, 93 kb/s Metadata: creation_time : 2008-10-08 18:07:47 Stream #0.1(eng): Video: h264 (Main), yuv420p, 1280x544, 3058 kb/s, 25 fps, 25 tbr, 600 tbn, 1200 tbc Metadata: creation_time : 2008-10-08 18:07:47 Stream #0.2(eng): Data: rtp / 0x20707472, 3179 kb/s Metadata: creation_time : 2008-10-08 18:07:47 Stream #0.3(eng): Data: rtp / 0x20707472, 96 kb/s Metadata: creation_time : 2008-10-08 18:07:47 File 'test1.ts' already exists. Overwrite ? [y/N] y [mpegts @ 0x10e4380] muxrate VBR, pcr every 60 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'test1.ts': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2008-10-08 18:07:47 encoder : Lavf53.0.3 Stream #0.0(eng): Video: avc1 / 0x31637661, yuv420p, 1280x544, q=2-31, 3058 kb/s, 90k tbn, 600 tbc Metadata: creation_time : 2008-10-08 18:07:47 Stream #0.1(eng): Audio: aac, 32000 Hz, stereo, 93 kb/s Metadata: creation_time : 2008-10-08 18:07:47 Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 4022 fps= 0 q=-1.0 Lsize= 67273kB time=160.80 bitrate=3427.2kbits/s video:60056kB audio:1832kB global headers:0kB muxing overhead 8.700379% $ mplayer test1.ts MPlayer SVN-r31097-4.4.3 (C) 2000-2010 MPlayer Team Playing test1.ts. TS file format detected. VIDEO H264(pid=256) AUDIO AAC(pid=257) NO SUBS (yet)! PROGRAM N. 1 FPS not specified in the header or invalid, use the -fps option. ========================================================================== Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding) Unsupported LATM configuration: 3 programs/ 56 subframes, 5 layers, allstreams: 0 Unsupported LATM configuration: 10 programs/ 25 subframes, 7 layers, allstreams: 0 FAAD: compressed input bitrate missing, assuming 128kbit/s! AUDIO: 32000 Hz, 2 ch, s16le, 128.0 kbit/12.50% (ratio: 16000->128000) Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio)) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... A: 2.8 (02.8) of 4305.4 ( 1:11:45.4) 0.8% MPlayer interrupted by signal 2 in module: play_audio A: 2.8 (02.8) of 4305.4 ( 1:11:45.4) 0.8% Exiting... (Quit) $ mediainfo test1.mp4 | grep -i "frame rate " Frame rate : 25.000 fps $ mediainfo test1.ts | grep -i "frame rate" $ From radioconfusion at gmail.com Tue Apr 26 14:36:44 2011 From: radioconfusion at gmail.com (Konfuusio Tonttu) Date: Tue, 26 Apr 2011 15:36:44 +0300 Subject: [FFmpeg-user] Remuxed TS-file doesn't have any frame rate In-Reply-To: References: Message-ID: Hi I have a problem when remuxing AVC+AAC in MP4 to MPEG-TS. The TS-file doesn't have any frame rate. I have tested with "-r" option but no luck. VLC plays broken file fine, but MPlayer doesn't show video without "-fps" option. Mediainfo doesn't show any frame rate to TS-file. How I can solve this problem? Thanks. Best Regards, Jussi $ ffmpeg -i test1.mp4 -acodec copy -vcodec copy -vbsf h264_mp4toannexb -f mpegts test1.ts ffmpeg version git-N-29386-g3788a3c, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 26 2011 14:50:11 with gcc 4.4.3 configuration: --prefix=/usr --enable-avfilter --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static --enable-libmp3lame libavutil 51. 0. 0 / 51. 0. 0 libavcodec 53. 1. 0 / 53. 1. 0 libavformat 53. 0. 3 / 53. 0. 3 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 0. 0 / 2. 0. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 1 codec frame rate differs from container frame rate: 1200.00 (1200/1) -> 25.00 (25/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test1.mp4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2008-10-08 18:07:47 Duration: 00:02:40.89, start: 0.000000, bitrate: 6434 kb/s Stream #0.0(eng): Audio: aac, 32000 Hz, stereo, s16, 93 kb/s Metadata: creation_time : 2008-10-08 18:07:47 Stream #0.1(eng): Video: h264 (Main), yuv420p, 1280x544, 3058 kb/s, 25 fps, 25 tbr, 600 tbn, 1200 tbc Metadata: creation_time : 2008-10-08 18:07:47 Stream #0.2(eng): Data: rtp / 0x20707472, 3179 kb/s Metadata: creation_time : 2008-10-08 18:07:47 Stream #0.3(eng): Data: rtp / 0x20707472, 96 kb/s Metadata: creation_time : 2008-10-08 18:07:47 File 'test1.ts' already exists. Overwrite ? [y/N] y [mpegts @ 0x10e4380] muxrate VBR, pcr every 60 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'test1.ts': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2008-10-08 18:07:47 encoder : Lavf53.0.3 Stream #0.0(eng): Video: avc1 / 0x31637661, yuv420p, 1280x544, q=2-31, 3058 kb/s, 90k tbn, 600 tbc Metadata: creation_time : 2008-10-08 18:07:47 Stream #0.1(eng): Audio: aac, 32000 Hz, stereo, 93 kb/s Metadata: creation_time : 2008-10-08 18:07:47 Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 4022 fps= 0 q=-1.0 Lsize= 67273kB time=160.80 bitrate=3427.2kbits/s video:60056kB audio:1832kB global headers:0kB muxing overhead 8.700379% $ mplayer test1.ts MPlayer SVN-r31097-4.4.3 (C) 2000-2010 MPlayer Team Playing test1.ts. TS file format detected. VIDEO H264(pid=256) AUDIO AAC(pid=257) NO SUBS (yet)! PROGRAM N. 1 FPS not specified in the header or invalid, use the -fps option. ========================================================================== Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding) Unsupported LATM configuration: 3 programs/ 56 subframes, 5 layers, allstreams: 0 Unsupported LATM configuration: 10 programs/ 25 subframes, 7 layers, allstreams: 0 FAAD: compressed input bitrate missing, assuming 128kbit/s! AUDIO: 32000 Hz, 2 ch, s16le, 128.0 kbit/12.50% (ratio: 16000->128000) Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio)) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory AO: [alsa] 48000Hz 2ch s16le (2 bytes per sample) Video: no video Starting playback... A: 2.8 (02.8) of 4305.4 ( 1:11:45.4) 0.8% MPlayer interrupted by signal 2 in module: play_audio A: 2.8 (02.8) of 4305.4 ( 1:11:45.4) 0.8% Exiting... (Quit) $ mediainfo test1.mp4 | grep -i "frame rate " Frame rate : 25.000 fps $ mediainfo test1.ts | grep -i "frame rate" $ From shacky83 at gmail.com Tue Apr 26 16:52:27 2011 From: shacky83 at gmail.com (shacky) Date: Tue, 26 Apr 2011 16:52:27 +0200 Subject: [FFmpeg-user] ffmpeg multi threading Message-ID: Hi. I'm using ffmpeg on my Windows machine to convert some videos from TS to FLV. I have a quadcore processor and I'm making some benchmark to have the conversion process as fast as possible. I'm using the binary win32 static version of ffmpeg from http://ffmpeg.arrozcru.org/autobuilds/. The command I use to convert from TS to FLV is the following: C:\Record\tv>c:\utils\ffmpeg-r26400\bin\ffmpeg.exe -i myfile.ts -y -f flv -s 320x240 -b 464k -ac 2 -ar 22050 -ab 48k myfile.flv But if I try to add the -threads 2 or -threads 4 parameter I get this error: Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height so I'm thinking that ffmpeg is not using all available cores of my CPU... Could you help me please? Thank you very much! Bye. From kaijun61 at yahoo.com Tue Apr 26 18:27:14 2011 From: kaijun61 at yahoo.com (Kaijun Tang) Date: Tue, 26 Apr 2011 09:27:14 -0700 (PDT) Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? Message-ID: <723878.64608.qm@web110209.mail.gq1.yahoo.com> I have tried -vbsf h264_mp4toannexb or -f rawvideo. These options don't work. Thanks, From ben at cardinalpeak.com Tue Apr 26 18:48:08 2011 From: ben at cardinalpeak.com (Ben Mesander) Date: Tue, 26 Apr 2011 09:48:08 -0700 (PDT) Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? In-Reply-To: <723878.64608.qm@web110209.mail.gq1.yahoo.com> References: <723878.64608.qm@web110209.mail.gq1.yahoo.com> Message-ID: <930517.989.qm@web161305.mail.bf1.yahoo.com> Here is how I did it: http://www.cardinalpeak.com/blog/?p=878 ffmpeg.exe -i Old\ Faithful.mp4 -vcodec copy -vbsf h264_mp4toannexb -an of.h264 I have also done this with .mov containers, which I assume is what you mean by a quicktime file. Regards, Ben ________________________________ From: Kaijun Tang To: ffmpeg-user at ffmpeg.org Sent: Tue, April 26, 2011 10:27:14 AM Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? I have tried -vbsf h264_mp4toannexb or -f rawvideo. These options don't work. Thanks, _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user 2i From pb at das-werkstatt.com Tue Apr 26 18:57:34 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 26 Apr 2011 18:57:34 +0200 Subject: [FFmpeg-user] YUV video to RGB image export color conversion issue In-Reply-To: <4D706231.3090809@gmail.com> References: <20110302092654.20701yye4sxg8hn2@webmail.tuwien.ac.at> <4D6E1FBC.6010501@gmail.com> <20110302153405.203369741w23ulwd@webmail.tuwien.ac.at> <20110302164503.167037ds6idenjsv@webmail.tuwien.ac.at> <4D706231.3090809@gmail.com> Message-ID: <4DB6F97E.9060405@das-werkstatt.com> Hello, I just wanted to ask the current status of this issue, since I saw that the problem still exists in the current git version. I have tested the patch suggested by Baptiste Coudurier and I'm using it extensively without problems. Is there anything I could do to help getting that fix into trunk? :) e.g. re-post this issue on ffmpeg-devel? Thanks and regards, Peter B. Baptiste Coudurier wrote: > On 03/02/2011 07:45 AM, Peter B. wrote: >> Quoting ?????? ???????? : >>>> I can imagine that this affects all image extractions from yuv422p >>>> material, or not? >>>> >>> You have a synthetic images (row color interlaced), I don't think >>> that on >>> regular images result will be so different... >> >> For regular video footage, you're right. >> In the end, it depends on the content, since for example generated >> overlays (e.g. text, effects, etc.) often produces completely different >> colors between fields. >> >> However, you're right: for regular images, it's bearable. ;) > > Try the attached patch. > > ------------------------------------------------------------------------ > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at mplayerhq.hu > https://lists.mplayerhq.hu/mailman/listinfo/ffmpeg-user From pb at das-werkstatt.com Tue Apr 26 18:58:57 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 26 Apr 2011 18:58:57 +0200 Subject: [FFmpeg-user] YUV video to RGB image export color conversion issue In-Reply-To: <4D706231.3090809@gmail.com> References: <20110302092654.20701yye4sxg8hn2@webmail.tuwien.ac.at> <4D6E1FBC.6010501@gmail.com> <20110302153405.203369741w23ulwd@webmail.tuwien.ac.at> <20110302164503.167037ds6idenjsv@webmail.tuwien.ac.at> <4D706231.3090809@gmail.com> Message-ID: <4DB6F9D1.7040603@das-werkstatt.com> Baptiste Coudurier wrote: > Try the attached patch. oops. sorry for top-posting previously. Peter B. From kaijun61 at yahoo.com Tue Apr 26 19:00:52 2011 From: kaijun61 at yahoo.com (Kaijun Tang) Date: Tue, 26 Apr 2011 10:00:52 -0700 (PDT) Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? In-Reply-To: <930517.989.qm@web161305.mail.bf1.yahoo.com> References: <723878.64608.qm@web110209.mail.gq1.yahoo.com> <930517.989.qm@web161305.mail.bf1.yahoo.com> Message-ID: <169522.26481.qm@web110215.mail.gq1.yahoo.com> I have tried that for both .mov and video only. But I got Unable to find a suitable output format for 'of.264', here is my ffmpeg info built on Dec 9 2010 15:18:49 with gcc 4.4.5 20101112 (Red Hat 4.4.5-2) configuration: libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.11. 0 / 0.11. 0 Kind Regards Kai ________________________________ From: Ben Mesander To: FFmpeg user questions and RTFMs Sent: Tue, 26 April, 2011 17:48:08 Subject: Re: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? Here is how I did it: http://www.cardinalpeak.com/blog/?p=878 ffmpeg.exe -i Old\ Faithful.mp4 -vcodec copy -vbsf h264_mp4toannexb -an of.h264 I have also done this with .mov containers, which I assume is what you mean by a quicktime file. Regards, Ben ________________________________ From: Kaijun Tang To: ffmpeg-user at ffmpeg.org Sent: Tue, April 26, 2011 10:27:14 AM Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? I have tried -vbsf h264_mp4toannexb or -f rawvideo. These options don't work. Thanks, _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user 2i _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From jogga at bitfield.se Tue Apr 26 19:06:59 2011 From: jogga at bitfield.se (=?iso-8859-1?Q?Isaksson_J=F6rgen?=) Date: Tue, 26 Apr 2011 19:06:59 +0200 Subject: [FFmpeg-user] Problems converting image only movies Message-ID: <88BD0891-A444-4E47-A4A7-4BC68BA6707F@bitfield.se> Hi List, I'm trying to overlay a QuickTime movie that consists of PNG images on top of another movie using FFMEG. I have noticed that FFMPEG doesn't seem to handle such an image only movie correctly. To illustrate the problem I have made a simple example movie that has one image sample for each second with a timecode written to it. When I try to convert this movie and change the framerate FFMPEG seems misinterpret samples randomly. Download the sample movie from http://dl.dropbox.com/u/659266/timecodes.mov and try with the following simple command line: ffmpeg -y -i timecodes.mov -r 25 timecodes_copy.mov You will see that FFMPEG will shorten the first sample to 1 frame instead of 25? The sample that should appear on frame 26 will instead appear on frame 2? What could be wrong? Did I find a bug? Looking forward to a reply / Joergen From ben at cardinalpeak.com Tue Apr 26 19:27:34 2011 From: ben at cardinalpeak.com (Ben Mesander) Date: Tue, 26 Apr 2011 10:27:34 -0700 (PDT) Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? In-Reply-To: <169522.26481.qm@web110215.mail.gq1.yahoo.com> References: <723878.64608.qm@web110209.mail.gq1.yahoo.com> <930517.989.qm@web161305.mail.bf1.yahoo.com> <169522.26481.qm@web110215.mail.gq1.yahoo.com> Message-ID: <719233.53612.qm@web161302.mail.bf1.yahoo.com> Hi Kai, Mine is a bit newer, maybe that is the issue? $ ./ffmpeg Hyper fast Audio and Video encoder usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfi le}... FFmpeg version git-N-29131-gf4bc923, Copyright (c) 2000-2011 the FFmpeg develope rs built on Apr 15 2011 13:17:53 with gcc 4.5.2 configuration: --enable-gpl --enable-version3 --enable-runtime-cpudetect --ena ble-memalign-hack --enable-avisynth --enable-bzlib --enable-frei0r --enable-libo pencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm -- enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enabl e-libx264 --enable-libxavs --enable-libxvid --enable-zlib --cross-prefix=i686-w6 4-mingw32- --target-os=mingw32 --arch=x86_32 --extra-cflags=-I/home/kyle/softwar e/ffmpeg/external-libraries/win32/include --extra-ldflags=-L/home/kyle/software/ ffmpeg/external-libraries/win32/lib --pkg-config=pkg-config libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.119. 1 / 52.119. 1 libavformat 52.108. 0 / 52.108. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 78. 0 / 1. 78. 0 libswscale 0. 13. 0 / 0. 13. 0 Use -h to get full help or, even better, run 'man ffmpeg' Regards, Ben ________________________________ From: Kaijun Tang To: FFmpeg user questions and RTFMs Sent: Tue, April 26, 2011 11:00:52 AM Subject: Re: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? I have tried that for both .mov and video only. But I got Unable to find a suitable output format for 'of.264', here is my ffmpeg info built on Dec 9 2010 15:18:49 with gcc 4.4.5 20101112 (Red Hat 4.4.5-2) configuration: libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.11. 0 / 0.11. 0 Kind Regards Kai ________________________________ From: Ben Mesander To: FFmpeg user questions and RTFMs Sent: Tue, 26 April, 2011 17:48:08 Subject: Re: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? Here is how I did it: http://www.cardinalpeak.com/blog/?p=878 ffmpeg.exe -i Old\ Faithful.mp4 -vcodec copy -vbsf h264_mp4toannexb -an of.h264 I have also done this with .mov containers, which I assume is what you mean by a quicktime file. Regards, Ben ________________________________ From: Kaijun Tang To: ffmpeg-user at ffmpeg.org Sent: Tue, April 26, 2011 10:27:14 AM Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? I have tried -vbsf h264_mp4toannexb or -f rawvideo. These options don't work. Thanks, _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user 2i _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From kaijun61 at yahoo.com Tue Apr 26 19:53:02 2011 From: kaijun61 at yahoo.com (Kaijun Tang) Date: Tue, 26 Apr 2011 10:53:02 -0700 (PDT) Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? In-Reply-To: <169522.26481.qm@web110215.mail.gq1.yahoo.com> References: <723878.64608.qm@web110209.mail.gq1.yahoo.com> <930517.989.qm@web161305.mail.bf1.yahoo.com> <169522.26481.qm@web110215.mail.gq1.yahoo.com> Message-ID: <440962.74673.qm@web110203.mail.gq1.yahoo.com> I have looked at the source code, it provides like ftyp header for packetized h264. I have not seen nal, sps and pps headers for writing streams. Kind Regards Kai ________________________________ From: Kaijun Tang To: FFmpeg user questions and RTFMs Sent: Tue, 26 April, 2011 18:00:52 Subject: Re: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? I have tried that for both .mov and video only. But I got Unable to find a suitable output format for 'of.264', here is my ffmpeg info built on Dec 9 2010 15:18:49 with gcc 4.4.5 20101112 (Red Hat 4.4.5-2) configuration: libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.11. 0 / 0.11. 0 Kind Regards Kai ________________________________ From: Ben Mesander To: FFmpeg user questions and RTFMs Sent: Tue, 26 April, 2011 17:48:08 Subject: Re: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? Here is how I did it: http://www.cardinalpeak.com/blog/?p=878 ffmpeg.exe -i Old\ Faithful.mp4 -vcodec copy -vbsf h264_mp4toannexb -an of.h264 I have also done this with .mov containers, which I assume is what you mean by a quicktime file. Regards, Ben ________________________________ From: Kaijun Tang To: ffmpeg-user at ffmpeg.org Sent: Tue, April 26, 2011 10:27:14 AM Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? I have tried -vbsf h264_mp4toannexb or -f rawvideo. These options don't work. Thanks, _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user 2i _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From lou at fakeoutdoorsman.com Tue Apr 26 20:14:53 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Tue, 26 Apr 2011 10:14:53 -0800 Subject: [FFmpeg-user] ffmpeg multi threading In-Reply-To: References: Message-ID: <20110426101453.6a933c3c@lrcd.com> On Tue, 26 Apr 2011 16:52:27 +0200 shacky wrote: > Hi. > I'm using ffmpeg on my Windows machine to convert some videos from TS > to FLV. I have a quadcore processor and I'm making some benchmark to > have the conversion process as fast as possible. > > I'm using the binary win32 static version of ffmpeg from > http://ffmpeg.arrozcru.org/autobuilds/. Newer Windows builds are located here: http://hawkeye.arrozcru.org/ > The command I use to convert from TS to FLV is the following: > > C:\Record\tv>c:\utils\ffmpeg-r26400\bin\ffmpeg.exe -i myfile.ts -y -f > flv -s 320x240 -b 464k -ac 2 -ar 22050 -ab 48k myfile.flv > > But if I try to add the -threads 2 or -threads 4 parameter I get this > error: > > Error while opening encoder for output stream #0.0 - maybe incorrect > parameters such as bit_rate, rate, width or height > > so I'm thinking that ffmpeg is not using all available cores of my > CPU... > > Could you help me please? > > Thank you very much! > Bye. flv (the encoder, not the container) does not support multithreaded encoding. Use libx264 if you want to output to flv and have multithreaded encoding. Example: ffmpeg -i input -vcodec libx264 -preset medium -crf 24 -threads 0 \ -vf scale=320:-1 -acodec libmp3lame -ar 44100 -aq 5 output.flv From kaijun61 at yahoo.com Tue Apr 26 22:58:41 2011 From: kaijun61 at yahoo.com (Kaijun Tang) Date: Tue, 26 Apr 2011 13:58:41 -0700 (PDT) Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? In-Reply-To: <719233.53612.qm@web161302.mail.bf1.yahoo.com> References: <723878.64608.qm@web110209.mail.gq1.yahoo.com> <930517.989.qm@web161305.mail.bf1.yahoo.com> <169522.26481.qm@web110215.mail.gq1.yahoo.com> <719233.53612.qm@web161302.mail.bf1.yahoo.com> Message-ID: <411348.96335.qm@web110201.mail.gq1.yahoo.com> Hi Ben, I just downloaded latest ffmpeg. It still doesn't work if I use .264 as output file. But if I use .mov file, it is packetized stream. Do you have the first start code of 0x0000000109 or 0x000000000109 at the beginning of the stream? Kind Regards Kaijun Tang ________________________________ From: Ben Mesander To: FFmpeg user questions and RTFMs Sent: Tue, 26 April, 2011 18:27:34 Subject: Re: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? Hi Kai, Mine is a bit newer, maybe that is the issue? $ ./ffmpeg Hyper fast Audio and Video encoder usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfi le}... FFmpeg version git-N-29131-gf4bc923, Copyright (c) 2000-2011 the FFmpeg develope rs built on Apr 15 2011 13:17:53 with gcc 4.5.2 configuration: --enable-gpl --enable-version3 --enable-runtime-cpudetect --ena ble-memalign-hack --enable-avisynth --enable-bzlib --enable-frei0r --enable-libo pencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm -- enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enabl e-libx264 --enable-libxavs --enable-libxvid --enable-zlib --cross-prefix=i686-w6 4-mingw32- --target-os=mingw32 --arch=x86_32 --extra-cflags=-I/home/kyle/softwar e/ffmpeg/external-libraries/win32/include --extra-ldflags=-L/home/kyle/software/ ffmpeg/external-libraries/win32/lib --pkg-config=pkg-config libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.119. 1 / 52.119. 1 libavformat 52.108. 0 / 52.108. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 78. 0 / 1. 78. 0 libswscale 0. 13. 0 / 0. 13. 0 Use -h to get full help or, even better, run 'man ffmpeg' Regards, Ben ________________________________ From: Kaijun Tang To: FFmpeg user questions and RTFMs Sent: Tue, April 26, 2011 11:00:52 AM Subject: Re: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? I have tried that for both .mov and video only. But I got Unable to find a suitable output format for 'of.264', here is my ffmpeg info built on Dec 9 2010 15:18:49 with gcc 4.4.5 20101112 (Red Hat 4.4.5-2) configuration: libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.11. 0 / 0.11. 0 Kind Regards Kai ________________________________ From: Ben Mesander To: FFmpeg user questions and RTFMs Sent: Tue, 26 April, 2011 17:48:08 Subject: Re: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? Here is how I did it: http://www.cardinalpeak.com/blog/?p=878 ffmpeg.exe -i Old\ Faithful.mp4 -vcodec copy -vbsf h264_mp4toannexb -an of.h264 I have also done this with .mov containers, which I assume is what you mean by a quicktime file. Regards, Ben ________________________________ From: Kaijun Tang To: ffmpeg-user at ffmpeg.org Sent: Tue, April 26, 2011 10:27:14 AM Subject: [FFmpeg-user] how to extract h264 nal bitstream into a file from h264 quicktime file? I have tried -vbsf h264_mp4toannexb or -f rawvideo. These options don't work. Thanks, _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user 2i _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From research at the10thfloor.com Wed Apr 27 02:43:01 2011 From: research at the10thfloor.com (Shalom Levytam) Date: Tue, 26 Apr 2011 20:43:01 -0400 Subject: [FFmpeg-user] AAC Downsample Doesn't Work With -f mpegts Message-ID: Hi, I was wondering if someone could help me with downsampling aac audio when converting to a .ts file. Currently, I'm trying: ffmpeg -y -i mc497b98.mp4 -f mpegts -acodec libfaac -ar 22050 -ac 2 -ab 64k -vcodec libx264 ..... The converted file is missing audio. Stream #0.1[0x101](und): Audio: aac, 0 channels, s16 If I upsample (ie -ar 48000) or remove -ar entirely it works fine. If I leave the -ar but remove -f mpegts it also works fine. Any thoughts? Thanks, slevytam From smouli60 at gmail.com Wed Apr 27 06:30:13 2011 From: smouli60 at gmail.com (mouli s) Date: Wed, 27 Apr 2011 10:00:13 +0530 Subject: [FFmpeg-user] building ffmpeg with flvtool2 Message-ID: Hello all can anyone tell me how to take a build for ffmpeg with flvtool2 for windows..... any wiki for tat???? thanks for any replies... From smouli60 at gmail.com Wed Apr 27 06:30:13 2011 From: smouli60 at gmail.com (mouli s) Date: Wed, 27 Apr 2011 10:00:13 +0530 Subject: [FFmpeg-user] building ffmpeg with flvtool2 Message-ID: Hello all can anyone tell me how to take a build for ffmpeg with flvtool2 for windows..... any wiki for tat???? thanks for any replies... From jankarlitos at hotmail.com Wed Apr 27 08:30:14 2011 From: jankarlitos at hotmail.com (Jk .) Date: Wed, 27 Apr 2011 06:30:14 +0000 Subject: [FFmpeg-user] Enable HE-AACv2 Message-ID: Hello, I'm trying to add HE-AACv2 encoding support to FFmpeg through libaacplus. I'm using this patch, and the 2011-04-25 snapshot of FFmpeg in Fedora 14. But i'm getting this error when compiling: libavcodec/libavcodec.a(allcodecs.o): In function `avcodec_register_all' libavcodec/allcodecs.c:369: undefined reference to `ff_libaacplus_encoder' collect2: ld returned 1 exit status I will appreciate any help. Thanks. From shacky83 at gmail.com Wed Apr 27 08:43:31 2011 From: shacky83 at gmail.com (shacky) Date: Wed, 27 Apr 2011 08:43:31 +0200 Subject: [FFmpeg-user] ffmpeg multi threading In-Reply-To: <20110426101453.6a933c3c@lrcd.com> References: <20110426101453.6a933c3c@lrcd.com> Message-ID: Thank you very much for your answer. I've never used the libx264 codec, so I'm looking to its options. I tried your command but I got the "Unrecognized option 'preset'" error. I tried to remove the -presets parameter but I got this error: Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height Could you help me please? 2011/4/26 Lou : > On Tue, 26 Apr 2011 16:52:27 +0200 > shacky wrote: > >> Hi. >> I'm using ffmpeg on my Windows machine to convert some videos from TS >> to FLV. I have a quadcore processor and I'm making some benchmark to >> have the conversion process as fast as possible. >> >> I'm using the binary win32 static version of ffmpeg from >> http://ffmpeg.arrozcru.org/autobuilds/. > > Newer Windows builds are located here: > http://hawkeye.arrozcru.org/ > >> The command I use to convert from TS to FLV is the following: >> >> C:\Record\tv>c:\utils\ffmpeg-r26400\bin\ffmpeg.exe -i myfile.ts -y -f >> flv -s 320x240 -b 464k -ac 2 -ar 22050 -ab 48k myfile.flv >> >> But if I try to add the -threads 2 or -threads 4 parameter I get this >> error: >> >> Error while opening encoder for output stream #0.0 - maybe incorrect >> parameters such as bit_rate, rate, width or height >> >> so I'm thinking that ffmpeg is not using all available cores of my >> CPU... >> >> Could you help me please? >> >> Thank you very much! >> Bye. > > flv (the encoder, not the container) does not support multithreaded > encoding. > > Use libx264 if you want to output to flv and have multithreaded > encoding. Example: > > ffmpeg -i input -vcodec libx264 -preset medium -crf 24 -threads 0 \ > -vf scale=320:-1 -acodec libmp3lame -ar 44100 -aq 5 output.flv > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From h.reindl at thelounge.net Wed Apr 27 09:19:18 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Wed, 27 Apr 2011 09:19:18 +0200 Subject: [FFmpeg-user] building ffmpeg with flvtool2 In-Reply-To: References: Message-ID: <4DB7C376.70501@thelounge.net> flvtool2 has nothing to do with ffmpeg it is a standalone application Am 27.04.2011 06:30, schrieb mouli s: > Hello all can anyone tell me how to take a build for ffmpeg with flvtool2 > for windows..... any wiki for tat???? thanks for any replies... > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From shacky83 at gmail.com Wed Apr 27 09:39:58 2011 From: shacky83 at gmail.com (shacky) Date: Wed, 27 Apr 2011 09:39:58 +0200 Subject: [FFmpeg-user] ffmpeg multi threading In-Reply-To: References: <20110426101453.6a933c3c@lrcd.com> Message-ID: I solved the problem on the x264 preset. I made a benchmark between the x264 with -threads 0 option and the flv codec with the command I specified in my first message, but the flv codec is faster than the x264 one. How is it possible? 2011/4/27 shacky : > Thank you very much for your answer. > I've never used the libx264 codec, so I'm looking to its options. > I tried your command but I got the "Unrecognized option 'preset'" error. > I tried to remove the -presets parameter but I got this error: > > Stream mapping: > ?Stream #0.0 -> #0.0 > ?Stream #0.1 -> #0.1 > Error while opening encoder for output stream #0.0 - maybe incorrect parameters > such as bit_rate, rate, width or height > > Could you help me please? > > > 2011/4/26 Lou : >> On Tue, 26 Apr 2011 16:52:27 +0200 >> shacky wrote: >> >>> Hi. >>> I'm using ffmpeg on my Windows machine to convert some videos from TS >>> to FLV. I have a quadcore processor and I'm making some benchmark to >>> have the conversion process as fast as possible. >>> >>> I'm using the binary win32 static version of ffmpeg from >>> http://ffmpeg.arrozcru.org/autobuilds/. >> >> Newer Windows builds are located here: >> http://hawkeye.arrozcru.org/ >> >>> The command I use to convert from TS to FLV is the following: >>> >>> C:\Record\tv>c:\utils\ffmpeg-r26400\bin\ffmpeg.exe -i myfile.ts -y -f >>> flv -s 320x240 -b 464k -ac 2 -ar 22050 -ab 48k myfile.flv >>> >>> But if I try to add the -threads 2 or -threads 4 parameter I get this >>> error: >>> >>> Error while opening encoder for output stream #0.0 - maybe incorrect >>> parameters such as bit_rate, rate, width or height >>> >>> so I'm thinking that ffmpeg is not using all available cores of my >>> CPU... >>> >>> Could you help me please? >>> >>> Thank you very much! >>> Bye. >> >> flv (the encoder, not the container) does not support multithreaded >> encoding. >> >> Use libx264 if you want to output to flv and have multithreaded >> encoding. Example: >> >> ffmpeg -i input -vcodec libx264 -preset medium -crf 24 -threads 0 \ >> -vf scale=320:-1 -acodec libmp3lame -ar 44100 -aq 5 output.flv >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > From ranjiniraguu at gmail.com Wed Apr 27 12:23:24 2011 From: ranjiniraguu at gmail.com (ranjini raguu) Date: Wed, 27 Apr 2011 15:53:24 +0530 Subject: [FFmpeg-user] Conversion fails if file is empty Message-ID: Hello im using ffmpeg for conversion....Conversion fails if file is empty... so wat i did is found the size of the file and if its 0 then stoped the execution with popup but now some files is empty but it shows 15bytes how its possible??? n how to handle this??? From ranjiniraguu at gmail.com Wed Apr 27 12:23:24 2011 From: ranjiniraguu at gmail.com (ranjini raguu) Date: Wed, 27 Apr 2011 15:53:24 +0530 Subject: [FFmpeg-user] Conversion fails if file is empty Message-ID: Hello im using ffmpeg for conversion....Conversion fails if file is empty... so wat i did is found the size of the file and if its 0 then stoped the execution with popup but now some files is empty but it shows 15bytes how its possible??? n how to handle this??? From smouli60 at gmail.com Wed Apr 27 12:27:46 2011 From: smouli60 at gmail.com (mouli s) Date: Wed, 27 Apr 2011 15:57:46 +0530 Subject: [FFmpeg-user] building ffmpeg with flvtool2 In-Reply-To: <4DB7C376.70501@thelounge.net> References: <4DB7C376.70501@thelounge.net> Message-ID: Hello Rein, Thanks for ur reply but how to solve my prob???? my prob is exact to the prob which is posted in this link http://stackoverflow.com/questions/1532703/video-length-missing-in-flv-converted-by-ffmpeg-php tats y i thought of using flvtool2 to my application.... but dnt find steps how to use it.. :( tats y posted in forum...thanks for any help.. On Wed, Apr 27, 2011 at 12:49 PM, Reindl Harald wrote: > flvtool2 has nothing to do with ffmpeg > it is a standalone application > > Am 27.04.2011 06:30, schrieb mouli s: > > Hello all can anyone tell me how to take a build for ffmpeg with flvtool2 > > for windows..... any wiki for tat???? thanks for any replies... > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From h.reindl at thelounge.net Wed Apr 27 12:37:07 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Wed, 27 Apr 2011 12:37:07 +0200 Subject: [FFmpeg-user] building ffmpeg with flvtool2 In-Reply-To: References: <4DB7C376.70501@thelounge.net> Message-ID: <4DB7F1D3.9010401@thelounge.net> Where is the problem? run "flvtool2 -U file.flv" the same way you call ffmpeg after conversion Am 27.04.2011 12:27, schrieb mouli s: > Hello Rein, > Thanks for ur reply but how to solve my prob???? my prob is > exact to the prob which is posted in this link > > http://stackoverflow.com/questions/1532703/video-length-missing-in-flv-converted-by-ffmpeg-php > > tats y i thought of using flvtool2 to my application.... but dnt find steps > how to use it.. :( tats y posted in forum...thanks for any help.. > > On Wed, Apr 27, 2011 at 12:49 PM, Reindl Harald wrote: > >> flvtool2 has nothing to do with ffmpeg >> it is a standalone application >> >> Am 27.04.2011 06:30, schrieb mouli s: >>> Hello all can anyone tell me how to take a build for ffmpeg with flvtool2 >>> for windows..... any wiki for tat???? thanks for any replies... >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> >> > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- Mit besten Gr??en, Reindl Harald the lounge interactive design GmbH A-1060 Vienna, Hofm?hlgasse 17 CTO / software-development / cms-solutions p: +43 (1) 595 3999 33, m: +43 (676) 40 221 40 icq: 154546673, http://www.thelounge.net/ http://www.thelounge.net/signature.asc.what.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From jogga at me.com Wed Apr 27 13:39:07 2011 From: jogga at me.com (=?iso-8859-1?Q?J=F6rgen_Isaksson?=) Date: Wed, 27 Apr 2011 13:39:07 +0200 Subject: [FFmpeg-user] Problems converting image only movies Message-ID: <8ACEE508-18FC-459A-A25F-A78A82B3BEC1@me.com> Hi List, I'm trying to overlay a QuickTime movie that consists of PNG images on top of another movie using FFMEG. I have noticed that FFMPEG doesn't seem to handle such an image only movie correctly. To illustrate the problem I have made a simple example movie that has one image sample for each second with a timecode written to it. When I try to convert this movie and change the framerate FFMPEG seems misinterpret samples randomly. Download the sample movie from http://dl.dropbox.com/u/659266/timecodes.mov and try with the following simple command line: ffmpeg -y -i timecodes.mov -r 25 timecodes_copy.mov You will see that FFMPEG will shorten the first sample to 1 frame instead of 25? The sample that should appear on frame 26 will instead appear on frame 2? What could be wrong? Did I find a bug? Looking forward to a reply / Joergen From frederiksunne at gmail.com Wed Apr 27 14:05:25 2011 From: frederiksunne at gmail.com (Frederik Dam Sunne) Date: Wed, 27 Apr 2011 14:05:25 +0200 Subject: [FFmpeg-user] ffmpeg multi threading In-Reply-To: References: <20110426101453.6a933c3c@lrcd.com> Message-ID: On Wed, Apr 27, 2011 at 09:39, shacky wrote: > I solved the problem on the x264 preset. > I made a benchmark between the x264 with -threads 0 option and the flv > codec with the command I specified in my first message, but the flv > codec is faster than the x264 one. > > How is it possible? > As I remember it, the default codec used for .flv files is H.263, which is a less CPU intensive codec with a much less efficient compression ratio compared to H.264. Frederik From coniophora at gmail.com Wed Apr 27 16:40:20 2011 From: coniophora at gmail.com (Jim Worrall) Date: Wed, 27 Apr 2011 08:40:20 -0600 Subject: [FFmpeg-user] keeping two subtitle streams Message-ID: I'm converting a .mkv to .m2ts and having subtitle trouble. The source .mkv has both English and Chinese subtitles and I would like to keep both. ffmpeg seems to recognize both in the source file, but is only making one subtitle stream in the output. I tried -scodec copy 2x but didn't make a difference. Doesn't seem to be a mapping issue since there is only one subtitle stream in the output. Would appreciate a pointer on how to specify that I want two subtitle streams. Thank you! ffmpeg -y -i 01.mkv -f mpegts \ > -vcodec libx264 -s hd720 -r 23.976 -b 1800K -vbsf h264_mp4toannexb \ > -acodec ac3 -ab 640K -ar 48K \ > -scodec copy \ > 01.m2ts FFmpeg version UNKNOWN, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 21 2011 18:00:17 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --arch=x86_64 --enable-runtime-cpudetect libavutil 51. 0. 0 / 51. 0. 0 libavcodec 53. 0. 0 / 53. 0. 0 libavformat 53. 0. 0 / 53. 0. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 0. 0 / 2. 0. 0 libswscale 0. 13. 0 / 0. 13. 0 [matroska,webm @ 0x101812400] max_analyze_duration reached [matroska,webm @ 0x101812400] Estimating duration from bitrate, this may be inaccurate Seems stream 0 codec frame rate differs from container frame rate: 47.95 (5000000/104271) -> 23.98 (24000/1001) Input #0, matroska,webm, from '01.mkv': Duration: 00:42:43.58, start: 0.000000, bitrate: N/A Stream #0.0(eng): Video: h264 (High), yuv420p, 1024x576, PAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default) Stream #0.1(eng): Audio: aac, 48000 Hz, 5.1, s16 (default) Stream #0.2(chi): Subtitle: [0][0][0][0] / 0x0000 (default) Stream #0.3(eng): Subtitle: [0][0][0][0] / 0x0000 [buffer @ 0x101306040] w:1024 h:576 pixfmt:yuv420p [scale @ 0x10132feb0] w:1024 h:576 fmt:yuv420p -> w:1280 h:720 fmt:yuv420p flags:0x4 [libx264 @ 0x1019a2600] Default settings detected, using medium profile [libx264 @ 0x1019a2600] using SAR=1/1 [libx264 @ 0x1019a2600] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x1019a2600] profile High, level 3.1 [mpegts @ 0x10185a200] muxrate VBR, pcr every 2 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to '01.m2ts': Metadata: encoder : Lavf53.0.0 Stream #0.0(eng): Video: libx264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], q=2-31, 1800 kb/s, 90k tbn, 23.98 tbc (default) Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 640 kb/s (default) Stream #0.2(eng): Subtitle: [0][0][0][0] / 0x0000 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Stream #0.3 -> #0.2 From poliva at wtelecom.es Wed Apr 27 18:48:24 2011 From: poliva at wtelecom.es (Pedro Oliva) Date: Wed, 27 Apr 2011 18:48:24 +0200 Subject: [FFmpeg-user] RTSP recording stopped if there is no motion Message-ID: Hi, I'm recording rtsp streams from IP cameras using this command: ffmpeg -v 3 -i rtsp://my_url -f mp4 -b 400k -r 24 -vcodec libx264 -ab 126k -ar 48k -acodec libfaac -vpre fast my_file.mp4 It seems like if there is no motion taking place in the video stream, ffmpeg stops recording. Is it possible to avoid this behavior? Thanks From lou at fakeoutdoorsman.com Wed Apr 27 19:10:52 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Wed, 27 Apr 2011 09:10:52 -0800 Subject: [FFmpeg-user] ffmpeg multi threading In-Reply-To: References: <20110426101453.6a933c3c@lrcd.com> Message-ID: <20110427091052.53f448b5@lrcd.com> On Wed, 27 Apr 2011 09:39:58 +0200 shacky wrote: > I solved the problem on the x264 preset. > I made a benchmark between the x264 with -threads 0 option and the flv > codec with the command I specified in my first message, but the flv > codec is faster than the x264 one. > > How is it possible? You could use a faster preset. Show your FFmpeg command and the complete terminal output. > 2011/4/27 shacky : > > Thank you very much for your answer. > > I've never used the libx264 codec, so I'm looking to its options. > > I tried your command but I got the "Unrecognized option 'preset'" > > error. I tried to remove the -presets parameter but I got this > > error: > > > > Stream mapping: > > ?Stream #0.0 -> #0.0 > > ?Stream #0.1 -> #0.1 > > Error while opening encoder for output stream #0.0 - maybe > > incorrect parameters such as bit_rate, rate, width or height > > > > Could you help me please? > > > > > > 2011/4/26 Lou : > >> On Tue, 26 Apr 2011 16:52:27 +0200 > >> shacky wrote: > >> > >>> Hi. > >>> I'm using ffmpeg on my Windows machine to convert some videos > >>> from TS to FLV. I have a quadcore processor and I'm making some > >>> benchmark to have the conversion process as fast as possible. > >>> > >>> I'm using the binary win32 static version of ffmpeg from > >>> http://ffmpeg.arrozcru.org/autobuilds/. > >> > >> Newer Windows builds are located here: > >> http://hawkeye.arrozcru.org/ > >> > >>> The command I use to convert from TS to FLV is the following: > >>> > >>> C:\Record\tv>c:\utils\ffmpeg-r26400\bin\ffmpeg.exe -i myfile.ts > >>> -y -f flv -s 320x240 -b 464k -ac 2 -ar 22050 -ab 48k myfile.flv > >>> > >>> But if I try to add the -threads 2 or -threads 4 parameter I get > >>> this error: > >>> > >>> Error while opening encoder for output stream #0.0 - maybe > >>> incorrect parameters such as bit_rate, rate, width or height > >>> > >>> so I'm thinking that ffmpeg is not using all available cores of my > >>> CPU... > >>> > >>> Could you help me please? > >>> > >>> Thank you very much! > >>> Bye. > >> > >> flv (the encoder, not the container) does not support multithreaded > >> encoding. > >> > >> Use libx264 if you want to output to flv and have multithreaded > >> encoding. Example: > >> > >> ffmpeg -i input -vcodec libx264 -preset medium -crf 24 -threads 0 \ > >> -vf scale=320:-1 -acodec libmp3lame -ar 44100 -aq 5 output.flv From koxaniy at mail.ru Wed Apr 27 22:40:40 2011 From: koxaniy at mail.ru (=?utf-8?Q?=D0=9F=D0=B0=D0=BF=D0=B0_=D0=9B=D1=8E=D0=B1=D1=8F=D1=89=D0=B8=D0=B9?=) Date: Thu, 28 Apr 2011 00:40:40 +0400 Subject: [FFmpeg-user] Problem with FFmpeg git-a304071 Message-ID: I use FFmpeg git-a304071 for convert dv.avi to dv???.mov Command line: -f mov? -vcodec dvvideo -r 25 -aspect 5:4 -s 720x576 -pix_fmt yuv411p -vtag dvpp? -acodec pcm_s16be -ar 48000 -ac 2 ??? FFmpeg version git-N-29181-ga304071, Copyright (c) 2000-2011 the FFmpeg developers ? built on Apr 18 2011 21:40:04 with gcc 4.5.2 ? configuration: --enable-gpl --enable-version3 --enable-runtime-cpudetect --enable-memalign-hack --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32 --arch=x86_64 --extra-cflags=-I/home/kyle/software/ffmpeg/external-libraries/win64/include --extra-ldflags=-L/home/kyle/software/ffmpeg/external-libraries/win64/lib --pkg-config=pkg-config ? libavutil??? 50. 40. 1 / 50. 40. 1 ? libavcodec?? 52.120. 0 / 52.120. 0 ? libavformat? 52.108. 0 / 52.108. 0 ? libavdevice? 52.? 4. 0 / 52.? 4. 0 ? libavfilter?? 1. 79. 0 /? 1. 79. 0 ? libswscale??? 0. 13. 0 /? 0. 13. 0 Input #0, avi, from 'D:\DVINPUT.avi': ? Duration: 00:01:58.96, start: 0.000000, bitrate: 30351 kb/s ??? Stream #0.0: Video: dvvideo, yuv420p, 720x576 [PAR 16:15 DAR 4:3], 25 fps, 25 tbr, 25 tbn, 25 tbc ??? Stream #0.1: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s [buffer @ 0000000001958A90] w:720 h:576 pixfmt:yuv420p [ffsink @ 00000000040833F0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 00000000040836A0] w:720 h:576 fmt:yuv420p -> w:720 h:576 fmt:yuv411p flags:0x4 Output #0, mov, to 'D:\DVOUTPUT.mov': ? Metadata: ??? encoder???????? : Lavf52.108.0 ??? Stream #0.0: Video: dvvideo, yuv411p, 720x576 [PAR 1:1 DAR 5:4], q=2-31, 200 kb/s, 25 tbn, 25 tbc ??? Stream #0.1: Audio: pcm_s16be, 48000 Hz, 2 channels, s16, 1536 kb/s Stream mapping: ? Stream #0.0 -> #0.0 ? Stream #0.1 -> #0.1 Press [q] to stop encoding frame=?? 65 fps=? 0 q=0.0 size=??? 9480kB time=2.56 bitrate=30336.1kbits/s??? frame=? 128 fps=126 q=0.0 size=?? 14220kB time=3.84 bitrate=30336.1kbits/s??? ...... But i have bad picture after encoding , see in? http://img822.imageshack.us/i/problemffmpeg.jpg/??? What is it ?? Its a bad build ? Or ?????? From baptiste.coudurier at gmail.com Wed Apr 27 23:26:39 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Wed, 27 Apr 2011 14:26:39 -0700 Subject: [FFmpeg-user] Problem with FFmpeg git-a304071 In-Reply-To: References: Message-ID: <4DB88A0F.8030509@ffmpeg.org> Hi, On 04/27/2011 01:40 PM, ???? ??????? wrote: > I use FFmpeg git-a304071 for convert dv.avi to dv?.mov > > Command line: > -f mov? -vcodec dvvideo -r 25 -aspect 5:4 -s 720x576 -pix_fmt yuv411p -vtag dvpp? -acodec pcm_s16be -ar 48000 -ac 2 ? > > FFmpeg version git-N-29181-ga304071, Copyright (c) 2000-2011 the FFmpeg developers > ? built on Apr 18 2011 21:40:04 with gcc 4.5.2 > ? configuration: --enable-gpl --enable-version3 --enable-runtime-cpudetect --enable-memalign-hack --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib --cross-prefix=x86_64-w64-mingw32- --target-os=mingw32 --arch=x86_64 --extra-cflags=-I/home/kyle/software/ffmpeg/external-libraries/win64/include --extra-ldflags=-L/home/kyle/software/ffmpeg/external-libraries/win64/lib --pkg-config=pkg-config > ? libavutil??? 50. 40. 1 / 50. 40. 1 > ? libavcodec?? 52.120. 0 / 52.120. 0 > ? libavformat? 52.108. 0 / 52.108. 0 > ? libavdevice? 52.? 4. 0 / 52.? 4. 0 > ? libavfilter?? 1. 79. 0 /? 1. 79. 0 > > ? libswscale??? 0. 13. 0 /? 0. 13. 0 > Input #0, avi, from 'D:\DVINPUT.avi': > ? Duration: 00:01:58.96, start: 0.000000, bitrate: 30351 kb/s > ??? Stream #0.0: Video: dvvideo, yuv420p, 720x576 [PAR 16:15 DAR 4:3], 25 fps, 25 tbr, 25 tbn, 25 tbc > ??? Stream #0.1: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s > [buffer @ 0000000001958A90] w:720 h:576 pixfmt:yuv420p > [ffsink @ 00000000040833F0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' > [scale @ 00000000040836A0] w:720 h:576 fmt:yuv420p -> w:720 h:576 fmt:yuv411p flags:0x4 > Output #0, mov, to 'D:\DVOUTPUT.mov': > ? Metadata: > ??? encoder???????? : Lavf52.108.0 > ??? Stream #0.0: Video: dvvideo, yuv411p, 720x576 [PAR 1:1 DAR 5:4], q=2-31, 200 kb/s, 25 tbn, 25 tbc > ??? Stream #0.1: Audio: pcm_s16be, 48000 Hz, 2 channels, s16, 1536 kb/s > Stream mapping: > ? Stream #0.0 -> #0.0 > ? Stream #0.1 -> #0.1 > Press [q] to stop encoding > frame=?? 65 fps=? 0 q=0.0 size=??? 9480kB time=2.56 bitrate=30336.1kbits/s??? > frame=? 128 fps=126 q=0.0 size=?? 14220kB time=3.84 bitrate=30336.1kbits/s? > ...... > > > But i have bad picture after encoding , see in? http://img822.imageshack.us/i/problemffmpeg.jpg/? > What is it ?? Its a bad build ? Or ???? Can you please upload your source file somewhere ? Thanks -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From koxaniy at mail.ru Thu Apr 28 00:54:23 2011 From: koxaniy at mail.ru (=?utf-8?Q?=D0=9F=D0=B0=D0=BF=D0=B0_=D0=9B=D1=8E=D0=B1=D1=8F=D1=89=D0=B8=D0=B9?=) Date: Thu, 28 Apr 2011 02:54:23 +0400 Subject: [FFmpeg-user] =?utf-8?q?Problem_with_FFmpeg_git-a304071?= In-Reply-To: <4DB88A0F.8030509@ffmpeg.org> References: <4DB88A0F.8030509@ffmpeg.org> Message-ID: Hi. Take please file from my camcoder , and create dv.mov from him with http://hawkeye.arrozcru.org/builds/win64/static/ffmpeg-git-a304071-win64-static.7z http://www.filefactory.com/file/cbf224e/n/00021.MTS I have Win7 64bit system i use command line : -f mov -vcodec dvvideo -r 25 -aspect 5:4 -s 720x576 -pix_fmt yuv411p -vtag dvpp -acodec pcm_s16be -ar 48000 -ac 2 -y and received this : http://img713.imageshack.us/i/ffmpegbug.jpg/ Thank you for your attention Wed, 27 Apr 2011 14:26:39 -0700 ?????? ?? Baptiste Coudurier : > Hi, > Can you please upload your source file somewhere ? > Thanks From baptiste at ffmpeg.org Thu Apr 28 02:04:40 2011 From: baptiste at ffmpeg.org (Baptiste Coudurier) Date: Wed, 27 Apr 2011 17:04:40 -0700 Subject: [FFmpeg-user] Problem with FFmpeg git-a304071 In-Reply-To: References: <4DB88A0F.8030509@ffmpeg.org> Message-ID: <4DB8AF18.8000100@ffmpeg.org> On 04/27/2011 03:54 PM, ???? ??????? wrote: > Hi. > > Take please file from my camcoder , and create dv.mov from him with http://hawkeye.arrozcru.org/builds/win64/static/ffmpeg-git-a304071-win64-static.7z > > http://www.filefactory.com/file/cbf224e/n/00021.MTS > > I have Win7 64bit system > i use command line : -f mov -vcodec dvvideo -r 25 -aspect 5:4 -s 720x576 -pix_fmt yuv411p -vtag dvpp -acodec pcm_s16be -ar 48000 -ac 2 -y > and received this : > > http://img713.imageshack.us/i/ffmpegbug.jpg/ > Can you please upload the avi file ? Thanks. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From grant.smith at envent-tech.com Wed Apr 27 06:09:56 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Wed, 27 Apr 2011 00:09:56 -0400 Subject: [FFmpeg-user] Fwd: Installation Problems In-Reply-To: References: Message-ID: I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial found @ http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 I get to the portion regarding the actual install of ffmpeg and I get the following error: ERROR: libmp3lame >= 3.98.3 not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. I don't know of a good way to get my config.log from my Linux box to my Windows box through PuTTY. Any advice on this problem would be GREATLY appreciated. Thank you, Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 From 2suyashk at gmail.com Wed Apr 27 14:14:41 2011 From: 2suyashk at gmail.com (Suyash K) Date: Wed, 27 Apr 2011 17:44:41 +0530 Subject: [FFmpeg-user] Latest compilation of ffmpeg win32 Message-ID: I would like to get the recent win32 compilation of ffmpeg (git 26th April, 2011). Pl help. Regards, libcr2 From hardik.sharma22 at yahoo.com Thu Apr 28 04:09:37 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Wed, 27 Apr 2011 19:09:37 -0700 (PDT) Subject: [FFmpeg-user] Problem in encoding yuv raw video file to h.264 Message-ID: <96619.46189.qm@web46204.mail.sp1.yahoo.com> Hi, I am trying to encode yuv video file to h.264 on ubuntu 10.10 but unable to get it done. My input video is silent.yuv cif format. I am using following commands and getting following errors- $ffmpeg -i silent_cif.yuv -acodec libfaac -aq 100 -vcodec libx264 -b 256k -preset slow -crf 22 -threads 0 output.mp4FFmpeg version git-N-29265-g3eb15d9, Copyright (c) 2000-2011 the FFmpeg developers ? built on Apr 21 2011 11:44:48 with gcc 4.4.5 ? configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab ? libavutil??? 51.? 0. 0 / 51.? 0. 0 ? libavcodec?? 53.? 0. 0 / 53.? 0. 0 ? libavformat? 53.? 0. 0 / 53.? 0. 0 ? libavdevice? 53.? 0. 0 / 53.? 0. 0 ? libavfilter?? 2.? 0. 0 /? 2.? 0. 0 ? libswscale??? 0. 13. 0 /? 0. 13. 0 ? libpostproc? 51.? 2. 0 / 51.? 2. 0 [IMGUTILS @ 0xbfa66b94] Picture size 0x0 is invalid [IMGUTILS @ 0xbfa66a94] Picture size 0x0 is invalid [rawvideo @ 0x9e8dbb0] Could not find codec parameters (Video: rawvideo, yuv420p) [rawvideo @ 0x9e8dbb0] Estimating duration from bitrate, this may be inaccurate Please help me with this issue. Thanks, Hardik Sharma From grant.smith at envent-tech.com Thu Apr 28 04:17:16 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Wed, 27 Apr 2011 22:17:16 -0400 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: Message-ID: I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial found @ http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 I get to the portion regarding the actual install of ffmpeg and I get the following error: ERROR: libmp3lame >= 3.98.3 not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. The link to my config.log file is http://pastebin.com/wsJYFe5e. It's set to expire in ~20 hours. Any advice on this problem would be GREATLY appreciated. Thank you, Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 From lou at fakeoutdoorsman.com Thu Apr 28 04:37:34 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Wed, 27 Apr 2011 18:37:34 -0800 Subject: [FFmpeg-user] Latest compilation of ffmpeg win32 In-Reply-To: References: Message-ID: <20110427183734.7d031e6f@lrcd.com> On Wed, 27 Apr 2011 17:44:41 +0530 Suyash K <2suyashk at gmail.com> wrote: > I would like to get the recent win32 compilation of ffmpeg (git 26th > April, 2011). > > Pl help. > > Regards, > > libcr2 Did you see this? HawkEye's FFmpeg Windows Builds http://hawkeye.arrozcru.org/ From kshawkeye at gmail.com Thu Apr 28 06:49:51 2011 From: kshawkeye at gmail.com (Kyle Schwarz) Date: Thu, 28 Apr 2011 00:49:51 -0400 Subject: [FFmpeg-user] Latest compilation of ffmpeg win32 In-Reply-To: <20110427183734.7d031e6f@lrcd.com> References: <20110427183734.7d031e6f@lrcd.com> Message-ID: On Wed, Apr 27, 2011 at 10:37 PM, Lou wrote: > On Wed, 27 Apr 2011 17:44:41 +0530 > Suyash K <2suyashk at gmail.com> wrote: > >> I would like to get the recent win32 compilation of ffmpeg (git 26th >> April, 2011). >> >> Pl help. >> >> Regards, >> >> libcr2 > > Did you see this? > > HawkEye's FFmpeg Windows Builds > http://hawkeye.arrozcru.org/ > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > The builds currently haven't been updated in a while due to computer issues. I explained this in detail here: http://ffmpeg.arrozcru.org/forum/viewtopic.php?f=15&t=1691&start=15#p6144 Builds should be back Tuesdays or Wednesday of next week, hang in there. From verem at m1stereo.tv Thu Apr 28 07:23:38 2011 From: verem at m1stereo.tv (Maksym Veremeyenko) Date: Thu, 28 Apr 2011 08:23:38 +0300 Subject: [FFmpeg-user] Problem with FFmpeg git-a304071 In-Reply-To: References: <4DB88A0F.8030509@ffmpeg.org> Message-ID: <4DB8F9DA.9020900@m1stereo.tv> 28.04.11 01:54, ???? ??????? ???????(??): > Hi. > > Take please file from my camcoder , and create dv.mov from him with > http://hawkeye.arrozcru.org/builds/win64/static/ffmpeg-git-a304071-win64-static.7z i think the problem is win64 build, try 32-bit build. i tested against latest win32 ffmpeg and libav builds (http://research.m1stereo.tv/ffmpeg/snapshots/) and found no problem with a picture... -- ________________________________________ Maksym Veremeyenko From saurabhg84 at gmail.com Thu Apr 28 07:33:31 2011 From: saurabhg84 at gmail.com (Saurabh Gandhi) Date: Thu, 28 Apr 2011 11:03:31 +0530 Subject: [FFmpeg-user] FFMPEG windows issues (H.264 RTSP stream decoding) In-Reply-To: References: Message-ID: *I am resending this mail since earlier I wasn't subscribed to the mailing lists and hence the mail got rejected. -- Regards, Saurabh Gandhi * *On Thu, Apr 28, 2011 at 10:42 AM, Saurabh Gandhi wrote: * > > *Hello all,* > * > * > *I am a windows user (unfortunately) and am trying to use FFMPEG for > decoding H.264 based RTSP stream. I have downloaded the latest windows build > available on the FFMPEG website (http://hawkeye.arrozcru.org/) and am > trying to use ffplay.exe.* > * > * > *The problem is that the output is very much pixelated / distorted when > there is some motion in the frame. The same stream when viewed using VLC > works perfectly fine. Another catch is that when I tried viewing the same > stream on ubuntu using FFMPEG itselg it works perfectly fine. What could be > the issue?* > * > * > *I tried running ffplay.exe on multiple hardware and software platforms > which include:* > > 1. *i3 (4GB RAM) with Windows 7* > 2. *core 2 duo (1GB RAM) with windows xp* > 3. *intel xeon (2GB RAM) with windows server 2003* > > *However the problem still persists. When on ubuntu, the problem does not > appear at all on all machines. I tried recompiling FFMPEG by self assuming > that it must be some issues related to the windows build supplied on the > FFMPEG site. However, even this fails to resolve the problem.* > * > * > *Are there any other windows users (read unfortunate) like me who have > faced / found a workaround for this issue.* > *Awaiting your reply. Thank you.* > * > * > *P.S.: I was wondering how no one has complained of this issue? Has no one > out here tried using the latest ffplay.exe from the windows build of ffmpeg > for decoding H.264 based RTSP stream?* > * > * > *Another apprehension that I had is how come vlc player is able to get > over the issue on windows when vlc itself uses ffmpeg (upto the best of my > knowledge).* > * > * > *Troubled windows user,* > *Saurabh Gandhi* > * > -- > Regards, > Saurabh Gandhi > > * > From dashing.meng at gmail.com Thu Apr 28 09:30:42 2011 From: dashing.meng at gmail.com (littlebat) Date: Thu, 28 Apr 2011 15:30:42 +0800 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: Message-ID: <20110428153042.0cb6a1be.dashing.meng@gmail.com> On Wed, 27 Apr 2011 22:17:16 -0400 Grant Smith wrote: > I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial found > @ http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 > > I get to the portion regarding the actual install of ffmpeg and I get > the following error: > > ERROR: libmp3lame >= 3.98.3 not found > > If you think configure made a mistake, make sure you are using the > latest version from Git. If the latest version fails, report the > problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on > irc.freenode.net. Include the log file "config.log" produced by > configure as this will help solving the problem. > > The link to my config.log file is http://pastebin.com/wsJYFe5e. It's > set to expire in ~20 hours. > > Any advice on this problem would be GREATLY appreciated. > > Thank you, > > Grant Smith > A+, Network+, MCP x 2, BSIT/VC, MIS > > Phone: +1.317.560.4457 > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user I have Compiled ffmpeg under Ubuntu 10.04 with the steps: 1, purge some package: sudo apt-get purge libavutil-dev libavutil50 libavutil-extra-50 \ libavcodec-dev libavcodec52 libavcodec-extra-52 libavformat-dev \ libavformat52 libavformat-extra-52 libavdevice-dev libavdevice52 \ libavdevice-extra-52 libavfilter-dev libavfilter1 libavfilter-extra-1 \ libswscale-dev libswscale0 libswscale-extra-0 libpostproc-dev \ libpostproc51 libpostproc-extra-51 gnome-desktop-environment ffmpeg 2, install some codecs dev packages and libsdl dev package: sudo apt-get install libx264-dev libxvidcore-dev libopencore-amrwb-dev \ libopencore-amrnb-dev libfaad-dev libfaac-dev libmp3lame-dev \ libtwolame-dev liba52-0.7.4-dev libcddb2-dev libcdaudio-dev \ libcdio-cdda-dev libvorbis-dev libopenjpeg-dev libsdl1.2-dev "git clone git://git.videolan.org/x264.git" and compiled libx264 from source 3, configure and install ./configure --enable-nonfree --enable-gpl --enable-version3 \ --enable-shared --enable-postproc --enable-libmp3lame \ --enable-libopenjpeg --enable-libvorbis --enable-libopencore-amrnb \ --enable-libopencore-amrwb --enable-libxvid --enable-libx264 \ --enable-libfaac --enable-pthreads From dashing.meng at gmail.com Thu Apr 28 09:38:30 2011 From: dashing.meng at gmail.com (littlebat) Date: Thu, 28 Apr 2011 15:38:30 +0800 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: Message-ID: <20110428153830.9eb8f055.dashing.meng@gmail.com> On Wed, 27 Apr 2011 22:17:16 -0400 Grant Smith wrote: > I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial found > @ http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 > > I get to the portion regarding the actual install of ffmpeg and I get > the following error: > > ERROR: libmp3lame >= 3.98.3 not found > > If you think configure made a mistake, make sure you are using the > latest version from Git. If the latest version fails, report the > problem to the ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on > irc.freenode.net. Include the log file "config.log" produced by > configure as this will help solving the problem. > > The link to my config.log file is http://pastebin.com/wsJYFe5e. It's > set to expire in ~20 hours. > > Any advice on this problem would be GREATLY appreciated. > > Thank you, > > Grant Smith > A+, Network+, MCP x 2, BSIT/VC, MIS > > Phone: +1.317.560.4457 > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user I have Compiled ffmpeg under Ubuntu 10.10 with the steps: 1, purge some package: sudo apt-get purge libavutil-dev libavutil50 libavutil-extra-50 \ libavcodec-dev libavcodec52 libavcodec-extra-52 libavformat-dev \ libavformat52 libavformat-extra-52 libavdevice-dev libavdevice52 \ libavdevice-extra-52 libavfilter-dev libavfilter1 libavfilter-extra-1 \ libswscale-dev libswscale0 libswscale-extra-0 libpostproc-dev \ libpostproc51 libpostproc-extra-51 gnome-desktop-environment ffmpeg note: purge gnome-desktop-environment will remove totem, gnash, sound-juicer, etc.. If you don't purge them, maybe found such as "ffmpeg: relocation error: /opt/ffmpeg20110404/lib/libavfilter.so.1: symbol av_expr_free, version LIBAVUTIL_50 not defined in file libavutil.so.50 with link time reference" when you run ffmpeg after compiled. 2, install some codecs dev packages and libsdl dev package: sudo apt-get install libx264-dev libxvidcore-dev libopencore-amrwb-dev \ libopencore-amrnb-dev libfaad-dev libfaac-dev libmp3lame-dev \ libtwolame-dev liba52-0.7.4-dev libcddb2-dev libcdaudio-dev \ libcdio-cdda-dev libvorbis-dev libopenjpeg-dev libsdl1.2-dev "git clone git://git.videolan.org/x264.git" and compiled libx264 from source 3, configure and install ./configure --enable-nonfree --enable-gpl --enable-version3 \ --enable-shared --enable-postproc --enable-libmp3lame \ --enable-libopenjpeg --enable-libvorbis --enable-libopencore-amrnb \ --enable-libopencore-amrwb --enable-libxvid --enable-libx264 \ --enable-libfaac --enable-pthreads From james.darnley at gmail.com Thu Apr 28 10:08:40 2011 From: james.darnley at gmail.com (James Darnley) Date: Thu, 28 Apr 2011 10:08:40 +0200 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: Message-ID: On 28/04/2011, Grant Smith wrote: > I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial found @ > http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 > > I get to the portion regarding the actual install of ffmpeg and I get the > following error: > > ERROR: libmp3lame >= 3.98.3 not found > > If you think configure made a mistake, make sure you are using the latest > version from Git. If the latest version fails, report the problem to the > ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. > Include the log file "config.log" produced by configure as this will help > solving the problem. > > The link to my config.log file is http://pastebin.com/wsJYFe5e. It's set to > expire in ~20 hours. > > Any advice on this problem would be GREATLY appreciated. Install lame 3.98.3 or newer. From james.darnley at gmail.com Thu Apr 28 10:20:44 2011 From: james.darnley at gmail.com (James Darnley) Date: Thu, 28 Apr 2011 10:20:44 +0200 Subject: [FFmpeg-user] Problem in encoding yuv raw video file to h.264 In-Reply-To: <96619.46189.qm@web46204.mail.sp1.yahoo.com> References: <96619.46189.qm@web46204.mail.sp1.yahoo.com> Message-ID: On 28/04/2011, Hardik Sharma wrote: > $ffmpeg -i silent_cif.yuv -acodec libfaac -aq 100 -vcodec libx264 -b 256k -preset slow -crf 22 -threads 0 output.mp4 How is ffmpeg supposed to know anything about your input file? You need to give the frame size, pixel format, frame rate. As a side note. You can't use both bitrate and crf rate control when encoding. From grant.smith at envent-tech.com Thu Apr 28 12:43:32 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Thu, 28 Apr 2011 06:43:32 -0400 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: Message-ID: I have already installed lame 3.98.4.1 as per the referenced tutorial. Thanks, Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 On Thu, Apr 28, 2011 at 4:08 AM, James Darnley wrote: > On 28/04/2011, Grant Smith wrote: > > I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial found @ > > http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 > > > > I get to the portion regarding the actual install of ffmpeg and I get the > > following error: > > > > ERROR: libmp3lame >= 3.98.3 not found > > > > If you think configure made a mistake, make sure you are using the latest > > version from Git. If the latest version fails, report the problem to the > > ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. > > Include the log file "config.log" produced by configure as this will help > > solving the problem. > > > > The link to my config.log file is http://pastebin.com/wsJYFe5e. It's set > to > > expire in ~20 hours. > > > > Any advice on this problem would be GREATLY appreciated. > > Install lame 3.98.3 or newer. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From saurabhg84 at gmail.com Thu Apr 28 07:12:44 2011 From: saurabhg84 at gmail.com (Saurabh Gandhi) Date: Thu, 28 Apr 2011 10:42:44 +0530 Subject: [FFmpeg-user] FFMPEG windows issues (H.264 RTSP stream decoding) Message-ID: Hello all, I am a windows user (unfortunately) and am trying to use FFMPEG for decoding H.264 based RTSP stream. I have downloaded the latest windows build available on the FFMPEG website (http://hawkeye.arrozcru.org/) and am trying to use ffplay.exe. The problem is that the output is very much pixelated / distorted when there is some motion in the frame. The same stream when viewed using VLC works perfectly fine. Another catch is that when I tried viewing the same stream on ubuntu using FFMPEG itselg it works perfectly fine. What could be the issue? I tried running ffplay.exe on multiple hardware and software platforms which include: 1. i3 (4GB RAM) with Windows 7 2. core 2 duo (1GB RAM) with windows xp 3. intel xeon (2GB RAM) with windows server 2003 However the problem still persists. When on ubuntu, the problem does not appear at all on all machines. I tried recompiling FFMPEG by self assuming that it must be some issues related to the windows build supplied on the FFMPEG site. However, even this fails to resolve the problem. Are there any other windows users (read unfortunate) like me who have faced / found a workaround for this issue. Awaiting your reply. Thank you. P.S.: I was wondering how no one has complained of this issue? Has no one out here tried using the latest ffplay.exe from the windows build of ffmpeg for decoding H.264 based RTSP stream? Another apprehension that I had is how come vlc player is able to get over the issue on windows when vlc itself uses ffmpeg (upto the best of my knowledge). Troubled windows user, Saurabh Gandhi -- Regards, Saurabh Gandhi From h.reindl at thelounge.net Thu Apr 28 12:52:48 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 28 Apr 2011 12:52:48 +0200 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: Message-ID: <4DB94700.40602@thelounge.net> You need the devel-package! Am 28.04.2011 12:43, schrieb Grant Smith: > I have already installed lame 3.98.4.1 as per the referenced tutorial. > > Thanks, > > Grant Smith > A+, Network+, MCP x 2, BSIT/VC, MIS > > Phone: +1.317.560.4457 > > > On Thu, Apr 28, 2011 at 4:08 AM, James Darnley wrote: > >> On 28/04/2011, Grant Smith wrote: >>> I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial found @ >>> http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 >>> >>> I get to the portion regarding the actual install of ffmpeg and I get the >>> following error: >>> >>> ERROR: libmp3lame >= 3.98.3 not found >>> >>> If you think configure made a mistake, make sure you are using the latest >>> version from Git. If the latest version fails, report the problem to the >>> ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. >>> Include the log file "config.log" produced by configure as this will help >>> solving the problem. >>> >>> The link to my config.log file is http://pastebin.com/wsJYFe5e. It's set >> to >>> expire in ~20 hours. >>> >>> Any advice on this problem would be GREATLY appreciated. >> >> Install lame 3.98.3 or newer. >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- Mit besten Gr??en, Reindl Harald the lounge interactive design GmbH A-1060 Vienna, Hofm?hlgasse 17 CTO / software-development / cms-solutions p: +43 (1) 595 3999 33, m: +43 (676) 40 221 40 icq: 154546673, http://www.thelounge.net/ http://www.thelounge.net/signature.asc.what.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From koxaniy at mail.ru Thu Apr 28 13:21:59 2011 From: koxaniy at mail.ru (=?utf-8?Q?=D0=9F=D0=B0=D0=BF=D0=B0_=D0=9B=D1=8E=D0=B1=D1=8F=D1=89=D0=B8=D0=B9?=) Date: Thu, 28 Apr 2011 15:21:59 +0400 Subject: [FFmpeg-user] =?utf-8?q?Problem_with_FFmpeg_git-a304071?= In-Reply-To: <4DB8AF18.8000100@ffmpeg.org> References: <4DB8AF18.8000100@ffmpeg.org> Message-ID: Yesterday he could not lay out, but today - here : http://www.filefactory.com/file/cbf40e0/n/DVINPUT.avi Wed, 27 Apr 2011 17:04:40 -0700 ?????? ?? Baptiste Coudurier : > Can you please upload the avi file ? > Thanks. > Baptiste COUDURIER From coniophora at gmail.com Thu Apr 28 15:47:16 2011 From: coniophora at gmail.com (Jim Worrall) Date: Thu, 28 Apr 2011 07:47:16 -0600 Subject: [FFmpeg-user] can't copy or transcode subtitles Message-ID: [Sorry for not posting to the related thread I started, but the mail list does not send me copies of my mails (though I have that selection checked in the options page), and I received no reply] I am having trouble getting subtitles to pass through in a file I am converting from mkv to m2ts. I can't tell what format the subtitles are in. The program VideoSpec just names them as "UTF-8". When I try to extract them using ffmpeg, I only get any success when extracting them into an srt file as follows ffmpeg -i 02.mkv -scodec copy -an -vn 02.srt -map 0.3 but the resulting file contains only the text lines, no time data. There are two subtitle streams and I would like to keep them both. I finally got the output file to have two subtitle streams by creating a second one with -newsubtitle and mapping, but ffmpeg gives the error below when it gets to the subtitles (doesn't matter whether the chinese or english one). Thanks for any reply. Here is the command and result: ffmpeg -y -i 02.mkv -f mpegts -threads -0 \ > -vcodec libx264 -s hd720 -r 23.976 -b 1800K -vbsf h264_mp4toannexb \ > -acodec ac3 -ab 640K -ar 48K \ > -scodec copy -scodec copy \ > 02.m2ts -newsubtitle -map 0.0 -map 0.1 -map 0.2 -map 0.3 FFmpeg version UNKNOWN, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 21 2011 18:00:17 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads --enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --arch=x86_64 --enable-runtime-cpudetect libavutil 51. 0. 0 / 51. 0. 0 libavcodec 53. 0. 0 / 53. 0. 0 libavformat 53. 0. 0 / 53. 0. 0 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 0. 0 / 2. 0. 0 libswscale 0. 13. 0 / 0. 13. 0 [matroska,webm @ 0x101812400] max_analyze_duration reached [matroska,webm @ 0x101812400] Estimating duration from bitrate, this may be inaccurate Seems stream 0 codec frame rate differs from container frame rate: 47.95 (5000000/104271) -> 23.98 (24000/1001) Input #0, matroska,webm, from '02.mkv': Duration: 00:42:43.58, start: 0.000000, bitrate: N/A Stream #0.0(eng): Video: h264 (High), yuv420p, 1024x576, PAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 1k tbn, 47.95 tbc (default) Stream #0.1(eng): Audio: aac, 48000 Hz, 5.1, s16 (default) Stream #0.2(chi): Subtitle: [0][0][0][0] / 0x0000 (default) Stream #0.3(eng): Subtitle: [0][0][0][0] / 0x0000 [buffer @ 0x101317490] w:1024 h:576 pixfmt:yuv420p [scale @ 0x1013061b0] w:1024 h:576 fmt:yuv420p -> w:1280 h:720 fmt:yuv420p flags:0x4 [libx264 @ 0x1019a2600] Default settings detected, using medium profile [libx264 @ 0x1019a2600] using SAR=1/1 [libx264 @ 0x1019a2600] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x1019a2600] profile High, level 3.1 Output #0, mpegts, to '02.m2ts': Stream #0.0(eng): Video: libx264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], q=2-31, 1800 kb/s, 90k tbn, 23.98 tbc (default) Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 640 kb/s (default) Stream #0.2(chi): Subtitle: srt (default) Stream #0.3(eng): Subtitle: [0][0][0][0] / 0x0000, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Stream #0.2 -> #0.2 Stream #0.3 -> #0.3 Error while opening encoder for output stream #0.2 - maybe incorrect parameters such as bit_rate, rate, width or height From grant.smith at envent-tech.com Thu Apr 28 16:32:40 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Thu, 28 Apr 2011 10:32:40 -0400 Subject: [FFmpeg-user] Installation Problems In-Reply-To: <4DB94700.40602@thelounge.net> References: <4DB94700.40602@thelounge.net> Message-ID: Reindl Harald,Could you please elaborate on this? I'm not as well-versed in Linux as I would like. How can I go about getting the devel package? Thank you!! Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 On Thu, Apr 28, 2011 at 6:52 AM, Reindl Harald wrote: > You need the devel-package! > > Am 28.04.2011 12:43, schrieb Grant Smith: > > I have already installed lame 3.98.4.1 as per the referenced tutorial. > > > > Thanks, > > > > Grant Smith > > A+, Network+, MCP x 2, BSIT/VC, MIS > > > > Phone: +1.317.560.4457 > > > > > > On Thu, Apr 28, 2011 at 4:08 AM, James Darnley >wrote: > > > >> On 28/04/2011, Grant Smith wrote: > >>> I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial found @ > >>> http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 > >>> > >>> I get to the portion regarding the actual install of ffmpeg and I get > the > >>> following error: > >>> > >>> ERROR: libmp3lame >= 3.98.3 not found > >>> > >>> If you think configure made a mistake, make sure you are using the > latest > >>> version from Git. If the latest version fails, report the problem to > the > >>> ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net > . > >>> Include the log file "config.log" produced by configure as this will > help > >>> solving the problem. > >>> > >>> The link to my config.log file is http://pastebin.com/wsJYFe5e. It's > set > >> to > >>> expire in ~20 hours. > >>> > >>> Any advice on this problem would be GREATLY appreciated. > >> > >> Install lame 3.98.3 or newer. > >> _______________________________________________ > >> ffmpeg-user mailing list > >> ffmpeg-user at ffmpeg.org > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > > Mit besten Gr??en, Reindl Harald > the lounge interactive design GmbH > A-1060 Vienna, Hofm?hlgasse 17 > CTO / software-development / cms-solutions > p: +43 (1) 595 3999 33, m: +43 (676) 40 221 40 > icq: 154546673, http://www.thelounge.net/ > > http://www.thelounge.net/signature.asc.what.htm > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From dave at avpreserve.com Thu Apr 28 17:25:23 2011 From: dave at avpreserve.com (Dave Rice) Date: Thu, 28 Apr 2011 11:25:23 -0400 Subject: [FFmpeg-user] getting non-AV stream from input to output Message-ID: Hi all, I have a QuickTime file with a caption track that ffmpeg identifies as Stream #0.1(eng): Data: c608 / 0x38303663, 4 kb/s I want to transcode the AV data to another QuickTime as an output but pass this track as-is to the output. Is there anyway to copy the caption track to the output from the input QuickTime file to the output QuickTime. I've been testing with -map and looking for other means without success. Full ffmpeg -i output http://pastebin.com/Y4mpkwvX Dave Rice From h.reindl at thelounge.net Thu Apr 28 18:47:10 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 28 Apr 2011 18:47:10 +0200 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: <4DB94700.40602@thelounge.net> Message-ID: <4DB99A0E.4010200@thelounge.net> I do not know what distribution you use you should get firm with you package-manager on fedora it is "yum install lame-devel" becuase there are seperate packages since most peopole do not need devel-files [root at buildserver:~]$ rpm -qa | grep lame lame-devel-3.98.4-5.fc13.rh.20110312.x86_64 lame-libs-3.98.4-5.fc13.rh.20110312.x86_64 lame-3.98.4-5.fc13.rh.20110312.x86_64 Am 28.04.2011 16:32, schrieb Grant Smith: > Reindl Harald,Could you please elaborate on this? I'm not as well-versed in > Linux as I would like. How can I go about getting the devel package? > > Thank you!! > > Grant Smith > A+, Network+, MCP x 2, BSIT/VC, MIS > > Phone: +1.317.560.4457 > > > On Thu, Apr 28, 2011 at 6:52 AM, Reindl Harald wrote: > >> You need the devel-package! >> >> Am 28.04.2011 12:43, schrieb Grant Smith: >>> I have already installed lame 3.98.4.1 as per the referenced tutorial. >>> >>> Thanks, >>> >>> Grant Smith >>> A+, Network+, MCP x 2, BSIT/VC, MIS >>> >>> Phone: +1.317.560.4457 >>> >>> >>> On Thu, Apr 28, 2011 at 4:08 AM, James Darnley >> wrote: >>> >>>> On 28/04/2011, Grant Smith wrote: >>>>> I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial found @ >>>>> http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 >>>>> >>>>> I get to the portion regarding the actual install of ffmpeg and I get >> the >>>>> following error: >>>>> >>>>> ERROR: libmp3lame >= 3.98.3 not found >>>>> >>>>> If you think configure made a mistake, make sure you are using the >> latest >>>>> version from Git. If the latest version fails, report the problem to >> the >>>>> ffmpeg-user at ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net >> . >>>>> Include the log file "config.log" produced by configure as this will >> help >>>>> solving the problem. >>>>> >>>>> The link to my config.log file is http://pastebin.com/wsJYFe5e. It's >> set >>>> to >>>>> expire in ~20 hours. >>>>> >>>>> Any advice on this problem would be GREATLY appreciated. >>>> >>>> Install lame 3.98.3 or newer. >>>> _______________________________________________ >>>> ffmpeg-user mailing list >>>> ffmpeg-user at ffmpeg.org >>>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From lou at fakeoutdoorsman.com Thu Apr 28 19:44:01 2011 From: lou at fakeoutdoorsman.com (Lou) Date: Thu, 28 Apr 2011 09:44:01 -0800 Subject: [FFmpeg-user] Installation Problems In-Reply-To: <4DB94700.40602@thelounge.net> References: <4DB94700.40602@thelounge.net> Message-ID: <20110428094401.0a237b1e@lrcd.com> On Thu, 28 Apr 2011 12:52:48 +0200 Reindl Harald wrote: > You need the devel-package! The -dev package from the Ubuntu repository will not help. libmp3lame-dev for Ubuntu Lucid is 3.98.2. FFmpeg requires >= 3.98.3. > Am 28.04.2011 12:43, schrieb Grant Smith: > > I have already installed lame 3.98.4.1 as per the referenced > > tutorial. > > > > Thanks, > > > > Grant Smith > > A+, Network+, MCP x 2, BSIT/VC, MIS > > > > Phone: +1.317.560.4457 > > > > > > On Thu, Apr 28, 2011 at 4:08 AM, James Darnley > > wrote: > > > >> On 28/04/2011, Grant Smith wrote: > >>> I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial > >>> found @ > >>> http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 > >>> > >>> I get to the portion regarding the actual install of ffmpeg and I > >>> get the following error: > >>> > >>> ERROR: libmp3lame >= 3.98.3 not found > >>> > >>> If you think configure made a mistake, make sure you are using > >>> the latest version from Git. If the latest version fails, report > >>> the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC > >>> #ffmpeg on irc.freenode.net. Include the log file "config.log" > >>> produced by configure as this will help solving the problem. > >>> > >>> The link to my config.log file is http://pastebin.com/wsJYFe5e. > >>> It's set > >> to > >>> expire in ~20 hours. > >>> > >>> Any advice on this problem would be GREATLY appreciated. > >> > >> Install lame 3.98.3 or newer. From h.reindl at thelounge.net Thu Apr 28 19:47:30 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Thu, 28 Apr 2011 19:47:30 +0200 Subject: [FFmpeg-user] Installation Problems In-Reply-To: <20110428094401.0a237b1e@lrcd.com> References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> Message-ID: <4DB9A832.2060502@thelounge.net> then learn how to build a ubunto-package as i did with fedora and build all needed pakacges the same way before or use ffmpeg from available repos - there is nothing between Am 28.04.2011 19:44, schrieb Lou: > On Thu, 28 Apr 2011 12:52:48 +0200 > Reindl Harald wrote: > >> You need the devel-package! > > The -dev package from the Ubuntu repository will not help. > libmp3lame-dev for Ubuntu Lucid is 3.98.2. FFmpeg requires >= 3.98.3. > >> Am 28.04.2011 12:43, schrieb Grant Smith: >>> I have already installed lame 3.98.4.1 as per the referenced >>> tutorial. >>> >>> Thanks, >>> >>> Grant Smith >>> A+, Network+, MCP x 2, BSIT/VC, MIS >>> >>> Phone: +1.317.560.4457 >>> >>> >>> On Thu, Apr 28, 2011 at 4:08 AM, James Darnley >>> wrote: >>> >>>> On 28/04/2011, Grant Smith wrote: >>>>> I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial >>>>> found @ >>>>> http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 >>>>> >>>>> I get to the portion regarding the actual install of ffmpeg and I >>>>> get the following error: >>>>> >>>>> ERROR: libmp3lame >= 3.98.3 not found >>>>> >>>>> If you think configure made a mistake, make sure you are using >>>>> the latest version from Git. If the latest version fails, report >>>>> the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC >>>>> #ffmpeg on irc.freenode.net. Include the log file "config.log" >>>>> produced by configure as this will help solving the problem. >>>>> >>>>> The link to my config.log file is http://pastebin.com/wsJYFe5e. >>>>> It's set >>>> to >>>>> expire in ~20 hours. >>>>> >>>>> Any advice on this problem would be GREATLY appreciated. >>>> >>>> Install lame 3.98.3 or newer. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- Mit besten Gr??en, Reindl Harald the lounge interactive design GmbH A-1060 Vienna, Hofm?hlgasse 17 CTO / software-development / cms-solutions p: +43 (1) 595 3999 33, m: +43 (676) 40 221 40 icq: 154546673, http://www.thelounge.net/ http://www.thelounge.net/signature.asc.what.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From grant.smith at envent-tech.com Thu Apr 28 20:00:20 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Thu, 28 Apr 2011 14:00:20 -0400 Subject: [FFmpeg-user] Installation Problems In-Reply-To: <4DB9A832.2060502@thelounge.net> References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> Message-ID: Reindl Harald,The tutorial has me building everything. The problem is that ffmpeg won't build. Please read my original email and look at my pastebin and you would see this. Thank you for your assistance. Yours, Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 On Thu, Apr 28, 2011 at 1:47 PM, Reindl Harald wrote: > then learn how to build a ubunto-package as i did with fedora > and build all needed pakacges the same way before or use ffmpeg > from available repos - there is nothing between > > Am 28.04.2011 19:44, schrieb Lou: > > On Thu, 28 Apr 2011 12:52:48 +0200 > > Reindl Harald wrote: > > > >> You need the devel-package! > > > > The -dev package from the Ubuntu repository will not help. > > libmp3lame-dev for Ubuntu Lucid is 3.98.2. FFmpeg requires >= 3.98.3. > > > >> Am 28.04.2011 12:43, schrieb Grant Smith: > >>> I have already installed lame 3.98.4.1 as per the referenced > >>> tutorial. > >>> > >>> Thanks, > >>> > >>> Grant Smith > >>> A+, Network+, MCP x 2, BSIT/VC, MIS > >>> > >>> Phone: +1.317.560.4457 > >>> > >>> > >>> On Thu, Apr 28, 2011 at 4:08 AM, James Darnley > >>> wrote: > >>> > >>>> On 28/04/2011, Grant Smith wrote: > >>>>> I'm trying to install ffmpeg on Ubuntu 10.04 using the tutorial > >>>>> found @ > >>>>> http://ubuntuforums.org/showpost.php?p=9868359&postcount=1289 > >>>>> > >>>>> I get to the portion regarding the actual install of ffmpeg and I > >>>>> get the following error: > >>>>> > >>>>> ERROR: libmp3lame >= 3.98.3 not found > >>>>> > >>>>> If you think configure made a mistake, make sure you are using > >>>>> the latest version from Git. If the latest version fails, report > >>>>> the problem to the ffmpeg-user at ffmpeg.org mailing list or IRC > >>>>> #ffmpeg on irc.freenode.net. Include the log file "config.log" > >>>>> produced by configure as this will help solving the problem. > >>>>> > >>>>> The link to my config.log file is http://pastebin.com/wsJYFe5e. > >>>>> It's set > >>>> to > >>>>> expire in ~20 hours. > >>>>> > >>>>> Any advice on this problem would be GREATLY appreciated. > >>>> > >>>> Install lame 3.98.3 or newer. > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > > Mit besten Gr??en, Reindl Harald > the lounge interactive design GmbH > A-1060 Vienna, Hofm?hlgasse 17 > CTO / software-development / cms-solutions > p: +43 (1) 595 3999 33, m: +43 (676) 40 221 40 > icq: 154546673, http://www.thelounge.net/ > > http://www.thelounge.net/signature.asc.what.htm > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From batguano999 at hotmail.com Thu Apr 28 20:32:46 2011 From: batguano999 at hotmail.com (bat guano) Date: Thu, 28 Apr 2011 18:32:46 +0000 Subject: [FFmpeg-user] re Installation problems. Message-ID: The tutorial has me building everything. The problem is that ffmpeg won't build. Please read my original email and look at my pastebin and you would see this. Thank you for your assistance. Yours, Grant Smith ************************ Hi I had some problems with lame when compiling FFmpeg for Karmic Ubuntu. The alternative Henrikx method in post #1286 of the tutorial worked for me. From batguano999 at hotmail.com Thu Apr 28 21:03:44 2011 From: batguano999 at hotmail.com (bat guano) Date: Thu, 28 Apr 2011 12:03:44 -0700 (PDT) Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> Message-ID: <1304017424642-3481884.post@n4.nabble.com> Hi I had some problems with lame when compiling FFmpeg for Karmic Ubuntu. The alternative Henrikx method in post #1286 of the tutorial worked for me. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Installation-Problems-tp3479957p3481884.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From grant.smith at envent-tech.com Thu Apr 28 21:16:58 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Thu, 28 Apr 2011 15:16:58 -0400 Subject: [FFmpeg-user] Installation Problems In-Reply-To: <1304017424642-3481884.post@n4.nabble.com> References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> <1304017424642-3481884.post@n4.nabble.com> Message-ID: I'll give it a look, thanks. Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 On Thu, Apr 28, 2011 at 3:03 PM, bat guano wrote: > > Hi > I had some problems with lame when compiling FFmpeg for Karmic Ubuntu. > The alternative Henrikx method in post #1286 of the tutorial worked for me. > > > -- > View this message in context: > http://ffmpeg-users.933282.n4.nabble.com/Installation-Problems-tp3479957p3481884.html > Sent from the FFmpeg-users mailing list archive at Nabble.com. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From coniophora at gmail.com Thu Apr 28 21:28:38 2011 From: coniophora at gmail.com (Jim Worrall) Date: Thu, 28 Apr 2011 13:28:38 -0600 Subject: [FFmpeg-user] getting non-AV stream from input to output In-Reply-To: References: Message-ID: <0E80DBC3-8529-4A84-BDA5-94D94D7D01F7@gmail.com> Did you try -scodec copy ? On Apr 28, 2011, at 9:25 AM, Dave Rice wrote: > Hi all, > > I have a QuickTime file with a caption track that ffmpeg identifies as > Stream #0.1(eng): Data: c608 / 0x38303663, 4 kb/s > > I want to transcode the AV data to another QuickTime as an output but pass this track as-is to the output. Is there anyway to copy the caption track to the output from the input QuickTime file to the output QuickTime. I've been testing with -map and looking for other means without success. > > Full ffmpeg -i output > http://pastebin.com/Y4mpkwvX > > Dave Rice > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From developer at noknok.net Fri Apr 29 01:48:57 2011 From: developer at noknok.net (NokNok Developer) Date: Thu, 28 Apr 2011 19:48:57 -0400 Subject: [FFmpeg-user] BOUNTY OFFERED: AAC 5.1 -> AAC Stereo In-Reply-To: <4DB05895.6010600@noknok.net> References: <4DB05895.6010600@noknok.net> Message-ID: <4DB9FCE9.90501@noknok.net> Bump.. If in fact it takes more then the below amount, suggest something that does work. On 4/21/2011 12:17 PM, NokNok Developer wrote: > Over the years we have seen various patches by folks to enable various > forms of 6 Channel to Stereo downsampling, non of which ever were > implemented in the core code base for AAC (internal) processing. > > While the solution to this problem can range from a simple mixing of > channels, adjusting volumes, etc to more advanced, our goal is to find > a solution acceptable to the ffmpeg team in any form at this point. > > Many devices support AAC but in a stereo mode only vs 6 channel, etc. > > We would like to offer such a bounty to get such a solution in the > code base which will enable downsampling of AAC 5.1 to AAC stereo when > encoding. An example of two previous variations: > > http://muzso.hu/2009/02/25/downsampling-multichannel-audio-5.1-into-stereo-2-channels-with-ffmpeg > > > Its important that the solution be acceptable in a form which it can > be included in the standard code base for ffmpeg so we are not having > to patch, etc. Using the internal AAC encoder (-strict experimental) > > Depending on the solution we would place the bounty anywhere from > $50-$250. > > If you are interested in doing this, or do this and looking for > payment, please email ffmpegbounty [at] bitsling.com > > Not sure where best in placing this bounty, so if not on the proper > list, please let us know as well. > > Thank You > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From firefighter44x at yahoo.com.au Fri Apr 29 02:36:15 2011 From: firefighter44x at yahoo.com.au (firefighter44x) Date: Thu, 28 Apr 2011 17:36:15 -0700 (PDT) Subject: [FFmpeg-user] Audio channel mapping from ac3 to aac In-Reply-To: <4CE7B8FB.8050105@achim-hofmann.com> References: <4CE7B8FB.8050105@achim-hofmann.com> Message-ID: <1304037375804-3482614.post@n4.nabble.com> Had the same problem, this works for me: add the following at line 91 of libavcodec/libfaac.c if (avctx->channels == 6) { faac_cfg->channel_map[0] = 2; faac_cfg->channel_map[1] = 0; faac_cfg->channel_map[2] = 1; faac_cfg->channel_map[3] = 4; faac_cfg->channel_map[4] = 5; faac_cfg->channel_map[5] = 3; av_log(avctx, AV_LOG_ERROR, "setting channel map to 2 0 1 4 5 3 / %d\n", avctx->channels); } -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Audio-channel-mapping-from-ac3-to-aac-tp3051477p3482614.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From hardik.sharma22 at yahoo.com Fri Apr 29 03:09:27 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Thu, 28 Apr 2011 18:09:27 -0700 (PDT) Subject: [FFmpeg-user] Problem in encoding yuv raw video file to h.264 Message-ID: <690175.82308.qm@web46201.mail.sp1.yahoo.com> Thanks James for your reply but even after putting frame rate and frame size and other information, I am getting same error as follows- ffmpeg -i silent_cif.yuv -f yuv420p -acodec libfaac -r 30 -s 352x288 -vcodec libx264 -g 10 -b 256k -s 352x288 -preset slow -threads 0 output.mp4 FFmpeg version git-N-29265-g3eb15d9, Copyright (c) 2000-2011 the FFmpeg developers ? built on Apr 21 2011 11:44:48 with gcc 4.4.5 ? configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab ? libavutil??? 51.? 0. 0 / 51.? 0. 0 ? libavcodec?? 53.? 0. 0 / 53.? 0. 0 ? libavformat? 53.? 0. 0 / 53.? 0. 0 ? libavdevice? 53.? 0. 0 / 53.? 0. 0 ? libavfilter?? 2.? 0. 0 /? 2.? 0. 0 ? libswscale??? 0. 13. 0 /? 0. 13. 0 ? libpostproc? 51.? 2. 0 / 51.? 2. 0 [IMGUTILS @ 0xbff4f234] Picture size 0x0 is invalid [IMGUTILS @ 0xbff4f134] Picture size 0x0 is invalid [rawvideo @ 0xa001bb0] Could not find codec parameters (Video: rawvideo, yuv420p) [rawvideo @ 0xa001bb0] Estimating duration from bitrate, this may be inaccurate silent_cif.yuv: could not find codec parameters What kind of more codec parameters and picture size I should put over here. I am getting same error after putting different other parameters. I really appreciate any kind of help as I am trying to encode one yuv video sequence from long time but didn't get success. Thanks in advance. Regards, Hardik Sharma? ? ? ? --- On Thu, 28/4/11, Hardik Sharma wrote: From: Hardik Sharma Subject: Problem in encoding yuv raw video file to h.264 To: ffmpeg-user at ffmpeg.org Date: Thursday, 28 April, 2011, 2:09 AM Hi, I am trying to encode yuv video file to h.264 on ubuntu 10.10 but unable to get it done. My input video is silent.yuv cif format. I am using following commands and getting following errors- $ffmpeg -i silent_cif.yuv -acodec libfaac -aq 100 -vcodec libx264 -b 256k -preset slow -crf 22 -threads 0 output.mp4FFmpeg version git-N-29265-g3eb15d9, Copyright (c) 2000-2011 the FFmpeg developers ? built on Apr 21 2011 11:44:48 with gcc 4.4.5 ? configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab ? libavutil??? 51.? 0. 0 / 51.? 0. 0 ? libavcodec?? 53.? 0. 0 / 53.? 0. 0 ? libavformat? 53.? 0. 0 / 53.? 0. 0 ? libavdevice? 53.? 0. 0 / 53.? 0. 0 ? libavfilter?? 2.? 0. 0 /? 2.? 0. 0 ? libswscale??? 0. 13. 0 /? 0. 13. 0 ? libpostproc? 51.? 2. 0 / 51.? 2. 0 [IMGUTILS @ 0xbfa66b94] Picture size 0x0 is invalid [IMGUTILS @ 0xbfa66a94] Picture size 0x0 is invalid [rawvideo @ 0x9e8dbb0] Could not find codec parameters (Video: rawvideo, yuv420p) [rawvideo @ 0x9e8dbb0] Estimating duration from bitrate, this may be inaccurate Please help me with this issue. Thanks, Hardik Sharma From leo.izen at gmail.com Fri Apr 29 03:42:35 2011 From: leo.izen at gmail.com (Leo Izen) Date: Thu, 28 Apr 2011 21:42:35 -0400 Subject: [FFmpeg-user] Good Free Prediction Algorithms Message-ID: I noticed that the only good lossless video out there is H.264 4:4:4 Predictive Profile, which has huge patent issues. No matter how high I set the bitrate, I can see obvious compression artifacts with libvpx, and theora takes an enormous bitrate to make it seem virtually lossless. There are free codecs like ffv1 and snow which pretty much have died due to lack of support. What caught my eye was huffyuv: lossless, and free (specification is here http://multimedia.cx/huffyuv.txt). The only issue that it had was that it used simple intra-only prediction. According to the spec, it uses simple intra-only prediction. This seems nice in terms of speed but isn't as nice in terms of compression ratio. I'm looking for a better FREE (emphasis on FREE) prediction algorithm, preferable inter-prediction. If anybody knows one please reply. Don't send me ones encumbered by patents. I need one that is FREE. Free as in free software and free as in freedom. I am thinking of revising the huffyuv spec and creating a new codec for a new algorithm, one with better prediction, and maybe using markov coding instead of huffman (or both, depending on how much better using both makes it). My thinking behind markov is that zlib is LZ+Huffman and lzma is LZ+Markov, and lzma is way better than zlib. From cfaf at hotmail.com Fri Apr 29 03:43:57 2011 From: cfaf at hotmail.com (christian fafard) Date: Fri, 29 Apr 2011 01:43:57 +0000 Subject: [FFmpeg-user] ffmpeg in After Effects / AVID / IMX30 broadcasting environment Message-ID: Hi, Is there a way to change the value of the metadata 'minor_version' when encoding a file? The first thing that comes to mind is -metadata minor_version="xxx" but it's not working. This value seems to be set automatically by ffmpeg. Quicktime movie files created by AVID or after effects have a typical metadata block like this, Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'avid.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-04-21 17:47:25 and are imported in AVID correctly. ffmpeg encoded files give this Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testout.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt creation_time : 1970-01-01 00:00:00 encoder : Lavf52.106.0 I'm pretty sure that this value of 512 is what's preventing an AVID system from importing correctly an ffmpeg encoded IMX30(D10) video file. Because once the good file re-wrap with a command line like this, ffmpeg -i avid.mov -metadata minor_version="537199360" -vcodec copy -acodec copy testout.mov. the minor_version field reset to 512 and the file doesn't get imported correctly. Thanks Christian Fafard From baptiste at ffmpeg.org Fri Apr 29 04:00:21 2011 From: baptiste at ffmpeg.org (Baptiste Coudurier) Date: Thu, 28 Apr 2011 19:00:21 -0700 Subject: [FFmpeg-user] ffmpeg in After Effects / AVID / IMX30 broadcasting environment In-Reply-To: References: Message-ID: <4DBA1BB5.8020104@ffmpeg.org> Hi Christian, On 04/28/2011 06:43 PM, christian fafard wrote: > > Hi, > > Is there a way to change the value of the metadata 'minor_version' when encoding a file? > The first thing that comes to mind is -metadata minor_version="xxx" but it's not working. > This value seems to be set automatically by ffmpeg. > > Quicktime movie files created by AVID or after effects have a typical metadata block like this, > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'avid.mov': > Metadata: > major_brand : qt > minor_version : 537199360 > compatible_brands: qt > creation_time : 2011-04-21 17:47:25 > > and are imported in AVID correctly. > > > ffmpeg encoded files give this > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testout.mov': > Metadata: > major_brand : qt > minor_version : 512 > compatible_brands: qt > creation_time : 1970-01-01 00:00:00 > encoder : Lavf52.106.0 > > I'm pretty sure that this value of 512 is what's preventing an AVID system from > importing correctly an ffmpeg encoded IMX30(D10) video file. > Because once the good file re-wrap with a command line like this, > ffmpeg -i avid.mov -metadata minor_version="537199360" -vcodec copy -acodec copy testout.mov. > the minor_version field reset to 512 and the file doesn't get imported correctly. Unfortunately it's more complicated than that. This won't work out of the box because AVID uses some particular info in the .mov file that AVID expects. FFmpeg does not recreate these currently. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From seandarcy2 at gmail.com Fri Apr 29 04:24:13 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Thu, 28 Apr 2011 22:24:13 -0400 Subject: [FFmpeg-user] Problem in encoding yuv raw video file to h.264 In-Reply-To: <690175.82308.qm@web46201.mail.sp1.yahoo.com> References: <690175.82308.qm@web46201.mail.sp1.yahoo.com> Message-ID: On 04/28/2011 09:09 PM, Hardik Sharma wrote: > > Thanks James for your reply but even after putting frame rate and frame size and other information, I am getting same error as follows- > > ffmpeg -i silent_cif.yuv -f yuv420p -acodec libfaac -r 30 -s 352x288 -vcodec libx264 -g 10 -b 256k -s 352x288 -preset slow -threads 0 output.mp4 > FFmpeg version git-N-29265-g3eb15d9, Copyright (c) 2000-2011 the FFmpeg developers > built on Apr 21 2011 11:44:48 with gcc 4.4.5 > configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab > libavutil 51. 0. 0 / 51. 0. 0 > libavcodec 53. 0. 0 / 53. 0. 0 > libavformat 53. 0. 0 / 53. 0. 0 > libavdevice 53. 0. 0 / 53. 0. 0 > libavfilter 2. 0. 0 / 2. 0. 0 > libswscale 0. 13. 0 / 0. 13. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [IMGUTILS @ 0xbff4f234] Picture size 0x0 is invalid > [IMGUTILS @ 0xbff4f134] Picture size 0x0 is invalid > [rawvideo @ 0xa001bb0] Could not find codec parameters (Video: rawvideo, yuv420p) > [rawvideo @ 0xa001bb0] Estimating duration from bitrate, this may be inaccurate > silent_cif.yuv: could not find codec parameters > > What kind of more codec parameters and picture size I should put over here. I am getting same error after putting different other parameters. I really appreciate any kind of help as I am trying to encode one yuv video sequence from long time but didn't get success. Thanks in advance. > > Regards, > Hardik Sharma > > --- On Thu, 28/4/11, Hardik Sharma wrote: > > From: Hardik Sharma > Subject: Problem in encoding yuv raw video file to h.264 > To: ffmpeg-user at ffmpeg.org > Date: Thursday, 28 April, 2011, 2:09 AM > > Hi, > I am trying to encode yuv video file to h.264 on ubuntu 10.10 but unable to get it done. My input video is silent.yuv cif format. I am using following commands and getting following errors- > $ffmpeg -i silent_cif.yuv -acodec libfaac -aq 100 -vcodec libx264 -b 256k -preset > slow -crf 22 -threads 0 output.mp4FFmpeg version git-N-29265-g3eb15d9, Copyright (c) 2000-2011 the FFmpeg developers > built on Apr 21 2011 11:44:48 with gcc 4.4.5 > configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab > libavutil 51. 0. 0 / 51. 0. 0 > libavcodec 53. 0. 0 / 53. 0. 0 > libavformat 53. 0. 0 / 53. 0. 0 > libavdevice 53. 0. 0 / 53. 0. 0 > libavfilter 2. 0. 0 / 2. 0. 0 > libswscale 0. 13. 0 / 0. 13. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [IMGUTILS @ 0xbfa66b94] Picture size 0x0 is invalid > [IMGUTILS @ > 0xbfa66a94] Picture size 0x0 is invalid > [rawvideo @ 0x9e8dbb0] Could not find codec parameters (Video: rawvideo, yuv420p) > [rawvideo @ 0x9e8dbb0] Estimating duration from bitrate, this may be inaccurate > > Please help me with this issue. > > Thanks, > Hardik Sharma One obvious issue is that any description of the input stream has to come before the input! ffmpeg -i input ...... sean From koxaniy at mail.ru Fri Apr 29 04:40:37 2011 From: koxaniy at mail.ru (Tuuls) Date: Thu, 28 Apr 2011 19:40:37 -0700 (PDT) Subject: [FFmpeg-user] Joining MTS videos together In-Reply-To: <35D954C42FB57C4FB551EC8CA508EA004F7294DAAD@L1VS01SBS08.rgs.local> References: <35D954C42FB57C4FB551EC8CA508EA004F7294DAAD@L1VS01SBS08.rgs.local> Message-ID: <1304044837748-3482762.post@n4.nabble.com> in windows use command line : copy /b file1.mts+file2.mts+file3.mts joinfile123.mts result file corrupt duration info , but normal encoded by ffmpeg and play in vlc players -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Joining-MTS-videos-together-tp3307469p3482762.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From grant.smith at envent-tech.com Fri Apr 29 05:01:31 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Thu, 28 Apr 2011 23:01:31 -0400 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> <1304017424642-3481884.post@n4.nabble.com> Message-ID: No dice... I get the same error trying to build ffmpeg as I did before trying the suggested alternative method.... Here's my latest pastebin: http://pastebin.com/aiueMJxA Thanks, Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 On Thu, Apr 28, 2011 at 3:16 PM, Grant Smith wrote: > I'll give it a look, thanks. > > > Grant Smith > A+, Network+, MCP x 2, BSIT/VC, MIS > > Phone: +1.317.560.4457 > > > On Thu, Apr 28, 2011 at 3:03 PM, bat guano wrote: > >> >> Hi >> I had some problems with lame when compiling FFmpeg for Karmic Ubuntu. >> The alternative Henrikx method in post #1286 of the tutorial worked for >> me. >> >> >> -- >> View this message in context: >> http://ffmpeg-users.933282.n4.nabble.com/Installation-Problems-tp3479957p3481884.html >> Sent from the FFmpeg-users mailing list archive at Nabble.com. >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From bahamutzero8825 at gmail.com Fri Apr 29 07:50:46 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Fri, 29 Apr 2011 00:50:46 -0500 Subject: [FFmpeg-user] Joining MTS videos together In-Reply-To: <1304044837748-3482762.post@n4.nabble.com> References: <35D954C42FB57C4FB551EC8CA508EA004F7294DAAD@L1VS01SBS08.rgs.local> <1304044837748-3482762.post@n4.nabble.com> Message-ID: <4DBA51B6.6000909@gmail.com> On 2011.04.28 09:40 PM, Tuuls wrote: > in windows use command line : > > copy /b file1.mts+file2.mts+file3.mts joinfile123.mts > > result file corrupt duration info , but normal encoded by ffmpeg and play in > vlc players tsMuxeR works beautifully for manipulating MPEG-TS containers. Unfortunately, download links seem to have been removed from the SmartLabs website. You could probably find it on a download site like CNET, though. From ankuj2004 at gmail.com Fri Apr 29 08:46:13 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Fri, 29 Apr 2011 12:16:13 +0530 Subject: [FFmpeg-user] Conversion to mp4 container Message-ID: Hi!! I am converting a video elementary mpeg4 stream to mp4 container format . I used the following command ffmpeg -i test043.avi -vcodec copy -an test043.avi.mp4 I am assuming that it will use the same codec for video as it is present in the mpeg4 file. Is my assumption correct or first ffmpeg will decode it and then re encode it again ? Regards Ankuj From smouli60 at gmail.com Fri Apr 29 08:57:08 2011 From: smouli60 at gmail.com (mouli s) Date: Fri, 29 Apr 2011 12:27:08 +0530 Subject: [FFmpeg-user] building ffmpeg with flvtool2 In-Reply-To: <4DB7F1D3.9010401@thelounge.net> References: <4DB7C376.70501@thelounge.net> <4DB7F1D3.9010401@thelounge.net> Message-ID: my prob is in adding tat exe to my application and using tat Rein can u plz provide me steps how to add tat .exe and use it to my ffmpeg !!!!!!!!! ???as im a begginer finding difficult... thanks for any replies. On Wed, Apr 27, 2011 at 4:07 PM, Reindl Harald wrote: > Where is the problem? > > run "flvtool2 -U file.flv" the same way you call > ffmpeg after conversion > > Am 27.04.2011 12:27, schrieb mouli s: > > Hello Rein, > > Thanks for ur reply but how to solve my prob???? my prob is > > exact to the prob which is posted in this link > > > > > http://stackoverflow.com/questions/1532703/video-length-missing-in-flv-converted-by-ffmpeg-php > > > > tats y i thought of using flvtool2 to my application.... but dnt find > steps > > how to use it.. :( tats y posted in forum...thanks for any help.. > > > > On Wed, Apr 27, 2011 at 12:49 PM, Reindl Harald >wrote: > > > >> flvtool2 has nothing to do with ffmpeg > >> it is a standalone application > >> > >> Am 27.04.2011 06:30, schrieb mouli s: > >>> Hello all can anyone tell me how to take a build for ffmpeg with > flvtool2 > >>> for windows..... any wiki for tat???? thanks for any replies... > >>> _______________________________________________ > >>> ffmpeg-user mailing list > >>> ffmpeg-user at ffmpeg.org > >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > >> > >> _______________________________________________ > >> ffmpeg-user mailing list > >> ffmpeg-user at ffmpeg.org > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > >> > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > > Mit besten Gr??en, Reindl Harald > the lounge interactive design GmbH > A-1060 Vienna, Hofm?hlgasse 17 > CTO / software-development / cms-solutions > p: +43 (1) 595 3999 33, m: +43 (676) 40 221 40 > icq: 154546673, http://www.thelounge.net/ > > http://www.thelounge.net/signature.asc.what.htm > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From koxaniy at mail.ru Fri Apr 29 09:03:27 2011 From: koxaniy at mail.ru (Tuuls) Date: Fri, 29 Apr 2011 00:03:27 -0700 (PDT) Subject: [FFmpeg-user] Joining MTS videos together In-Reply-To: <4DBA51B6.6000909@gmail.com> References: <35D954C42FB57C4FB551EC8CA508EA004F7294DAAD@L1VS01SBS08.rgs.local> <1304044837748-3482762.post@n4.nabble.com> <4DBA51B6.6000909@gmail.com> Message-ID: <1304060607656-3483071.post@n4.nabble.com> >tsMuxeR works beautifully for manipulating MPEG-TS containers. Why ? tsMuxeR not a command line utility -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Joining-MTS-videos-together-tp3307469p3483071.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From bahamutzero8825 at gmail.com Fri Apr 29 09:06:55 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Fri, 29 Apr 2011 02:06:55 -0500 Subject: [FFmpeg-user] Conversion to mp4 container In-Reply-To: References: Message-ID: <4DBA638F.5040003@gmail.com> On 2011.04.29 01:46 AM, Ankuj Gupta wrote: > ffmpeg -i test043.avi -vcodec copy -an test043.avi.mp4 > > I am assuming that it will use the same codec for video as it is present in > the mpeg4 file. Is my assumption correct or first ffmpeg will decode it and > then re encode it again ? If you tell FFmpeg to copy the video stream, it will do so. FFmpeg can't reencode if you don't specify a codec anyway (it wouldn't know what to do). From bahamutzero8825 at gmail.com Fri Apr 29 09:08:42 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Fri, 29 Apr 2011 02:08:42 -0500 Subject: [FFmpeg-user] Joining MTS videos together In-Reply-To: <1304060607656-3483071.post@n4.nabble.com> References: <35D954C42FB57C4FB551EC8CA508EA004F7294DAAD@L1VS01SBS08.rgs.local> <1304044837748-3482762.post@n4.nabble.com> <4DBA51B6.6000909@gmail.com> <1304060607656-3483071.post@n4.nabble.com> Message-ID: <4DBA63FA.5080006@gmail.com> On 2011.04.29 02:03 AM, Tuuls wrote: > >tsMuxeR works beautifully for manipulating MPEG-TS containers. > > Why ? tsMuxeR not a command line utility Why what? tsMuxeR.exe is a command-line program. tsMuxerGUI.exe is the GUI version. From ankuj2004 at gmail.com Fri Apr 29 09:20:55 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Fri, 29 Apr 2011 12:50:55 +0530 Subject: [FFmpeg-user] Conversion to mp4 container In-Reply-To: <4DBA638F.5040003@gmail.com> References: <4DBA638F.5040003@gmail.com> Message-ID: But when viewed through Hex Editor there is difference in between the video elementary stream given as input and the one extracted from mp4 On Fri, Apr 29, 2011 at 12:36 PM, Andrew Berg wrote: > On 2011.04.29 01:46 AM, Ankuj Gupta wrote: > > ffmpeg -i test043.avi -vcodec copy -an test043.avi.mp4 > > > > I am assuming that it will use the same codec for video as it is present > in > > the mpeg4 file. Is my assumption correct or first ffmpeg will decode it > and > > then re encode it again ? > If you tell FFmpeg to copy the video stream, it will do so. FFmpeg can't > reencode if you don't specify a codec anyway (it wouldn't know what to do). > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From ankuj2004 at gmail.com Fri Apr 29 09:24:16 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Fri, 29 Apr 2011 12:54:16 +0530 Subject: [FFmpeg-user] Conversion to mp4 container In-Reply-To: References: <4DBA638F.5040003@gmail.com> Message-ID: The original stream has multiple Visual Object Sequence which have been removed by ffmpeg. On Fri, Apr 29, 2011 at 12:50 PM, Ankuj Gupta wrote: > But when viewed through Hex Editor there is difference in between the video > elementary stream given as input and the one extracted from mp4 > > > > > On Fri, Apr 29, 2011 at 12:36 PM, Andrew Berg wrote: > >> On 2011.04.29 01:46 AM, Ankuj Gupta wrote: >> > ffmpeg -i test043.avi -vcodec copy -an test043.avi.mp4 >> > >> > I am assuming that it will use the same codec for video as it is present >> in >> > the mpeg4 file. Is my assumption correct or first ffmpeg will decode it >> and >> > then re encode it again ? >> If you tell FFmpeg to copy the video stream, it will do so. FFmpeg can't >> reencode if you don't specify a codec anyway (it wouldn't know what to >> do). >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > > From koxaniy at mail.ru Fri Apr 29 09:25:44 2011 From: koxaniy at mail.ru (Tuuls) Date: Fri, 29 Apr 2011 00:25:44 -0700 (PDT) Subject: [FFmpeg-user] Joining MTS videos together In-Reply-To: <4DBA63FA.5080006@gmail.com> References: <35D954C42FB57C4FB551EC8CA508EA004F7294DAAD@L1VS01SBS08.rgs.local> <1304044837748-3482762.post@n4.nabble.com> <4DBA51B6.6000909@gmail.com> <1304060607656-3483071.post@n4.nabble.com> <4DBA63FA.5080006@gmail.com> Message-ID: <1304061944398-3483111.post@n4.nabble.com> > Why what? Why use it? Russian tirelessly to support her :) -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Joining-MTS-videos-together-tp3307469p3483111.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From bahamutzero8825 at gmail.com Fri Apr 29 09:29:33 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Fri, 29 Apr 2011 02:29:33 -0500 Subject: [FFmpeg-user] Joining MTS videos together In-Reply-To: <1304061944398-3483111.post@n4.nabble.com> References: <35D954C42FB57C4FB551EC8CA508EA004F7294DAAD@L1VS01SBS08.rgs.local> <1304044837748-3482762.post@n4.nabble.com> <4DBA51B6.6000909@gmail.com> <1304060607656-3483071.post@n4.nabble.com> <4DBA63FA.5080006@gmail.com> <1304061944398-3483111.post@n4.nabble.com> Message-ID: <4DBA68DD.6040901@gmail.com> On 2011.04.29 02:25 AM, Tuuls wrote: > Why use it? Because it's extremely useful for creating properly split or joined MPEG-TS containers. > Russian tirelessly to support her :) I have no idea what this means. From bahamutzero8825 at gmail.com Fri Apr 29 09:34:54 2011 From: bahamutzero8825 at gmail.com (Andrew Berg) Date: Fri, 29 Apr 2011 02:34:54 -0500 Subject: [FFmpeg-user] Conversion to mp4 container In-Reply-To: References: <4DBA638F.5040003@gmail.com> Message-ID: <4DBA6A1E.4010205@gmail.com> On 2011.04.29 02:20 AM, Ankuj Gupta wrote: > But when viewed through Hex Editor there is difference in between the video > elementary stream given as input and the one extracted from mp4 It's not unusual for a program to modify headers when remuxing. If you specify copy as the -vcodec option, encoding is out of context. From dashing.meng at gmail.com Fri Apr 29 09:46:02 2011 From: dashing.meng at gmail.com (littlebat) Date: Fri, 29 Apr 2011 15:46:02 +0800 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> <1304017424642-3481884.post@n4.nabble.com> Message-ID: <20110429154602.a9e617db.dashing.meng@gmail.com> If you make sure ffmpeg has been compiled and installed ok, then try "sudo ldconfig -v" On Thu, 28 Apr 2011 23:01:31 -0400 Grant Smith wrote: > No dice... I get the same error trying to build ffmpeg as I did before > trying the suggested alternative method.... > > Here's my latest pastebin: http://pastebin.com/aiueMJxA > > Thanks, > > Grant Smith > A+, Network+, MCP x 2, BSIT/VC, MIS > > Phone: +1.317.560.4457 > > > On Thu, Apr 28, 2011 at 3:16 PM, Grant Smith > wrote: > > > I'll give it a look, thanks. > > > > > > Grant Smith > > A+, Network+, MCP x 2, BSIT/VC, MIS > > > > Phone: +1.317.560.4457 > > > > > > On Thu, Apr 28, 2011 at 3:03 PM, bat guano > > wrote: > > > >> > >> Hi > >> I had some problems with lame when compiling FFmpeg for Karmic > >> Ubuntu. The alternative Henrikx method in post #1286 of the > >> tutorial worked for me. > >> > >> > >> -- > >> View this message in context: > >> http://ffmpeg-users.933282.n4.nabble.com/Installation-Problems-tp3479957p3481884.html > >> Sent from the FFmpeg-users mailing list archive at Nabble.com. > >> _______________________________________________ > >> ffmpeg-user mailing list > >> ffmpeg-user at ffmpeg.org > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > >> > > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From h.reindl at thelounge.net Fri Apr 29 10:01:49 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Fri, 29 Apr 2011 10:01:49 +0200 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> <1304017424642-3481884.post@n4.nabble.com> Message-ID: <4DBA706D.3070508@thelounge.net> Am 29.04.2011 05:01, schrieb Grant Smith: > No dice... I get the same error trying to build ffmpeg as I did before > trying the suggested alternative method.... > > Here's my latest pastebin: http://pastebin.com/aiueMJxA /home/ubuntu/lame-3.98.4/include/lame.h why should configure look there? personal build have to be in /usr/local/ substrcuture and not dropped overall the system and if you leave the standard-paths you must say configure where your stuff is hidden -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From marian.daniel at ext.alerant.hu Fri Apr 29 11:16:08 2011 From: marian.daniel at ext.alerant.hu (=?iso-8859-2?Q?Mari=E1n_D=E1niel?=) Date: Fri, 29 Apr 2011 11:16:08 +0200 Subject: [FFmpeg-user] User reference needed - FFMPEG - Really Important Message-ID: Hi to all on the Mailing list! We are currently working on a project, wich needs references from the users of FFMPEG. Our team really needs your help, we have a deadline on monday, so if you have one minute please send for me the following: Your data will not be publicated or abused on any way! . Your name and e-mail address . The name of the company, where you using FFMPEG . The project's name in wich you use the FFMPEG Thank You! Daniel Marian Alerant Inc. 1117 Budapest, Hungary Infopark s?t?ny 1. Tel.: +36-30-4780634 Fax: +36-1-2050056 E-mail: marian.daniel at ext.alerant.hu From jdmls at yahoo.com Fri Apr 29 11:21:46 2011 From: jdmls at yahoo.com (John Doe) Date: Fri, 29 Apr 2011 02:21:46 -0700 (PDT) Subject: [FFmpeg-user] building ffmpeg with flvtool2 In-Reply-To: References: <4DB7C376.70501@thelounge.net> <4DB7F1D3.9010401@thelounge.net> Message-ID: <565402.16332.qm@web114704.mail.gq1.yahoo.com> From: mouli s > my prob is in adding tat exe to my application and using tat Rein can u plz > provide me steps how to add tat .exe and use it to my ffmpeg !!!!!!!!! > ???as im a begginer finding difficult... thanks for any replies. http://tinyurl.com/6dr8uuo JD From bouke at editb.nl Fri Apr 29 11:25:38 2011 From: bouke at editb.nl (Bouke) Date: Fri, 29 Apr 2011 11:25:38 +0200 Subject: [FFmpeg-user] User reference needed - FFMPEG - Really Important References: Message-ID: <003d01cc064f$6a35b730$4301a8c0@hpkantoor> ----- Original Message ----- From: "Mari?n D?niel" To: Cc: "T?zs?r ?ron" ; "Darmai G?bor" Sent: Friday, April 29, 2011 11:16 AM Subject: [FFmpeg-user] User reference needed - FFMPEG - Really Important > Hi to all on the Mailing list! > > > > We are currently working on a project, wich needs references from the > users > of FFMPEG. > > > > Our team really needs your help, we have a deadline on monday, so if you > have one minute please send for me the following: > > > > Your data will not be publicated or abused on any way! > > > > . Your name and e-mail address > > . The name of the company, where you using FFMPEG > > . The project's name in wich you use the FFMPEG > > > > Thank You! This is an unreasonable request for a stranger to ask me. But if you need a big name to convince your client, use the BBC. Bouke > > Daniel Marian > > Alerant Inc. > > 1117 Budapest, Hungary > > Infopark s?t?ny 1. > Tel.: +36-30-4780634 > > Fax: +36-1-2050056 > > E-mail: marian.daniel at ext.alerant.hu > > > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From h.reindl at thelounge.net Fri Apr 29 11:27:34 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Fri, 29 Apr 2011 11:27:34 +0200 Subject: [FFmpeg-user] building ffmpeg with flvtool2 In-Reply-To: <565402.16332.qm@web114704.mail.gq1.yahoo.com> References: <4DB7C376.70501@thelounge.net> <4DB7F1D3.9010401@thelounge.net> <565402.16332.qm@web114704.mail.gq1.yahoo.com> Message-ID: <4DBA8486.6080402@thelounge.net> Am 29.04.2011 11:21, schrieb John Doe: > From: mouli s > >> my prob is in adding tat exe to my application and using tat Rein can u plz >> provide me steps how to add tat .exe and use it to my ffmpeg !!!!!!!!! >> ???as im a begginer finding difficult... thanks for any replies. there is nothing to help call it manually if your application does not support THE LAST TIME: flvtool2 has NOTHING to do with the ffmpeg-binary hwo should i help you? we are using self-developed software calling ffmpeg per command-line and supporting postprocessing calling flvtool2 if the target is FLV and qt-faststart if the output is MP4 -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From ankuj2004 at gmail.com Fri Apr 29 12:08:18 2011 From: ankuj2004 at gmail.com (Ankuj Gupta) Date: Fri, 29 Apr 2011 15:38:18 +0530 Subject: [FFmpeg-user] Conversion to mp4 container In-Reply-To: <4DBA6A1E.4010205@gmail.com> References: <4DBA638F.5040003@gmail.com> <4DBA6A1E.4010205@gmail.com> Message-ID: While converting some streams I am getting non monotone timestamps 1 >= 0 What is the exact cause of this error ? On Fri, Apr 29, 2011 at 1:04 PM, Andrew Berg wrote: > On 2011.04.29 02:20 AM, Ankuj Gupta wrote: > > But when viewed through Hex Editor there is difference in between the > video > > elementary stream given as input and the one extracted from mp4 > It's not unusual for a program to modify headers when remuxing. If you > specify copy as the -vcodec option, encoding is out of context. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From shailendra at sdi.la Fri Apr 29 12:40:18 2011 From: shailendra at sdi.la (Shailendra) Date: Fri, 29 Apr 2011 16:10:18 +0530 Subject: [FFmpeg-user] How to edit or write in metadata of video in c++ Message-ID: Hi Alll, Can anybody guide me how to edit/write in metadata of the video? Thanks, Shailendra From rens at onlinemedia.nl Fri Apr 29 13:38:11 2011 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Fri, 29 Apr 2011 13:38:11 +0200 (CEST) Subject: [FFmpeg-user] h264 not decoding In-Reply-To: <7852048.1416541304076407705.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <9130306.1416561304077091567.JavaMail.root@webmail.onlinemedia.nl> Hi, i would like to convert this TEST.MTS file to something more usable in Avid Mediacomposer but sofar all programs tested (Avid, VLC, FFmpeg, FFmbc, MpegStreamClip Mplayer & Quicktime ) failed to decode the file properly, most of the time you get some decode macroblocks at the top of the picture but about 80 % remains green, we have some audio but decoding is poor. ffprobe gives the following output. # ffprobe -pretty TEST.MTS FFprobe version 0.6.1, Copyright (c) 2007-2010 the FFmpeg developers built on Feb 4 2011 11:57:14 with gcc 4.2.1 (Apple Inc. build 5664) configuration: arch=x86_64 --enable-libmp3lame --enable-libfaac --enable-libvpx --enable-libx264 --enable-gpl --enable-nonfree --enable-pthreads --enable-shared libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.11. 0 / 0.11. 0 [h264 @ 0x101009200]top block unavailable for requested intra4x4 mode -1 at 67 0 [h264 @ 0x101009200]error while decoding MB 67 0, bytestream (312813) Input #0, mpegts, from 'TEST.MTS': Duration: 00:11:53.01, start: 1.000033, bitrate: 23846 kb/s Program 1 Stream #0.0[0x1011]: Video: h264, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 50 fps, 2250 tbr, 90k tbn, 50 tbc Stream #0.1[0x1100]: Audio: pcm_bluray, 48000 Hz, stereo, s16, 1536 kb/s Any ideas what to try Rens Dijkshoorn From grant.smith at envent-tech.com Fri Apr 29 13:49:44 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Fri, 29 Apr 2011 07:49:44 -0400 Subject: [FFmpeg-user] Installation Problems In-Reply-To: <20110429154602.a9e617db.dashing.meng@gmail.com> References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> <1304017424642-3481884.post@n4.nabble.com> <20110429154602.a9e617db.dashing.meng@gmail.com> Message-ID: littlebat, FFmpeg has /not/ been compiled and installed ok. That's the problem. It keeps failing, saying that it cannot find lame. Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 On Fri, Apr 29, 2011 at 3:46 AM, littlebat wrote: > If you make sure ffmpeg has been compiled and installed ok, > then try "sudo ldconfig -v" > > On Thu, 28 Apr 2011 23:01:31 -0400 > Grant Smith wrote: > > > No dice... I get the same error trying to build ffmpeg as I did before > > trying the suggested alternative method.... > > > > Here's my latest pastebin: http://pastebin.com/aiueMJxA > > > > Thanks, > > > > Grant Smith > > A+, Network+, MCP x 2, BSIT/VC, MIS > > > > Phone: +1.317.560.4457 > > > > > > On Thu, Apr 28, 2011 at 3:16 PM, Grant Smith > > wrote: > > > > > I'll give it a look, thanks. > > > > > > > > > Grant Smith > > > A+, Network+, MCP x 2, BSIT/VC, MIS > > > > > > Phone: +1.317.560.4457 > > > > > > > > > On Thu, Apr 28, 2011 at 3:03 PM, bat guano > > > wrote: > > > > > >> > > >> Hi > > >> I had some problems with lame when compiling FFmpeg for Karmic > > >> Ubuntu. The alternative Henrikx method in post #1286 of the > > >> tutorial worked for me. > > >> > > >> > > >> -- > > >> View this message in context: > > >> > http://ffmpeg-users.933282.n4.nabble.com/Installation-Problems-tp3479957p3481884.html > > >> Sent from the FFmpeg-users mailing list archive at Nabble.com. > > >> _______________________________________________ > > >> ffmpeg-user mailing list > > >> ffmpeg-user at ffmpeg.org > > >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > >> > > > > > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From grant.smith at envent-tech.com Fri Apr 29 13:51:57 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Fri, 29 Apr 2011 07:51:57 -0400 Subject: [FFmpeg-user] Installation Problems In-Reply-To: <4DBA706D.3070508@thelounge.net> References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> <1304017424642-3481884.post@n4.nabble.com> <4DBA706D.3070508@thelounge.net> Message-ID: Reindl Harald, I'm doing everything according to the tutorial and I haven't done anything custom, i.e. everything goes to the paths it's set to go to without any intervention from me. I can give it a try from the root of the filesystem tonight and see if that helps. Thank you for your input. Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 On Fri, Apr 29, 2011 at 4:01 AM, Reindl Harald wrote: > > Am 29.04.2011 05:01, schrieb Grant Smith: > > No dice... I get the same error trying to build ffmpeg as I did before > > trying the suggested alternative method.... > > > > Here's my latest pastebin: http://pastebin.com/aiueMJxA > > /home/ubuntu/lame-3.98.4/include/lame.h > > why should configure look there? > > personal build have to be in /usr/local/ substrcuture and > not dropped overall the system and if you leave the standard-paths > you must say configure where your stuff is hidden > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From h.reindl at thelounge.net Fri Apr 29 13:59:47 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Fri, 29 Apr 2011 13:59:47 +0200 Subject: [FFmpeg-user] Installation Problems In-Reply-To: References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> <1304017424642-3481884.post@n4.nabble.com> <4DBA706D.3070508@thelounge.net> Message-ID: <4DBAA833.3030209@thelounge.net> Am 29.04.2011 13:51, schrieb Grant Smith: > Reindl Harald, > > I'm doing everything according to the tutorial and I haven't done anything > custom, i.e. everything goes to the paths it's set to go to without any > intervention from me. if your tutorial ends in "/home/ubuntu/lame-3.98.4/include/lame.h" and the configure from thesame tutorial is called without any informations wehre inlcudes are the tutorial is crap i can not really help on ubuntu because i will never touch and deb-based distribution in my life, but again: you should learn how to build deb-apackages instead of compiling installing somewhere manually, this is a dirty way and will bring al lot of troubles sooner or later since builds of new versions do not remove orphaned files - that is why pakacge-managment was invented -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From phil_rhodes at rocketmail.com Fri Apr 29 14:00:39 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Fri, 29 Apr 2011 13:00:39 +0100 Subject: [FFmpeg-user] User reference needed - FFMPEG - Really Important In-Reply-To: <003d01cc064f$6a35b730$4301a8c0@hpkantoor> References: <003d01cc064f$6a35b730$4301a8c0@hpkantoor> Message-ID: > This is an unreasonable request for a stranger to ask me. No, it isn't. Don't be so standoffish. If you don't want to do it, don't do it, but asking for references is a perfectly normal thing to do. Apologies to Mari?n, don't worry, not all opensource people are like that... P From bouke at editb.nl Fri Apr 29 14:55:33 2011 From: bouke at editb.nl (Bouke) Date: Fri, 29 Apr 2011 14:55:33 +0200 Subject: [FFmpeg-user] User reference needed - FFMPEG - Really Important References: <003d01cc064f$6a35b730$4301a8c0@hpkantoor> Message-ID: <00a601cc066c$bd4bfc50$4301a8c0@hpkantoor> ---- Original Message ----- From: "Phil Rhodes" To: "FFmpeg user questions and RTFMs" Sent: Friday, April 29, 2011 2:00 PM Subject: Re: [FFmpeg-user] User reference needed - FFMPEG - Really Important >> This is an unreasonable request for a stranger to ask me. > > No, it isn't. Don't be so standoffish. If you don't want to do it, don't > do it, but asking for references is a perfectly normal thing to do. Phil, Asking for company names and e-mail adresses is something used in practices called 'spamming', and other stuff. Now Mari?n is new. I totally agree that it is a different story if it is somebody one knows (at least a bit.) And i could indeed totally ignore it and do nothing, but i believe i've given some decent advice while i was at it... Bouke > Apologies to Mari?n, don't worry, not all opensource people are like > that... > > P > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From bouke at editb.nl Fri Apr 29 14:58:07 2011 From: bouke at editb.nl (Bouke) Date: Fri, 29 Apr 2011 14:58:07 +0200 Subject: [FFmpeg-user] h264 not decoding References: <9130306.1416561304077091567.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <00a901cc066d$1c488b60$4301a8c0@hpkantoor> ----- Original Message ----- From: "Rens Dijkshoorn" To: "ffmpeg-user" Sent: Friday, April 29, 2011 1:38 PM Subject: [FFmpeg-user] h264 not decoding > Hi, > > i would like to convert this TEST.MTS file to something more usable in > Avid Mediacomposer > but sofar all programs tested (Avid, VLC, FFmpeg, FFmbc, MpegStreamClip > Mplayer & Quicktime ) > failed to decode the file properly, most of the time you get some decode > macroblocks at the > top of the picture but about 80 % remains green, we have some audio but > decoding is poor. Eeer, at risk of stating the obvious, if NOTHING can decode the file, are you sure the file isn't suffering from bitrot? What does play back the file then? Bouke > ffprobe gives the following output. > > # ffprobe -pretty TEST.MTS > > FFprobe version 0.6.1, Copyright (c) 2007-2010 the FFmpeg developers > built on Feb 4 2011 11:57:14 with gcc 4.2.1 (Apple Inc. build 5664) > configuration: > arch=x86_64 --enable-libmp3lame --enable-libfaac --enable-libvpx --enable-libx264 > --enable-gpl > --enable-nonfree --enable-pthreads --enable-shared > > libavutil 50.15. 1 / 50.15. 1 > libavcodec 52.72. 2 / 52.72. 2 > libavformat 52.64. 2 / 52.64. 2 > libavdevice 52. 2. 0 / 52. 2. 0 > libswscale 0.11. 0 / 0.11. 0 > > [h264 @ 0x101009200]top block unavailable for requested intra4x4 mode -1 > at 67 0 > [h264 @ 0x101009200]error while decoding MB 67 0, bytestream (312813) > > Input #0, mpegts, from 'TEST.MTS': > Duration: 00:11:53.01, start: 1.000033, bitrate: 23846 kb/s > Program 1 > Stream #0.0[0x1011]: Video: h264, yuv420p, 1920x1080 [PAR 1:1 DAR > 16:9], 50 fps, 2250 tbr, 90k tbn, 50 tbc > Stream #0.1[0x1100]: Audio: pcm_bluray, 48000 Hz, stereo, s16, 1536 > kb/s > > Any ideas what to try > > > Rens Dijkshoorn > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From rens at onlinemedia.nl Fri Apr 29 15:16:00 2011 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Fri, 29 Apr 2011 15:16:00 +0200 (CEST) Subject: [FFmpeg-user] h264 not decoding In-Reply-To: <26044611.1417131304082699340.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <20882493.1417241304082960530.JavaMail.root@webmail.onlinemedia.nl> Hi Bouke, i'am told Windows Media Player 11 can play without any problems en free-mts-converter found at http://www.topsevenreviews.com/free-mts-converter.html can convert into something more usefull, but i would like to handle the files in my ffmpeg based batch converter on a Mac. Regards Rens > Hi, > > i would like to convert this TEST.MTS file to something more usable in > Avid Mediacomposer > but sofar all programs tested (Avid, VLC, FFmpeg, FFmbc, MpegStreamClip > Mplayer & Quicktime ) > failed to decode the file properly, most of the time you get some decode > macroblocks at the > top of the picture but about 80 % remains green, we have some audio but > decoding is poor. Eeer, at risk of stating the obvious, if NOTHING can decode the file, are you sure the file isn't suffering from bitrot? What does play back the file then? Bouke > ffprobe gives the following output. > > # ffprobe -pretty TEST.MTS > > FFprobe version 0.6.1, Copyright (c) 2007-2010 the FFmpeg developers > built on Feb 4 2011 11:57:14 with gcc 4.2.1 (Apple Inc. build 5664) > configuration: > arch=x86_64 --enable-libmp3lame --enable-libfaac --enable-libvpx --enable-libx264 > --enable-gpl > --enable-nonfree --enable-pthreads --enable-shared > > libavutil 50.15. 1 / 50.15. 1 > libavcodec 52.72. 2 / 52.72. 2 > libavformat 52.64. 2 / 52.64. 2 > libavdevice 52. 2. 0 / 52. 2. 0 > libswscale 0.11. 0 / 0.11. 0 > > [h264 @ 0x101009200]top block unavailable for requested intra4x4 mode -1 > at 67 0 > [h264 @ 0x101009200]error while decoding MB 67 0, bytestream (312813) > > Input #0, mpegts, from 'TEST.MTS': > Duration: 00:11:53.01, start: 1.000033, bitrate: 23846 kb/s > Program 1 > Stream #0.0[0x1011]: Video: h264, yuv420p, 1920x1080 [PAR 1:1 DAR > 16:9], 50 fps, 2250 tbr, 90k tbn, 50 tbc > Stream #0.1[0x1100]: Audio: pcm_bluray, 48000 Hz, stereo, s16, 1536 > kb/s > > Any ideas what to try > > > Rens Dijkshoorn > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From bouke at editb.nl Fri Apr 29 15:21:58 2011 From: bouke at editb.nl (Bouke) Date: Fri, 29 Apr 2011 15:21:58 +0200 Subject: [FFmpeg-user] h264 not decoding References: <20882493.1417241304082960530.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <00ee01cc0670$6e2f7710$4301a8c0@hpkantoor> ----- Original Message ----- From: "Rens Dijkshoorn" To: "FFmpeg user questions and RTFMs" Sent: Friday, April 29, 2011 3:16 PM Subject: Re: [FFmpeg-user] h264 not decoding > Hi Bouke, > > i'am told Windows Media Player 11 can play without any problems > en free-mts-converter found at > > http://www.topsevenreviews.com/free-mts-converter.html > > can convert into something more usefull, but i would like to handle > the files in my ffmpeg based batch converter on a Mac. Rens, Perhaps copy protection stuff that is messing things up? Where did you get the file? And if it is a one-off, and you have means to convert it, why not use that and just be done with it? Bouke > Regards Rens > > >> Hi, >> >> i would like to convert this TEST.MTS file to something more usable in >> Avid Mediacomposer >> but sofar all programs tested (Avid, VLC, FFmpeg, FFmbc, MpegStreamClip >> Mplayer & Quicktime ) >> failed to decode the file properly, most of the time you get some decode >> macroblocks at the >> top of the picture but about 80 % remains green, we have some audio but >> decoding is poor. > > Eeer, at risk of stating the obvious, > if NOTHING can decode the file, are you sure the file isn't suffering from > bitrot? > What does play back the file then? > > Bouke > > > >> ffprobe gives the following output. >> >> # ffprobe -pretty TEST.MTS >> >> FFprobe version 0.6.1, Copyright (c) 2007-2010 the FFmpeg developers >> built on Feb 4 2011 11:57:14 with gcc 4.2.1 (Apple Inc. build 5664) >> configuration: >> arch=x86_64 --enable-libmp3lame --enable-libfaac --enable-libvpx --enable-libx264 >> --enable-gpl >> --enable-nonfree --enable-pthreads --enable-shared >> >> libavutil 50.15. 1 / 50.15. 1 >> libavcodec 52.72. 2 / 52.72. 2 >> libavformat 52.64. 2 / 52.64. 2 >> libavdevice 52. 2. 0 / 52. 2. 0 >> libswscale 0.11. 0 / 0.11. 0 >> >> [h264 @ 0x101009200]top block unavailable for requested intra4x4 mode -1 >> at 67 0 >> [h264 @ 0x101009200]error while decoding MB 67 0, bytestream (312813) >> >> Input #0, mpegts, from 'TEST.MTS': >> Duration: 00:11:53.01, start: 1.000033, bitrate: 23846 kb/s >> Program 1 >> Stream #0.0[0x1011]: Video: h264, yuv420p, 1920x1080 [PAR 1:1 DAR >> 16:9], 50 fps, 2250 tbr, 90k tbn, 50 tbc >> Stream #0.1[0x1100]: Audio: pcm_bluray, 48000 Hz, stereo, s16, 1536 >> kb/s >> >> Any ideas what to try >> >> >> Rens Dijkshoorn >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From rens at onlinemedia.nl Fri Apr 29 15:33:38 2011 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Fri, 29 Apr 2011 15:33:38 +0200 (CEST) Subject: [FFmpeg-user] h264 not decoding In-Reply-To: <17254216.1417501304083660473.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <13263332.1417541304084018781.JavaMail.root@webmail.onlinemedia.nl> Hi Bouke its not that simple, file is recorded by some sort of flash based camera i'm still waiting on the details about settings type of camera etc. And more files are underway so i would like to bring into our standard workflow. If i just use video copy to h264 the resulting video is not playable at all. During conversion i get these kind of error messages... [h264 @ 0x10100c200] st:0 error, non monotone timestamps 100800 >= 99040 [NULL @ 0x101014200] non-existing SPS 2 referenced in buffering periodbits/s [h264 @ 0x10100c200] st:0 error, non monotone timestamps 9882000 >= 9882000/s [h264 @ 0x10100c200] st:0 error, non monotone timestamps 10011600 >= 10011600 [NULL @ 0x101014200] non-existing SPS 2 referenced in buffering periodkbits/s [h264 @ 0x10100c200] st:0 error, non monotone timestamps 50425200 >= 50411200 sofar no luck Rens > Hi Bouke, > > i'am told Windows Media Player 11 can play without any problems > en free-mts-converter found at > > http://www.topsevenreviews.com/free-mts-converter.html > > can convert into something more usefull, but i would like to handle > the files in my ffmpeg based batch converter on a Mac. Rens, Perhaps copy protection stuff that is messing things up? Where did you get the file? And if it is a one-off, and you have means to convert it, why not use that and just be done with it? Bouke > Regards Rens > > >> Hi, >> >> i would like to convert this TEST.MTS file to something more usable in >> Avid Mediacomposer >> but sofar all programs tested (Avid, VLC, FFmpeg, FFmbc, MpegStreamClip >> Mplayer & Quicktime ) >> failed to decode the file properly, most of the time you get some decode >> macroblocks at the >> top of the picture but about 80 % remains green, we have some audio but >> decoding is poor. > > Eeer, at risk of stating the obvious, > if NOTHING can decode the file, are you sure the file isn't suffering from > bitrot? > What does play back the file then? > > Bouke > > > >> ffprobe gives the following output. >> >> # ffprobe -pretty TEST.MTS >> >> FFprobe version 0.6.1, Copyright (c) 2007-2010 the FFmpeg developers >> built on Feb 4 2011 11:57:14 with gcc 4.2.1 (Apple Inc. build 5664) >> configuration: >> arch=x86_64 --enable-libmp3lame --enable-libfaac --enable-libvpx --enable-libx264 >> --enable-gpl >> --enable-nonfree --enable-pthreads --enable-shared >> >> libavutil 50.15. 1 / 50.15. 1 >> libavcodec 52.72. 2 / 52.72. 2 >> libavformat 52.64. 2 / 52.64. 2 >> libavdevice 52. 2. 0 / 52. 2. 0 >> libswscale 0.11. 0 / 0.11. 0 >> >> [h264 @ 0x101009200]top block unavailable for requested intra4x4 mode -1 >> at 67 0 >> [h264 @ 0x101009200]error while decoding MB 67 0, bytestream (312813) >> >> Input #0, mpegts, from 'TEST.MTS': >> Duration: 00:11:53.01, start: 1.000033, bitrate: 23846 kb/s >> Program 1 >> Stream #0.0[0x1011]: Video: h264, yuv420p, 1920x1080 [PAR 1:1 DAR >> 16:9], 50 fps, 2250 tbr, 90k tbn, 50 tbc >> Stream #0.1[0x1100]: Audio: pcm_bluray, 48000 Hz, stereo, s16, 1536 >> kb/s >> >> Any ideas what to try >> >> >> Rens Dijkshoorn >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From bouke at editb.nl Fri Apr 29 15:51:59 2011 From: bouke at editb.nl (Bouke) Date: Fri, 29 Apr 2011 15:51:59 +0200 Subject: [FFmpeg-user] h264 not decoding References: <13263332.1417541304084018781.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <010b01cc0674$9fa40000$4301a8c0@hpkantoor> hmm, Action / sports / helmet cam? Do share the model / make, as i want to buy one myself :-) Does a straight import in Avid work? And just for fun, can i have a look at one of the files? Bouke ----- Original Message ----- From: "Rens Dijkshoorn" To: "FFmpeg user questions and RTFMs" Sent: Friday, April 29, 2011 3:33 PM Subject: Re: [FFmpeg-user] h264 not decoding > Hi Bouke > > its not that simple, file is recorded by some sort > of flash based camera i'm still waiting on the details > about settings type of camera etc. > > And more files are underway so i would like to bring into > our standard workflow. > > If i just use video copy to h264 the resulting video is > not playable at all. During conversion i get these kind > of error messages... > > [h264 @ 0x10100c200] st:0 error, non monotone timestamps 100800 >= 99040 > [NULL @ 0x101014200] non-existing SPS 2 referenced in buffering > periodbits/s > [h264 @ 0x10100c200] st:0 error, non monotone timestamps 9882000 >= > 9882000/s > [h264 @ 0x10100c200] st:0 error, non monotone timestamps 10011600 >= > 10011600 > [NULL @ 0x101014200] non-existing SPS 2 referenced in buffering > periodkbits/s > [h264 @ 0x10100c200] st:0 error, non monotone timestamps 50425200 >= > 50411200 > > sofar no luck > > Rens > > >> Hi Bouke, >> >> i'am told Windows Media Player 11 can play without any problems >> en free-mts-converter found at >> >> http://www.topsevenreviews.com/free-mts-converter.html >> >> can convert into something more usefull, but i would like to handle >> the files in my ffmpeg based batch converter on a Mac. > > Rens, > Perhaps copy protection stuff that is messing things up? > Where did you get the file? > And if it is a one-off, and you have means to convert it, why not use that > and just be done with it? > > Bouke > > >> Regards Rens >> >> >>> Hi, >>> >>> i would like to convert this TEST.MTS file to something more usable in >>> Avid Mediacomposer >>> but sofar all programs tested (Avid, VLC, FFmpeg, FFmbc, MpegStreamClip >>> Mplayer & Quicktime ) >>> failed to decode the file properly, most of the time you get some decode >>> macroblocks at the >>> top of the picture but about 80 % remains green, we have some audio but >>> decoding is poor. >> >> Eeer, at risk of stating the obvious, >> if NOTHING can decode the file, are you sure the file isn't suffering >> from >> bitrot? >> What does play back the file then? >> >> Bouke >> >> >> >>> ffprobe gives the following output. >>> >>> # ffprobe -pretty TEST.MTS >>> >>> FFprobe version 0.6.1, Copyright (c) 2007-2010 the FFmpeg developers >>> built on Feb 4 2011 11:57:14 with gcc 4.2.1 (Apple Inc. build 5664) >>> configuration: >>> arch=x86_64 --enable-libmp3lame --enable-libfaac --enable-libvpx --enable-libx264 >>> --enable-gpl >>> --enable-nonfree --enable-pthreads --enable-shared >>> >>> libavutil 50.15. 1 / 50.15. 1 >>> libavcodec 52.72. 2 / 52.72. 2 >>> libavformat 52.64. 2 / 52.64. 2 >>> libavdevice 52. 2. 0 / 52. 2. 0 >>> libswscale 0.11. 0 / 0.11. 0 >>> >>> [h264 @ 0x101009200]top block unavailable for requested intra4x4 mode -1 >>> at 67 0 >>> [h264 @ 0x101009200]error while decoding MB 67 0, bytestream (312813) >>> >>> Input #0, mpegts, from 'TEST.MTS': >>> Duration: 00:11:53.01, start: 1.000033, bitrate: 23846 kb/s >>> Program 1 >>> Stream #0.0[0x1011]: Video: h264, yuv420p, 1920x1080 [PAR 1:1 DAR >>> 16:9], 50 fps, 2250 tbr, 90k tbn, 50 tbc >>> Stream #0.1[0x1100]: Audio: pcm_bluray, 48000 Hz, stereo, s16, 1536 >>> kb/s >>> >>> Any ideas what to try >>> >>> >>> Rens Dijkshoorn >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From rens at onlinemedia.nl Fri Apr 29 16:01:28 2011 From: rens at onlinemedia.nl (Rens Dijkshoorn) Date: Fri, 29 Apr 2011 16:01:28 +0200 (CEST) Subject: [FFmpeg-user] h264 not decoding In-Reply-To: <18399433.1417721304085386366.JavaMail.root@webmail.onlinemedia.nl> Message-ID: <12819338.1417741304085688077.JavaMail.root@webmail.onlinemedia.nl> Hi Bouke, I will ask the content owner, the smallest one i have is 1.5 Gb Regards Rens hmm, Action / sports / helmet cam? Do share the model / make, as i want to buy one myself :-) Does a straight import in Avid work? And just for fun, can i have a look at one of the files? Bouke ----- Original Message ----- From: "Rens Dijkshoorn" To: "FFmpeg user questions and RTFMs" Sent: Friday, April 29, 2011 3:33 PM Subject: Re: [FFmpeg-user] h264 not decoding > Hi Bouke > > its not that simple, file is recorded by some sort > of flash based camera i'm still waiting on the details > about settings type of camera etc. > > And more files are underway so i would like to bring into > our standard workflow. > > If i just use video copy to h264 the resulting video is > not playable at all. During conversion i get these kind > of error messages... > > [h264 @ 0x10100c200] st:0 error, non monotone timestamps 100800 >= 99040 > [NULL @ 0x101014200] non-existing SPS 2 referenced in buffering > periodbits/s > [h264 @ 0x10100c200] st:0 error, non monotone timestamps 9882000 >= > 9882000/s > [h264 @ 0x10100c200] st:0 error, non monotone timestamps 10011600 >= > 10011600 > [NULL @ 0x101014200] non-existing SPS 2 referenced in buffering > periodkbits/s > [h264 @ 0x10100c200] st:0 error, non monotone timestamps 50425200 >= > 50411200 > > sofar no luck > > Rens > > >> Hi Bouke, >> >> i'am told Windows Media Player 11 can play without any problems >> en free-mts-converter found at >> >> http://www.topsevenreviews.com/free-mts-converter.html >> >> can convert into something more usefull, but i would like to handle >> the files in my ffmpeg based batch converter on a Mac. > > Rens, > Perhaps copy protection stuff that is messing things up? > Where did you get the file? > And if it is a one-off, and you have means to convert it, why not use that > and just be done with it? > > Bouke > > >> Regards Rens >> >> >>> Hi, >>> >>> i would like to convert this TEST.MTS file to something more usable in >>> Avid Mediacomposer >>> but sofar all programs tested (Avid, VLC, FFmpeg, FFmbc, MpegStreamClip >>> Mplayer & Quicktime ) >>> failed to decode the file properly, most of the time you get some decode >>> macroblocks at the >>> top of the picture but about 80 % remains green, we have some audio but >>> decoding is poor. >> >> Eeer, at risk of stating the obvious, >> if NOTHING can decode the file, are you sure the file isn't suffering >> from >> bitrot? >> What does play back the file then? >> >> Bouke >> >> >> >>> ffprobe gives the following output. >>> >>> # ffprobe -pretty TEST.MTS >>> >>> FFprobe version 0.6.1, Copyright (c) 2007-2010 the FFmpeg developers >>> built on Feb 4 2011 11:57:14 with gcc 4.2.1 (Apple Inc. build 5664) >>> configuration: >>> arch=x86_64 --enable-libmp3lame --enable-libfaac --enable-libvpx --enable-libx264 >>> --enable-gpl >>> --enable-nonfree --enable-pthreads --enable-shared >>> >>> libavutil 50.15. 1 / 50.15. 1 >>> libavcodec 52.72. 2 / 52.72. 2 >>> libavformat 52.64. 2 / 52.64. 2 >>> libavdevice 52. 2. 0 / 52. 2. 0 >>> libswscale 0.11. 0 / 0.11. 0 >>> >>> [h264 @ 0x101009200]top block unavailable for requested intra4x4 mode -1 >>> at 67 0 >>> [h264 @ 0x101009200]error while decoding MB 67 0, bytestream (312813) >>> >>> Input #0, mpegts, from 'TEST.MTS': >>> Duration: 00:11:53.01, start: 1.000033, bitrate: 23846 kb/s >>> Program 1 >>> Stream #0.0[0x1011]: Video: h264, yuv420p, 1920x1080 [PAR 1:1 DAR >>> 16:9], 50 fps, 2250 tbr, 90k tbn, 50 tbc >>> Stream #0.1[0x1100]: Audio: pcm_bluray, 48000 Hz, stereo, s16, 1536 >>> kb/s >>> >>> Any ideas what to try _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From joolzg at btinternet.com Fri Apr 29 16:27:36 2011 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Fri, 29 Apr 2011 15:27:36 +0100 (BST) Subject: [FFmpeg-user] ffmpeg and x264 In-Reply-To: <61386.97286.qm@web86408.mail.ird.yahoo.com> References: <61386.97286.qm@web86408.mail.ird.yahoo.com> Message-ID: <237821.66460.qm@web86407.mail.ird.yahoo.com> Have i missed something about the new ffmpeg. Just installed a clean git version of ffmpeg and x264 and issued my line with "-vcodec libx264 -vpre faster" and get the error that there is not ffpreset for faster, looking into the ffpreset directory there are only a few linx264-lossless_xxxx.ffpreset and none of the older ones. Anybody shed some light on the missing slow/medium/fast/faster/superfast/veryfast and ultrafast joolz From jklimek at gmail.com Fri Apr 29 16:59:07 2011 From: jklimek at gmail.com (John Klimek) Date: Fri, 29 Apr 2011 10:59:07 -0400 Subject: [FFmpeg-user] av_interleaved_write_frame - operation not permitted Message-ID: I'm trying to stream some video to my iPhone (using FFmpeg and an open source HTTP segmenter) and FFmpeg is giving an error: av_interleaved_write_frame(): Operation not permitted I've done some research and it seems like this appears because of invalid timestamps but I'm a bit confused because this file does playback in every media player just fine. The script I'm using (iStreamdev) to execute FFmpeg starts up like this: (trap "rm -f ./ffmpeg.pid; rm -f ./fifo" EXIT HUP INT TERM ABRT; \ $FFMPEGPREFIX | $FFPATH -i "$STREAM" -deinterlace -f mpegts -acodec libmp3lame -ab $ARATE -ac 2 -s $XY -vcodec libx264 -b $VRATE -flags +loop \ -cmp \+chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 \ -sc_threshold 40 -i_qfactor 0.71 -bt $VRATE -maxrate $VRATE -bufsize $VRATE -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 \ -qmin 10 -qmax 51 -qdiff 4 -level 30 -g 30 -async 2 -threads 4 - 2>$FFMPEGLOG 1>./fifo) & Here is my FFmpeg log: ffmpeg version git-N-29391-gd84f191, Copyright (c) 2000-2011 the FFmpeg developers built on Apr 27 2011 09:52:06 with gcc 4.4.5 configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libop encore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --ena ble-libvpx --disable-vaapi libavutil 51. 0. 0 / 51. 0. 0 libavcodec 53. 1. 0 / 53. 1. 0 libavformat 53. 0. 3 / 53. 0. 3 libavdevice 53. 0. 0 / 53. 0. 0 libavfilter 2. 0. 0 / 2. 0. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 [h264 @ 0x18a7df0] mmco: unref short failure Last message repeated 1 times [h264 @ 0x18a7df0] number of reference frames exceeds max (probably corrupt input), discarding one [mpegts @ 0x18a2b70] max_analyze_duration reached [mpegts @ 0x18a2b70] Estimating duration from bitrate, this may be inaccurate Seems stream 0 codec frame rate differs from container frame rate: 59.94 (60000/1001) -> 59.94 (60000/1001) Input #0, mpegts, from 'pipe:': Duration: N/A, start: 61552.826133, bitrate: 384 kb/s Program 132 Stream #0.0[0x1122]: Video: h264 (High), yuv420p, 1440x1080 [PAR 4:3 DAR 16:9], 59.96 fps, 59.94 tbr, 90k tbn, 59.94 tbc Stream #0.1[0x1123](eng): Audio: ac3, 48000 Hz, stereo, s16, 384 kb/s [buffer @ 0x18a62b0] w:1440 h:1080 pixfmt:yuv420p [scale @ 0x18cf3a0] w:1440 h:1080 fmt:yuv420p -> w:408 h:272 fmt:yuv420p flags:0xa0000004 [libx264 @ 0x18a6720] using SAR=4/3 [libx264 @ 0x18a6720] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 0x18a6720] profile Constrained Baseline, level 3.0 [mpegts @ 0x18aa6d0] muxrate VBR, pcr every 5 pkts, sdt every 200, pat/pmt every 40 pkts Output #0, mpegts, to 'pipe:': Metadata: encoder : Lavf53.0.3 Stream #0.0: Video: libx264, yuv420p, 408x272 [PAR 4:3 DAR 2:1], q=10-51, 350 kb/s, 90k tbn, 59.94 tbc Stream #0.1(eng): Audio: libmp3lame, 48000 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 [h264 @ 0x18a7df0] mmco: unref short failure Last message repeated 1 times [h264 @ 0x18a7df0] number of reference frames exceeds max (probably corrupt input), discarding one frame= 82 fps= 0 q=38.0 size= 41kB time=0.60 bitrate= 559.0kbits/s dup=64 drop=0 ^Mframe= 124 fps=120 q=36.0 size= 77kB time=1.22 bitrate= 516.1kbits/s dup=85 drop=0 ^Mframe= 166 fps=108 q=36.0 size= 123kB time=1.85 bitrate= 546.9kbits/s dup=106 drop=0 ^Mframe= 206 fps=100 q=36.0 size= 175kB time=2.64 bitrate= 544.1kbits/s dup=126 drop=0 ^Mframe= 246 fps= 96 q=36.0 size= 220kB time=3.35 bitrate= 537.8kbits/s dup=146 drop=0 ^Mframe= 286 fps= 93 q=37.0 size= 273kB time=4.02 bitrate= 555.9kbits/s dup=166 drop=0 ^Mframe= 328 fps= 92 q=37.0 size= 310kB time=4.72 bitrate= 537.7kbits/s dup=187 drop=0 ^Mav_interleaved_write_frame(): Operation not permitted From Dave.Pope at myxer.com Fri Apr 29 17:37:47 2011 From: Dave.Pope at myxer.com (Dave Pope) Date: Fri, 29 Apr 2011 11:37:47 -0400 Subject: [FFmpeg-user] ffmpeg and x264 In-Reply-To: <237821.66460.qm@web86407.mail.ird.yahoo.com> References: <61386.97286.qm@web86408.mail.ird.yahoo.com> <237821.66460.qm@web86407.mail.ird.yahoo.com> Message-ID: Yeah, I hit this one too. Use "-preset" and "-profile" switches instead of -vpre. "-preset fast -profile baseline" works OK for me. Annoying that it's not reflected in the --help output; I don't see a way to list the available values yet either. -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of JULIAN GARDNER Sent: Friday, April 29, 2011 10:28 AM To: FFmpeg user questions and RTFMs Subject: [FFmpeg-user] ffmpeg and x264 Have i missed something about the new ffmpeg. Just installed a clean git version of ffmpeg and x264 and issued my line with "-vcodec libx264 -vpre faster" and get the error that there is not ffpreset for faster, looking into the ffpreset directory there are only a few linx264-lossless_xxxx.ffpreset and none of the older ones. Anybody shed some light on the missing slow/medium/fast/faster/superfast/veryfast and ultrafast joolz _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From Dave.Pope at myxer.com Fri Apr 29 17:37:47 2011 From: Dave.Pope at myxer.com (Dave Pope) Date: Fri, 29 Apr 2011 11:37:47 -0400 Subject: [FFmpeg-user] ffmpeg and x264 In-Reply-To: <237821.66460.qm@web86407.mail.ird.yahoo.com> References: <61386.97286.qm@web86408.mail.ird.yahoo.com> <237821.66460.qm@web86407.mail.ird.yahoo.com> Message-ID: Yeah, I hit this one too. Use "-preset" and "-profile" switches instead of -vpre. "-preset fast -profile baseline" works OK for me. Annoying that it's not reflected in the --help output; I don't see a way to list the available values yet either. -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of JULIAN GARDNER Sent: Friday, April 29, 2011 10:28 AM To: FFmpeg user questions and RTFMs Subject: [FFmpeg-user] ffmpeg and x264 Have i missed something about the new ffmpeg. Just installed a clean git version of ffmpeg and x264 and issued my line with "-vcodec libx264 -vpre faster" and get the error that there is not ffpreset for faster, looking into the ffpreset directory there are only a few linx264-lossless_xxxx.ffpreset and none of the older ones. Anybody shed some light on the missing slow/medium/fast/faster/superfast/veryfast and ultrafast joolz _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From hardik.sharma22 at yahoo.com Fri Apr 29 18:07:25 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Fri, 29 Apr 2011 09:07:25 -0700 (PDT) Subject: [FFmpeg-user] Problem in encoding yuv raw video file to h.264 Message-ID: <744292.56743.qm@web46216.mail.sp1.yahoo.com> Thank you very much James, its working now. ? --- On Fri, 29/4/11, Hardik Sharma wrote: From: Hardik Sharma Subject: Re: Problem in encoding yuv raw video file to h.264 To: ffmpeg-user at ffmpeg.org Cc: james.darnley at gmail.com Date: Friday, 29 April, 2011, 1:09 AM Thanks James for your reply but even after putting frame rate and frame size and other information, I am getting same error as follows- ffmpeg -i silent_cif.yuv -f yuv420p -acodec libfaac -r 30 -s 352x288 -vcodec libx264 -g 10 -b 256k -s 352x288 -preset slow -threads 0 output.mp4 FFmpeg version git-N-29265-g3eb15d9, Copyright (c) 2000-2011 the FFmpeg developers ? built on Apr 21 2011 11:44:48 with gcc 4.4.5 ? configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab ? libavutil??? 51.? 0. 0 / 51.? 0. 0 ? libavcodec?? 53.? 0. 0 / 53.? 0. 0 ? libavformat? 53.? 0. 0 / 53.? 0. 0 ? libavdevice? 53.? 0. 0 / 53.? 0. 0 ? libavfilter?? 2.? 0. 0 /? 2.? 0. 0 ? libswscale??? 0. 13. 0 /? 0. 13. 0 ? libpostproc? 51.? 2. 0 / 51.? 2. 0 [IMGUTILS @ 0xbff4f234] Picture size 0x0 is invalid [IMGUTILS @ 0xbff4f134] Picture size 0x0 is invalid [rawvideo @ 0xa001bb0] Could not find codec parameters (Video: rawvideo, yuv420p) [rawvideo @ 0xa001bb0] Estimating duration from bitrate, this may be inaccurate silent_cif.yuv: could not find codec parameters What kind of more codec parameters and picture size I should put over here. I am getting same error after putting different other parameters. I really appreciate any kind of help as I am trying to encode one yuv video sequence from long time but didn't get success. Thanks in advance. Regards, Hardik Sharma? ? ? ? --- On Thu, 28/4/11, Hardik Sharma wrote: From: Hardik Sharma Subject: Problem in encoding yuv raw video file to h.264 To: ffmpeg-user at ffmpeg.org Date: Thursday, 28 April, 2011, 2:09 AM Hi, I am trying to encode yuv video file to h.264 on ubuntu 10.10 but unable to get it done. My input video is silent.yuv cif format. I am using following commands and getting following errors- $ffmpeg -i silent_cif.yuv -acodec libfaac -aq 100 -vcodec libx264 -b 256k -preset slow -crf 22 -threads 0 output.mp4FFmpeg version git-N-29265-g3eb15d9, Copyright (c) 2000-2011 the FFmpeg developers ? built on Apr 21 2011 11:44:48 with gcc 4.4.5 ? configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab ? libavutil??? 51.? 0. 0 / 51.? 0. 0 ? libavcodec?? 53.? 0. 0 / 53.? 0. 0 ? libavformat? 53.? 0. 0 / 53.? 0. 0 ? libavdevice? 53.? 0. 0 / 53.? 0. 0 ? libavfilter?? 2.? 0. 0 /? 2.? 0. 0 ? libswscale??? 0. 13. 0 /? 0. 13. 0 ? libpostproc? 51.? 2. 0 / 51.? 2. 0 [IMGUTILS @ 0xbfa66b94] Picture size 0x0 is invalid [IMGUTILS @ 0xbfa66a94] Picture size 0x0 is invalid [rawvideo @ 0x9e8dbb0] Could not find codec parameters (Video: rawvideo, yuv420p) [rawvideo @ 0x9e8dbb0] Estimating duration from bitrate, this may be inaccurate Please help me with this issue. Thanks, Hardik Sharma From grant.smith at envent-tech.com Fri Apr 29 19:22:29 2011 From: grant.smith at envent-tech.com (Grant Smith) Date: Fri, 29 Apr 2011 13:22:29 -0400 Subject: [FFmpeg-user] Installation Problems In-Reply-To: <4DBAA833.3030209@thelounge.net> References: <4DB94700.40602@thelounge.net> <20110428094401.0a237b1e@lrcd.com> <4DB9A832.2060502@thelounge.net> <1304017424642-3481884.post@n4.nabble.com> <4DBA706D.3070508@thelounge.net> <4DBAA833.3030209@thelounge.net> Message-ID: The tutorial builds packages. They can then be removed through use of dpkg (-r) to uninstall. As I stated earlier, if you take a look at the tutorial, you'd see what I'm talking about. Thanks, Grant Smith A+, Network+, MCP x 2, BSIT/VC, MIS Phone: +1.317.560.4457 On Fri, Apr 29, 2011 at 7:59 AM, Reindl Harald wrote: > > > Am 29.04.2011 13:51, schrieb Grant Smith: > > Reindl Harald, > > > > I'm doing everything according to the tutorial and I haven't done > anything > > custom, i.e. everything goes to the paths it's set to go to without any > > intervention from me. > > if your tutorial ends in "/home/ubuntu/lame-3.98.4/include/lame.h" > and the configure from thesame tutorial is called without any > informations wehre inlcudes are the tutorial is crap > > i can not really help on ubuntu because i will never touch > and deb-based distribution in my life, but again: > > you should learn how to build deb-apackages instead of compiling > installing somewhere manually, this is a dirty way and will bring > al lot of troubles sooner or later since builds of new versions > do not remove orphaned files - that is why pakacge-managment was invented > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From arnstrb at yahoo.com Fri Apr 29 19:28:37 2011 From: arnstrb at yahoo.com (Arnie Bearak) Date: Fri, 29 Apr 2011 10:28:37 -0700 (PDT) Subject: [FFmpeg-user] h264/rtsp encoding question Message-ID: <314723.86456.qm@web121706.mail.ne1.yahoo.com> I would like to get some information on how to encode a series of pgm files into an mp4 file that I can later stream to a remote player. When I use the following command ffmpeg -r 1 -b 1800 -i frame_%05d.pgm -f rtsp -vcodec libx264 -vpre slow Arn.mp4 I have an mp4 file that I can copy to a remote system, and play the file with VLC player or QuickTime. However, when I try to stream that file from my linux box using the Darwin Streaming server to the remote machine using VLC or Quick time, I am getting an invalid format. However, I can sucessfully stream the sampls Darwin stream mp4 files. Is there a command line option I need to add to my ffmpeg command to have ffmpeg insert whatever framing tags into the resultant file. I tried -f rtsp, but that seems to be for use if ffmpeg is talking to ffserver. Thanks, Arnie From benoit at friry.net Fri Apr 29 19:53:03 2011 From: benoit at friry.net (Benoit) Date: Fri, 29 Apr 2011 19:53:03 +0200 Subject: [FFmpeg-user] deinterlacing failed (webm encoding) Message-ID: <20110429175303.GA15426@friry.net> Hello, I try to save VHS films. 1st step: capturing from Dazzle DVC100 My box is not speed enough to encode directly. $ ffmpeg \ -f alsa -i plughw:0,0 \ -f video4linux2 -tvstd SECAM -i /dev/video0 \ -acodec pcm_s16le -ac 2 -ar 44100 \ -vcodec mjpeg -qscale 1 -s 720x576 -r 25 -sameq \ -f avi video.avi Avi file is huge, but seems ok. Readable with ffplay. 2nd step: cut and sync I use avidemux to cut unwanted time at the beginning and at the end, and to offset sound so as to resync. 3rd step: reencode $ ffmpeg \ -i video.avi \ -acodec libvorbis -aq 4 -ac 2 \ -vcodec libvpx \ -deinterlace \ -croptop 6 -cropbottom 2 -cropleft 16 -cropright 6 \ -vb 1M \ -g 360 -keyint_min 0 \ -rc_lookahead 16 \ -skip_threshold 0 -level 116 \ -f webm video.webm output is: FFmpeg version 0.6.2-4:0.6.2-1, Copyright (c) 2000-2010 the Libav developers built on Mar 29 2011 16:14:25 with gcc 4.5.2 configuration: --extra-version=4:0.6.2-1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-vaapi --enable-libopenjpeg --enable-gpl --enable-postproc --enable-x11grab --enable-libdirac --enable-libfaad --enable-librtmp --enable-libdc1394 --enable-shared --disable-static libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, avi, from 'video.avi': Duration: 00:31:23.68, start: 0.000000, bitrate: 25231 kb/s Stream #0.0: Video: mjpeg, yuvj420p, 720x576, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s [libvpx @ 0xc78e50]v0.9.6 Output #0, webm, to 'video.webm': Metadata: encoder : Lavf52.64.2 Stream #0.0: Video: libvpx, yuv420p, 698x568, q=2-31, 1000 kb/s, 1k tbn, 25 tbc Stream #0.1: Audio: libvorbis, 44100 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding Deinterlacing failed Deinterlacing failed Deinterlacing failed Deinterlacing failed Deinterlacing failedq=0.0 size= 69kB time=0.09 bitrate=6046.5kbits/s Deinterlacing failed Deinterlacing failed Deinterlacing failedq=0.0 size= 69kB time=0.18 bitrate=3141.5kbits/s Deinterlacing failed Deinterlacing failed Deinterlacing failedq=0.0 size= 101kB time=0.29 bitrate=2794.8kbits/s Deinterlacing failed Deinterlacing failed Deinterlacing failedq=0.0 size= 133kB time=0.43 bitrate=2503.7kbits/s Deinterlacing failed Deinterlacing failedq=0.0 size= 133kB time=0.50 bitrate=2156.0kbits/s Deinterlacing failed Deinterlacing failed Deinterlacing failedq=0.0 size= 165kB time=0.62 bitrate=2175.4kbits/s frame= 19 fps= 4 q=0.0 Lsize= 195kB time=0.72 bitrate=2216.6kbits/s video:182kB audio:8kB global headers:4kB muxing overhead 0.508158% Final webm video is not deinterlaced. I've got the same errors with first file (before avidemux) Any idea about the "Deinterlacing failed"? (I'm also interested by any hint about my process. Is MJPEG a good intermediate?) Thanks, benoit From research at the10thfloor.com Fri Apr 29 20:45:52 2011 From: research at the10thfloor.com (Shalom Levytam) Date: Fri, 29 Apr 2011 14:45:52 -0400 Subject: [FFmpeg-user] AAC Downsample Doesn't Work With -f mpegts In-Reply-To: References: Message-ID: <8A469E70-C997-4998-9226-F3A1E89A05BC@the10thfloor.com> Any suggestions at all ??? On 2011-04-26, at 8:43 PM, Shalom Levytam wrote: > Hi, > > I was wondering if someone could help me with downsampling aac audio when converting to a .ts file. > > Currently, I'm trying: > > ffmpeg -y -i mc497b98.mp4 -f mpegts -acodec libfaac -ar 22050 -ac 2 -ab 64k -vcodec libx264 ..... > > The converted file is missing audio. > Stream #0.1[0x101](und): Audio: aac, 0 channels, s16 > > If I upsample (ie -ar 48000) or remove -ar entirely it works fine. > > If I leave the -ar but remove -f mpegts it also works fine. > > Any thoughts? > > Thanks, > > slevytam > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From joolzg at btinternet.com Fri Apr 29 21:00:29 2011 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Fri, 29 Apr 2011 20:00:29 +0100 (BST) Subject: [FFmpeg-user] ffmpeg and x264 In-Reply-To: References: <61386.97286.qm@web86408.mail.ird.yahoo.com> <237821.66460.qm@web86407.mail.ird.yahoo.com> Message-ID: <783169.26532.qm@web86405.mail.ird.yahoo.com> Thanks for the quick update, im looking at posting a new patch to allow UDP to use a task for receiving data and wanted to check it with a new git pull first. Now i can test it and post the patch. joolz ----- Original Message ----- From: Dave Pope To: FFmpeg user questions and RTFMs ; FFmpeg user questions and RTFMs Cc: Sent: Friday, 29 April 2011, 17:37 Subject: Re: [FFmpeg-user] ffmpeg and x264 Yeah, I hit this one too.? Use "-preset" and "-profile" switches instead of -vpre.? "-preset fast -profile baseline" works OK for me.? Annoying that it's not reflected in the --help output; I don't see a way to list the available values yet either. -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of JULIAN GARDNER Sent: Friday, April 29, 2011 10:28 AM To: FFmpeg user questions and RTFMs Subject: [FFmpeg-user] ffmpeg and x264 Have i missed something about the new ffmpeg. Just installed a clean git version of ffmpeg and x264 and issued my line with "-vcodec libx264 -vpre faster" and get the error that there is not ffpreset for faster, looking into the ffpreset directory there are only a few linx264-lossless_xxxx.ffpreset and none of the older ones. Anybody shed some light on the missing slow/medium/fast/faster/superfast/veryfast and ultrafast joolz _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From arnstrb at yahoo.com Sat Apr 30 01:01:02 2011 From: arnstrb at yahoo.com (Arnie Bearak) Date: Fri, 29 Apr 2011 16:01:02 -0700 (PDT) Subject: [FFmpeg-user] Why is my mp4 file larger when I do an encode with -threads 0 Message-ID: <321309.68566.qm@web121704.mail.ne1.yahoo.com> I did an encode of a set of pgm files with the following command ffmpeg -r 1 -b -I frame_%05d.pgm -b 18000 -vcodec libx264 -vpre slow Arn.mp4 it generated an pm4 file of about 56K But when I repeated the test and did ffmpeg -r 1 -b -I frame_%05d.pgm -b 18000 - threads 0 -vcodec libx264 -vpre slow Arn.mp4 it generated a file of about 102K bytes. Can anyone tell me why the generated file is larger when using -threads 0? Thanks From hardik.sharma22 at yahoo.com Sat Apr 30 02:16:27 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Fri, 29 Apr 2011 17:16:27 -0700 (PDT) Subject: [FFmpeg-user] Unable to get correct decoded file Message-ID: <98850.77544.qm@web46215.mail.sp1.yahoo.com> Hi, I am a new bee with Ubuntu and ffmpeg both so please help me with this issue. I encoded the yuv video to h264 by following command-? ?ffmpeg -f rawvideo -r 30 -b 256k -s 352x288 -i silent_cif.yuv -vcodec libx264 -b 256k -s 352x288 -preset slow -f h264 -threads 0 silent264.h264 But I don't think I am getting correct encoded file as after decoding back to yuv video, I am only getting 2 frames and that too wrong frames. Command line for decoding- ?ffmpeg -r 30 -b 256k -s 352x288 -i output.h264 -vcodec libx264 -b 256k -s 352x288 -preset slow -f rawvideo -threads 0 slient.yuv Please tell me my mistake in command line. I really appreciate your help. Thanks. Regards, Hardik Sharma? From leo.izen at gmail.com Sat Apr 30 02:47:28 2011 From: leo.izen at gmail.com (Leo Izen) Date: Fri, 29 Apr 2011 20:47:28 -0400 Subject: [FFmpeg-user] Unable to get correct decoded file In-Reply-To: <98850.77544.qm@web46215.mail.sp1.yahoo.com> References: <98850.77544.qm@web46215.mail.sp1.yahoo.com> Message-ID: On Fri, Apr 29, 2011 at 8:16 PM, Hardik Sharma wrote: > > Command line for decoding- > > ffmpeg -r 30 -b 256k -s 352x288 -i output.h264 -vcodec libx264 -b 256k > -s 352x288 -preset slow -f rawvideo -threads 0 slient.yuv > You should use -vcodec rawvideo for decoding. I also suggest this: Your full command line for decoding should be this: ffmpeg -i in.h264 -vcodec rawvideo -f yuv4mpegpipe out.y4m I have removed most unnecessary options, but left some to show a point. Generally you should store rawvideo that is in some yuv pixel format in the yuv4mpegpipe (y4m) container, as it is simply a wrapper for the video, but has data like size and frame rate, so the user of the program reading it doesn't have to specify this data. Note: this only stores these pixel formats: yuv444p, yuv422p, yuv420p, yuv411p and gray. y4m files also can't contain audio, or any non-raw video. But it is very useful for video. From baptiste at ffmpeg.org Sat Apr 30 03:50:47 2011 From: baptiste at ffmpeg.org (Baptiste Coudurier) Date: Fri, 29 Apr 2011 18:50:47 -0700 Subject: [FFmpeg-user] av_interleaved_write_frame - operation not permitted In-Reply-To: References: Message-ID: <4DBB6AF7.2010100@ffmpeg.org> Hi John, On 04/29/2011 07:59 AM, John Klimek wrote: > I'm trying to stream some video to my iPhone (using FFmpeg and an open > source HTTP segmenter) and FFmpeg is giving an error: > > av_interleaved_write_frame(): Operation not permitted > > I've done some research and it seems like this appears because of > invalid timestamps but I'm a bit confused because this file does > playback in every media player just fine. > > The script I'm using (iStreamdev) to execute FFmpeg starts up like this: > > (trap "rm -f ./ffmpeg.pid; rm -f ./fifo" EXIT HUP INT TERM ABRT; \ > $FFMPEGPREFIX | $FFPATH -i "$STREAM" -deinterlace -f mpegts -acodec > libmp3lame -ab $ARATE -ac 2 -s $XY -vcodec libx264 -b $VRATE -flags > +loop \ > -cmp \+chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 > -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 \ > -sc_threshold 40 -i_qfactor 0.71 -bt $VRATE -maxrate $VRATE -bufsize > $VRATE -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 \ > -qmin 10 -qmax 51 -qdiff 4 -level 30 -g 30 -async 2 -threads 4 - > 2>$FFMPEGLOG 1>./fifo) & Use -preset when encoding with libx264. -async 1 should be enough. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From rodney.baker at iinet.net.au Sat Apr 30 04:12:30 2011 From: rodney.baker at iinet.net.au (Rodney Baker) Date: Sat, 30 Apr 2011 11:42:30 +0930 Subject: [FFmpeg-user] ffmpeg and x264 In-Reply-To: References: <237821.66460.qm@web86407.mail.ird.yahoo.com> Message-ID: <201104301142.30578.rodney.baker@iinet.net.au> On Sat, 30 Apr 2011 01:07:47 Dave Pope wrote: > Yeah, I hit this one too. Use "-preset" and "-profile" switches instead > of -vpre. "-preset fast -profile baseline" works OK for me. Annoying > that it's not reflected in the --help output; I don't see a way to list > the available values yet either. > Details of this change were posted on this list a couple of weeks ago (check the list archives). Details were also supposed to be added to the news section of the ffmpeg home page (a patch was submitted on ffmpeg-devel) but it does not appear to have been updated yet. -- =================================================== Rodney Baker VK5ZTV rodney.baker at iinet.net.au =================================================== From bmurphy1976 at gmail.com Sat Apr 30 04:38:39 2011 From: bmurphy1976 at gmail.com (Bryan Murphy) Date: Fri, 29 Apr 2011 21:38:39 -0500 Subject: [FFmpeg-user] BOUNTY OFFERED: AAC 5.1 -> AAC Stereo In-Reply-To: <4DB9FCE9.90501@noknok.net> References: <4DB05895.6010600@noknok.net> <4DB9FCE9.90501@noknok.net> Message-ID: I'm sorry, but I just don't think anybody is going to take you seriously. $250 is barely two hours of work at reasonable consulting rates and this is seems to be a very hard problem. Maybe if you offered 10x that amount... Bryan On Thu, Apr 28, 2011 at 6:48 PM, NokNok Developer wrote: > Bump.. ?If in fact it takes more then the below amount, suggest something > that does work. > > On 4/21/2011 12:17 PM, NokNok Developer wrote: >> >> Over the years we have seen various patches by folks to enable various >> forms of 6 Channel to Stereo downsampling, non of which ever were >> implemented in the core code base for AAC (internal) processing. >> >> While the solution to this problem can range from a simple mixing of >> channels, adjusting volumes, etc to more advanced, our goal is to find a >> solution acceptable to the ffmpeg team in any form at this point. >> >> Many devices support AAC but in a stereo mode only vs 6 channel, etc. >> >> We would like to offer such a bounty to get such a solution in the code >> base which will enable downsampling of AAC 5.1 to AAC stereo when encoding. >> ? ?An example of two previous variations: >> >> >> http://muzso.hu/2009/02/25/downsampling-multichannel-audio-5.1-into-stereo-2-channels-with-ffmpeg >> >> Its important that the solution be acceptable in a form which it can be >> included in the standard code base for ffmpeg so we are not having to patch, >> etc. ?Using the internal AAC encoder (-strict experimental) >> >> Depending on the solution we would place the bounty anywhere from >> $50-$250. >> >> If you are interested in doing this, or do this and looking for payment, >> please email ffmpegbounty [at] bitsling.com >> >> Not sure where best in placing this bounty, so if not on the proper list, >> please let us know as well. >> >> Thank You >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From forum at epifanio.org Sat Apr 30 05:59:45 2011 From: forum at epifanio.org (2407365) Date: Fri, 29 Apr 2011 20:59:45 -0700 (PDT) Subject: [FFmpeg-user] FFmpeg from avi to 3gp Message-ID: <1304135985158-3485307.post@n4.nabble.com> Hi all, I'm new, I was trying to convert an avi file to a 3gp file, but when I try to do so with next line: ffmpeg -strict experimental -i file.avi -vcodec libx264 -s qvga -b 6438912 -r 16 -acodec aac -ab 393216 -ar 32000 -ac 1 -y file.3gp I get next message: [libx264 @ 0x9f1e7f0] broken ffmpeg default settings detected [libx264 @ 0x9f1e7f0] use an encoding preset (e.g. -vpre medium) [libx264 @ 0x9f1e7f0] preset usage: -vpre -vpre [libx264 @ 0x9f1e7f0] speed presets are listed in x264 --help [libx264 @ 0x9f1e7f0] profile is optional; x264 defaults to high Output #0, 3gp, to '3gp.3gp': Stream #0.0: Video: libx264, yuv420p, 320x240 [PAR 1:1 DAR 4:3], q=2-31, 6438 kb/s, 90k tbn, 16 tbc Stream #0.1: Audio: libfaac, 32000 Hz, 1 channels, s16, 393 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height I already installed the code x264, I'm running an Ubuntu. I'm mainly interested in converting AVI and WMV file to 3gp, 3g2 and mp4 formats and I think next are the right parameters, since I want to convert the videos to three different mobile phone formats. 3gp 3g2 mp4 bitrate 786 kb/s ? 6438912 bits 252 kb/s 785 kb/s Video format h264 mpeg4 h264 Compression yuv420p yuv420p yuv420p Size 320x240 320x240 320x240 Audio codec aac aac aac Hz 32000 24000 32000 channels stereo stereo stereo framerate 16s 16s 16s bitrate 48 kb/s -393216 bits 64 kb/s 48 kb/s I'm new, so I don't really have experience encoding any video at all, till now I just wrote a perl script (daemon) for managing the ffmpeg software automatically, but now I'm dealling with the parameters. Any help please? Thank you -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFmpeg-from-avi-to-3gp-tp3485307p3485307.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From developer at noknok.net Sat Apr 30 07:05:18 2011 From: developer at noknok.net (NokNok Developer) Date: Sat, 30 Apr 2011 01:05:18 -0400 Subject: [FFmpeg-user] BOUNTY OFFERED: AAC 5.1 -> AAC Stereo In-Reply-To: References: <4DB05895.6010600@noknok.net> <4DB9FCE9.90501@noknok.net> Message-ID: <4DBB988E.9020105@noknok.net> Bryan I guess its a matter of how its done. As one person has suggested in a private email, integrating this into libavfilter, which is a larger undertaking, but also a better solution overall. One can also simply implement a rudimentary channel mix which previous patches already exist (http://muzso.hu/2009/02/25/downsampling-multichannel-audio-5.1-into-stereo-2-channels-with-ffmpeg) , and would only need to be adapted to current code. So, yes the range of implementation could range from 2 hrs, to 20 hours. I would also say, as I do not code within the ffmpeg base or libs, I have no clue as to the task at hand, only the previous patches as reference material. Additionally, being an open source and community project, the ability I am sure to resample AAC to other channel selections would be something of a benefit to all, and not so much as a one-off task for say just me. I light of what of what I have seen in the code base (looking this past week), and other, I bumped and asked for a suggestion, and while most work is done on ffmpeg out of interest by someone (contributor), I have specifically asked that whatever work is to be done, be acceptable to the maintainers, as I would like this to be a benefit for everyone and included in the normal branch. To that end, I am not sure how many people do offer compensation for features to ffmpeg, vs just bitching in the mailing that something doesnt work, or when will ffmpeg support "x" or "y". Thru a previous discussion with another party, it was also suggested the amount should be in $1000 area. So again, the reason I asked for suggestions. You also suggest "reasonable consulting rates". If I was looking to do this, and and keep the patch to ourselves, etc, I would agree, we could just about go and find any in the know consultant with ffmpeg/libav libs, and have it done. But part of the purpose of offering the bounty is to both give back to the developers who have made ffmpeg a terrific tool for many many people, as well keep it open source for everyone. With that concept in mind, we had hoped there was someone else with the same need, capable of doing the job, and we could come to some fair amount. Soooooo... with all that being said... I'll make the bounty $1000, and if someone is interested, let me know. If not, of course we will end up going the other route and consult it out on freelancer or something, which doesnt help anyone :( but our need. Shawn On 4/29/2011 10:38 PM, Bryan Murphy wrote: > I'm sorry, but I just don't think anybody is going to take you > seriously. $250 is barely two hours of work at reasonable consulting > rates and this is seems to be a very hard problem. Maybe if you > offered 10x that amount... > > Bryan > > On Thu, Apr 28, 2011 at 6:48 PM, NokNok Developer wrote: >> Bump.. If in fact it takes more then the below amount, suggest something >> that does work. >> >> On 4/21/2011 12:17 PM, NokNok Developer wrote: >>> Over the years we have seen various patches by folks to enable various >>> forms of 6 Channel to Stereo downsampling, non of which ever were >>> implemented in the core code base for AAC (internal) processing. >>> >>> While the solution to this problem can range from a simple mixing of >>> channels, adjusting volumes, etc to more advanced, our goal is to find a >>> solution acceptable to the ffmpeg team in any form at this point. >>> >>> Many devices support AAC but in a stereo mode only vs 6 channel, etc. >>> >>> We would like to offer such a bounty to get such a solution in the code >>> base which will enable downsampling of AAC 5.1 to AAC stereo when encoding. >>> An example of two previous variations: >>> >>> >>> http://muzso.hu/2009/02/25/downsampling-multichannel-audio-5.1-into-stereo-2-channels-with-ffmpeg >>> >>> Its important that the solution be acceptable in a form which it can be >>> included in the standard code base for ffmpeg so we are not having to patch, >>> etc. Using the internal AAC encoder (-strict experimental) >>> >>> Depending on the solution we would place the bounty anywhere from >>> $50-$250. >>> >>> If you are interested in doing this, or do this and looking for payment, >>> please email ffmpegbounty [at] bitsling.com >>> >>> Not sure where best in placing this bounty, so if not on the proper list, >>> please let us know as well. >>> >>> Thank You >>> >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From leegold at speedymail.org Sat Apr 30 07:59:31 2011 From: leegold at speedymail.org (leegold at speedymail.org) Date: Fri, 29 Apr 2011 22:59:31 -0700 Subject: [FFmpeg-user] How to convert mp3 to flv Message-ID: <1304143171.2579.1446692701@webmail.messagingengine.com> Hi, Is there a way to convert an audio only mp3 file to a Flash flv file? Is there a way to add an image to the flv to show while it plays? Thank you. Lee G. From krueger at signal7.de Sat Apr 30 10:04:36 2011 From: krueger at signal7.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Sat, 30 Apr 2011 10:04:36 +0200 Subject: [FFmpeg-user] ffmpeg and x264 In-Reply-To: References: <61386.97286.qm@web86408.mail.ird.yahoo.com> <237821.66460.qm@web86407.mail.ird.yahoo.com> Message-ID: On Apr 29, 2011, at 17:37 , Dave Pope wrote: > Yeah, I hit this one too. Use "-preset" and "-profile" switches instead > of -vpre. "-preset fast -profile baseline" works OK for me. Annoying > that it's not reflected in the --help output; I don't see a way to list > the available values yet either. > Check x264 docs. As far as I understood the purpose of the change was not to duplicate efforts from the x264 project and that is documented quite well. try x264 --fullhelp From blacktrash at gmx.net Sat Apr 30 10:49:31 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Sat, 30 Apr 2011 09:49:31 +0100 Subject: [FFmpeg-user] How to convert mp3 to flv In-Reply-To: <1304143171.2579.1446692701@webmail.messagingengine.com> References: <1304143171.2579.1446692701@webmail.messagingengine.com> Message-ID: <20110430084931.GC672@krille.blacktrash.org> * leegold at speedymail.org on Friday, April 29, 2011 at 22:59:31 -0700 > Is there a way to convert an audio only mp3 file to a Flash flv file? Is > there a way to add an image to the flv to show while it plays? Thank > you. I use something like this to create a fake flv with a minimal image clip when I want to pseudostream an mp3: ffmpeg -shortest -loop_input -f image2 -i -i \ -f flv -acodec copy -vb 10k -s 10x10 out.flv \ && flvmeta out.flv out-meta.flv c -- Was hei?t hier Dogma, ich bin Underdogma! [ What the hell do you mean dogma, I am underdogma. ] free movies --->>> http://www.blacktrash.org/underdogma http://itunes.apple.com/podcast/underdogma-movies/id363423596 From stefano.sabatini-lala at poste.it Sat Apr 30 13:38:24 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sat, 30 Apr 2011 13:38:24 +0200 Subject: [FFmpeg-user] BOUNTY OFFERED: AAC 5.1 -> AAC Stereo In-Reply-To: <4DBB988E.9020105@noknok.net> References: <4DB05895.6010600@noknok.net> <4DB9FCE9.90501@noknok.net> <4DBB988E.9020105@noknok.net> Message-ID: <20110430113824.GA17356@geppetto> On date Saturday 2011-04-30 01:05:18 -0400, NokNok Developer encoded: > Bryan > > I guess its a matter of how its done. As one person has suggested > in a private email, integrating this into libavfilter, which is a > larger undertaking, but also a better solution overall. One can > also simply implement a rudimentary channel mix which previous > patches already exist (http://muzso.hu/2009/02/25/downsampling-multichannel-audio-5.1-into-stereo-2-channels-with-ffmpeg) > , and would only need to be adapted to current code. So, yes the > range of implementation could range from 2 hrs, to 20 hours. > > I would also say, as I do not code within the ffmpeg base or libs, I > have no clue as to the task at hand, only the previous patches as > reference material. Additionally, being an open source and > community project, the ability I am sure to resample AAC to other > channel selections would be something of a benefit to all, and not > so much as a one-off task for say just me. > > I light of what of what I have seen in the code base (looking this > past week), and other, I bumped and asked for a suggestion, and > while most work is done on ffmpeg out of interest by someone > (contributor), I have specifically asked that whatever work is to be > done, be acceptable to the maintainers, as I would like this to be a > benefit for everyone and included in the normal branch. To that > end, I am not sure how many people do offer compensation for > features to ffmpeg, vs just bitching in the mailing that something > doesnt work, or when will ffmpeg support "x" or "y". Thru a > previous discussion with another party, it was also suggested the > amount should be in $1000 area. So again, the reason I asked for > suggestions. > > You also suggest "reasonable consulting rates". If I was looking to > do this, and and keep the patch to ourselves, etc, I would agree, we > could just about go and find any in the know consultant with > ffmpeg/libav libs, and have it done. But part of the purpose of > offering the bounty is to both give back to the developers who have > made ffmpeg a terrific tool for many many people, as well keep it > open source for everyone. With that concept in mind, we had hoped > there was someone else with the same need, capable of doing the job, > and we could come to some fair amount. > > Soooooo... with all that being said... I'll make the bounty $1000, > and if someone is interested, let me know. If not, of course we > will end up going the other route and consult it out on freelancer > or something, which doesnt help anyone :( but our need. > > Shawn Please file an issue on the FFmpeg issue tracker: https://ffmpeg.org/trac/ffmpeg/report mentioning that you're willing to sponsor the change and which are your timeline constraints. That looks like a change needing some major rework and IIRC that's a long time standing issue, but maybe there is some hope for getting it fixed properly (and I have an almost working audio filtering branch which could possibly help). Thanks, regard. From d_elia at hotmail.com Thu Apr 28 13:30:55 2011 From: d_elia at hotmail.com (Daniele Elia) Date: Thu, 28 Apr 2011 13:30:55 +0200 Subject: [FFmpeg-user] use ffmpeg to create stremable content In-Reply-To: References: Message-ID: Hi all, I'm trying to convert a video file to mp4 with h264 codec but I'm having a problem with the result. The problem is that resulting file seams not to be streamable with some devices like android 2.2. Can someone help me ? This is my command line: ffmpeg -i inputFile.mp4 -y -f mp4 -vcodec libx264 -vpre default -crf 22 -maxrate 1000k -minrate 700k -acodec libfaac -ab 128k -ac 2 outputFile.mp4 Thanks -- Daniele Elia d_elia at hotmail.com From ludovic.artru at etu.univ-lyon1.fr Thu Apr 28 17:31:03 2011 From: ludovic.artru at etu.univ-lyon1.fr (ARTRU LUDOVIC p0802080) Date: Thu, 28 Apr 2011 17:31:03 +0200 Subject: [FFmpeg-user] Question about the LGPL Message-ID: <3F0605D5DEBC204F91753FF87A147DC9022ED5F551@BV-LAME-13.univ-lyon1.fr> Hello I'm a French student doing a work experience placement and I have to program a software using videos. But the videos are in .avi and I have to convert it to .flv That's why I found your software. I download it with Windows XP and it works (I haven't already writte a line of code) But the finality of my work experience placement is commercial software. I have some question about the right of using your software. Can I use the ffmpeg.exe if I only use it to convert videos to .flv? (The software will be write in VB.net). Thanks. ARTRU Ludovic IUT Informatique 2G2 IEM From tobias_petry at hotmail.com Thu Apr 28 20:42:50 2011 From: tobias_petry at hotmail.com (TobiasP) Date: Thu, 28 Apr 2011 11:42:50 -0700 (PDT) Subject: [FFmpeg-user] libx264 keyframes ? Message-ID: <1304016170099-3481848.post@n4.nabble.com> Hi, i have got an m2ts h264 fullhd camera video with the only keyframe at frame 1, and want to cut hundreds of small parts of this video. For fast time completion i first transcode a small chunk (skipping video to some point, transcoding a few seconds with keyframes every full second): ffmpeg -ss 300 -i video.m2ts -r 25 -g 25 -t 15 -b 750k -s 480x352 -vpre libx264-medium -acodec copy -threads 0 -y video_part1_tmp.mp4 The problem now is, that the first keyframe in video_part1_tmp.mp4 is not at frame 1 and then every full second - as i excepted and a 2 years old built did. Did i make anything wrong or is there any special trick to accomplish having frames 1, 26, 51, ... as keyframe for correct cutting? At the moment the keyframes are at position 17 + i * 25 Tobias -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/libx264-keyframes-tp3481848p3481848.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From tobias_petry at hotmail.com Thu Apr 28 20:43:42 2011 From: tobias_petry at hotmail.com (TobiasP) Date: Thu, 28 Apr 2011 11:43:42 -0700 (PDT) Subject: [FFmpeg-user] libx264 keyframes ? In-Reply-To: <1304016170099-3481848.post@n4.nabble.com> References: <1304016170099-3481848.post@n4.nabble.com> Message-ID: <1304016222969-3481849.post@n4.nabble.com> Forget to mention, i am using the latest HawkEye [1] build. [1] http://hawkeye.arrozcru.org/ -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/libx264-keyframes-tp3481848p3481849.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From h.reindl at thelounge.net Sat Apr 30 13:47:29 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sat, 30 Apr 2011 13:47:29 +0200 Subject: [FFmpeg-user] Question about the LGPL In-Reply-To: <3F0605D5DEBC204F91753FF87A147DC9022ED5F551@BV-LAME-13.univ-lyon1.fr> References: <3F0605D5DEBC204F91753FF87A147DC9022ED5F551@BV-LAME-13.univ-lyon1.fr> Message-ID: <4DBBF6D1.9050505@thelounge.net> Am 28.04.2011 17:31, schrieb ARTRU LUDOVIC p0802080: > Hello > I'm a French student doing a work experience placement and I have to program a software using videos. > But the videos are in .avi and I have to convert it to .flv > That's why I found your software. > I download it with Windows XP and it works (I haven't already writte a line of code) > But the finality of my work experience placement is commercial software. > I have some question about the right of using your software. > Can I use the ffmpeg.exe if I only use it to convert videos to .flv? > (The software will be write in VB.net) sure, especially if you use the unmodified source because the LGPL/GPL affects linking/code and not the call of a independent binary with LGPL you can even use the unmodified source-headers for linking in your application code, with the GPL in this case your software must be GPL too, this is the difference of LGPL -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature URL: From Dave.Pope at myxer.com Sat Apr 30 17:39:25 2011 From: Dave.Pope at myxer.com (Dave Pope) Date: Sat, 30 Apr 2011 11:39:25 -0400 Subject: [FFmpeg-user] ffmpeg and x264 References: <61386.97286.qm@web86408.mail.ird.yahoo.com><237821.66460.qm@web86407.mail.ird.yahoo.com> Message-ID: Thanks! The change makes perfect sense, I just had to dig around for while before figuring it out. Do you know if there's a patch in the works to make 'ffmpeg --help' report something about these? It seems like it might require building some "handshaking" between them that might not already exist in general form. ________________________________ From: ffmpeg-user-bounces at ffmpeg.org on behalf of Robert Kr?ger Sent: Sat 4/30/2011 4:04 AM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] ffmpeg and x264 On Apr 29, 2011, at 17:37 , Dave Pope wrote: > Yeah, I hit this one too. Use "-preset" and "-profile" switches instead > of -vpre. "-preset fast -profile baseline" works OK for me. Annoying > that it's not reflected in the --help output; I don't see a way to list > the available values yet either. > Check x264 docs. As far as I understood the purpose of the change was not to duplicate efforts from the x264 project and that is documented quite well. try x264 --fullhelp _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4254 bytes Desc: not available URL: From leo.izen at gmail.com Sat Apr 30 19:06:47 2011 From: leo.izen at gmail.com (Leo Izen) Date: Sat, 30 Apr 2011 13:06:47 -0400 Subject: [FFmpeg-user] FFmpeg from avi to 3gp In-Reply-To: <1304135985158-3485307.post@n4.nabble.com> References: <1304135985158-3485307.post@n4.nabble.com> Message-ID: On Fri, Apr 29, 2011 at 11:59 PM, 2407365 wrote: > > [libx264 @ 0x9f1e7f0] broken ffmpeg default settings detected > [libx264 @ 0x9f1e7f0] use an encoding preset (e.g. -vpre medium) > [libx264 @ 0x9f1e7f0] preset usage: -vpre -vpre > [libx264 @ 0x9f1e7f0] speed presets are listed in x264 --help > [libx264 @ 0x9f1e7f0] profile is optional; x264 defaults to high > Just like libx264 says, use and encoding preset. This means specify -vpre after specifying -vcodec libx264. The presets are ultrafast, veryfast, faster, fast, medium, slow, slower, veryslow, and placebo. Also, to constrain to the main profile instead of high use -vpre main AFTER the speed preset, and to constrain to the baseline profile use -vpre baseline AFTER the speed preset. Slower speeds yield better quality for the same bitrate, but placebo is pointlessly slow and has a less than 1% visual difference from veryslow, so use veryslow instead of placebo in any case. Also, when use x264, there's a unique option -crf, to set quality. -crf 15 is virtually lossless, but as the number increases it gets more visual, so -crf 25 has noticeable compression artifacts. You still need to specify a preset, but with -crf generally better quality/bitrate is achieved. Slower speeds with crf give a lower bitrate for the same quality. So: an example is: ffmpeg -strict experimental -i file.avi -vcodec libx264 -vpre slow -s qvga -b 6438912 -r 16 -acodec aac -ab 393216 -ar 32000 -ac 1 -y file.3gp Notice the -vpre slow in there. Also, 6.4M is a waaaaay unnecessarily high bitrate for a video with a frame rate of 16 and qvga size, and 393k is a waaaaay huge audio bitrate for 32 kHz single channel audio. Consider reducing these, and quality won't really drop at all.