From atskiisotona at gmail.com Mon Aug 1 08:35:05 2011 From: atskiisotona at gmail.com (Dmitry Gorbunov) Date: Sun, 31 Jul 2011 23:35:05 -0700 (PDT) Subject: [FFmpeg-user] Web camera and microphone capture via dshow Message-ID: <1312180505914-3708971.post@n4.nabble.com> I'm trying to capture and encode video/audio from camera and microphone using libavcodec, libavformat and libavdevice. It works, video and audio captures, but after encoding it all is messed: everything is extremely speed up (like you recorded it at 25 fps and then played at 100). I think I don't get something about all that pts/dts stuff and synchronization, but I honestly don't know where to look. ffmpeg unfortunately is very poorly documented in terms of programming (why? it's so mature, why someone won't write at least how library works?). I took muxing example from ffmpeg git and modified it to feed encoders with data from webcam and microphone using av_read_frame and av_decode_video/audio. Any ideas? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Web-camera-and-microphone-capture-via-dshow-tp3708971p3708971.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rhodri at kynesim.co.uk Mon Aug 1 15:21:17 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Mon, 01 Aug 2011 14:21:17 +0100 Subject: [FFmpeg-user] Web camera and microphone capture via dshow In-Reply-To: <1312180505914-3708971.post@n4.nabble.com> References: <1312180505914-3708971.post@n4.nabble.com> Message-ID: On Mon, 01 Aug 2011 07:35:05 +0100, Dmitry Gorbunov wrote: > I'm trying to capture and encode video/audio from camera and microphone > using > libavcodec, libavformat and libavdevice. It works, video and audio > captures, > but after encoding it all is messed: everything is extremely speed up > (like > you recorded it at 25 fps and then played at 100). I think I don't get > something about all that pts/dts stuff and synchronization, but I > honestly > don't know where to look. ffmpeg unfortunately is very poorly documented > in > terms of programming (why? it's so mature, why someone won't write at > least > how library works?). I took muxing example from ffmpeg git and modified > it > to feed encoders with data from webcam and microphone using av_read_frame > and av_decode_video/audio. Any ideas? Check the clock speed of your web cam. I've seen some that clock at 30kHz rather than 90KHz, which will mess up your PTS calculations. -- Rhodri James Kynesim Ltd From rhodri at kynesim.co.uk Mon Aug 1 16:11:25 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Mon, 01 Aug 2011 15:11:25 +0100 Subject: [FFmpeg-user] Web camera and microphone capture via dshow In-Reply-To: <1312180505914-3708971.post@n4.nabble.com> References: <1312180505914-3708971.post@n4.nabble.com> Message-ID: On Mon, 01 Aug 2011 07:35:05 +0100, Dmitry Gorbunov wrote: > I'm trying to capture and encode video/audio from camera and microphone > using > libavcodec, libavformat and libavdevice. It works, video and audio > captures, > but after encoding it all is messed: everything is extremely speed up > (like > you recorded it at 25 fps and then played at 100). I think I don't get > something about all that pts/dts stuff and synchronization, but I > honestly > don't know where to look. ffmpeg unfortunately is very poorly documented > in > terms of programming (why? it's so mature, why someone won't write at > least > how library works?). I took muxing example from ffmpeg git and modified > it > to feed encoders with data from webcam and microphone using av_read_frame > and av_decode_video/audio. Any ideas? Check the clock speed of your web cam. I've seen some that clock at 30kHz rather than 90KHz, which will mess up your PTS calculations. -- Rhodri James Kynesim Ltd From atskiisotona at gmail.com Mon Aug 1 17:31:49 2011 From: atskiisotona at gmail.com (Dmitry Gorbunov) Date: Mon, 1 Aug 2011 08:31:49 -0700 (PDT) Subject: [FFmpeg-user] Web camera and microphone capture via dshow In-Reply-To: References: <1312180505914-3708971.post@n4.nabble.com> Message-ID: <1312212709947-3709999.post@n4.nabble.com> Rhodri James wrote: > > Check the clock speed of your web cam. I've seen some that clock at > 30kHz rather than 90KHz, which will mess up your PTS calculations. > Thank you, that would be a good hint if I only knew how and where to calculate all that pts/dts stuff and where to put it. I don't understand the whole process of 1. Take video stream from webcam (av_get_frame?) - probably here some pts stuff 2. Re-encode it (av_decode_video/av_encode_video?) - and here 3. Write to file/stream over the network (av_interleaved_write_frame?) I'm going to dig in ffmpeg.c a bit more, but it's really hard to guess what's going on, because there are very few comments. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Web-camera-and-microphone-capture-via-dshow-tp3708971p3709999.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From sreemnpy at gmail.com Mon Aug 1 11:49:07 2011 From: sreemnpy at gmail.com (sreerag r) Date: Mon, 1 Aug 2011 15:19:07 +0530 Subject: [FFmpeg-user] Error in ffmpeg-git-78accb8 Message-ID: Hi, When i am compiling ffmpeg-git-78accb8 in MinGW, i am getting the following errors. errors: " 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." Please help me..how i can solve this problem? with thanks and regards Sreerag R From thomas at pixelpartner.de Mon Aug 1 23:13:42 2011 From: thomas at pixelpartner.de (Thomas Kumlehn) Date: Mon, 1 Aug 2011 23:13:42 +0200 Subject: [FFmpeg-user] subtitle update In-Reply-To: <7E7A6FF0-CC0D-46A7-B362-8BD0DE8EB0AE@gmail.com> References: <9AB93297-4166-41A3-9BDD-A515E76E7291@gmail.com> <1312009154.8162.YahooMailNeo@web86408.mail.ird.yahoo.com> <7E7A6FF0-CC0D-46A7-B362-8BD0DE8EB0AE@gmail.com> Message-ID: <765C6103-E624-4764-BDBA-89BB1BC1AF5D@pixelpartner.de> On the windows platform I worked with a combination of AviSynth and ffmpeg. AviSynth has established subtitle support and ffmpeg configured with --enable-avisynth can process these *.avs scripts. Just a hint, because I didn't do hard subs yet. Thomas Kumlehn PIXEL PARTNER (R) Send from my iPad 3-D http://www.pixelpartner.de Am 30.07.2011 um 10:38 schrieb "Rick C." : > > On Jul 30, 2011, at 2:59 PM, JULIAN GARDNER wrote: > >> >> >> >> >> ----- Original Message ----- >>> From: Rick C. >>> To: FFmpeg user questions and RTFMs >>> Cc: >>> Sent: Friday, 29 July 2011, 12:13 >>> Subject: [FFmpeg-user] subtitle update >>> >>> Hello, >>> >>> I just wanted to see if there was any update to hard-coding subtitles (like >>> .srt) into an encoding? So let's say going from .mkv to .avi I would like >>> to hard-code the subtitles in the .avi. I tried to see if drawtext would do it >>> and while I was able to get text to appear I didn't see any way to actually >>> deal with an entire subtitle file. I've seen some talking recently in the >>> developer list about this that's why I thought I'd ask. If there's >>> something experimental I'd be glad to give it a try. Thanks! >>> >>> rc >>> _______________________________________________ >>> ffmpeg-user mailing list >>> ffmpeg-user at ffmpeg.org >>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >>> >> Iv been testing this for the last month and everything seems to be working fine, but there are some caveats >> >> 1, DVB Subtitles - Working >> 2. DIVX - Working, and can be transcoded to either Hard Subs or DVB Subtitles >> 3. MKV - Cant test as i cannot get ASS working >> 4. AVI/SRT/SUB - Problems are that the SRT/SUB file give the wrong Timestamp so they appear at the wrong place, As an example, the IDX says 2.26.880 but the sub file says it starts earlier. As the index file is not used i would need to add in some offsetting >> >> Also a couple of more things ive asked about but not had any answers/ideas is that when transcoding to DVB Subtitles and the screen is not a full 720x576, which is what DVB Subs wants, you get subtitles off screen, so not viewable >> >> joolz > > Thank you for the reply. So at this time it looks like this feature will be included in the future (there's no way to access this now that I know of). That's great and all the best on this project! :-) > > rc > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From joolzg at btinternet.com Mon Aug 1 23:36:11 2011 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Mon, 1 Aug 2011 22:36:11 +0100 (BST) Subject: [FFmpeg-user] subtitle update In-Reply-To: <765C6103-E624-4764-BDBA-89BB1BC1AF5D@pixelpartner.de> References: <9AB93297-4166-41A3-9BDD-A515E76E7291@gmail.com> <1312009154.8162.YahooMailNeo@web86408.mail.ird.yahoo.com> <7E7A6FF0-CC0D-46A7-B362-8BD0DE8EB0AE@gmail.com> <765C6103-E624-4764-BDBA-89BB1BC1AF5D@pixelpartner.de> Message-ID: <1312234571.90166.YahooMailNeo@web86406.mail.ird.yahoo.com> >________________________________ >From: Thomas Kumlehn >To: FFmpeg user questions and RTFMs >Sent: Monday, 1 August 2011, 22:13 >Subject: Re: [FFmpeg-user] subtitle update > >On the windows platform I worked with a combination of AviSynth and ffmpeg. >AviSynth has established subtitle support and ffmpeg configured with --enable-avisynth can process these *.avs scripts. >Just a hint, because I didn't do hard subs yet. > >Thomas Kumlehn >PIXEL PARTNER (R) > >Send from my iPad 3-D >http://www.pixelpartner.de > >Am 30.07.2011 um 10:38 schrieb "Rick C." : > >> >> On Jul 30, 2011, at 2:59 PM, JULIAN GARDNER wrote: >> >>> >>> >>> >>> >>> ----- Original Message ----- >>>> From: Rick C. >>>> To: FFmpeg user questions and RTFMs >>>> Cc: >>>> Sent: Friday, 29 July 2011, 12:13 >>>> Subject: [FFmpeg-user] subtitle update >>>> >>>> Hello, >>>> >>>> I just wanted to see if there was any update to hard-coding subtitles (like >>>> .srt) into an encoding?? So let's say going from .mkv to .avi I would like >>>> to hard-code the subtitles in the .avi.? I tried to see if drawtext would do it >>>> and while I was able to get text to appear I didn't see any way to actually >>>> deal with an entire subtitle file.? I've seen some talking recently in the >>>> developer list about this that's why I thought I'd ask.? If there's >>>> something experimental I'd be glad to give it a try.? Thanks! >>>> >>>> rc >>>> _______________________________________________ >>>> ffmpeg-user mailing list >>>> ffmpeg-user at ffmpeg.org >>>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >>>> >>> Iv been testing this for the last month and everything seems to be working fine, but there are some caveats >>> >>> 1, DVB Subtitles - Working >>> 2. DIVX? ? ? ? ? ? ???- Working, and can be transcoded to either Hard Subs or DVB Subtitles >>> 3. MKV? ? ? ? ? ? ? ? - Cant test as i cannot get ASS working >>> 4. AVI/SRT/SUB - Problems are that the SRT/SUB file give the wrong Timestamp so they appear at the wrong place, As an example, the IDX says 2.26.880 but the sub file says it starts earlier. As the index file is not used i would need to add in some offsetting >>> >>> Also a couple of more things ive asked about but not had any answers/ideas is that when transcoding to DVB Subtitles and the screen is not a full 720x576, which is what DVB Subs wants, you get subtitles off screen, so not viewable >>> >>> joolz >> >> Thank you for the reply.? So at this time it looks like this feature will be included in the future (there's no way to access this now that I know of).? That's great and all the best on this project! :-) >> >> rc >> >> _______________________________________________ >> 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 > As stated before Hard Subs works, but there are come caveats. If people can put up with it then ok, but what i dont want is to post a patch then have loads of reports of problems, which i already know about. joolz ?states From rickcorteza at gmail.com Tue Aug 2 07:23:38 2011 From: rickcorteza at gmail.com (Rick C.) Date: Tue, 2 Aug 2011 13:23:38 +0800 Subject: [FFmpeg-user] merging or joining videos Message-ID: Hello, I'm wondering if someone can point me in right direction about how I should join together multiple videos? So if I have video1.avi & video2.avi how do I turn them into joinedvideo.avi? It doesn't matter to me if I join them after converting or during converting (I'm assuming I have to do it during?). I'm familiar with cat but I believe this only works on .mpg and I would like a way to work on many files such as .avi or .mp4. I've also read the instructions in the documentation but I was thinking there's a more updated way than having to convert everything to .mpg first? Anyways, if someone could point me in the right direction that would be great. Thanks! rc From stefano.sabatini-lala at poste.it Tue Aug 2 14:48:56 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Tue, 2 Aug 2011 14:48:56 +0200 Subject: [FFmpeg-user] Error in ffmpeg-git-78accb8 In-Reply-To: References: Message-ID: <20110802124856.GA11760@geppetto> On date Monday 2011-08-01 15:19:07 +0530, sreerag r encoded: > Hi, > When i am compiling ffmpeg-git-78accb8 in MinGW, i am getting the > following errors. > > errors: > " 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." > > Please help me..how i can solve this problem? Where is your config.log? Did you check that you have gcc installed? (what's the output of gcc --version?) From stefano.sabatini-lala at poste.it Tue Aug 2 14:51:07 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Tue, 2 Aug 2011 14:51:07 +0200 Subject: [FFmpeg-user] merging or joining videos In-Reply-To: References: Message-ID: <20110802125107.GB11760@geppetto> On date Tuesday 2011-08-02 13:23:38 +0800, Rick C. encoded: > Hello, > > I'm wondering if someone can point me in right direction about how I > should join together multiple videos? So if I have video1.avi & > video2.avi how do I turn them into joinedvideo.avi? It doesn't > matter to me if I join them after converting or during converting > (I'm assuming I have to do it during?). I'm familiar with cat but I > believe this only works on .mpg and I would like a way to work on > many files such as .avi or .mp4. I've also read the instructions in > the documentation but I was thinking there's a more updated way than > having to convert everything to .mpg first? Anyways, if someone > could point me in the right direction that would be great. Thanks! In general the response is "there is no generic way", the best that you can achieve right now is to convert to a cattable format (mpeg, rawvideo) and physically cat or use the concat protocol. Logical join is not yet possible, although there is a gsoc task concerning this (can't say about the status of the task itself). -- ffmpeg-user random tip #5 FFmpeg documentation: http://www.ffmpeg.org/documentation.html From snthdiueoa at gmail.com Tue Aug 2 17:02:39 2011 From: snthdiueoa at gmail.com (Tom) Date: Tue, 2 Aug 2011 17:02:39 +0200 Subject: [FFmpeg-user] [lavc rc] Error: 2pass curve failed to converge Message-ID: Hi, I'm trying to convert a series of jpegs to a movie, and I get an error (log below). I've done this earlier, without issues, for other jpegs, but for this particular set, i get the error. I don't know how. I'm using the latest version of ffmpeg (i built it last week from git sources), on ubuntu 8.04. My images can be downloaded on: http://www.compulated.info/ffmpegtest_images.zip (6 MB) The error I get is (see pass 2): ubuntu at ip-10-244-10-134:~$ ffmpeg -i 7-%4d.jpg -t 1 -r 25 -b 4000k -f mpeg -vcodec mpeg2video -an -pass 1 -passlogfile pass.txt -y video_14.mpg 2>&1 ffmpeg version git-Thu Jul 28 14:59:54 2011 +0200-956c901, Copyright (c) 2000-2011 the FFmpeg developers built on Jul 28 2011 15:55:33 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libxvid --enable-x11grab --enable-pthreads libavutil 51. 11. 1 / 51. 11. 1 libavcodec 53. 9. 0 / 53. 9. 0 libavformat 53. 6. 0 / 53. 6. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 27. 3 / 2. 27. 3 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, image2, from '7-%4d.jpg': Duration: 00:00:01.00, start: 0.000000, bitrate: N/A Stream #0.0: Video: mjpeg, yuvj444p, 640x480 [SAR 1:1 DAR 4:3], 25 fps, 25 tbr, 25 tbn, 25 tbc Unrecognized option 'passlogfile' Incompatible pixel format 'yuvj444p' for codec 'mpeg2video', auto-selecting format 'yuv420p' [buffer @ 0x8de0880] w:640 h:480 pixfmt:yuvj444p tb:1/1000000 sar:1/1 sws_param: [buffersink @ 0x8de6de0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x8de7160] w:640 h:480 fmt:yuvj444p -> w:640 h:480 fmt:yuv420p flags:0x4 [mpeg @ 0x8ddf500] VBV buffer size not set, muxing may fail Output #0, mpeg, to 'video_14.mpg': Metadata: encoder : Lavf53.6.0 Stream #0.0: Video: mpeg2video, yuv420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, pass 1, 4000 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop, [?] for help frame= 25 fps= 0 q=4.4 Lsize= 680kB time=00:00:00.96 bitrate=5802.7kbits/s video:677kB audio:0kB global headers:0kB muxing overhead 0.481541% ubuntu at ip-10-244-10-134:~$ ffmpeg -i 7-%4d.jpg -t 1 -r 25 -b 4000k -f mpeg -vcodec mpeg2video -an -pass 2 -passlogfile pass.txt -y video_14.mpg 2>&1 ffmpeg version git-Thu Jul 28 14:59:54 2011 +0200-956c901, Copyright (c) 2000-2011 the FFmpeg developers built on Jul 28 2011 15:55:33 with gcc 4.2.4 (Ubuntu 4.2.4-1ubuntu4) configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libxvid --enable-x11grab --enable-pthreads libavutil 51. 11. 1 / 51. 11. 1 libavcodec 53. 9. 0 / 53. 9. 0 libavformat 53. 6. 0 / 53. 6. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 27. 3 / 2. 27. 3 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, image2, from '7-%4d.jpg': Duration: 00:00:01.00, start: 0.000000, bitrate: N/A Stream #0.0: Video: mjpeg, yuvj444p, 640x480 [SAR 1:1 DAR 4:3], 25 fps, 25 tbr, 25 tbn, 25 tbc Unrecognized option 'passlogfile' Incompatible pixel format 'yuvj444p' for codec 'mpeg2video', auto-selecting format 'yuv420p' [buffer @ 0x8de0880] w:640 h:480 pixfmt:yuvj444p tb:1/1000000 sar:1/1 sws_param: [buffersink @ 0x8de6de0] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 0x8de7160] w:640 h:480 fmt:yuvj444p -> w:640 h:480 fmt:yuv420p flags:0x4 [mpeg2video @ 0x8ddf9e0] [lavc rc] Error: 2pass curve failed to converge Output #0, mpeg, to 'video_14.mpg': Stream #0.0: Video: mpeg2video, yuv420p, 640x480 [SAR 1:1 DAR 4:3], q=2-31, pass 2, 4000 kb/s, 90k tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height Can anyone help? I've searched google, but all errors of this kind were solved in 2006-2007 apparently, so it doesn't seem to be valid for me... Cheers, Tom From rickcorteza at gmail.com Wed Aug 3 05:22:56 2011 From: rickcorteza at gmail.com (Rick C.) Date: Wed, 3 Aug 2011 11:22:56 +0800 Subject: [FFmpeg-user] merging or joining videos In-Reply-To: <20110802125107.GB11760@geppetto> References: <20110802125107.GB11760@geppetto> Message-ID: <2923D23E-3E1F-46AA-AFFE-5C151EA19B95@gmail.com> On Aug 2, 2011, at 8:51 PM, Stefano Sabatini wrote: > On date Tuesday 2011-08-02 13:23:38 +0800, Rick C. encoded: >> Hello, >> >> I'm wondering if someone can point me in right direction about how I >> should join together multiple videos? So if I have video1.avi & >> video2.avi how do I turn them into joinedvideo.avi? It doesn't >> matter to me if I join them after converting or during converting >> (I'm assuming I have to do it during?). I'm familiar with cat but I >> believe this only works on .mpg and I would like a way to work on >> many files such as .avi or .mp4. I've also read the instructions in >> the documentation but I was thinking there's a more updated way than >> having to convert everything to .mpg first? Anyways, if someone >> could point me in the right direction that would be great. Thanks! > > In general the response is "there is no generic way", the best that > you can achieve right now is to convert to a cattable format (mpeg, > rawvideo) and physically cat or use the concat protocol. > > Logical join is not yet possible, although there is a gsoc task > concerning this (can't say about the status of the task itself). > -- > ffmpeg-user random tip #5 > FFmpeg documentation: > http://www.ffmpeg.org/documentation.html > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user Thanks for that I will work on this! From bfallik at bamboom.com Wed Aug 3 06:21:16 2011 From: bfallik at bamboom.com (Brian Fallik) Date: Wed, 3 Aug 2011 00:21:16 -0400 Subject: [FFmpeg-user] av_interleaved_write_frame(): Operation not permitted. Message-ID: Hi, I'm trying to use ffmpeg to strip one audio stream from a mpegts file containing 1 video streams and 2 audio. The ffprobe output from the input file is: [mpegts @ 0x1b702a0] max_analyze_duration reached Input #0, mpegts, from 'in.ts': Duration: 00:02:45.97, start: 69498.987267, bitrate: 1819 kb/s Program 1 Stream #0.0[0x44]: Video: h264 (High), yuv420p, 720x720 [PAR 16:9 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0.1[0x45]: Audio: aac, 44100 Hz, mono, s16, 81 kb/s Stream #0.2[0x46]: Audio: aac, 44100 Hz, mono, s16, 79 kb/s When I run: $ ffmpeg -i in.ts -map 0.0 -map 0.1 -acodec copy -vcodec copy out.ts I get an error: [mpegts @ 0x1ced340] aac bitstream not in adts format and extradata missing. av_interleaved_write_frame(): Operation not permitted. Any ideas on what I'm doing wrong? If the problem is likely in the input file, any ideas on what might be wrong with it? The input file is captured from a third-party proprietary transcoder. Thanks, brian From pb at das-werkstatt.com Wed Aug 3 10:51:53 2011 From: pb at das-werkstatt.com (Peter B.) Date: Wed, 03 Aug 2011 10:51:53 +0200 Subject: [FFmpeg-user] merging or joining videos In-Reply-To: References: Message-ID: <4E390C29.6040603@das-werkstatt.com> Rick C. wrote: > I'm familiar with cat but I believe this only works on .mpg and I would like a way to work on many files such as .avi or .mp4. I'm not 100% sure, but I think it also depends on which codec you're using (.avi is just the container). > I've also read the instructions in the documentation but I was thinking there's a more updated way than having to convert everything to .mpg first? Anyways, if someone could point me in the right direction that would be great. > I've also had the same problem: All documentation I found mentioned MPEGs and "cat" or yuv2mpeg for piping things together :( This doesn't work at all for AVI files. I found a solution that even works bitproof for audio and video, by using "mencoder" to merge the videos back together. We're using this method daily in the Austrian national video archive (=?sterreichische Mediathek [1][2]) - but only with intraframe codecs (FFv1 / DV) and PCM for audio. Additionally, it's also possible to merge *and* transcode the files in one step! :) Details can be found in a forum post on "das-werkstatt.com" [3]. Additionally, there's an article on "linuxreview.org" about merging video files [4] - but I must say that I haven't had a good experience with "avimerge". Good luck! Peter B. == References: [1] http://www.mediathek.at/ [2] http://sourceforge.net/projects/dva-profession/ [3] http://www.das-werkstatt.com/forum/werkstatt/viewtopic.php?f=7&t=1793 [4] http://en.linuxreviews.org/HOWTO_Merge_video_files From x2305andy2305x at yahoo.com Wed Aug 3 10:57:17 2011 From: x2305andy2305x at yahoo.com (Andy Andy) Date: Wed, 3 Aug 2011 01:57:17 -0700 (PDT) Subject: [FFmpeg-user] Has anything happened with loop_input? Message-ID: <1312361837.43049.YahooMailNeo@web32507.mail.mud.yahoo.com> Hi, I'm trying to create an mjpeg from a still image by looping input. Using the same command i've been using for a long time, but now anything i do results in a 1 frame output ffmpeg -y -loop_input -vframes 10 -r 25 -i ~/media_samples/arminFrame.bmp Desktop/MJPEGGED.mjpeg What i did do is to upgrade ffmpeg to latest git version because of a bug in rotating inputs to conver to mjepg output that i've posted and was fixed. What could be wrong? Regards, DAV From kct.venkat at gmail.com Wed Aug 3 11:25:38 2011 From: kct.venkat at gmail.com (Venkatasubramaniam R) Date: Wed, 3 Aug 2011 11:25:38 +0200 Subject: [FFmpeg-user] merging or joining videos In-Reply-To: <4E390C29.6040603@das-werkstatt.com> References: <4E390C29.6040603@das-werkstatt.com> Message-ID: Hi, Once I had used MP4Box package in Linux for concatenating smaller mp4 files to get a big one. It worked pretty well, but I had not tried it on AVI files. May be you could try MP4Box (which comes as a command line tool when installing gpac package : http://gpac.wp.institut-telecom.fr/mp4box/). All the best, -Venkat. On Wed, Aug 3, 2011 at 10:51 AM, Peter B. wrote: > Rick C. wrote: > > I'm familiar with cat but I believe this only works on .mpg and I would > like a way to work on many files such as .avi or .mp4. > I'm not 100% sure, but I think it also depends on which codec you're > using (.avi is just the container). > > > I've also read the instructions in the documentation but I was thinking > there's a more updated way than having to convert everything to .mpg first? > Anyways, if someone could point me in the right direction that would be > great. > > > I've also had the same problem: All documentation I found mentioned > MPEGs and "cat" or yuv2mpeg for piping things together :( This doesn't > work at all for AVI files. > > I found a solution that even works bitproof for audio and video, by > using "mencoder" to merge the videos back together. We're using this > method daily in the Austrian national video archive (=?sterreichische > Mediathek [1][2]) - but only with intraframe codecs (FFv1 / DV) and PCM > for audio. Additionally, it's also possible to merge *and* transcode the > files in one step! :) > Details can be found in a forum post on "das-werkstatt.com" [3]. > > Additionally, there's an article on "linuxreview.org" about merging > video files [4] - but I must say that I haven't had a good experience > with "avimerge". > > Good luck! > Peter B. > > == References: > [1] http://www.mediathek.at/ > [2] http://sourceforge.net/projects/dva-profession/ > [3] http://www.das-werkstatt.com/forum/werkstatt/viewtopic.php?f=7&t=1793 > [4] http://en.linuxreviews.org/HOWTO_Merge_video_files > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From sreemnpy at gmail.com Wed Aug 3 12:17:13 2011 From: sreemnpy at gmail.com (sreerag r) Date: Wed, 3 Aug 2011 15:47:13 +0530 Subject: [FFmpeg-user] How to add 'dshow' format in FFMPEG? Message-ID: Hi, Now i am working with FFMPEG. I compiled it on windows platform. But that compiled version is missing with the format 'dshow'. How we can add 'dshow' format to this FFMPEG. If anyone knows the answer please help me... With Thanks and Regards Sreerag R From batguano999 at hotmail.com Wed Aug 3 16:53:45 2011 From: batguano999 at hotmail.com (bat guano) Date: Wed, 3 Aug 2011 14:53:45 +0000 Subject: [FFmpeg-user] Has anything happened with loop_input? In-Reply-To: <1312361837.43049.YahooMailNeo@web32507.mail.mud.yahoo.com> References: <1312361837.43049.YahooMailNeo@web32507.mail.mud.yahoo.com> Message-ID: > ffmpeg -y -loop_input -vframes 10 -r 25 -i ~/media_samples/arminFrame.bmp Desktop/MJPEGGED.mjpeg > > What i did do is to upgrade ffmpeg to latest git version because of a bug in rotating inputs to conver to mjepg output that i've posted and was fixed. > > What could be wrong? Hi Do you see the error message in console "-loop_input is deprecated, use -loop 1"? If so, change your command to use "-loop 1" something like this:- ffmpeg -y -loop 1 -vframes 200 -r 25 -i ~/media_samples/arminFrame.bmp Desktop/MJPEGGED.mjpeg From haakon.riiser at fys.uio.no Wed Aug 3 17:53:57 2011 From: haakon.riiser at fys.uio.no (Haakon Riiser) Date: Wed, 03 Aug 2011 17:53:57 +0200 Subject: [FFmpeg-user] =?utf-8?q?Custom_URLProtocols_deprecated=3F?= Message-ID: <095ce2fa00e39be7dc373989e3538d61@ulrik.uio.no> Adding custom URLProtocols through av_register_protocol2() is deprecated in recent versions of FFmpeg. What is the currently recommended alternative? Thanks, -hr From x2305andy2305x at yahoo.com Wed Aug 3 18:30:25 2011 From: x2305andy2305x at yahoo.com (Andy Andy) Date: Wed, 3 Aug 2011 09:30:25 -0700 (PDT) Subject: [FFmpeg-user] Has anything happened with loop_input? In-Reply-To: References: <1312361837.43049.YahooMailNeo@web32507.mail.mud.yahoo.com> Message-ID: <1312389025.24920.YahooMailNeo@web32501.mail.mud.yahoo.com> Hi, I saw no error in the console so didn't know about the deprecation. Another devloper, Carl Eugen Hoyos answered me also regarding this matter and fixed the latest git version to still be able to use loop_input. I need to know, will this change, will "loop 1" be the only supported switch? Regards, DAV ________________________________ From: bat guano To: ffmpeg-user at ffmpeg.org Sent: Wednesday, August 3, 2011 5:53 PM Subject: Re: [FFmpeg-user] Has anything happened with loop_input? > ffmpeg -y -loop_input -vframes 10 -r 25 -i ~/media_samples/arminFrame.bmp Desktop/MJPEGGED.mjpeg > > What i did do is to upgrade ffmpeg to latest git version because of a bug in rotating inputs to conver to mjepg output that i've posted and was fixed. > > What could be wrong? Hi Do you see the error message in console "-loop_input is deprecated, use -loop 1"? If so, change your command to use "-loop 1" something like this:- ffmpeg -y -loop 1 -vframes 200 -r 25 -i ~/media_samples/arminFrame.bmp Desktop/MJPEGGED.mjpeg ??? ??? ??? ? ??? ??? ? _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From stefano.sabatini-lala at poste.it Wed Aug 3 18:59:15 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Wed, 3 Aug 2011 18:59:15 +0200 Subject: [FFmpeg-user] Has anything happened with loop_input? In-Reply-To: <1312389025.24920.YahooMailNeo@web32501.mail.mud.yahoo.com> References: <1312361837.43049.YahooMailNeo@web32507.mail.mud.yahoo.com> <1312389025.24920.YahooMailNeo@web32501.mail.mud.yahoo.com> Message-ID: <20110803165915.GA15066@geppetto> On date Wednesday 2011-08-03 09:30:25 -0700, Andy Andy encoded: > Hi, > > I saw no error in the console so didn't know about the deprecation. > > Another devloper, Carl Eugen Hoyos answered me also regarding this > matter and fixed the latest git version to still be able to use > loop_input. I need to know, will this change, will "loop 1" be the > only supported switch? In the long term, yes, as we try to reduce the burden that we have to maintain (but usually just after a major release). From hcwilli at hcwilli.at Wed Aug 3 20:24:12 2011 From: hcwilli at hcwilli.at (Hans-Christian Willibald) Date: Wed, 3 Aug 2011 20:24:12 +0200 Subject: [FFmpeg-user] differend par / dar output made by ffmpeg compared to divx plus converter Message-ID: Hi ive installed the latest Versions from divx Plus Converter and ffmpeg on windows 7. i tried to reproduce the output from divx plus converter because i think its most compatible (with hardware solutions) and for pcs ive digitized some home videos with an dvd recorder, and renamed the .vob files in .mpg then i converted the mpg with divx plus converter and ffmpeg. analyzing the videos with ffmpeg,theres the output[1] [2] [4] i am confused about the differend par / dar output made by ffmpeg, (with the options [3]) [PAR 67:66 DAR4:3] is the output generated by ffmpeg ok, or is there an error? thanks in advance for answering [1]>>>>>>>>output for the vob, renamed in in.mpeg <<<<<<<<<<<<<<< Input #0, mpeg, from 'in.mpg': Duration: 00:30:30.92, start: 0.218044, bitrate: 4691 kb/s Stream #0.0[0x80]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s Stream #0.1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [PAR 16:15 DA R 4:3], 8860 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc [buffer @ 01C5FF80] w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:16/15 sws_param: [scale @ 02F33E20] w:720 h:576 fmt:yuv420p -> w:720 h:540 fmt:yuv420p flags:0x4 [2]>>>>>>output for the divx, made by the divx plus converter:<<<<<<<<< Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) -> 25.00 (25/1) Input #0, avi, from 'divxplusconverter.divx': Duration: 00:30:30.84, start: 0.000000, bitrate: 1261 kb/s Stream #0.0: Video: mpeg4, yuv420p, 704x536 [PAR 1:1 DAR 88:67], 25 fps, 25 tbr, 25 tbn, 30k tbc Metadata: title : Video Stream #0.1: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s Metadata: title : Audio [3] >>>>>>>>>>>>ffmpeg commandline<<<<<<<<<<<<<<< c:\ffmpeg\bin\ffmpeg -i in.mpg -t 120 -mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1 -aspect 4:3 -deinterlace -vtag DIVX -f avi -vcodec mpeg4 -s 704x536 -b 996000 -acodec ac3 -ab 256000 -ar 48000 -pass 1 out.divx [4] >>>>>>>>>>>output for the divx, made by ffmpeg<<<<<<<<<<<< this leads to the following divx: Input #0, avi, from 'out.divx': Metadata: encoder : Lavf53.6.0 Duration: 00:02:00.00, start: 0.000000, bitrate: 1274 kb/s Stream #0.0: Video: mpeg4 (Simple Profile), yuv420p, 704x536 [PAR 67:66 DAR 4:3], 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s From venuiyer at yahoo.com Wed Aug 3 15:30:08 2011 From: venuiyer at yahoo.com (venuiyer) Date: Wed, 3 Aug 2011 06:30:08 -0700 (PDT) Subject: [FFmpeg-user] FFMPEG encoding for H264 Baseline 3 Profile Message-ID: <1312378208315-3715448.post@n4.nabble.com> Hello experts, I am encoding a video file for H264Baseline 3.0 profile using FFMPEG. Am using windows OS for FFMPEG Below is the command am using. ffmpeg -i -f mpegts -acodec aac -ar -ab -strict experimental -s -vcodec libx264 -b -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 -maxrate -minrate -bufsize 100k -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect 320:240 -g 30 -r -strict experimental -async 2 .ts Kindly help me with the below queries 1. How do I ensure that am encoding for Baseline 3.0 profile with the above command? 2. Is the command line am using correct one? Regards Venugopal G -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFMPEG-encoding-for-H264-Baseline-3-Profile-tp3715448p3715448.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From omkiran.for.wiki at gmail.com Wed Aug 3 20:44:44 2011 From: omkiran.for.wiki at gmail.com (Omkiran Sharma) Date: Thu, 4 Aug 2011 00:14:44 +0530 Subject: [FFmpeg-user] FFMPEG encoding for H264 Baseline 3 Profile In-Reply-To: <1312378208315-3715448.post@n4.nabble.com> References: <1312378208315-3715448.post@n4.nabble.com> Message-ID: Ensure your toolsets match baseline and your resolution/framerate/bitrate match level 3.0: For reference: http://blog.mediacoderhq.com/h264-profiles-and-levels/ Then you will be assured of baseline profile with level 3.0! On Wed, Aug 3, 2011 at 7:00 PM, venuiyer wrote: > Hello experts, > > ?I am encoding a video file for H264Baseline 3.0 profile using FFMPEG. Am > using windows OS for FFMPEG > ?Below is the command am using. > > > ffmpeg -i -f mpegts -acodec aac -ar -ab > -strict experimental ?-s -vcodec libx264 -b > -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 ?-maxrate > ?-minrate ?-bufsize 100k -rc_eq > 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect > 320:240 -g 30 -r -strict experimental -async 2 > .ts > > Kindly help me with the below queries > > 1. How do I ensure that am encoding for Baseline 3.0 profile with the above > command? > 2. Is the command line am using correct one? > > > Regards > Venugopal G > > > -- > View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFMPEG-encoding-for-H264-Baseline-3-Profile-tp3715448p3715448.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 Wed Aug 3 20:59:41 2011 From: coniophora at gmail.com (Jim Worrall) Date: Wed, 3 Aug 2011 12:59:41 -0600 Subject: [FFmpeg-user] FFMPEG encoding for H264 Baseline 3 Profile In-Reply-To: <1312378208315-3715448.post@n4.nabble.com> References: <1312378208315-3715448.post@n4.nabble.com> Message-ID: On Aug 3, 2011, at 7:30 AM, venuiyer wrote: > Hello experts, > > I am encoding a video file for H264Baseline 3.0 profile using FFMPEG. Am > using windows OS for FFMPEG > Below is the command am using. > > > ffmpeg -i -f mpegts -acodec aac -ar -ab > -strict experimental -s -vcodec libx264 -b > -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 -maxrate > -minrate -bufsize 100k -rc_eq > 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect > 320:240 -g 30 -r -strict experimental -async 2 > .ts > > Kindly help me with the below queries > > 1. How do I ensure that am encoding for Baseline 3.0 profile with the above > command? > 2. Is the command line am using correct one? > I have been using the -profile command in ffmpeg (I used high as an argument but presumably baseline would work too). Then I send commands directly to x264 by issuing the following before running ffmpeg: XOPTS="level=4.1 . . ." then include the following in the ffmpeg command line: -x264opts $XOPTS I'm not sure why some work through one avenue and some the other, but this seems to work. So you can explicitly set both profile and level. I've no idea what happens if you issue other commands that conflict with the effect of these. Jim From pssturges at gmail.com Thu Aug 4 01:53:17 2011 From: pssturges at gmail.com (pssturges) Date: Wed, 3 Aug 2011 16:53:17 -0700 (PDT) Subject: [FFmpeg-user] Interlaced avi h264 to mp4 In-Reply-To: <1309863240275-3645627.post@n4.nabble.com> References: <1309863240275-3645627.post@n4.nabble.com> Message-ID: <1312415597626-3717262.post@n4.nabble.com> It's amazing (and very frustrating) how when you are trying to solve a problem your very first attempt can be sooo close to the final solution. Yet it takes many different attempts, trying many varied things, only to go full circle and find the solution is a minor variation on your very first attempt. In this case, after several months working on and off on this, asking in several forums, the very simple solution is below: mkvmerge -o video.mkv video.avi ffmpeg -i video.mkv -vcodec copy -acodec libfaac -ac 2 -ab 160k video.mp4 -acodec copy -newaudio I don't know why muxing the original avi to mkv allows ffmpeg to be able to handle it better, but it seems that it does. Perhaps there's a bug with how ffmpeg handles avi's? Anyhow, this one is now (finally) solved. Thanks for all the help, Phil -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Interlaced-avi-h264-to-mp4-tp3645627p3717262.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rickcorteza at gmail.com Thu Aug 4 02:37:33 2011 From: rickcorteza at gmail.com (Rick C.) Date: Thu, 4 Aug 2011 08:37:33 +0800 Subject: [FFmpeg-user] merging or joining videos In-Reply-To: References: <4E390C29.6040603@das-werkstatt.com> Message-ID: <406C05B6-B517-4BAE-982A-D4E5D4321B1A@gmail.com> On Aug 3, 2011, at 5:25 PM, Venkatasubramaniam R wrote: > Hi, > > Once I had used MP4Box package in Linux for concatenating smaller mp4 files > to get a big one. It worked pretty well, but I had not tried it on AVI > files. > > May be you could try MP4Box (which comes as a command line tool when > installing gpac package : http://gpac.wp.institut-telecom.fr/mp4box/). > > All the best, > -Venkat. > > On Wed, Aug 3, 2011 at 10:51 AM, Peter B. wrote: > >> Rick C. wrote: >>> I'm familiar with cat but I believe this only works on .mpg and I would >> like a way to work on many files such as .avi or .mp4. >> I'm not 100% sure, but I think it also depends on which codec you're >> using (.avi is just the container). >> >>> I've also read the instructions in the documentation but I was thinking >> there's a more updated way than having to convert everything to .mpg first? >> Anyways, if someone could point me in the right direction that would be >> great. >>> >> I've also had the same problem: All documentation I found mentioned >> MPEGs and "cat" or yuv2mpeg for piping things together :( This doesn't >> work at all for AVI files. >> >> I found a solution that even works bitproof for audio and video, by >> using "mencoder" to merge the videos back together. We're using this >> method daily in the Austrian national video archive (=?sterreichische >> Mediathek [1][2]) - but only with intraframe codecs (FFv1 / DV) and PCM >> for audio. Additionally, it's also possible to merge *and* transcode the >> files in one step! :) >> Details can be found in a forum post on "das-werkstatt.com" [3]. >> >> Additionally, there's an article on "linuxreview.org" about merging >> video files [4] - but I must say that I haven't had a good experience >> with "avimerge". >> >> Good luck! >> Peter B. >> >> == References: >> [1] http://www.mediathek.at/ >> [2] http://sourceforge.net/projects/dva-profession/ >> [3] http://www.das-werkstatt.com/forum/werkstatt/viewtopic.php?f=7&t=1793 >> [4] http://en.linuxreviews.org/HOWTO_Merge_video_files >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> Great thanks again I'll be working on this. I know converting to .mpg first will work but that makes the process longer and than I would have to double convert... From rickcorteza at gmail.com Thu Aug 4 02:39:00 2011 From: rickcorteza at gmail.com (Rick C.) Date: Thu, 4 Aug 2011 08:39:00 +0800 Subject: [FFmpeg-user] subtitle update In-Reply-To: <1312234571.90166.YahooMailNeo@web86406.mail.ird.yahoo.com> References: <9AB93297-4166-41A3-9BDD-A515E76E7291@gmail.com> <1312009154.8162.YahooMailNeo@web86408.mail.ird.yahoo.com> <7E7A6FF0-CC0D-46A7-B362-8BD0DE8EB0AE@gmail.com> <765C6103-E624-4764-BDBA-89BB1BC1AF5D@pixelpartner.de> <1312234571.90166.YahooMailNeo@web86406.mail.ird.yahoo.com> Message-ID: <55794277-32B7-466B-92F4-1E1581A0ACAF@gmail.com> On Aug 2, 2011, at 5:36 AM, JULIAN GARDNER wrote: > > > > > >> ________________________________ >> From: Thomas Kumlehn >> To: FFmpeg user questions and RTFMs >> Sent: Monday, 1 August 2011, 22:13 >> Subject: Re: [FFmpeg-user] subtitle update >> >> On the windows platform I worked with a combination of AviSynth and ffmpeg. >> AviSynth has established subtitle support and ffmpeg configured with --enable-avisynth can process these *.avs scripts. >> Just a hint, because I didn't do hard subs yet. >> >> Thomas Kumlehn >> PIXEL PARTNER (R) >> >> Send from my iPad 3-D >> http://www.pixelpartner.de >> >> Am 30.07.2011 um 10:38 schrieb "Rick C." : >> >>> >>> On Jul 30, 2011, at 2:59 PM, JULIAN GARDNER wrote: >>> >>>> >>>> >>>> >>>> >>>> ----- Original Message ----- >>>>> From: Rick C. >>>>> To: FFmpeg user questions and RTFMs >>>>> Cc: >>>>> Sent: Friday, 29 July 2011, 12:13 >>>>> Subject: [FFmpeg-user] subtitle update >>>>> >>>>> Hello, >>>>> >>>>> I just wanted to see if there was any update to hard-coding subtitles (like >>>>> .srt) into an encoding? So let's say going from .mkv to .avi I would like >>>>> to hard-code the subtitles in the .avi. I tried to see if drawtext would do it >>>>> and while I was able to get text to appear I didn't see any way to actually >>>>> deal with an entire subtitle file. I've seen some talking recently in the >>>>> developer list about this that's why I thought I'd ask. If there's >>>>> something experimental I'd be glad to give it a try. Thanks! >>>>> >>>>> rc >>>>> _______________________________________________ >>>>> ffmpeg-user mailing list >>>>> ffmpeg-user at ffmpeg.org >>>>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >>>>> >>>> Iv been testing this for the last month and everything seems to be working fine, but there are some caveats >>>> >>>> 1, DVB Subtitles - Working >>>> 2. DIVX - Working, and can be transcoded to either Hard Subs or DVB Subtitles >>>> 3. MKV - Cant test as i cannot get ASS working >>>> 4. AVI/SRT/SUB - Problems are that the SRT/SUB file give the wrong Timestamp so they appear at the wrong place, As an example, the IDX says 2.26.880 but the sub file says it starts earlier. As the index file is not used i would need to add in some offsetting >>>> >>>> Also a couple of more things ive asked about but not had any answers/ideas is that when transcoding to DVB Subtitles and the screen is not a full 720x576, which is what DVB Subs wants, you get subtitles off screen, so not viewable >>>> >>>> joolz >>> >>> Thank you for the reply. So at this time it looks like this feature will be included in the future (there's no way to access this now that I know of). That's great and all the best on this project! :-) >>> >>> rc >>> >>> _______________________________________________ >>> 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 >> > > > As stated before Hard Subs works, but there are come caveats. > > If people can put up with it then ok, but what i dont want is to post a patch then have loads of reports of problems, which i already know about. > > joolz > > > states I'm not sure if I'm in a position to say post a patch. :-) I would definitely try something if it was available but for now I guess I'll have to go another direction. Hopefully it will be available one day! From ronag89 at gmail.com Thu Aug 4 10:44:42 2011 From: ronag89 at gmail.com (Robert Nagy) Date: Thu, 4 Aug 2011 10:44:42 +0200 Subject: [FFmpeg-user] AVFrame extend life beyond next call to avcodec_decode_video2 Message-ID: Hi I would like to extend the life of AVFrame provided by decode_video beyond the next call to decode_video. I have tried overriding release_buffer and defer the release_buffer call to after I no longer need the buffer, however this crashes the application. e.g. static void null_release_buffer(AVCodecContext *codec, AVFrame *pic){} ... old_release_buffer = codec->release_buffer; codec->release_buffer = null_release_buffer; ... several decode_video calls for all frames: old_release_buffer(&frame); As a mentioned this strategy doesn't seems to work as the application crashes. Any suggestions? From gavr.mail at gmail.com Thu Aug 4 10:55:24 2011 From: gavr.mail at gmail.com (Kirill Gavrilov) Date: Thu, 4 Aug 2011 12:55:24 +0400 Subject: [FFmpeg-user] AVFrame extend life beyond next call to avcodec_decode_video2 In-Reply-To: References: Message-ID: Hi, I would like to extend the life of AVFrame provided by decode_video beyond > the next call to decode_video. > static void null_release_buffer(AVCodecContext *codec, AVFrame *pic){} > I'm not sure but I don't think that this could works. Decoder probably can reuse the same buffer for next frames. As a mentioned this strategy doesn't seems to work as the application > crashes. > > Any suggestions? > that is an *alien* buffer. You should make a copy if you need. On Thu, Aug 4, 2011 at 12:44 PM, Robert Nagy wrote: > Hi > > I would like to extend the life of AVFrame provided by decode_video beyond > the next call to decode_video. > > I have tried overriding release_buffer and defer the release_buffer call to > after I no longer need the buffer, however this crashes the application. > > e.g. > > static void null_release_buffer(AVCodecContext *codec, AVFrame *pic){} > > ... > > old_release_buffer = codec->release_buffer; > codec->release_buffer = null_release_buffer; > > ... several decode_video calls > > for all frames: > old_release_buffer(&frame); > > As a mentioned this strategy doesn't seems to work as the application > crashes. > > Any suggestions? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > ----------------------------------------------- Kirill Gavrilov, Software designer. From ronag89 at gmail.com Thu Aug 4 11:20:51 2011 From: ronag89 at gmail.com (Robert Nagy) Date: Thu, 4 Aug 2011 11:20:51 +0200 Subject: [FFmpeg-user] AVFrame extend life beyond next call to avcodec_decode_video2 In-Reply-To: References: Message-ID: > that is an *alien* buffer. You should make a copy if you need. Well the point of this is to avoid making an unnecessary copy. From gavr.mail at gmail.com Thu Aug 4 12:34:53 2011 From: gavr.mail at gmail.com (Kirill Gavrilov) Date: Thu, 4 Aug 2011 14:34:53 +0400 Subject: [FFmpeg-user] AVFrame extend life beyond next call to avcodec_decode_video2 In-Reply-To: References: Message-ID: On Thu, Aug 4, 2011 at 1:20 PM, Robert Nagy wrote: > Well the point of this is to avoid making an unnecessary copy. > As I already noticed decoder can reuse the same buffer for next frames so the only way to get control of this - to create buffers for decoder by yourself (and create several such buffers to switch between them). But decoding maybe much more complicated than you think. However I didn't find the way for that in current API (only in encoding you can provide your own frames). Maybe you will be more lucky but in general I recommend you just do copy if you really need it ;) On Thu, Aug 4, 2011 at 1:20 PM, Robert Nagy wrote: > > that is an *alien* buffer. You should make a copy if you need. > > Well the point of this is to avoid making an unnecessary copy. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > ----------------------------------------------- Kirill Gavrilov, Software designer. From ronag89 at gmail.com Thu Aug 4 13:25:22 2011 From: ronag89 at gmail.com (Robert Nagy) Date: Thu, 4 Aug 2011 13:25:22 +0200 Subject: [FFmpeg-user] AVFrame extend life beyond next call to avcodec_decode_video2 In-Reply-To: References: Message-ID: I see. Thank you for your explanation. I guess I'll have to live with the copying. Best Regards From cmhjones at gmail.com Fri Aug 5 18:59:50 2011 From: cmhjones at gmail.com (cameron) Date: Fri, 5 Aug 2011 09:59:50 -0700 (PDT) Subject: [FFmpeg-user] distortive video effects Message-ID: <1312563590774-3721703.post@n4.nabble.com> hi, i'm looking for some filters which can be used to distort a source video to produce some 'funky' effects. i've created some mirror effects using transpose filter, and can get greyscale through applying formats, but are there any other filters or configurations which produce distortive, yet appealing effects? I was thinking of creating half tone, sepia or swapping the color channels but it would seem i need to use eq2 fin mencoder for these types of color manipulations? is there anything like this possible from ffmpegs filters? i was also looking at creating a 'pixelated' effect, but it would need to be relatively clean, not so much as a result of lossy processing, is this possible? thanks! cam -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/distortive-video-effects-tp3721703p3721703.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From luj125 at gmail.com Fri Aug 5 20:20:42 2011 From: luj125 at gmail.com (James Lu) Date: Fri, 5 Aug 2011 14:20:42 -0400 Subject: [FFmpeg-user] distortive video effects In-Reply-To: <1312563590774-3721703.post@n4.nabble.com> References: <1312563590774-3721703.post@n4.nabble.com> Message-ID: On Fri, Aug 5, 2011 at 12:59 PM, cameron wrote: > hi, > > i'm looking for some filters which can be used to distort a source video to > produce some 'funky' effects. i've created some mirror effects using > transpose filter, and can get greyscale through applying formats, but are > there any other filters or configurations which produce distortive, yet > appealing effects? > > I was thinking of creating half tone, sepia or swapping the color channels > but it would seem i need to use eq2 fin mencoder for these types of color > manipulations? is there anything like this possible from ffmpegs filters? > > i was also looking at creating a 'pixelated' effect, but it would need to > be > relatively clean, not so much as a result of lossy processing, is this > possible? > > thanks! > cam > > -- > View this message in context: > http://ffmpeg-users.933282.n4.nabble.com/distortive-video-effects-tp3721703p3721703.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 > Hey Cameron, Initial research leads to me frei0r http://ffmpeg.org/libavfilter.html#SEC24 Hope this gets you towards the right direction, it looks like that libavfilter can connect to the frei0r filter API, so ffmpeg's -vf frei0r=*effect:parameters *seems able to perform some more complicated filters. ~James From rogerdpack2 at gmail.com Sat Aug 6 01:06:42 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Fri, 5 Aug 2011 17:06:42 -0600 Subject: [FFmpeg-user] bgr24 -> yuv420p darkens? Message-ID: Hello all. First time here, and first of all thank you for a marvelous program in ffmpeg. Everybody everywhere uses it! If I could donate some money to it, I would (I know somewhere it said you don't take donations though). Ok, observations. With this bmp (bgr24) [1], if I use ffplay on it, it comes out "darkened" and/or "with a red hue" for whatever reason. [1] http://rogerdpack.t28.net/incoming/mostly_white.bmp the "white" outermost backgrounded window looks somewhat red http://rogerdpack.t28.net/incoming/red_hue.bmp within the ffplay window. Here's the command line output (mingw build, but it seems to reproduce with any platform/version). $ ffplay version N-31706-g335bbe4, Copyright (c) 2003-2011 the FFmpeg developers built on Jul 31 2011 18:52:06 with gcc 4.6.1 configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable-runtime-cpudetect --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 libavutil 51. 11. 1 / 51. 11. 1 libavcodec 53. 9. 0 / 53. 9. 0 libavformat 53. 6. 0 / 53. 6. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 27. 5 / 2. 27. 5 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, image2, from 'mostly_white.bmp': Duration: 00:00:00.04, start: 0.000000, bitrate: N/A Stream #0.0: Video: bmp, bgr24, 1680x1050, 25 tbr, 25 tbn, 25 tbc [buffersink @ 03C05D40] auto-inserting filter 'auto-inserted scaler 0' between the filter 'src' and the filter 'out' [scale @ 03C06280] w:1680 h:1050 fmt:bgr24 -> w:1680 h:1050 fmt:yuv420p flags:0x4 Bug? Thank you. -roger- From h.reindl at thelounge.net Sat Aug 6 04:01:24 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sat, 06 Aug 2011 04:01:24 +0200 Subject: [FFmpeg-user] File for preset 'baseline' not found Message-ID: <4E3CA074.50502@thelounge.net> Hi N-31788-ge1e3d79 says "File for preset 'baseline' not found" without twopass-encoding fails with "maybe incorrect parameters such as bit_rate, rate, width or height" it is a little bit frustrating that vpre is permanently changed because if you write wrappers it is impossible to say on what foreign builds they can be used :-( has anybody an idea what on the two command-lines will break 2pass encoding and what made the additional -vpre baseline on both of them different to be able remove vpre completly? /usr/bin/ffmpeg -i '/Volumes/dune/buildserver/autotest/parts/ffmpeg/demo.mp4' -y -vb '448k' -bt '64k' -r '25' -vf 'scale=480:266,setdar=16:9,pad=480:270:0:2:000000' -pass '1' -passlogfile '4bc0ff09ef706758bc91a44e289127f5-mp4' -maxrate '544k' -f 'ipod' -vcodec 'libx264' -flags '+loop+mv4' -flags2 '+bpyramid+wpred+mixed_refs+dct8x8' -cmp '+chroma' -partitions '-parti8x8-parti4x4-partp8x8-partb8x8' -me_method 'dia' -me_range '16' -subq '2' -trellis '0' -refs '1' -bf '0' -coder '0' -g '250' -keyint_min '25' -sc_threshold '40' -i_qfactor '0.71' -qcomp '0.6' -qmin '0' -qmax '69' -qdiff '4' -bufsize '2M' -directpred '1' -wpredp '2' -rc_lookahead '30' -threads '2' -timestamp 'now' -an /dev/null >> '/Volumes/dune/buildserver/autotest/parts/ffmpeg/logs/mp4_mp4-x264.log' 2>> '/Volumes/dune/buildserver/autotest/parts/ffmpeg/logs/mp4_mp4-x264.log' /usr/bin/ffmpeg -i '/Volumes/dune/buildserver/autotest/parts/ffmpeg/demo.mp4' -y -vb '448k' -ab '96k' -bt '64k' -ar '44100' -ac '2' -r '25' -vf 'scale=480:266,setdar=16:9,pad=480:270:0:2:000000' -pass '2' -passlogfile '4bc0ff09ef706758bc91a44e289127f5-mp4' -maxrate '544k' -f 'ipod' -vcodec 'libx264' -acodec 'libfaac' -flags '+loop+mv4' -flags2 '+bpyramid+wpred+mixed_refs+dct8x8' -cmp '256' -partitions '+parti4x4+parti8x8+partp4x4+partp8x8+partb8x8' -me_method 'hex' -me_range '16' -subq '7' -trellis '1' -refs '5' -bf '0' -coder '0' -g '250' -keyint_min '25' -sc_threshold '40' -i_qfactor '0.71' -qcomp '0.6' -qmin '0' -qmax '69' -qdiff '4' -bufsize '2M' -directpred '3' -wpredp '0' -rc_lookahead '50' -threads '2' -timestamp 'now' '/Volumes/dune/buildserver/autotest/parts/ffmpeg/targets/test_2p.x264.mp4' >> '/Volumes/dune/buildserver/autotest/parts/ffmpeg/logs/mp4_mp4-x264.log' 2>> '/Volumes/dune/buildserver/autotest/parts/ffmpeg/logs/mp4_mp4-x264.log' -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From stefano.sabatini-lala at poste.it Sat Aug 6 12:57:25 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sat, 6 Aug 2011 12:57:25 +0200 Subject: [FFmpeg-user] distortive video effects In-Reply-To: <1312563590774-3721703.post@n4.nabble.com> References: <1312563590774-3721703.post@n4.nabble.com> Message-ID: <20110806105725.GA27839@geppetto> On date Friday 2011-08-05 09:59:50 -0700, cameron encoded: > hi, > > i'm looking for some filters which can be used to distort a source video to > produce some 'funky' effects. i've created some mirror effects using > transpose filter, and can get greyscale through applying formats, but are > there any other filters or configurations which produce distortive, yet > appealing effects? > > I was thinking of creating half tone, sepia or swapping the color channels > but it would seem i need to use eq2 fin mencoder for these types of color > manipulations? is there anything like this possible from ffmpegs filters? As mentioned, the frei0r filter allows to access all the frei0r effects, and many Mplayer/mencoder filters are available through the mp wrapper. And I'd be glad to help someone porting more filters (e.g. from virtualdub/effectv). > i was also looking at creating a 'pixelated' effect, but it would need to be > relatively clean, not so much as a result of lossy processing, is this > possible? Maybe -vf frei0r=pixeliz0r? From prashant at b-one.net Sun Aug 7 14:23:26 2011 From: prashant at b-one.net (Prashant Rathi) Date: Sun, 7 Aug 2011 16:23:26 +0400 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software Message-ID: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> Hi FFMpeg Users, 1) Is it allowed to use FFMpeg.exe (build with GPL enabled)command line tool from proprietary software? 2) Can I distribute this FFMpeg.exe command line tool with my proprietary software? Please provide answers to above two queries.. Thanks Prashant From h.reindl at thelounge.net Sun Aug 7 14:30:12 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sun, 07 Aug 2011 14:30:12 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> Message-ID: <4E3E8554.8050009@thelounge.net> Am 07.08.2011 14:23, schrieb Prashant Rathi: > Hi FFMpeg Users, > > 1) Is it allowed to use FFMpeg.exe (build with GPL enabled)command line > tool from proprietary software? > > 2) Can I distribute this FFMpeg.exe command line tool with my > proprietary software? > > Please provide answers to above two queries.. just think a second about what GPL means do you touch ffmpeg or include any headers? no you do not if you call it only via CLI http://www.ffmpeg.org/legal.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From prashant at b-one.net Sun Aug 7 15:18:08 2011 From: prashant at b-one.net (Prashant Rathi) Date: Sun, 7 Aug 2011 17:18:08 +0400 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <4E3E8554.8050009@thelounge.net> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> Message-ID: <002501cc5504$73c0b120$5b421360$@b-one.net> Thanks Reindl, Since I am new to this. To take the safer side I wanted to be more specific about my implementation. I am using FFMpeg.exe command line tool programmatically and running FFMpeg.exe commands programmatically. For this I need to distribute FFMpeg.exe command line tool with my proprietary software. I hope this will not cause any Legal issue.. Please reply.. Thanks Prashant -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Reindl Harald Sent: Sunday, August 07, 2011 4:30 PM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software Am 07.08.2011 14:23, schrieb Prashant Rathi: > Hi FFMpeg Users, > > 1) Is it allowed to use FFMpeg.exe (build with GPL enabled)command line > tool from proprietary software? > > 2) Can I distribute this FFMpeg.exe command line tool with my > proprietary software? > > Please provide answers to above two queries.. just think a second about what GPL means do you touch ffmpeg or include any headers? no you do not if you call it only via CLI http://www.ffmpeg.org/legal.html From h.reindl at thelounge.net Sun Aug 7 15:23:56 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sun, 07 Aug 2011 15:23:56 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <002501cc5504$73c0b120$5b421360$@b-one.net> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> Message-ID: <4E3E91EC.6040207@thelounge.net> as long as you only call a binary application via cli it does not matter under what license the called cli-app is because you do not touch it, no matter if it is GPL, LGPL, Commerical..... the other thing is copying the binary-app and this is a totally different topic, sure you can bundle a binary ffmpeg but you should provide the source code from which it was compiled and not only link to the homepage because in the last years the command lines often changed and the sources a year later me be useless for your tool Am 07.08.2011 15:18, schrieb Prashant Rathi: > Thanks Reindl, > > Since I am new to this. To take the safer side I wanted to be more specific > about my implementation. > > I am using FFMpeg.exe command line tool programmatically and running > FFMpeg.exe commands programmatically. > For this I need to distribute FFMpeg.exe command line tool with my > proprietary software. > > I hope this will not cause any Legal issue.. > > Please reply.. > > Thanks > Prashant -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From prashant at b-one.net Sun Aug 7 15:44:29 2011 From: prashant at b-one.net (Prashant Rathi) Date: Sun, 7 Aug 2011 17:44:29 +0400 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <4E3E91EC.6040207@thelounge.net> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> Message-ID: <003001cc5508$21e4bdc0$65ae3940$@b-one.net> Thanks for your prompt reply. Regards Prashant -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Reindl Harald Sent: Sunday, August 07, 2011 5:24 PM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software as long as you only call a binary application via cli it does not matter under what license the called cli-app is because you do not touch it, no matter if it is GPL, LGPL, Commerical..... the other thing is copying the binary-app and this is a totally different topic, sure you can bundle a binary ffmpeg but you should provide the source code from which it was compiled and not only link to the homepage because in the last years the command lines often changed and the sources a year later me be useless for your tool Am 07.08.2011 15:18, schrieb Prashant Rathi: > Thanks Reindl, > > Since I am new to this. To take the safer side I wanted to be more > specific about my implementation. > > I am using FFMpeg.exe command line tool programmatically and running > FFMpeg.exe commands programmatically. > For this I need to distribute FFMpeg.exe command line tool with my > proprietary software. > > I hope this will not cause any Legal issue.. > > Please reply.. > > Thanks > Prashant From philip at turmel.org Sun Aug 7 15:45:06 2011 From: philip at turmel.org (Phil Turmel) Date: Sun, 07 Aug 2011 09:45:06 -0400 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <002501cc5504$73c0b120$5b421360$@b-one.net> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> Message-ID: <4E3E96E2.2060609@turmel.org> Good morning Prashant, [Please don't top-post on this list, and do trim your replies.] On 08/07/2011 09:18 AM, Prashant Rathi wrote: > Thanks Reindl, > > Since I am new to this. To take the safer side I wanted to be more specific > about my implementation. > > I am using FFMpeg.exe command line tool programmatically and running > FFMpeg.exe commands programmatically. Calling FFmpeg via its command line interface is generally accepted as a way to keep your software proprietary. Some purists disagree, but their's no court decisions I'm aware of that have addressed the question. > For this I need to distribute FFMpeg.exe command line tool with my > proprietary software. You could point your customers to ffmpeg.org, with detailed instructions, and have them install it themselves. You might discover that some will already have it installed. > I hope this will not cause any Legal issue.. It might. Any time you distribute GPL software, you must offer your customers the source code and build instructions for the version you ship. If they can compile a new copy of ffmpeg.exe to use with your software, you've given them enough material. I hope that it is obvious that you need to compile FFmpeg.exe yourself, to be sure that your customers will also be able to. Similarly, as the GPL version of FFmpeg includes other GPL libraries, you need to do the same with all of those libraries. It is *not* sufficient to point your customers at the same website you got a copy of FFmpeg.exe from. > http://www.ffmpeg.org/legal.html Most of the legal advice on the link above applies to "LGPL" usage of FFmpeg, which you are avoiding. But the patent mini-FAQ would still apply. Of course, I Am Not A Lawyer, so the above is solely my opinion. I suggest you read the actual GPL and make sure you understand it. If you can't, you should hire a real lawyer who has experience in this area. HTH, Phil From phil_rhodes at rocketmail.com Sun Aug 7 15:49:42 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Sun, 07 Aug 2011 14:49:42 +0100 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <4E3E91EC.6040207@thelounge.net> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> Message-ID: > as long as you only call a binary application via > cli it does not matter under what license the called > cli-app is because you do not touch it, no matter if > it is GPL, LGPL, Commerical..... This would be one of the reasons why the GPL and its derivatives make absolutely no sense whatsoever. What's the functional difference?! What's the situation with including an ffmpeg CLI binary with rental equipment? If I do that, if it's an embedded system, how do I make the required source code offer? None of this has been thought through very carefully, has it... P From h.reindl at thelounge.net Sun Aug 7 15:55:53 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sun, 07 Aug 2011 15:55:53 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> Message-ID: <4E3E9969.2030609@thelounge.net> Am 07.08.2011 15:49, schrieb Phil Rhodes: >> as long as you only call a binary application via >> cli it does not matter under what license the called >> cli-app is because you do not touch it, no matter if >> it is GPL, LGPL, Commerical..... > > This would be one of the reasons why the GPL and its derivatives make > absolutely no sense whatsoever for YOU if YOU want use the code like YOU WANT it makes sense if the developer DECIDED that every derived code has to be available too and he decided this be selecting the license so what do you try to tell us? that YOU do not like the license fine, nobody is forcing you to use anything you do not like -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From googol at wolke7.net Sun Aug 7 16:16:24 2011 From: googol at wolke7.net (lovelove) Date: Sun, 7 Aug 2011 07:16:24 -0700 (PDT) Subject: [FFmpeg-user] Scene detection In-Reply-To: <354784.24180.qm@web110607.mail.gq1.yahoo.com> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> Message-ID: <1312726584077-3724931.post@n4.nabble.com> Hi, can someone please help a desperate soul like me? I am in the same situation as Fori Pepe Fori Pepe wrote: > > I would like to export/extract images from videos at every scene changes. > Is it possible to use ffmpeg to create images on scene changes? > It would be the most comfortable to use it like this: > ffmpeg -i test.mpg --detect_scene_changes newscene[00-99].png > ... i.e. export first frame of each new scene as an image. This effectively generates a storyboard. It also would be an enormous help in identifying added and deleted scenes when comparing 2 videos. this was the last answer in that thread: Stefano Sabatini wrote: > > The fake syntax for a lavfi filter [...] would be something of the kind: > > ffmpeg -i in.avi -vfilters "[IN] split [SC_DETECT] [OUT], [SC_DETECT] > sc_detect=PARAMS, process_sc_frames=PARAMS" /dev/null > > As you see you would need two filters, one detecting scene changes and > outputting the detected scene change frames to the output, and a > filter to furtherly processes the information of the frames, for > example it could print to a file the timestamp of the file or to > render as an image file the processed frames. > > The sc_detect filter implementation is the tricky part, but > a naive implementation shouldn't be too hard to accomplish. > I am really a beginner, so I don't quite know what to do with that. All I know is I have been looking for >1 year on the www, from top to bottom and could not find anything. I would be so thankful to anybody of you here, if you could tell me what to do. with my best regards, Christian Davideck -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Scene-detection-tp941845p3724931.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From mark at mdsh.com Sun Aug 7 16:47:15 2011 From: mark at mdsh.com (Mark Himsley) Date: Sun, 07 Aug 2011 15:47:15 +0100 Subject: [FFmpeg-user] Scene detection In-Reply-To: <1312726584077-3724931.post@n4.nabble.com> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> Message-ID: <4E3EA573.9010109@mdsh.com> On 07/08/2011 15:16, lovelove wrote: > Hi, can someone please help a desperate soul like me? > > I am in the same situation as Fori Pepe > > > Fori Pepe wrote: >> >> I would like to export/extract images from videos at every scene changes. >> Is it possible to use ffmpeg to create images on scene changes? >> It would be the most comfortable to use it like this: >> ffmpeg -i test.mpg --detect_scene_changes newscene[00-99].png >> > > ... i.e. export first frame of each new scene as an image. This effectively > generates a storyboard. > It also would be an enormous help in identifying added and deleted scenes > when comparing 2 videos. > > this was the last answer in that thread: > > > Stefano Sabatini wrote: >> >> The fake syntax for a lavfi filter [...] would be something of the kind: >> >> ffmpeg -i in.avi -vfilters "[IN] split [SC_DETECT] [OUT], [SC_DETECT] >> sc_detect=PARAMS, process_sc_frames=PARAMS" /dev/null >> >> As you see you would need two filters, one detecting scene changes and >> outputting the detected scene change frames to the output, and a >> filter to furtherly processes the information of the frames, for >> example it could print to a file the timestamp of the file or to >> render as an image file the processed frames. >> >> The sc_detect filter implementation is the tricky part, but >> a naive implementation shouldn't be too hard to accomplish. I wonder if any code from the BBC R&D's open source (LGPL) Video Shot Change Detector could be used? http://www.bbc.co.uk/opensource/projects/shot_change/ -- Mark From googol at wolke7.net Sun Aug 7 17:08:02 2011 From: googol at wolke7.net (lovelove) Date: Sun, 7 Aug 2011 08:08:02 -0700 (PDT) Subject: [FFmpeg-user] Scene detection In-Reply-To: <4E3EA573.9010109@mdsh.com> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <4E3EA573.9010109@mdsh.com> Message-ID: <1312729682524-3725015.post@n4.nabble.com> Mark Himsley wrote: > > I wonder if any code from the BBC R&D's open source (LGPL) Video Shot > Change Detector could be used? > > http://www.bbc.co.uk/opensource/projects/shot_change/ > The netiquette says no smalltalk, still I want to say how grateful I am for your reply. Hope that's ok. Here is other code I found for scene change detection: (being coding illiterate I don't know how userful it is) http://www.mathworks.com/products/computer-vision/demos.html?file=/products/demos/shipping/vision/videoscenechange.html http://www.catenary.com/howto/motion.html http://forum.doom9.org/showthread.php?p=268534#post268534 (of course I did NOT expect anyone to do coding work) proof-of-concept: http://www.youtube.com/watch?v=4988BhKUa9Q&NR=1 http://www.youtube.com/watch?v=4esP2nO1_kY -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Scene-detection-tp941845p3725015.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From phil_rhodes at rocketmail.com Sun Aug 7 22:08:41 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Sun, 07 Aug 2011 21:08:41 +0100 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <4E3E9969.2030609@thelounge.net> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <4E3E9969.2030609@thelounge.net> Message-ID: > fine, nobody is forcing you to use anything you do not like Yers. That's what they say about Windows. P From nicolas.george at normalesup.org Sun Aug 7 23:32:33 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Sun, 7 Aug 2011 23:32:33 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> Message-ID: <20110807213233.GA380@phare.normalesup.org> [ I know answering to this person is usually wasted time, but his message can fool earnest readers. ] Le decadi 20 thermidor, an CCXIX, Phil Rhodes a ?crit?: > >as long as you only call a binary application via > >cli it does not matter under what license the called > >cli-app is because you do not touch it, no matter if > >it is GPL, LGPL, Commerical..... > This would be one of the reasons why the GPL and its derivatives > make absolutely no sense whatsoever. > > What's the functional difference?! There is a reason, and it is a legal one. The legal basis that allows the viral clause of the GPL to work is that under copyright law, one can not distribute a works without the consent of its author, and the author can pose almost any condition to give his consent. When building a program using a library, even a dynamically linked one, part of the library is embedded in the final binary: the library's copyright owner can pose his condition on the distribution of said binary. On the other hand, a program that calls ffmpeg as an external program, no part of ffmpeg is embedded. The program can even be built on a system where ffmpeg is not installed and never was. Therefore, ffmpeg's copyright owners can not pose any condition. Unfortunately, a variation on this argument could allow anyone to distribute a proprietary program linked to GPL ffmpeg libraries; but I will not explain my reasoning publicly here, as I consider it a violation of the spirit of the license and do not want to encourage it. 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 phil_rhodes at rocketmail.com Mon Aug 8 01:09:19 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Mon, 08 Aug 2011 00:09:19 +0100 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <20110807213233.GA380@phare.normalesup.org> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> Message-ID: > There is a reason, and it is a legal one. Ah, that explains why it doesn't make any objective sense. P From h.reindl at thelounge.net Mon Aug 8 01:36:44 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Mon, 08 Aug 2011 01:36:44 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> Message-ID: <4E3F218C.6070204@thelounge.net> Am 08.08.2011 01:09, schrieb Phil Rhodes: > >> There is a reason, and it is a legal one. > > Ah, that explains why it doesn't make any objective sense can you stop this childish polemic? as developer you have the copyright and can define the rules under which your software/code is used and damned this is a legal thing and not a wish so what exactly is your problem? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From phil_rhodes at rocketmail.com Mon Aug 8 01:44:49 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Mon, 08 Aug 2011 00:44:49 +0100 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <4E3F218C.6070204@thelounge.net> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> Message-ID: > so what exactly is your problem? Simply that it doesn't make sense and isn't good for anyone. You are of course free to define your own rules, I'm just pointing out how stupid they are. P From h.reindl at thelounge.net Mon Aug 8 02:11:52 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Mon, 08 Aug 2011 02:11:52 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> Message-ID: <4E3F29C8.80508@thelounge.net> Am 08.08.2011 01:44, schrieb Phil Rhodes: > >> so what exactly is your problem? > > Simply that it doesn't make sense and isn't good for anyone. > You are of course free to define your own rules, I'm just pointing out how stupid they are this is typically the argumentation of people who are pissed off that there is code they would like too use, too lazy to write it self and not willing to accept that the developer decided he is willing to offer his code for free but not willing taht peopole take it, make money with it and does not give back improvements you call the rules stupid - i call you stupid and no i am not contributor to ffmpeg, this is a generally thing -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From hvidal at tesseract-tech.com Mon Aug 8 02:16:09 2011 From: hvidal at tesseract-tech.com (H. Vidal, Jr.) Date: Sun, 07 Aug 2011 20:16:09 -0400 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> Message-ID: <4E3F2AC9.3000902@tesseract-tech.com> On 08/07/2011 07:44 PM, Phil Rhodes wrote: > >> so what exactly is your problem? > > Simply that it doesn't make sense and isn't good for anyone. > > You are of course free to define your own rules, I'm just pointing out > how stupid they are. Respectfully, I have noticed, Mr. Rhodes, that although you are frequently one to contribute a comment to this list, your comments are hardly constructive and helpful to a set of users and developers intent on working with and on ffmpeg. You are, of course, free to continue your oft-repeated comments on the problems with ffmpeg, open-source and etc. Just please consider that this isn't truly helping the users and engineers on this code. I am certain that I speak for a few of us when I say, I really wish you would simply acknowledge that we acknowledge your reservations, and simply be done with it. I value your criticisms, but after a while, repetition changes signal to noise, in this case. With all due respect, hv From phil_rhodes at rocketmail.com Mon Aug 8 02:22:01 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Mon, 08 Aug 2011 01:22:01 +0100 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <4E3F2AC9.3000902@tesseract-tech.com> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> Message-ID: > Just please consider that this isn't truly helping > the users and engineers on this code. Well, it is intended to. The fact that what I'm saying is unpopular does not make it untrue. You could also consider the fact that nobody has answered the question - because, I suspect, it is unanswerable. If I build an embedded device with ffmpeg in it and rent or sell that device to someone, how do I comply? Who, at what postal address, is the legal representative of the ffmpeg project authorised to act on its behalf in terms of license negotiations? The answer to those two questions is, I suspect, "nobody knows", because under the current rules nobody can possibly know. P From hvidal at tesseract-tech.com Mon Aug 8 02:48:42 2011 From: hvidal at tesseract-tech.com (H. Vidal, Jr.) Date: Sun, 07 Aug 2011 20:48:42 -0400 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> Message-ID: <4E3F326A.2010403@tesseract-tech.com> On 08/07/2011 08:22 PM, Phil Rhodes wrote: > >> Just please consider that this isn't truly helping >> the users and engineers on this code. > > Well, it is intended to. Then perhaps we can be constructive about this. > > The fact that what I'm saying is unpopular does not make it untrue. Again the potential benefits of open discourse... > You could also consider the fact that nobody has answered the question - > because, I suspect, it is unanswerable. I assume this q is coming up and..... > If I build an embedded device with ffmpeg in it and rent or sell that > device to someone, how do I comply? It is my understanding, perhaps flawed, that if the ffmpeg technology is used at the 'source' level (meaning, actual C from ffmpeg is used in a product) then this product is subject to GPL. This means that (again, as I understand it( one (meaning the 'device' developer and seller) would be compelled to publish the code incorporating the ffmpeg sources. I may have this wrong, but I am trying..... :) If, however, the code is used as a set of 'linked libraries' then I (again, may be wrong) believe that the LGPL clause kicks in. (are ffmpeg libraries under LGPL?) Thus, one would have to supply attribution to ffmpeg ("....my code uses ffmpeg, copyright original authors, etc...") and also centrally publish source to ffmpeg or link to library as used, in question. I kind of hope I have this wrong, so that a better expert will note my errors and I can learn a bit from this exchange.... > Who, at what postal address, is the > legal representative of the ffmpeg project authorised to act on its > behalf in terms of license negotiations? It's my understanding that there are a group of core developers, no? At least under US law, the copyright is theirs. If they publish it under an open source license, and the terms of the license are violated, a civil suit can be filed against the breaching party. > > The answer to those two questions is, I suspect, "nobody knows", because > under the current rules nobody can possibly know. You may be right, but perhaps we can try to get this cleared up....let's give it a whirl. Any correcting comments most welcome. hv From lists at monopolis.org Mon Aug 8 03:32:37 2011 From: lists at monopolis.org (Evan Scanlan) Date: Sun, 7 Aug 2011 21:32:37 -0400 Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even when -sn switch used Message-ID: Hi, Apologies if this has been posted/solved elsewhere? I searched and did not find this particular issue. Also, apologies if I do not include the correct set of information. If I am missing something then please let me know and I will repost. I am using FFMPEG to encode from an Matroska video (mkv) to a Quicktime mov file. ?Issue is that mov file includes subtitle stream even when -sn switch is used. ?This occurs with both the Fedora 12 current packaged binary (ffmpeg-0.7-45_rc1.fc12.x86_64) and with a build from GIT source (about one month old). ?It also happens when using the MacPorts FFMPEG. ?This does not seem to be a problem when converting from other file formats to mov (e.g. m2ts to mov) Here is a little example using the precompiled binary Fedora package. Notice that the second time I run ffmpeg the subtitle stream appears. Any help or suggestions would be much appreciated. Thanks! Evan [evan at linux]$ ffmpeg -t 5 -i test.mkv -map 0:0 -map 0:1 -vcodec copy -acodec libmp3lame -ac 2 -async 2 -ar 48000 -ab 448k -sn -f mov test.mov ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth --enable-libopencv --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping 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. 77. 0 / 1. 77. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 [matroska,webm @ 0x1fa3f40] max_analyze_duration reached [matroska,webm @ 0x1fa3f40] Estimating duration from bitrate, this may be inaccurate Input #0, matroska,webm, from 'test.mkv': Duration: 00:51:10.04, start: 0.000000, bitrate: 448 kb/s Chapter #0.0: start 0.000000, end 531.739533 Metadata: title : Chapter 00 Chapter #0.1: start 531.739533, end 1150.858022 Metadata: title : Chapter 01 Chapter #0.2: start 1150.858022, end 1862.610733 Metadata: title : Chapter 02 Chapter #0.3: start 1862.610733, end 2475.014200 Metadata: title : Chapter 03 Chapter #0.4: start 2475.014200, end 2475.014200 Metadata: title : Chapter 04 Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 23.98 tbc (default) Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s (default) Metadata: title : 3/2+1 Stream #0.2(eng): Subtitle: pgssub (default) Output #0, mov, to 'test.mov': Metadata: encoder : Lavf52.108.0 Chapter #0.0: start 0.000000, end 5.000000 Metadata: title : Chapter 00 Stream #0.0(eng): Video: vc-1 / 0x312D6376, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], q=2-31, 24k tbn, 23.98 tbc (default) Stream #0.1(eng): Audio: libmp3lame, 48000 Hz, 2 channels, s16, 448 kb/s (default) Metadata: title : 3/2+1 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 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 [mov @ 0x1fdc3a0] pts has no value Last message repeated 54 times frame= 120 fps= 0 q=-1.0 Lsize= 651kB time=5.00 bitrate=1064.9kbits/s video:449kB audio:198kB global headers:0kB muxing overhead 0.572616% [evan at linux]$ ffmpeg -i test.mov ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth --enable-libopencv --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping 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. 77. 0 / 1. 77. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt creation_time : 1970-01-01 00:00:00 encoder : Lavf52.108.0 Duration: 00:00:05.06, start: 0.000000, bitrate: 1052 kb/s Chapter #0.0: start 0.000000, end 5.000000 Metadata: title : Chapter 00 Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 734 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 23.98 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Audio: mp3, 48000 Hz, 2 channels, s16, 320 kb/s Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.2(eng): Subtitle: text / 0x74786574, 0 kb/s Metadata: creation_time : 1970-01-01 00:00:00 At least one output file must be specified From humer4489 at gmail.com Mon Aug 8 07:55:21 2011 From: humer4489 at gmail.com (Patrick) Date: Mon, 8 Aug 2011 01:55:21 -0400 Subject: [FFmpeg-user] c implementation of a basic ffmpeg command Message-ID: <73A4FB20-6A86-4AE8-8043-39310504580B@gmail.com> I have successfully created an xcode project with ffmpeg. I am just learning c and would like to carry out some basic ffmpeg command line equivalents , such as: ffmpeg -y -ss 00:00:30.00 -t 59 -i songIn -ab 24k songOut How would this line be implemented in objective c, and is there any tutorials out there that would explain it. I have searched and found that all examples and tutorials are directed to video only. Thanks for your help. Pat From mathieu_g1 at gmx.fr Mon Aug 8 09:31:07 2011 From: mathieu_g1 at gmx.fr (Mathieu Goutfreind) Date: Mon, 08 Aug 2011 09:31:07 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> Message-ID: <4E3F90BB.2030104@gmx.fr> Le 08/08/2011 02:22, Phil Rhodes a ?crit : > >> Just please consider that this isn't truly helping >> the users and engineers on this code. > > Well, it is intended to. > > The fact that what I'm saying is unpopular does not make it untrue. > > You could also consider the fact that nobody has answered the question > - because, I suspect, it is unanswerable. > > If I build an embedded device with ffmpeg in it and rent or sell that > device to someone, how do I comply? Who, at what postal address, is > the legal representative of the ffmpeg project authorised to act on > its behalf in terms of license negotiations? > > The answer to those two questions is, I suspect, "nobody knows", > because under the current rules nobody can possibly know. > > P > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user To comply you can simply put on your website the source code used and put a link appearing in the product. Like a legal notice (ex : this product contain programs under the gpl licence : ....). you don't need to contact a legal representative of the project. -------------- next part -------------- A non-text attachment was scrubbed... Name: mathieu_g1.vcf Type: text/x-vcard Size: 186 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 898 bytes Desc: OpenPGP digital signature URL: From fcassia at gmail.com Mon Aug 8 10:03:08 2011 From: fcassia at gmail.com (Fernando Cassia) Date: Mon, 8 Aug 2011 05:03:08 -0300 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> Message-ID: On Sun, Aug 7, 2011 at 21:22, Phil Rhodes wrote: > If I build an embedded device with ffmpeg in it and rent or sell that device > to someone, how do I comply? This is stupid. AOL along with thousands of others have sold devices with GPL code embedded. The first one I bought was the "Gateway Connected Touch Pad" (GCTP) a device which ran Midori Linux and AOL?s proprietary client for Linux on top. Guess what? AOL made the full source code available, both the original Linux source code along with the modifications AOL had to do to make it run on the device http://web.archive.org/web/20040414212406/http://opensource.aol.com/ "This is the host site for Open Source software used in the Instant AOL device. Below are links to the extended version of Transmeta Corporation's MidoriTM Linux distribution. These images contain MidoriTM and the Instant AOL modifications to MidoriTM. A variety of Open Source licenses are used within MidoriTM(depending on the area), the most common one is the GNU General Public License. For full license details, see source code headers in individual Linux packages." So, to answer your question, that is how you?re supposed to comply. Another example is Linksys, whose Wi-Fi routers run Linux. And here is how they comply. http://homesupport.cisco.com/en-us/gplcodecenter But then you ask: "If I build an embedded device with ffmpeg in it and rent or sell that device to someone, how do I comply? Who, at what postal address, is the legal representative of the ffmpeg project authorised to act on its behalf in terms of license negotiations?" There is nothing to negotiate. You comply like Cisco or AOL did, by stating somewhere on the documentation and/or web site that the device includes GPL-licensed software,and make an offer to provide the source code to interested parties so if they want to fix a bug without your assistance on the GPL code, or re-use your GPL binary on a similar device, they can. If you do ship a device with GPL software but fail to provide the source, you will find that some hackers will eventually realize about this and you and your company listed on the GPL-violations site, with possible legal hassle. I respectfully suggest you take further GPL-related queries to the GPL-legal mailing list http://gpl-violations.org/mailinglists.html FC From erik.torlen at apicasystem.com Mon Aug 8 10:44:21 2011 From: erik.torlen at apicasystem.com (Erik Torlen) Date: Mon, 8 Aug 2011 08:44:21 +0000 Subject: [FFmpeg-user] Running ffplay without X Message-ID: <15328F14E709E14F86C3FF7392216A5B013E69B6@SESTIVDC02P.Apica.local> Hi, I would like to get out stats from a running stream like bytes sent/received, seconds played etc. I decided to use ffplay with its "-stats"-command to get information about a stream. The problem I'm having is that I'm trying to use ffplay on a debian without the x server, which doesn't work for me. I know that ffplay want to use a gui but I'm a bit unsure if it can be used without a gui and only command line? This is the error that I'm getting: "(*) DirectFB/Core: Single Application Core. (2008-08-18 12:36) (!) Direct/Util: opening '/dev/fb0' and '/dev/fb/0' failed --> No such file or directory (!) DirectFB/FBDev: Error opening framebuffer device! (!) DirectFB/FBDev: Use 'fbdev' option or set FRAMEBUFFER environment variable. (!) DirectFB/Core: Could not initialize 'system' core! --> Initialization error! Could not initialize SDL - DirectFBCreate: Initialization error!" Is it possible to use ffplay with only command line and how do I do that? Do you have any other recommendation of what to use (if ffplay is the wrong choice for my misson). I would like to check info/stats on streams using http/rtsp/rtmp (and all its variants). Regards Erik From dave at avpreserve.com Mon Aug 8 15:31:39 2011 From: dave at avpreserve.com (Dave Rice) Date: Mon, 8 Aug 2011 09:31:39 -0400 Subject: [FFmpeg-user] Scene detection In-Reply-To: <1312726584077-3724931.post@n4.nabble.com> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> Message-ID: <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> On Aug 7, 2011, at 10:16 AM, lovelove wrote: > Hi, can someone please help a desperate soul like me? > > I am in the same situation as Fori Pepe > > > Fori Pepe wrote: >> >> I would like to export/extract images from videos at every scene changes. >> Is it possible to use ffmpeg to create images on scene changes? >> It would be the most comfortable to use it like this: >> ffmpeg -i test.mpg --detect_scene_changes newscene[00-99].png >> > > ... i.e. export first frame of each new scene as an image. This effectively > generates a storyboard. > It also would be an enormous help in identifying added and deleted scenes > when comparing 2 videos. I don't know how to do this in ffmpeg along, but you could pipe yuv4mpegpipe output from ffmpeg to yuvdiff. See: http://www.silicontrip.net/~mark/view.php/lavtools/20100906-yuvdiff For instance: ffmpeg -i input_file.mov -pix_fmt yuv420p -f yuv4mpegpipe - | yuvdiff -g > frame_difference_values.txt You'll get a two column output of frame number and a number representing the difference between the current frame and the prior. Then determine a threshold for what is a scene cut, parse the output to get the frame numbers of cuts, then use those to cut the desired frames out. Dave Rice avpreserve.com From pb at das-werkstatt.com Mon Aug 8 16:12:54 2011 From: pb at das-werkstatt.com (Peter B.) Date: Mon, 08 Aug 2011 16:12:54 +0200 Subject: [FFmpeg-user] Scene detection In-Reply-To: <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> Message-ID: <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> Quoting Dave Rice : > For instance: > ffmpeg -i input_file.mov -pix_fmt yuv420p -f yuv4mpegpipe - | > yuvdiff -g > frame_difference_values.txt > > You'll get a two column output of frame number and a number > representing the difference between the current frame and the prior. > Then determine a threshold for what is a scene cut, parse the output > to get the frame numbers of cuts, then use those to cut the desired > frames out. For getting a list of detected scene cut offsets, I can also suggest using "shotdetect" by Johan Mathe: http://shotdetect.nonutc.fr/ (NOTE: The current release is quite old and contains some issues, but I've fixed some of them and Johan agreed to merge them back upstream - there's no official release containing those changes, but if you need them, let me know) ...you could take its output XML with the frame offset numbers, and use a XSLT stylesheet to generate the necessary ffmpeg commands which actually perform the cuts. However, that's just some implementation idea from the top of my head :) Or is there any way to feed some sort of "edit list" to ffmpeg? Regards, Peter B. From phil_rhodes at rocketmail.com Mon Aug 8 16:42:30 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Mon, 08 Aug 2011 15:42:30 +0100 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> Message-ID: > AOL along with thousands of others have sold devices > with GPL code embedded. I don't have AOL's legal team. > There is nothing to negotiate. So you say. No offence intended but this is exactly why it's important to have a point of contact. Whatever "the ffmpeg project" is, it's a sort of amorphous blob of people who claim to be involved in development, with no real definition of who's in charge, who can speak with authority, who can say "yes do that, that's OK" and have that actually be binding. Right now, anyone could say anything and someone else could countermand it a week later. Sure, I could go ask on gnu-violations.org, but who are they other than yet another third party with no legal relationship to the informal group of software engineers who appear, notionally, to be in charge. You see why this is a problem. P From hvidal at tesseract-tech.com Mon Aug 8 16:54:23 2011 From: hvidal at tesseract-tech.com (H. Vidal, Jr.) Date: Mon, 08 Aug 2011 10:54:23 -0400 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> Message-ID: <4E3FF89F.3060403@tesseract-tech.com> On 08/08/2011 10:42 AM, Phil Rhodes wrote: ...> > Sure, I could go ask on gnu-violations.org, but who are they other than > yet another third party with no legal relationship to the informal group > of software engineers who appear, notionally, to be in charge. > > You see why this is a problem. Perhaps it's useful to consider 'how' this is important.... (from me, this is partially because I am really trying to understand the point you are proposing....) So, allow us to understand in context.... The notion is, this is relevant in terms of a situation where there is a commercial interest in the technology? Then, because commercial interest must have a point of contact (the ffmpeg-interested party) there should be a corresponding point of contact in the ffmpeg community? Or am I wildly off-base ? I guess this kind of doesn't matter if it's a hobbyist.... it does not seem to matter if a party uses the ffmpeg tool in a shop for a bunch of stuff, free or commercial, but does not distribute binary or other code, just uses tool for a service.... Does not seem relevant to a student or research institute where investigation is being done/tool is being used, but there is no profit... (soft point here?) There may be other cases, but it seems that the only real point here is in a commercial context. Thoughts? hv From googol at wolke7.net Mon Aug 8 17:13:16 2011 From: googol at wolke7.net (googol at wolke7.net) Date: Mon, 08 Aug 2011 17:13:16 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> Message-ID: <4E3FFD0C.9030809@wolke7.net> > Whatever "the ffmpeg project" is, it's a sort of > amorphous blob of people who claim to be involved in development, with > no real definition of who's in charge, who can speak with authority, who > can say "yes do that, that's OK" and have that actually be binding. > Right now, anyone could say anything and someone else could countermand > it a week later. That is why from a legal perspective (which seems to be what you are interested in) it does not matter what someone *says*. For it to be legally binding, you would have to prove that you concluded an oral contract (or via e-mail). Good luck with that. Why would you want to do that? It's just calling for trouble. There already *is* a written contract (the licence). This is the set of rules which are legally binding. This is the legal side of the issue. Concerning the moral/ethical side there have been lots of replies already. From mathieu_g1 at gmx.fr Mon Aug 8 17:43:25 2011 From: mathieu_g1 at gmx.fr (Mathieu Goutfreind) Date: Mon, 08 Aug 2011 17:43:25 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> Message-ID: <4E40041D.2030106@gmx.fr> Le 08/08/2011 16:42, Phil Rhodes a ?crit : >> AOL along with thousands of others have sold devices >> with GPL code embedded. > > I don't have AOL's legal team. > >> There is nothing to negotiate. > > So you say. > > No offence intended but this is exactly why it's important to have a > point of contact. Whatever "the ffmpeg project" is, it's a sort of > amorphous blob of people who claim to be involved in development, with > no real definition of who's in charge, who can speak with authority, > who can say "yes do that, that's OK" and have that actually be > binding. Right now, anyone could say anything and someone else could > countermand it a week later. > > Sure, I could go ask on gnu-violations.org, but who are they other > than yet another third party with no legal relationship to the > informal group of software engineers who appear, notionally, to be in > charge. > > You see why this is a problem. > > P > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user A licence like GPL2 is a list of rights (and rules) given by the author(s) to the users, so you don't need to ask someone how to do things with FFMpeg just read the licence, in addition there are several case with the GPL licence it'll be easy to do a research and find what you want. See you soon -------------- next part -------------- A non-text attachment was scrubbed... Name: mathieu_g1.vcf Type: text/x-vcard Size: 186 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 898 bytes Desc: OpenPGP digital signature URL: From rogerdpack2 at gmail.com Mon Aug 8 18:03:17 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 8 Aug 2011 10:03:17 -0600 Subject: [FFmpeg-user] Running ffplay without X In-Reply-To: <15328F14E709E14F86C3FF7392216A5B013E69B6@SESTIVDC02P.Apica.local> References: <15328F14E709E14F86C3FF7392216A5B013E69B6@SESTIVDC02P.Apica.local> Message-ID: > The problem I'm having is that I'm trying to use ffplay on a debian without the x server, which doesn't work for me. xvfb maybe? From phil_rhodes at rocketmail.com Mon Aug 8 18:27:12 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Mon, 08 Aug 2011 17:27:12 +0100 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <4E40041D.2030106@gmx.fr> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> <4E40041D.2030106@gmx.fr> Message-ID: > you don't need to ask someone how to do > things with FFMpeg just read the licence This interpretation is, at best, optimistic. I'll repeat my example again: what if I build an embedded device, and rent it to people? The GPL (and similar licenses) don't talk about this. P From ben at cardinalpeak.com Mon Aug 8 18:32:33 2011 From: ben at cardinalpeak.com (Ben Mesander) Date: Mon, 8 Aug 2011 09:32:33 -0700 (PDT) Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> <4E40041D.2030106@gmx.fr> Message-ID: <1312821153.1937.YahooMailNeo@web161319.mail.bf1.yahoo.com> > I'll repeat my example again: what if I build an embedded device, and rent it to people? It's a non-issue because you're too busy trolling the mailing list to ever build an embedded device and rent it to anyone. Nobody forces you to use ffmpeg, if you find the licensing unsuitable, use something with a license to your liking. From erik.torlen at apicasystem.com Mon Aug 8 18:43:23 2011 From: erik.torlen at apicasystem.com (Erik Torlen) Date: Mon, 8 Aug 2011 16:43:23 +0000 Subject: [FFmpeg-user] Running ffplay without X In-Reply-To: References: <15328F14E709E14F86C3FF7392216A5B013E69B6@SESTIVDC02P.Apica.local>, Message-ID: Hi, Yes, I thought of that and it is my backup way of doing it. It would be better to have ffplay running plain without xvfb, if it is supported? Regards Erik 8 aug 2011 kl. 18:03 skrev "Roger Pack" : >> The problem I'm having is that I'm trying to use ffplay on a debian without the x server, which doesn't work for me. > > xvfb maybe? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From mathieu_g1 at gmx.fr Mon Aug 8 18:44:22 2011 From: mathieu_g1 at gmx.fr (Mathieu Goutfreind) Date: Mon, 08 Aug 2011 18:44:22 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> <4E40041D.2030106@gmx.fr> Message-ID: <4E401266.307@gmx.fr> Le 08/08/2011 18:27, Phil Rhodes a ?crit : >> you don't need to ask someone how to do >> things with FFMpeg just read the licence > > This interpretation is, at best, optimistic. > > I'll repeat my example again: what if I build an embedded device, and > rent it to people? > > The GPL (and similar licenses) don't talk about this. > > P > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user To rent is not to sell, an ISP called Free has done this and has never given the source code there was a case on this. The judge said that the device rented by Free was part of the Free network and there is no obligation to give the source code. -------------- next part -------------- A non-text attachment was scrubbed... Name: mathieu_g1.vcf Type: text/x-vcard Size: 186 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 898 bytes Desc: OpenPGP digital signature URL: From baptiste.coudurier at gmail.com Mon Aug 8 18:55:11 2011 From: baptiste.coudurier at gmail.com (Baptiste Coudurier) Date: Mon, 08 Aug 2011 09:55:11 -0700 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> <4E40041D.2030106@gmx.fr> Message-ID: <4E4014EF.6040403@gmail.com> On 08/08/2011 09:27 AM, Phil Rhodes wrote: >> you don't need to ask someone how to do >> things with FFMpeg just read the licence > > This interpretation is, at best, optimistic. > > I'll repeat my example again: what if I build an embedded device, and > rent it to people? > > The GPL (and similar licenses) don't talk about this. You can argue about the GPL, but always asking the same question when people have given reasonable answers is definitely trolling. This is not the first time this has happened. I'll give you a last warning. You will be banned. -- Baptiste COUDURIER Key fingerprint 8D77134D20CC9220201FC5DB0AC9325C5C1ABAAA FFmpeg maintainer http://www.ffmpeg.org From phil_rhodes at rocketmail.com Mon Aug 8 19:02:54 2011 From: phil_rhodes at rocketmail.com (Phil Rhodes) Date: Mon, 08 Aug 2011 18:02:54 +0100 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: <4E4014EF.6040403@gmail.com> References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> <4E40041D.2030106@gmx.fr> <4E4014EF.6040403@gmail.com> Message-ID: > You can argue about the GPL, but always asking the same question when > people have given reasonable answers is definitely trolling. > > This is not the first time this has happened. > I'll give you a last warning. You will be banned. No, it isn't the first time, so I will paraphrase the same response I gave before. Much as I regret upsetting one of the most professional and qualified members of this list, I'm not really in a position to stop you doing anything you want - if you feel that's the best response, then you should go ahead and do it. You are certainly in no position to demand anything of me. P From h.reindl at thelounge.net Mon Aug 8 19:06:44 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Mon, 08 Aug 2011 19:06:44 +0200 Subject: [FFmpeg-user] Can I use FFmpeg.exe command line in my proprietary software In-Reply-To: References: <000c01cc54fc$cf8148b0$6e83da10$@b-one.net> <4E3E8554.8050009@thelounge.net> <002501cc5504$73c0b120$5b421360$@b-one.net> <4E3E91EC.6040207@thelounge.net> <20110807213233.GA380@phare.normalesup.org> <4E3F218C.6070204@thelounge.net> <4E3F2AC9.3000902@tesseract-tech.com> <4E40041D.2030106@gmx.fr> <4E4014EF.6040403@gmail.com> Message-ID: <4E4017A4.5070107@thelounge.net> Am 08.08.2011 19:02, schrieb Phil Rhodes: > >> You can argue about the GPL, but always asking the same question when >> people have given reasonable answers is definitely trolling. >> >> This is not the first time this has happened. >> I'll give you a last warning. You will be banned. > > No, it isn't the first time, so I will paraphrase the same response I gave before. if you not smart enough to understand licenenses take some money in your hand and colsult peopole who are, if you are not smart enough and have no money you are in the wrong business > > Much as I regret upsetting one of the most professional and qualified > members of this list, I'm not really in a position to stop you doing > anything you want - if you feel that's the best response, then you should go > ahead and do it. > it is the only repsonse which makes sense to stupid trolls > You are certainly in no position to demand anything of me so why are you trolling here instead write your own software with your own rules if you are unwilling to accpet the rules of others and unwilling to understand them? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From seandarcy2 at gmail.com Mon Aug 8 19:32:15 2011 From: seandarcy2 at gmail.com (Sean Darcy) Date: Mon, 08 Aug 2011 13:32:15 -0400 Subject: [FFmpeg-user] latest git: Continuity Check Failed, PES packet size mismatch Message-ID: I've got an .m2t file that plays correctly in mplayer. I'm trying to convert it to an h.264/aac mp4 file. ffmpeg -i 2011-Dinner-Theater.m2t -an -pass 2 -vf yadif=1:0,hqdn3d -vcodec libx264 -preset veryslow -tune film -r 60000/1001 -timestamp now -b 4000k -threads 0 2011-Dinner-Theater.m4v Input #0, mpegts, from '2011-Dinner-Theater.m2t': Duration: 00:30:42.60, start: 0.937767, bitrate: 27001 kb/s Program 100 Stream #0.0[0x810]: Video: mpeg2video (Main), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], 25000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0.1[0x814]: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s Stream #0.2[0x815]: Data: [160][0][0][0] / 0x00A0 Stream #0.3[0x811]: Data: [161][0][0][0] / 0x00A1 I keep getting this warning/error: Continuity Check Failed PES packet size mismatch What can/should i do? sean From mandycyiu at yahoo.com.hk Mon Aug 8 21:15:55 2011 From: mandycyiu at yahoo.com.hk (tonykhlo) Date: Mon, 8 Aug 2011 12:15:55 -0700 (PDT) Subject: [FFmpeg-user] Using video card chipset to encode. In-Reply-To: <002201ca95d4$a46fb4b0$ed4f1e10$@com> References: <002201ca95d4$a46fb4b0$ed4f1e10$@com> Message-ID: <1312830955560-3727956.post@n4.nabble.com> Dear experts, I am working on a project to add video watermark to mpeg2 videos. I am using window server 2008 x64 and E3-12XX CPU. The watermark need to be moving text and png. I am using old version of ffmpeg with vhook to do that. Since I need to do them in volume. Each video need to have the name of the customer watermarked on it. So one video with have more than 1000 copies with different watermark on them. The ffmpeg vhook solution is working but I am looking for faster solution. 1) Is there anyway to do video watermark with newer version of ffmpeg (at least multi-threade)? vhook is not supported in the new ffmpeg release. I am stuck here withthe old version. 2) Is there anyone who can offer me a solution and even better if CUDA can be incorporated. Or anyone know how to use Intel ipp with ffmpeg? I am willing to pay for either of both of the above a solution. Please point me out which direction to find such experts. thanks -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Using-video-card-chipset-to-encode-tp1014723p3727956.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From stefano.sabatini-lala at poste.it Tue Aug 9 00:25:57 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Tue, 9 Aug 2011 00:25:57 +0200 Subject: [FFmpeg-user] Scene detection In-Reply-To: <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> Message-ID: <20110808222557.GC9924@geppetto> On date Monday 2011-08-08 16:12:54 +0200, Peter B. encoded: > Quoting Dave Rice : > >For instance: > >ffmpeg -i input_file.mov -pix_fmt yuv420p -f yuv4mpegpipe - | > >yuvdiff -g > frame_difference_values.txt > > > >You'll get a two column output of frame number and a number > >representing the difference between the current frame and the > >prior. Then determine a threshold for what is a scene cut, parse > >the output to get the frame numbers of cuts, then use those to cut > >the desired frames out. > > For getting a list of detected scene cut offsets, I can also suggest > using "shotdetect" by Johan Mathe: > http://shotdetect.nonutc.fr/ > > (NOTE: The current release is quite old and contains some issues, > but I've fixed some of them and Johan agreed to merge them back > upstream - there's no official release containing those changes, but > if you need them, let me know) > > ...you could take its output XML with the frame offset numbers, and > use a XSLT stylesheet to generate the necessary ffmpeg commands > which actually perform the cuts. However, that's just some > implementation idea from the top of my head :) > > Or is there any way to feed some sort of "edit list" to ffmpeg? Scene-detection looks like a much requested feature, please file a feature request on the issue tracker. I'll hope we'll be able to find some mechanisms for allowing users to (micro?)fund and vote on the issues/features they want to be fixed. On the technical side: this has been already discussed on ffmpeg-devel, an underkill solution may be as simple as tweaking the select filter and adopt a simple pixel-per-pixel average difference as metric. For images with moving objects that's not a proper solution, and you need motion estimation code, which is already integrated in libavcodec, so the ideal solution would be to refactor the ME code in libavcodec and move it to libavutil. By doing this you may also improve the code itself, and thus improve the codecs using it at the same time. -- 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 lists at monopolis.org Tue Aug 9 01:39:53 2011 From: lists at monopolis.org (Evan Scanlan) Date: Mon, 8 Aug 2011 19:39:53 -0400 Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even when -sn switch used Message-ID: Hi, Apologies if this has been posted/solved elsewhere? I searched and did not find this particular issue. Also, I tried to post this last night and it did not seem to go through -- sorry if this is a double-post. I am using FFMPEG to encode from an Matroska video (mkv) to a Quicktime mov file. Issue is that mov file includes subtitle stream even when -sn switch is used. This occurs with both the Fedora 12 current packaged binary (ffmpeg-0.7-45_rc1.fc12.x86_64) and with a build from GIT source (about one month old). It also happens when using the MacPorts FFMPEG. This does not seem to be a problem when converting from other file formats to mov (e.g. m2ts to mov) Here is a little example using the precompiled binary Fedora package. The second time I run ffmpeg the subtitle stream appears even though conversion was run with -sn the first time. Any help or suggestions would be much appreciated. Thanks! Evan [evan at linux]$ ffmpeg -t 5 -i test.mkv -map 0:0 -map 0:1 -vcodec copy -acodec libmp3lame -ac 2 -async 2 -ar 48000 -ab 448k -sn -f mov test.mov ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth --enable-libopencv --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping 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. 77. 0 / 1. 77. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 [matroska,webm @ 0x1fa3f40] max_analyze_duration reached [matroska,webm @ 0x1fa3f40] Estimating duration from bitrate, this may be inaccurate Input #0, matroska,webm, from 'test.mkv': Duration: 00:51:10.04, start: 0.000000, bitrate: 448 kb/s Chapter #0.0: start 0.000000, end 531.739533 Metadata: title : Chapter 00 Chapter #0.1: start 531.739533, end 1150.858022 Metadata: title : Chapter 01 Chapter #0.2: start 1150.858022, end 1862.610733 Metadata: title : Chapter 02 Chapter #0.3: start 1862.610733, end 2475.014200 Metadata: title : Chapter 03 Chapter #0.4: start 2475.014200, end 2475.014200 Metadata: title : Chapter 04 Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 23.98 tbc (default) Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s (default) Metadata: title : 3/2+1 Stream #0.2(eng): Subtitle: pgssub (default) Output #0, mov, to 'test.mov': Metadata: encoder : Lavf52.108.0 Chapter #0.0: start 0.000000, end 5.000000 Metadata: title : Chapter 00 Stream #0.0(eng): Video: vc-1 / 0x312D6376, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], q=2-31, 24k tbn, 23.98 tbc (default) Stream #0.1(eng): Audio: libmp3lame, 48000 Hz, 2 channels, s16, 448 kb/s (default) Metadata: title : 3/2+1 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 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 [mov @ 0x1fdc3a0] pts has no value Last message repeated 54 times frame= 120 fps= 0 q=-1.0 Lsize= 651kB time=5.00 bitrate=1064.9kbits/s video:449kB audio:198kB global headers:0kB muxing overhead 0.572616% [evan at linux]$ ffmpeg -i test.mov ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth --enable-libopencv --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping 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. 77. 0 / 1. 77. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt creation_time : 1970-01-01 00:00:00 encoder : Lavf52.108.0 Duration: 00:00:05.06, start: 0.000000, bitrate: 1052 kb/s Chapter #0.0: start 0.000000, end 5.000000 Metadata: title : Chapter 00 Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 734 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 23.98 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Audio: mp3, 48000 Hz, 2 channels, s16, 320 kb/s Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.2(eng): Subtitle: text / 0x74786574, 0 kb/s Metadata: creation_time : 1970-01-01 00:00:00 At least one output file must be specified From x2305andy2305x at yahoo.com Tue Aug 9 12:35:33 2011 From: x2305andy2305x at yahoo.com (Andy Andy) Date: Tue, 9 Aug 2011 03:35:33 -0700 (PDT) Subject: [FFmpeg-user] (no subject) Message-ID: <1312886133.18320.YahooMailNeo@web32506.mail.mud.yahoo.com> Hi guys, I have a simple uncompressed WAV file, which i want to split. The problem is that i amost never get the precise exact amount of sound i want, as per example: ffmpeg -i 17203588.wav? -ss 1 -t 1 17203588_cut.wav should cut out exactly one second from the stream, but: ffmpeg -i 17203588_cut.wav? outputs: ffmpeg version N-31743-g324b8ad, Copyright (c) 2000-2011 the FFmpeg developers ? built on Aug? 3 2011 15:13:54 with gcc 4.5.2 ..... Input #0, wav, from '/home/alexandru-david/Exporter/exportWAV/userclips/17203588_cut.wav': ? Duration: 00:00:00.99, bitrate: 1411 kb/s ??? Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s At least one output file must be specified This is a pretty happy case though, sometimes i'm missing more millis from the output. And it's very important to get the precise length i need because i'm concatting those WAVs and sometimes i get little pauses at concat time in the final sound. Any ideas, tips or tricks for me? Thanks in advance, Regards, DAV From bouke at editb.nl Tue Aug 9 12:53:52 2011 From: bouke at editb.nl (bouke) Date: Tue, 9 Aug 2011 12:53:52 +0200 Subject: [FFmpeg-user] (no subject) References: <1312886133.18320.YahooMailNeo@web32506.mail.mud.yahoo.com> Message-ID: <024a01cc5682$a399bac0$4301a8c0@hpkantoor> ----- Original Message ----- From: "Andy Andy" To: Sent: Tuesday, August 09, 2011 12:35 PM Subject: [FFmpeg-user] (no subject) > Hi guys, > > I have a simple uncompressed WAV file, which i want to split. The problem > is that i amost never get the precise exact amount of sound i want, as per > example: > > ffmpeg -i 17203588.wav -ss 1 -t 1 17203588_cut.wav > > should cut out exactly one second from the stream, but: > > ffmpeg -i 17203588_cut.wav > > outputs: > ffmpeg version N-31743-g324b8ad, Copyright (c) 2000-2011 the FFmpeg > developers > built on Aug 3 2011 15:13:54 with gcc 4.5.2 > ..... > Input #0, wav, from > '/home/alexandru-david/Exporter/exportWAV/userclips/17203588_cut.wav': > Duration: 00:00:00.99, bitrate: 1411 kb/s > Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s > At least one output file must be specified > > > This is a pretty happy case though, sometimes i'm missing more millis from > the output. And it's very important to get the precise length i need > because i'm concatting those WAVs and sometimes i get little pauses at > concat time in the final sound. > > Any ideas, tips or tricks for me? Not sure why FFmpeg would not work, but if you can simply copy bytes into a new file, and change a few bytes here and there, you can go sample accurate. Wave is a pretty simple format, see here: https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ So, copy the header, dive into the datastream and append as much as you need, then modify size according to the amount of bytes copied. If you really want to do it well, look around your desired cut point to see if you can find low volume values, or at least close to equal values with the previous/next clip, to avoid pops in the output. Or, have a look at SOX... hth, Bouke > Thanks in advance, > > Regards, > DAV > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From x2305andy2305x at yahoo.com Tue Aug 9 13:18:43 2011 From: x2305andy2305x at yahoo.com (Andy Andy) Date: Tue, 9 Aug 2011 04:18:43 -0700 (PDT) Subject: [FFmpeg-user] (no subject) In-Reply-To: <024a01cc5682$a399bac0$4301a8c0@hpkantoor> References: <1312886133.18320.YahooMailNeo@web32506.mail.mud.yahoo.com> <024a01cc5682$a399bac0$4301a8c0@hpkantoor> Message-ID: <1312888723.18231.YahooMailNeo@web32503.mail.mud.yahoo.com> Thanks, i did have SOX as an option, but actually i found a while ago that it too was inacurate. I have a newer version now and from initial tests it seems to do the job great. Will use that. Thanks again. Regards, DAV ________________________________ From: bouke To: FFmpeg user questions and RTFMs Sent: Tuesday, August 9, 2011 1:53 PM Subject: Re: [FFmpeg-user] (no subject) ----- Original Message ----- From: "Andy Andy" To: Sent: Tuesday, August 09, 2011 12:35 PM Subject: [FFmpeg-user] (no subject) > Hi guys, > > I have a simple uncompressed WAV file, which i want to split. The problem > is that i amost never get the precise exact amount of sound i want, as per > example: > > ffmpeg -i 17203588.wav -ss 1 -t 1 17203588_cut.wav > > should cut out exactly one second from the stream, but: > > ffmpeg -i 17203588_cut.wav > > outputs: > ffmpeg version N-31743-g324b8ad, Copyright (c) 2000-2011 the FFmpeg > developers > built on Aug 3 2011 15:13:54 with gcc 4.5.2 > ..... > Input #0, wav, from > '/home/alexandru-david/Exporter/exportWAV/userclips/17203588_cut.wav': > Duration: 00:00:00.99, bitrate: 1411 kb/s > Stream #0.0: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s > At least one output file must be specified > > > This is a pretty happy case though, sometimes i'm missing more millis from > the output. And it's very important to get the precise length i need > because i'm concatting those WAVs and sometimes i get little pauses at > concat time in the final sound. > > Any ideas, tips or tricks for me? Not sure why FFmpeg would not work, but if you can simply copy bytes into a new file,? and change a few bytes here and there, you can go sample accurate. Wave is a pretty simple format, see here: https://ccrma.stanford.edu/courses/422/projects/WaveFormat/ So, copy the header, dive into the datastream and append as much as you need, then modify size according to the amount of bytes copied. If you really want to do it well, look around your desired cut point to see if you can find low volume values, or at least close to equal values with the previous/next clip, to avoid pops in the output. Or, have a look at SOX... hth, Bouke > Thanks in advance, > > Regards, > DAV > _______________________________________________ > 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 pb at das-werkstatt.com Tue Aug 9 13:31:50 2011 From: pb at das-werkstatt.com (Peter B.) Date: Tue, 09 Aug 2011 13:31:50 +0200 Subject: [FFmpeg-user] Scene detection In-Reply-To: <20110808222557.GC9924@geppetto> References: <354784.24180.qm@web110607.mail.gq1.yahoo.com> <1312726584077-3724931.post@n4.nabble.com> <39659E17-5BE5-4C7C-8E1E-0E79DB34CCE5@avpreserve.com> <20110808161254.88776wg7sdrgb946@webmail.tuwien.ac.at> <20110808222557.GC9924@geppetto> Message-ID: <20110809133150.12654mahrbxwoskm@webmail.tuwien.ac.at> Quoting Stefano Sabatini : > On the technical side: this has been already discussed on > ffmpeg-devel, an underkill solution may be as simple as tweaking the > select filter and adopt a simple pixel-per-pixel average difference as > metric. > > For images with moving objects that's not a proper solution, and you > need motion estimation code, which is already integrated in > libavcodec, so the ideal solution would be to refactor the ME code in > libavcodec and move it to libavutil. By doing this you may also > improve the code itself, and thus improve the codecs using it at the > same time. For ideas/details about how to implement a scene-cut detection, I would suggest looking at the code of shotdetect (Johan Mathe is really nice, and the license is LGPL by the way). His approach is actually really simple and straightforward, and we've now been using it in our long-term archive solution for quite a while and could provide information based on hands-on experience about what to expect from that straight-forwards algorithm. I'm mentioning this, because I think Johan's algorithm is merely a variation of a "pixel-per-pixel average difference" and it works surprisingly well - on really different kinds of content. Pb From cmhjones at gmail.com Tue Aug 9 13:36:16 2011 From: cmhjones at gmail.com (cameron) Date: Tue, 9 Aug 2011 04:36:16 -0700 (PDT) Subject: [FFmpeg-user] distortive video effects In-Reply-To: <20110806105725.GA27839@geppetto> References: <1312563590774-3721703.post@n4.nabble.com> <20110806105725.GA27839@geppetto> Message-ID: <1312889776255-3729682.post@n4.nabble.com> thanks guys! i missed the frei0r plugin first time round but got it recompiled and the pixeliz0r filter is perfect! cam -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/distortive-video-effects-tp3721703p3729682.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From krueger at signal7.de Tue Aug 9 14:04:10 2011 From: krueger at signal7.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Tue, 9 Aug 2011 14:04:10 +0200 Subject: [FFmpeg-user] avfilter to reduce light flicker Message-ID: <2C620917-EFDF-41F5-A263-FEF6A9A46DD9@signal7.de> Hi, does anyone know/can recommend a filter (maybe mplayer, frei0r?) that can reduce flicker in video footage caused by flickering light (in my case a problem with a cheap LED video lamp)? Thanks in advance, Robert From rogerdpack2 at gmail.com Tue Aug 9 20:39:36 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 9 Aug 2011 12:39:36 -0600 Subject: [FFmpeg-user] avfilter to reduce light flicker In-Reply-To: <2C620917-EFDF-41F5-A263-FEF6A9A46DD9@signal7.de> References: <2C620917-EFDF-41F5-A263-FEF6A9A46DD9@signal7.de> Message-ID: > > does anyone know/can recommend a filter (maybe mplayer, frei0r?) that can reduce flicker in video footage caused by flickering light (in my case a problem with a cheap LED video lamp)? denoise3d's temporal? From marc at hallmarcwebsites.com Tue Aug 9 22:31:52 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Tue, 9 Aug 2011 16:31:52 -0400 Subject: [FFmpeg-user] mp4 settings In-Reply-To: <20110722162222.8FF7EF0B3A@avserver.banki.hu> References: <20110722162222.8FF7EF0B3A@avserver.banki.hu> Message-ID: Wow, what a learning experience this has been. First I need to throw out a round of apologies to the list for my attitude. I thought I had a working instance of ffmpeg, libx264 and qt-faststart on my server and it turns out that what I had was a mess instead. Rebuilt everything fresh and I know have a working version of all three with a nice collection of vpre libs for libx264. Looking into faststart.py next just to make it safe for the avg user. [>] What I had put up, minus the qscale - thanks for noticing that, actually should've worked yet did not. I never once tried to send the command through anything else but a php script which didn't provide any error reporting and I should've run it through a CLI. I know better and just got lazy and a little too comfortable with my PHP skills for my own good. Having said all of this I know have two questions: 1) how can I pipe the qt-faststart command into my php exec() or CLI statement so that ffmpeg finishes the transcoding before qt tries to do the atom move? 2) if you look (black screen) at this video, http://hallmarcwebsites.com/wrdp/wp-content/uploads/2011/08/final-stren.mp4 on a AT&T Motorola Backflip (with Android 2.1u1) you get audio and no video. Here are the ffmpeg settings I used to do the transcoding (let me know what I have left out) ffmpeg -i INPUT_FILE.mov -r 25 -b 864k -acodec libfaac -ab 96k -ar 44100 -ac 2 -vcodec libx264 -vpre lossless_max -async 1 OUTPUT_FILE.mp4 From krueger at signal7.de Tue Aug 9 22:32:05 2011 From: krueger at signal7.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Tue, 9 Aug 2011 22:32:05 +0200 Subject: [FFmpeg-user] avfilter to reduce light flicker In-Reply-To: References: <2C620917-EFDF-41F5-A263-FEF6A9A46DD9@signal7.de> Message-ID: On Aug 9, 2011, at 20:39 , Roger Pack wrote: >> >> does anyone know/can recommend a filter (maybe mplayer, frei0r?) that can reduce flicker in video footage caused by flickering light (in my case a problem with a cheap LED video lamp)? > > denoise3d's temporal? s**t, of course! Thanks!! The first test produced almost perfect results. You made my day :-)) From dave.bevan at bbc.co.uk Tue Aug 9 23:10:24 2011 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Tue, 9 Aug 2011 22:10:24 +0100 Subject: [FFmpeg-user] mp4 settings References: <20110722162222.8FF7EF0B3A@avserver.banki.hu> Message-ID: >Having said all of this I know have two questions: >1) how can I pipe the qt-faststart command into my php exec() or CLI >statement so that ffmpeg finishes the transcoding before qt tries to do the >atom move? Hi Marc. How about putting your ffmpeg and qt-faststart commands into a bash script, one after the other, and execing that instead? --D. 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. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4131 bytes Desc: not available URL: From marc at hallmarcwebsites.com Tue Aug 9 23:27:54 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Tue, 9 Aug 2011 17:27:54 -0400 Subject: [FFmpeg-user] mp4 settings In-Reply-To: References: <20110722162222.8FF7EF0B3A@avserver.banki.hu> Message-ID: Hmm that may be the thing and I have been considering it however, now that I have this running through a WordPress plugin environment and something I want to pass on for other WordPress users, bash file isn't doable. Need to stick with PHP exec(). I'm thinking I need to write a class that will check for a successful shell output and then fire off the qt-faststart. Marc Hall HallMarc Websites www.HallMarcWebsites.com 610-446-3346 From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Dave Bevan Sent: Tuesday, August 09, 2011 5:10 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] mp4 settings >Having said all of this I know have two questions: >1) how can I pipe the qt-faststart command into my php exec() or CLI >statement so that ffmpeg finishes the transcoding before qt tries to do the >atom move? Hi Marc. How about putting your ffmpeg and qt-faststart commands into a bash script, one after the other, and execing that instead? --D. 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. 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. -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 5394 bytes Desc: not available URL: From seandarcy2 at gmail.com Tue Aug 9 23:46:48 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Tue, 09 Aug 2011 17:46:48 -0400 Subject: [FFmpeg-user] latest git: Continuity Check Failed, PES packet size mismatch In-Reply-To: References: Message-ID: On 08/08/2011 01:32 PM, Sean Darcy wrote: > > I've got an .m2t file that plays correctly in mplayer. I'm trying to > convert it to an h.264/aac mp4 file. > > ffmpeg -i 2011-Dinner-Theater.m2t -an -pass 2 -vf yadif=1:0,hqdn3d > -vcodec libx264 -preset veryslow -tune film -r 60000/1001 -timestamp now > -b 4000k -threads 0 2011-Dinner-Theater.m4v > > Input #0, mpegts, from '2011-Dinner-Theater.m2t': > Duration: 00:30:42.60, start: 0.937767, bitrate: 27001 kb/s > Program 100 > Stream #0.0[0x810]: Video: mpeg2video (Main), yuv420p, 1440x1080 [SAR > 4:3 DAR 16:9], 25000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc > Stream #0.1[0x814]: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s > Stream #0.2[0x815]: Data: [160][0][0][0] / 0x00A0 > Stream #0.3[0x811]: Data: [161][0][0][0] / 0x00A1 > > > I keep getting this warning/error: > > Continuity Check Failed > PES packet size mismatch > > What can/should i do? > > sean The resulting mp4 file has a pronounced shudder every 10-15 seconds or so. I assume this is related. How do I fix it? sean From ja0335 at gmail.com Tue Aug 9 23:53:48 2011 From: ja0335 at gmail.com (ja0335) Date: Tue, 9 Aug 2011 14:53:48 -0700 (PDT) Subject: [FFmpeg-user] Get frame info into an ALIGNED DATA ARRAY. :D Message-ID: <1312926828087-3731378.post@n4.nabble.com> hii My question is simple, how can i get the aligned data from an AvFrame? the format needed is : char* img_data = new char[pFrame->width*pFrame->height*4] = { RED, GREEN, BLUE, ALPHA, RED, GREEN, BLUE, ALPHA, RED, GREEN, BLUE, ALPHA, ......}; I dont know how to extract the data in this format, please help me -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Get-frame-info-into-an-ALIGNED-DATA-ARRAY-D-tp3731378p3731378.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From stefano.sabatini-lala at poste.it Wed Aug 10 00:46:37 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Wed, 10 Aug 2011 00:46:37 +0200 Subject: [FFmpeg-user] Get frame info into an ALIGNED DATA ARRAY. :D In-Reply-To: <1312926828087-3731378.post@n4.nabble.com> References: <1312926828087-3731378.post@n4.nabble.com> Message-ID: <20110809224637.GB32177@geppetto> On date Tuesday 2011-08-09 14:53:48 -0700, ja0335 encoded: > hii > > My question is simple, how can i get the aligned data from an AvFrame? > > the format needed is : > > char* img_data = new char[pFrame->width*pFrame->height*4] = { > RED, GREEN, BLUE, ALPHA, > RED, GREEN, BLUE, ALPHA, > RED, GREEN, BLUE, ALPHA, ......}; > > > I dont know how to extract the data in this format, please help me You need to read from AVFrame.data[4] and .linesize[4], *the layout depends on the pixel format used*. Check libavutil/pixfmt.h, libavutil/pixdesc.h, and libavutil/imgutils.h for more information about each pixel format layout (and you can use libswscale for converting to the wanted format). From sreemnpy at gmail.com Wed Aug 10 06:35:03 2011 From: sreemnpy at gmail.com (sreerag) Date: Tue, 9 Aug 2011 21:35:03 -0700 (PDT) Subject: [FFmpeg-user] libopenjpeg not found Message-ID: <1312950903621-3732026.post@n4.nabble.com> Hi, I am a new guy to FFMPEG. When i am compiling FFMPEG on MinGW i am getting the following error ERROR:libopenjpeg not found. In the configuration file i am enabling the libopenjpeg(--enable-libopenjpeg). Please send me the solution if anyone konws.... Thanks in advance Sreerag R -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/libopenjpeg-not-found-tp3732026p3732026.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From sreemnpy at gmail.com Wed Aug 10 07:57:07 2011 From: sreemnpy at gmail.com (Sreerag) Date: Wed, 10 Aug 2011 05:57:07 +0000 (UTC) Subject: [FFmpeg-user] libopenjpeg not found Message-ID: Hi, I am a new guy to FFMPEG. When i am compiling FFMPEG on MinGW i am getting the following error ERROR:libopenjpeg not found. In the configuration file i am enabling the libopenjpeg(--enable-libopenjpeg). Please send me the solution if anyone konws.... Thanks in advance Sreerag R From gavr.mail at gmail.com Wed Aug 10 08:15:03 2011 From: gavr.mail at gmail.com (Kirill Gavrilov) Date: Wed, 10 Aug 2011 10:15:03 +0400 Subject: [FFmpeg-user] libopenjpeg not found In-Reply-To: References: Message-ID: Hi, On Wed, Aug 10, 2011 at 9:57 AM, Sreerag wrote: > I am a new guy to FFMPEG. When i am compiling FFMPEG on MinGW i am getting > the > following error > ERROR:libopenjpeg not found. > building tool say you explicitly that library 'libopenjpeg' doesn't found. But you request it. So - did you install / build this 3rd-party library? ----------------------------------------------- Kirill Gavrilov, Software designer. From sreemnpy at gmail.com Wed Aug 10 08:20:50 2011 From: sreemnpy at gmail.com (sreerag) Date: Tue, 9 Aug 2011 23:20:50 -0700 (PDT) Subject: [FFmpeg-user] libopenjpeg not found In-Reply-To: References: <1312950903621-3732026.post@n4.nabble.com> Message-ID: no, i didn't installed this 3rd party library and i don't know how to install that one. I searched for it but i have found only some source code files.please help me and thanks for the reply... On Wed, Aug 10, 2011 at 11:45 AM, ?????? ???????? [via FFmpeg-users] < ml-node+3732138-1438338174-256511 at n4.nabble.com> wrote: > Hi, > > On Wed, Aug 10, 2011 at 9:57 AM, Sreerag <[hidden email]> > wrote: > > > I am a new guy to FFMPEG. When i am compiling FFMPEG on MinGW i am > getting > > the > > following error > > ERROR:libopenjpeg not found. > > > > building tool say you explicitly that library 'libopenjpeg' doesn't found. > But you request it. So - did you install / build this 3rd-party library? > ----------------------------------------------- > Kirill Gavrilov, > Software designer. > <[hidden email] > > _______________________________________________ > ffmpeg-user mailing list > [hidden email] > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://ffmpeg-users.933282.n4.nabble.com/libopenjpeg-not-found-tp3732026p3732138.html > To unsubscribe from libopenjpeg not found, click here. > > -- with regards ?...S????AG...? ...?...Fate Determines Who Comes Into Our Lives, But Heart Determines Who Stays...? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/libopenjpeg-not-found-tp3732026p3732149.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From stalepie at hotmail.com Wed Aug 10 08:26:39 2011 From: stalepie at hotmail.com (Matt Dickinson) Date: Wed, 10 Aug 2011 02:26:39 -0400 Subject: [FFmpeg-user] libopenjpeg not found In-Reply-To: References: <1312950903621-3732026.post@n4.nabble.com>, , Message-ID: no! http://www.youtube.com/watch?v=WVRJTFNYqzc I'm not having it! > Date: Tue, 9 Aug 2011 23:20:50 -0700 > From: sreemnpy at gmail.com > To: ffmpeg-user at ffmpeg.org > Subject: Re: [FFmpeg-user] libopenjpeg not found > > no, i didn't installed this 3rd party library and i don't know how to > install that one. I searched for it but i have found only some source code > files.please help me and thanks for the reply... > > On Wed, Aug 10, 2011 at 11:45 AM, ?????? ???????? [via FFmpeg-users] < > ml-node+3732138-1438338174-256511 at n4.nabble.com> wrote: > > > Hi, > > > > On Wed, Aug 10, 2011 at 9:57 AM, Sreerag <[hidden email]> > > wrote: > > > > > I am a new guy to FFMPEG. When i am compiling FFMPEG on MinGW i am > > getting > > > the > > > following error > > > ERROR:libopenjpeg not found. > > > > > > > building tool say you explicitly that library 'libopenjpeg' doesn't found. > > But you request it. So - did you install / build this 3rd-party library? > > ----------------------------------------------- > > Kirill Gavrilov, > > Software designer. > > <[hidden email] > > > _______________________________________________ > > ffmpeg-user mailing list > > [hidden email] > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > ------------------------------ > > If you reply to this email, your message will be added to the discussion > > below: > > > > http://ffmpeg-users.933282.n4.nabble.com/libopenjpeg-not-found-tp3732026p3732138.html > > To unsubscribe from libopenjpeg not found, click here. > > > > > > > > -- > with regards > ?...S????AG...? > ...?...Fate Determines Who Comes Into Our Lives, But Heart Determines Who > Stays...? > > > -- > View this message in context: http://ffmpeg-users.933282.n4.nabble.com/libopenjpeg-not-found-tp3732026p3732149.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 ja0335 at gmail.com Wed Aug 10 12:53:01 2011 From: ja0335 at gmail.com (ja0335) Date: Wed, 10 Aug 2011 03:53:01 -0700 (PDT) Subject: [FFmpeg-user] Get frame info into an ALIGNED DATA ARRAY. :D In-Reply-To: <20110809224637.GB32177@geppetto> References: <1312926828087-3731378.post@n4.nabble.com> <20110809224637.GB32177@geppetto> Message-ID: <1312973581904-3732561.post@n4.nabble.com> hi you... i have a question.. I was reading and AvFrame.data have the structure AvFrame.data[0] = RED AvFrame.data[1] = GREEN AvFrame.data[2] = BLUE AvFrame.data[3] = ALPHA then if i read from data[4] ( i supouse you was refering data[3]), i will have only alpha, no?, or am i wrong about the AvFram.data structure? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Get-frame-info-into-an-ALIGNED-DATA-ARRAY-D-tp3731378p3732561.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From venuiyer at yahoo.com Wed Aug 10 14:05:19 2011 From: venuiyer at yahoo.com (venuiyer) Date: Wed, 10 Aug 2011 05:05:19 -0700 (PDT) Subject: [FFmpeg-user] FFMPEG encoding for H264 Baseline 3 Profile In-Reply-To: References: <1312378208315-3715448.post@n4.nabble.com> Message-ID: <1312977919802-3732721.post@n4.nabble.com> Hello, Thanks for the replies..So finally how should my command look like for encoding a video into H.264 BaseLine 3 profile? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFMPEG-encoding-for-H264-Baseline-3-Profile-tp3715448p3732721.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From seandarcy2 at gmail.com Wed Aug 10 16:34:24 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Wed, 10 Aug 2011 10:34:24 -0400 Subject: [FFmpeg-user] latest git: Continuity Check Failed, PES packet size mismatch In-Reply-To: References: Message-ID: On 08/09/2011 05:46 PM, sean darcy wrote: > On 08/08/2011 01:32 PM, Sean Darcy wrote: >> >> I've got an .m2t file that plays correctly in mplayer. I'm trying to >> convert it to an h.264/aac mp4 file. >> >> ffmpeg -i 2011-Dinner-Theater.m2t -an -pass 2 -vf yadif=1:0,hqdn3d >> -vcodec libx264 -preset veryslow -tune film -r 60000/1001 -timestamp now >> -b 4000k -threads 0 2011-Dinner-Theater.m4v >> >> Input #0, mpegts, from '2011-Dinner-Theater.m2t': >> Duration: 00:30:42.60, start: 0.937767, bitrate: 27001 kb/s >> Program 100 >> Stream #0.0[0x810]: Video: mpeg2video (Main), yuv420p, 1440x1080 [SAR >> 4:3 DAR 16:9], 25000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc >> Stream #0.1[0x814]: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s >> Stream #0.2[0x815]: Data: [160][0][0][0] / 0x00A0 >> Stream #0.3[0x811]: Data: [161][0][0][0] / 0x00A1 >> >> >> I keep getting this warning/error: >> >> Continuity Check Failed >> PES packet size mismatch >> >> What can/should i do? >> >> sean > > The resulting mp4 file has a pronounced shudder every 10-15 seconds or > so. I assume this is related. How do I fix it? > > sean Never did get ffmpeg to work: seems to be broken. But found: https://sites.google.com/site/linuxencoding/x264-encoding-guide and used: mplayer -nosound -benchmark -vo yuv4mpeg:file=>(x264 --demuxer y4m --crf 20 --threads auto --output output.264 - 2>x264.log) 2011-Dinner-Theater.m2t That works. No shudder. Seems odd since mplayer uses ffmpeg for its plumbing, but.... sean From stefano.sabatini-lala at poste.it Wed Aug 10 16:38:18 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Wed, 10 Aug 2011 16:38:18 +0200 Subject: [FFmpeg-user] Get frame info into an ALIGNED DATA ARRAY. :D In-Reply-To: <1312973581904-3732561.post@n4.nabble.com> References: <1312926828087-3731378.post@n4.nabble.com> <20110809224637.GB32177@geppetto> <1312973581904-3732561.post@n4.nabble.com> Message-ID: <20110810143818.GA12972@geppetto> On date Wednesday 2011-08-10 03:53:01 -0700, ja0335 encoded: > hi you... > > i have a question.. I was reading and AvFrame.data have the structure > AvFrame.data[0] = RED > AvFrame.data[1] = GREEN > AvFrame.data[2] = BLUE > AvFrame.data[3] = ALPHA > > then if i read from data[4] ( i supouse you was refering data[3]), i will > have only alpha, no? >, or am i wrong about the AvFram.data structure? C language adopts 0-indexing, so data[3] is alpha. Assuming you're reading from a PIX_FMT_RGBA image, you get: // first line of data data[0] = RGBARGBARGBA....LINE_PADDING...RGBARGBARGBA...LINE_PADDING...RGBARGBARGBA... so for accessing the first byte of pixel x,y you do: uint8_t v = data[0] + y * linesize[0] + x * linestep in this case linestep = 4 (how many bytes in a pixel) Note that: linesize != width * linestep since each line data needs to be aligned (the exact amount of alignment - and thus of padding bytes, depends on architecture). For other pixel formats the exact layout (components disposition in data, linestep etc.) may be different, and you need to read/understand pixfmt.h/pixdesc.h for dealing with them. From joolzg at btinternet.com Wed Aug 10 16:43:01 2011 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Wed, 10 Aug 2011 15:43:01 +0100 (BST) Subject: [FFmpeg-user] latest git: Continuity Check Failed, PES packet size mismatch In-Reply-To: References: Message-ID: <1312987381.14945.YahooMailNeo@web86401.mail.ird.yahoo.com> >________________________________ >From: sean darcy >To: ffmpeg-user at ffmpeg.org >Sent: Wednesday, 10 August 2011, 15:34 >Subject: Re: [FFmpeg-user] latest git: Continuity Check Failed, PES packet size mismatch > >On 08/09/2011 05:46 PM, sean darcy wrote: >> On 08/08/2011 01:32 PM, Sean Darcy wrote: >>> >>> I've got an .m2t file that plays correctly in mplayer. I'm trying to >>> convert it to an h.264/aac mp4 file. >>> >>> ffmpeg -i 2011-Dinner-Theater.m2t -an -pass 2 -vf yadif=1:0,hqdn3d >>> -vcodec libx264 -preset veryslow -tune film -r 60000/1001 -timestamp now >>> -b 4000k -threads 0 2011-Dinner-Theater.m4v >>> >>> Input #0, mpegts, from '2011-Dinner-Theater.m2t': >>> Duration: 00:30:42.60, start: 0.937767, bitrate: 27001 kb/s >>> Program 100 >>> Stream #0.0[0x810]: Video: mpeg2video (Main), yuv420p, 1440x1080 [SAR >>> 4:3 DAR 16:9], 25000 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc >>> Stream #0.1[0x814]: Audio: mp2, 48000 Hz, stereo, s16, 384 kb/s >>> Stream #0.2[0x815]: Data: [160][0][0][0] / 0x00A0 >>> Stream #0.3[0x811]: Data: [161][0][0][0] / 0x00A1 >>> >>> >>> I keep getting this warning/error: >>> >>> Continuity Check Failed >>> PES packet size mismatch >>> >>> What can/should i do? >>> >>> sean >> >> The resulting mp4 file has a pronounced shudder every 10-15 seconds or >> so. I assume this is related. How do I fix it? >> >> sean > >Never did get ffmpeg to work: seems to be broken. But found: > >https://sites.google.com/site/linuxencoding/x264-encoding-guide > >and used: > >? mplayer -nosound -benchmark -vo yuv4mpeg:file=>(x264 --demuxer y4m >--crf 20 --threads auto --output output.264 - 2>x264.log) >2011-Dinner-Theater.m2t > >That works. No shudder. > >Seems odd since mplayer uses ffmpeg for its plumbing, but.... > >sean > >_______________________________________________ >ffmpeg-user mailing list >ffmpeg-user at ffmpeg.org >http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > Im also getting ffmpeg dropouts, as in ffmpeg stopping working and dropping back to the cli, and i think its down to this extra code. going to investigate joolz From ja0335 at gmail.com Wed Aug 10 17:12:47 2011 From: ja0335 at gmail.com (ja0335) Date: Wed, 10 Aug 2011 08:12:47 -0700 (PDT) Subject: [FFmpeg-user] How use sws_scale to conver an AvFrame to PIX_FMT_BGR8? Message-ID: <1312989167361-3733200.post@n4.nabble.com> hiii. yes, that's the question. How use sws_scale to conver an AvFrame to PIX_FMT_BGR8? thanks in advance for your help -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-use-sws-scale-to-conver-an-AvFrame-to-PIX-FMT-BGR8-tp3733200p3733200.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ja0335 at gmail.com Wed Aug 10 17:47:30 2011 From: ja0335 at gmail.com (ja0335) Date: Wed, 10 Aug 2011 08:47:30 -0700 (PDT) Subject: [FFmpeg-user] Get frame info into an ALIGNED DATA ARRAY. :D In-Reply-To: <20110810143818.GA12972@geppetto> References: <1312926828087-3731378.post@n4.nabble.com> <20110809224637.GB32177@geppetto> <1312973581904-3732561.post@n4.nabble.com> <20110810143818.GA12972@geppetto> Message-ID: <1312991250033-3733343.post@n4.nabble.com> Hi Stefano in your sample data[0] = RGBARGBARGBA....LINE_PADDING...RGBARGBARGBA...LINE_PADDING...RGBARGBARGBA.. what you reffer LINE_PADDING? what is this? sorry my ingnorance but is the first time in image processing -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Get-frame-info-into-an-ALIGNED-DATA-ARRAY-D-tp3731378p3733343.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From stefano.sabatini-lala at poste.it Wed Aug 10 18:15:58 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Wed, 10 Aug 2011 18:15:58 +0200 Subject: [FFmpeg-user] Get frame info into an ALIGNED DATA ARRAY. :D In-Reply-To: <1312991250033-3733343.post@n4.nabble.com> References: <1312926828087-3731378.post@n4.nabble.com> <20110809224637.GB32177@geppetto> <1312973581904-3732561.post@n4.nabble.com> <20110810143818.GA12972@geppetto> <1312991250033-3733343.post@n4.nabble.com> Message-ID: <20110810161557.GA23699@geppetto> On date Wednesday 2011-08-10 08:47:30 -0700, ja0335 encoded: > Hi Stefano > > in your sample > data[0] = > RGBARGBARGBA....LINE_PADDING...RGBARGBARGBA...LINE_PADDING...RGBARGBARGBA.. > > what you reffer LINE_PADDING? what is this? > > sorry my ingnorance but is the first time in image processing Data that you should ignore, used for internal purposes. -- ffmpeg-user random tip #5 FFmpeg documentation: http://www.ffmpeg.org/documentation.html From rogerdpack2 at gmail.com Wed Aug 10 19:11:51 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 10 Aug 2011 11:11:51 -0600 Subject: [FFmpeg-user] libopenjpeg not found In-Reply-To: <1312950903621-3732026.post@n4.nabble.com> References: <1312950903621-3732026.post@n4.nabble.com> Message-ID: > In the configuration file i am enabling the > libopenjpeg(--enable-libopenjpeg). either disable it or install it first using "traditional" installation methods. From rogerdpack2 at gmail.com Wed Aug 10 19:12:59 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 10 Aug 2011 11:12:59 -0600 Subject: [FFmpeg-user] Using video card chipset to encode. In-Reply-To: <1312830955560-3727956.post@n4.nabble.com> References: <002201ca95d4$a46fb4b0$ed4f1e10$@com> <1312830955560-3727956.post@n4.nabble.com> Message-ID: > I am willing to pay for either of both of the above a solution. > > Please point me out which direction to find such experts. stackoverflow might get you some good advice... From marc at hallmarcwebsites.com Wed Aug 10 19:32:53 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Wed, 10 Aug 2011 13:32:53 -0400 Subject: [FFmpeg-user] mp4 settings In-Reply-To: References: <20110722162222.8FF7EF0B3A@avserver.banki.hu> Message-ID: OK, almost there still can't figure out why my Backflip plays the audio and no video. Any idea where I might look to try to correct his? here is what I have so far: ---BEGIN CODE--- $str = "ffmpeg -i ".$originalFileUrl." -vcodec libx264 -vpre lossless_medium -vpre baseline -acodec libfaac -ab 96k -ar 44100 -ac 2 ".$newFile; exec($str); ---END CODE--- then I run it through qt-faststart as the output from above is always set to mp4 and I want to make sure it can play on Smart devices Also, when I add these parameters, -s 480x320 -aspect 3:2, to the line above I get a file that is 4x's larger than the original. How can I limit this? From carlos at digitalartdesigners.com Wed Aug 10 12:53:18 2011 From: carlos at digitalartdesigners.com (=?ISO-8859-1?Q?Carlos_Fern=E1ndez_San_Mill=E1n?=) Date: Wed, 10 Aug 2011 11:53:18 +0100 Subject: [FFmpeg-user] FFmpeg configure fails Message-ID: Hi. I have been trying to run the following command withot sucess: ./configure --enable-nonfree --enable-gpl --enable-version3 --enable-gray --enable-runtime-cpudetect --enable-libfaac --enable-libmp3lame --enable-libvpx --enable-libx264 --enable-sram the output error is: ERROR: libvpx decoder version must be >=0.9.1 However, I have just downloaded and compiled from Git and have the following version: vp8 - WebM Project VP8 Decoder v0.9.7-4-gb84e8f2 Any idea? Thank you. [image: logo_firma.gif] *Carlos Fern?ndez San Mill?n*. *IT Manager.* *Phone Number: (+34) 928 706 117* www.digitalartdesigners.com *Facebook: *http://goo.gl/6yJ1W *Twitter: * http://goo.gl/n2s5n Este mensaje se dirige exclusivamente a su destinatario y puede contener informaci?n privilegiada o confidencial. Si no es vd. el destinatario indicado, queda notificado de que la utilizaci?n, divulgaci?n o copia sin autorizaci?n esta prohibida en virtud de la legislaci?n vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma v?a y proceda a su destrucci?n. This message is intended exclusively for its addressee and may contain information that is CONFIDENTIAL and protected by professional privilege. If you are not the intended recipient you are hereby notified that any dissemination, copy or disclosure of this communication is strictly prohibited by law. If this message has been received in error, please immediately notify us via e-mail and delete it. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 3162 bytes Desc: not available URL: From marc at hallmarcwebsites.com Wed Aug 10 21:38:09 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Wed, 10 Aug 2011 15:38:09 -0400 Subject: [FFmpeg-user] FFmpeg configure fails In-Reply-To: References: Message-ID: Have you seen this yet? http://multimedia.cx/eggs/libvpx-0-9-1-and-ffmpeg-0-6/ Marc Hall HallMarc Websites www.HallMarcWebsites.com 610-446-3346 From tomek at at-net.com.pl Wed Aug 10 22:44:22 2011 From: tomek at at-net.com.pl (=?UTF-8?B?VG9tYXN6IEfDs3JhbA==?=) Date: Wed, 10 Aug 2011 22:44:22 +0200 Subject: [FFmpeg-user] Get frame info into an ALIGNED DATA ARRAY. :D In-Reply-To: <1312991250033-3733343.post@n4.nabble.com> References: <1312926828087-3731378.post@n4.nabble.com> <20110809224637.GB32177@geppetto> <1312973581904-3732561.post@n4.nabble.com> <20110810143818.GA12972@geppetto> <1312991250033-3733343.post@n4.nabble.com> Message-ID: <4E42EDA6.5080909@at-net.com.pl> W dniu 2011-08-10 17:47, ja0335 pisze: > Hi Stefano > > in your sample > data[0] = > RGBARGBARGBA....LINE_PADDING...RGBARGBARGBA...LINE_PADDING...RGBARGBARGBA.. > > what you reffer LINE_PADDING? what is this? > > sorry my ingnorance but is the first time in image processing > > Hi, Memory is divided into segments, LINE_PADDING is aligned to the next memory segment. e.g. memory segment size is 2048 bytes, and one line video is 1536 bytes, the remaining bytes of the segment is called compensatory -- Tomasz G?ral AT-net http://www.at-net.com.pl - oprogramowanie na zam?wienie http://www.imprezylive.pl - transmisje na ?ywo w internecie Tel. +48 509 288 840 From rogerdpack2 at gmail.com Thu Aug 11 00:35:20 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 10 Aug 2011 16:35:20 -0600 Subject: [FFmpeg-user] How use sws_scale to conver an AvFrame to PIX_FMT_BGR8? In-Reply-To: <1312989167361-3733200.post@n4.nabble.com> References: <1312989167361-3733200.post@n4.nabble.com> Message-ID: > yes, that's the question. How use sws_scale to conver an AvFrame to > PIX_FMT_BGR8? maybe just -vf scale will match it with the desired output [?] From mirsev at cicese.mx Thu Aug 11 05:47:09 2011 From: mirsev at cicese.mx (Serguei Miridonov) Date: Wed, 10 Aug 2011 20:47:09 -0700 Subject: [FFmpeg-user] ffmpeg for DLNA Message-ID: <201108102047.10506.mirsev@cicese.mx> Hello, Probably this or similar question has been already asked... I'm trying to use minidlna server (https://sourceforge.net/projects/minidlna/) for playing back AVCHD video recorded by Panasonic camcorder on TV connected to DLNA capable Samsung BD-C6900 blu-ray player. In general, this setup works but... 1. Every short scene recorded by the camcorder is stored as separate file (h.264 + ac3 streams in a mpeg-ts container). 2. The player and minidlna server play every file separately. 3. Buffering time for every file is quite long (several seconds). So, when playing back complete unedited video, most of the time is spent for buffering between files. I would like to add a capability to minidlna server to represent a sequence of files as a continuous mpeg-ts stream, so that buffering will be only at the beginning of the playback or when jumping forth and back over video. As I see, this will require to read TS from the files, fix DTS and PTS for all video/audio packets to play them smoothly, and stream them to the player. Do I miss anything? My question is basically this: is ffmpeg library set a right thing to do this task? I have experience in programming but never dealt with any MPEG video. So, any links to similar software using ffmpeg will be appreciated. Thank you and best regards, Serguei. From acandido at hi-iberia.es Thu Aug 11 13:41:25 2011 From: acandido at hi-iberia.es (=?ISO-8859-1?Q?Andr=E9s_Gonz=E1lez?=) Date: Thu, 11 Aug 2011 13:41:25 +0200 Subject: [FFmpeg-user] RTMP connection long delay Message-ID: <4E43BFE5.6040901@hi-iberia.es> Hi, I'm trying to use ffmpeg to get images from a live video stream. I use a command like: ffmpeg -i rtmp://mylocation/myVideo -r 1 img%d.jpg The problem is ffmpeg waits even minutes after writing the first image. How could I avoid this delay? Regards, Andres -- Andr?s C. Gonz?lez T: 91 458 52 54 HI-IBERIA INGENIERIA Y PROYECTOS C/ Bolivia 5 - Madrid - 28016 From rogerdpack2 at gmail.com Thu Aug 11 15:13:56 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 11 Aug 2011 07:13:56 -0600 Subject: [FFmpeg-user] ffmpeg for DLNA In-Reply-To: <201108102047.10506.mirsev@cicese.mx> References: <201108102047.10506.mirsev@cicese.mx> Message-ID: > As I see, this will require to read TS from the files, fix DTS and PTS for all > video/audio packets to play them smoothly, and stream them to the player. Do I > miss anything? It might work with "broken" PTS times for the packets, as well. One thing related (though off topic) might be VLC's "sout-keep" and mencoder's "reindex" ffserver might work here, too. cheers! -roger- From mirsev at cicese.mx Thu Aug 11 16:12:43 2011 From: mirsev at cicese.mx (Serguei Miridonov) Date: Thu, 11 Aug 2011 07:12:43 -0700 Subject: [FFmpeg-user] ffmpeg for DLNA In-Reply-To: References: <201108102047.10506.mirsev@cicese.mx> Message-ID: <201108110712.44311.mirsev@cicese.mx> On Thursday 11 August 2011, Roger Pack wrote: > > As I see, this will require to read TS from the files, fix DTS and PTS > > for all video/audio packets to play them smoothly, and stream them to > > the player. Do I miss anything? > > It might work with "broken" PTS times for the packets, as well. I'm not sure. I tried just to concatenate files and play them but this does not work well. > One thing related (though off topic) might be VLC's "sout-keep" and > mencoder's "reindex" ffserver might work here, too. Thank you for your reply. Unfortunately, these words still mean nothing to me. Any links to the docs or some examples? I'm going to start to dig into ffmpeg APIs but they are huge. As I see, I need to check only one or two layers: stream container and packets/frames without video/audio decoding: 1. open an input file 2. get streams info 3. open output MPEG transport stream 4. loop: (get packet/frame, fix DTS/PTS, send it to the output stream) 5. open next input file, and so on. I would appreciate if someone familiar with ffmpeg API shows me a right direction: where to start. Is there anything like "ffmpeg for idiots"? :) From ronag89 at gmail.com Thu Aug 11 17:10:11 2011 From: ronag89 at gmail.com (Robert Nagy) Date: Thu, 11 Aug 2011 17:10:11 +0200 Subject: [FFmpeg-user] H264 Sliced Decoding Access Violation? Message-ID: I'm playing around a bit with thread decoding of H264. I would like to decode slices. However, I get an access violation inside of "thread_execute" at the line "int r = func(s, reinterpret_cast(arg) + n*size);". This works just fine with mpeg2 decoding. Any ideas as to why I'm getting an access violation? Is it a bug in the sliced h264 decoder, or am I doing something wrong? int thread_execute(AVCodecContext* s, int (*func)(AVCodecContext *c2, void *arg2), void* arg, int* ret, int count, int size) { for(size_t n = 0; n < count; ++n) // will be parallel_for { int r = func(s, reinterpret_cast(arg) + n*size); if(ret) ret[n] = r; } return 0; } void thread_init(AVCodecContext* s) { static const size_t MAX_THREADS = 16; // See mpegvideo.h static int dummy_opaque; s->active_thread_type = FF_THREAD_SLICE; s->thread_opaque = &dummy_opaque; s->execute = thread_execute; //s->execute2 = thread_execute2; not used s->thread_count = MAX_THREADS; // We are using a task-scheduler, so use as many "threads/tasks" as possible. } void thread_free(AVCodecContext* s) { if(!s->thread_opaque) return; s->thread_opaque = nullptr; } int custom_avcodec_open(AVCodecContext* avctx, AVCodec* codec) { avctx->thread_count = 1; if((codec->id == CODEC_ID_H264) && (codec->capabilities & CODEC_CAP_SLICE_THREADS) && (avctx->thread_type & FF_THREAD_SLICE)) { thread_init(avctx); } // ff_thread_init will not be executed since thread_count == 1. return avcodec_open(avctx, codec); } int custom_avcodec_close(AVCodecContext* avctx) { thread_free(avctx); // ff_thread_free will not be executed since thread_opaque == nullptr. return avcodec_close(avctx); } From ronag89 at gmail.com Thu Aug 11 21:18:47 2011 From: ronag89 at gmail.com (Robert Nagy) Date: Thu, 11 Aug 2011 21:18:47 +0200 Subject: [FFmpeg-user] H264 Sliced Decoding Access Violation? In-Reply-To: References: Message-ID: I also get same problem with access violation with "w32thread.c" From rogerdpack2 at gmail.com Thu Aug 11 21:37:46 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 11 Aug 2011 13:37:46 -0600 Subject: [FFmpeg-user] ffmpeg for DLNA In-Reply-To: <201108110712.44311.mirsev@cicese.mx> References: <201108102047.10506.mirsev@cicese.mx> <201108110712.44311.mirsev@cicese.mx> Message-ID: > I'm not sure. I tried just to concatenate files and play them but this does not > work well. Did you try concatenating them with ffmpeg or mencoder? (make a separate file). Cheers! -roger- From bfallik at bamboom.com Thu Aug 11 21:45:42 2011 From: bfallik at bamboom.com (Brian Fallik) Date: Thu, 11 Aug 2011 15:45:42 -0400 Subject: [FFmpeg-user] help filtering an audio stream from mpegts Message-ID: Hi, I'm trying to use lib(avcodec|format) to discard one audio stream (PID) from an input mpegts file. Using ffmpeg, the command line: $ ffmpeg -y -i infile.ts -vcodec copy -acodec copy outfile.ts -map 0.0 -map 0.2 does exactly what I want, but I want to avoid the fork&exec of ffmpeg and use the library interface directly. I can't figure out how to setup the input and output contexts. If I don't add the filtered stream to the output context, av_interleaved_write_frame() will segfault since it accesses stream[n] with n == 2, but the output context only contains a list of streams for ids 0 and 1. If I add the input stream to the output context, the PMT contains both audio PIDs, which isn't right. Does the 'discard' attribute of AVStream help? I tried setting it to AVDISCARD_ALL for the stream I want to remove but it doesn't seem to do much of anything. Can anyone shed some light on this? The ffmpeg binary does the right thing so I know it's possible. Thanks, brian From daigd at enveesoft.com Thu Aug 11 11:51:39 2011 From: daigd at enveesoft.com (=?gbk?B?tPq547Tv?=) Date: Thu, 11 Aug 2011 17:51:39 +0800 Subject: [FFmpeg-user] [ffmpeg] Encode Problem Message-ID: Hi, Sorry to bother. I have met a problem when using ffmpeg libs to generate a mp4 video, described as below. 1. I have a sequence of image frames, and a sequence of audio files (.caf format), I need to use them to generate a mp4 video. 2. There are two streams to operate, one for video and the other for audio. 3. Video stream works fine. 4. I need to encode the audio files into specified frames(based on video stream) to sync the video and audio. 5. Now, I cannot control which frame (or when) to add the audio files, the audio are played at the very beginning of the mp4 file continuously. Any suggestions would be grateful, thanks very much. ------------------ Dai Guang Da (???) Tel?13880465899 E-mail?daigd at enveesoft.com ???????????? Enveesoft Chengdu Co., Ltd. ???????????????A1?207 http://www.enveesoft.com From luj125 at gmail.com Thu Aug 11 22:43:48 2011 From: luj125 at gmail.com (James Lu) Date: Thu, 11 Aug 2011 16:43:48 -0400 Subject: [FFmpeg-user] [ffmpeg] Encode Problem In-Reply-To: References: Message-ID: 2011/8/11 ??? > Hi, > > > Sorry to bother. > > > I have met a problem when using ffmpeg libs to generate a mp4 video, > described as below. > > > 1. I have a sequence of image frames, and a sequence of audio files (.caf > format), I need to use them to generate a mp4 video. > 2. There are two streams to operate, one for video and the other for audio. > 3. Video stream works fine. > 4. I need to encode the audio files into specified frames(based on video > stream) to sync the video and audio. > 5. Now, I cannot control which frame (or when) to add the audio files, the > audio are played at the very beginning of the mp4 file continuously. > > > Any suggestions would be grateful, thanks very much. > > > ------------------ > Dai Guang Da (???) > Tel?13880465899 > E-mail?daigd at enveesoft.com > ???????????? > Enveesoft Chengdu Co., Ltd. > ???????????????A1?207 > > http://www.enveesoft.com > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > usually audio manipulations are better handled by another project called sox http://sox.sourceforge.net/ An alternative with a GUI would be audacity http://audacity.sourceforge.net/ I'm not quite sure how you want to arrange the audio and video, but I'm assuming you have image1, image2, image3, etc and you want audio1, audio2, audio3 to align respectively. In which case, I would suggest muxing the individual audio/video segments alone at first, and then concat the resultant videos together? I hope this helps, if it doesn't can you be a bit clearer on how you want to arrange the image and audio sequences? Also, a command line that you are currently working with may be relevant, so you could also include that? ~James From mirsev at cicese.mx Thu Aug 11 22:49:12 2011 From: mirsev at cicese.mx (Serguei Miridonov) Date: Thu, 11 Aug 2011 13:49:12 -0700 Subject: [FFmpeg-user] ffmpeg for DLNA In-Reply-To: References: <201108102047.10506.mirsev@cicese.mx> <201108110712.44311.mirsev@cicese.mx> Message-ID: <201108111349.13025.mirsev@cicese.mx> On Thursday 11 August 2011, Roger Pack wrote: > > I'm not sure. I tried just to concatenate files and play them but this > > does not work well. > > Did you try concatenating them with ffmpeg or mencoder? (make a separate > file). Cheers! No, here I meant the concatenation by cat command. This did not work even though files were just MPEG transport streams. They do not contain a header like AVI files, right? If I concatenate them using mencoder with -vc copy -ac copy options, it works quite well. But I believe that mencoder recalculates DTS and PTS to make frames really sequential, so this was not a simple concatenation of streams. From kev at xithing.com Fri Aug 12 12:42:23 2011 From: kev at xithing.com (Kevin) Date: Fri, 12 Aug 2011 12:42:23 +0200 Subject: [FFmpeg-user] Text in upper-right-corner of a .flv Message-ID: Hello, At this moment I use a php5-webpage to upload and convert regular-movie-files into .flv files. I there for use the commandline: exec("ffmpeg -i ".$path."/".$moviepath."/".$oldfilename." -acodec mp3 -ar ".$audrange." -ab ".$audbitr." -f flv -s ".$videosize." -b ".$videobit." ".$path."/".$flvpath."/".$newfilename.".flv"); Is it possible to add a text, for example: (c) THISCLIPISMINE.XYZ, in the upper-right-corner of the entire .flv-output-file? Related question: The PHP-script takes a picture from the movie, that I can use as a thumbnail. I also want on this picture a text, for example: (c) THISCLIPISMINE.XYZ, in the upper-right-corner. Is this possible? exec("ffmpeg -y -i ".$path."/".$moviepath."/".$foldileame." -vframes 1 -ss 00:00:02 -an -vcodec png -f rawvideo -s 320x240 ".$path."/".$pngpath."/".$pngfilename.".png"); Who can help me? What do I have to do? From sreemnpy at gmail.com Fri Aug 12 13:10:06 2011 From: sreemnpy at gmail.com (sreerag) Date: Fri, 12 Aug 2011 04:10:06 -0700 (PDT) Subject: [FFmpeg-user] How can we enable dshow format in ffmpeg when compiling on windows? Message-ID: <1313147406257-3738949.post@n4.nabble.com> Hi There is no dshow format avilable in ffmpeg when i compiled ffmpeg on windows. How can we enable dshow format in ffmpeg? Please provide a solution if anyone knows... Regards Sreerag -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-can-we-enable-dshow-format-in-ffmpeg-when-compiling-on-windows-tp3738949p3738949.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From raj4126 at yahoo.com Fri Aug 12 13:19:56 2011 From: raj4126 at yahoo.com (raj4126) Date: Fri, 12 Aug 2011 04:19:56 -0700 (PDT) Subject: [FFmpeg-user] adding 3D metadata to mp4: frame-packing=3 Message-ID: <1313147996974-3738964.post@n4.nabble.com> I need to add a metadata property: frame-packing=3 to a 3D mp4 file with ffmpeg. This is the command I use: ffmpeg -i input.mp4 -vcodec libx264 -metadata frame-packing=3 output.mp4 The above command does not add a new property called frame-packing to the output mp4 file. Could someone please tell me if I am doing anything wrong, or there is another way? Thanks, Raj -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/adding-3D-metadata-to-mp4-frame-packing-3-tp3738964p3738964.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From dave.bevan at bbc.co.uk Fri Aug 12 13:41:09 2011 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Fri, 12 Aug 2011 12:41:09 +0100 Subject: [FFmpeg-user] Text in upper-right-corner of a .flv References: Message-ID: > Is it possible to add a text, for example: (c) THISCLIPISMINE.XYZ, in the >upper-right-corner of the entire .flv-output-file? Consider using the DRAWTEXT filter - it will do /exactly/ what you want. --D. 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. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 2911 bytes Desc: not available URL: From kev at xithing.com Fri Aug 12 15:31:28 2011 From: kev at xithing.com (Kevin) Date: Fri, 12 Aug 2011 15:31:28 +0200 Subject: [FFmpeg-user] Text in upper-right-corner of a .flv In-Reply-To: References: Message-ID: > > Is it possible to add a text, for example: (c) THISCLIPISMINE.XYZ, in > the > >upper-right-corner of the entire .flv-output-file? > > Consider using the DRAWTEXT filter - it will do /exactly/ what you > want. > > --D. > Ok, that's what I'was looking for. So, I just have to add the next line into the existing exec( ) commandline? drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='(c)THISCLIPISMINE.XYZ' " And ofcourse I have to check on the server where FFMPEG is running: 1) if the libfreetype is enabled: (--enable libfree type ) and 2) if the the desired font is avaible. Kevin From dave.bevan at bbc.co.uk Fri Aug 12 15:50:40 2011 From: dave.bevan at bbc.co.uk (Dave Bevan) Date: Fri, 12 Aug 2011 14:50:40 +0100 Subject: [FFmpeg-user] Text in upper-right-corner of a .flv References: Message-ID: >> > Is it possible to add a text, for example: (c) THISCLIPISMINE.XYZ, in >> the >> >upper-right-corner of the entire .flv-output-file? >> >> Consider using the DRAWTEXT filter - it will do /exactly/ what you >> want. >> >> --D. >> >Ok, that's what I'was looking for. > >So, I just have to add the next line into the existing exec( ) commandline? > >drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf:text='(c)THISCLIPISMINE.XYZ' " > > >And ofcourse I have to check on the server where FFMPEG is running: > >1) if the libfreetype is enabled: (--enable libfree type ) >and >2) if the the desired font is avaible. > > >Kevin Yes (re libfreetype etc). And consult the documentation - http://www.ffmpeg.org/libavfilter.html#SEC20 -- 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. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 3443 bytes Desc: not available URL: From rogerdpack2 at gmail.com Fri Aug 12 22:22:55 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Fri, 12 Aug 2011 14:22:55 -0600 Subject: [FFmpeg-user] How can we enable dshow format in ffmpeg when compiling on windows? In-Reply-To: <1313147406257-3738949.post@n4.nabble.com> References: <1313147406257-3738949.post@n4.nabble.com> Message-ID: > How can we enable dshow format in ffmpeg? According to http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=3&t=27&start=10 (last post) it should "just work" though I haven't tried it myself, and apparently it isn't easy... From luj125 at gmail.com Fri Aug 12 22:36:54 2011 From: luj125 at gmail.com (James Lu) Date: Fri, 12 Aug 2011 16:36:54 -0400 Subject: [FFmpeg-user] How can we enable dshow format in ffmpeg when compiling on windows? In-Reply-To: References: <1313147406257-3738949.post@n4.nabble.com> Message-ID: On Fri, Aug 12, 2011 at 4:22 PM, Roger Pack wrote: > > How can we enable dshow format in ffmpeg? > According to > http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=3&t=27&start=10 > (last post) it should "just work" though I haven't tried it myself, > and apparently it isn't easy... > Make sure you are using the most recent libav* libraries in compiling It seems you also should try --enable-avisynth ? Not sure... ref: http://ffmpeg.org/faq.html#SEC26 If you're looking for an alternative, VLC has a directshow capture mode that works moderately well. Seeing as how VLC is built on the same libraries as ffmpeg, I'm assuming it does the same job as if you got the -f dshow to work on the command line tool. Also, vlc works as a command line tool itself, try using it's syntax found here: http://wiki.videolan.org/VLC_command-line_help if you can't get ffmpeg working? Hope one of these solutions works out for you, ~James From pyprog05 at gmail.com Sat Aug 13 00:44:11 2011 From: pyprog05 at gmail.com (PyProg PyProg) Date: Sat, 13 Aug 2011 00:44:11 +0200 Subject: [FFmpeg-user] Syntax problems with different versions of FFmpeg Message-ID: Hi, I try to apply a command with FFmpeg but I have a problem with different versions of FFmpeg. The problem comes from the -padleft ... -padright ... and a version of FFmpeg. It is reported that the syntax is no longer valid ... but in a more recent version of FFmpeg syntax in question works, and I'm wondering ... That's the problem: A version of FFmpeg under windows: C:\Documents and Settings\toto1>ffmpeg -i "E:\videos_pour_ekd\machines_sous_esc lier_divx.avi" -s 1280x720 -aspect 16:9 -padleft 312 -padright 312 -padbottom -padtop 0 -padcolor 808080 -sameq -y "C:\Documents and Settings\toto1\Mes docu ents\ekd_tests\vid?o\08_08_11_video_filtre_bandes_predef.avi" FFmpeg version git-c9e16a9-Sherpya, Copyright (c) 2000-2011 the FFmpeg develope s built on Feb 4 2011 07:04:01 with gcc 4.2.5 20090330 (prerelease) [Sherpya] libavutil 50. 37. 0 / 50. 37. 0 libavcore 0. 16. 1 / 0. 16. 1 libavcodec 52.109. 0 / 52.109. 0 libavformat 52. 95. 0 / 52. 95. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1. 74. 0 / 1. 74. 0 libswscale 0. 12. 0 / 0. 12. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, avi, from 'E:\videos_pour_ekd\machines_sous_escalier_divx.avi': Duration: 00:00:06.00, start: 0.000000, bitrate: 936 kb/s Stream #0.0: Video: msmpeg4, yuv420p, 384x320, 25 tbr, 25 tbn, 25 tbc Option 'padleft' has been removed, use the pad filter instead ffmpeg: failed to set value '312' for option 'padleft' I do not understand all that this request syntax as values ??(???): C:\Documents and Settings\toto1>ffmpeg -i "E:\videos_pour_ekd\machines_sous_esca lier_divx.avi" -s 864x720 -aspect 16:9 -vf pad=0:0:864:720:black -sameq -y "C:\ Documents and Settings\toto1\Mes documents\ekd_tests\vid?o\08_08_11_video_filtre _bandes_predef.avi" FFmpeg version git-c9e16a9-Sherpya, Copyright (c) 2000-2011 the FFmpeg developer s built on Feb 4 2011 07:04:01 with gcc 4.2.5 20090330 (prerelease) [Sherpya] libavutil 50. 37. 0 / 50. 37. 0 libavcore 0. 16. 1 / 0. 16. 1 libavcodec 52.109. 0 / 52.109. 0 libavformat 52. 95. 0 / 52. 95. 0 libavdevice 52. 2. 3 / 52. 2. 3 libavfilter 1. 74. 0 / 1. 74. 0 libswscale 0. 12. 0 / 0. 12. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, avi, from 'E:/videos_pour_ekd\machines_sous_escalier_divx.avi': Duration: 00:00:06.00, start: 0.000000, bitrate: 936 kb/s Stream #0.0: Video: msmpeg4, yuv420p, 384x320, 25 tbr, 25 tbn, 25 tbc [buffer @ 01ae1370] w:384 h:320 pixfmt:yuv420p [scale @ 01ae1530] w:384 h:320 fmt:yuv420p -> w:864 h:720 fmt:yuv420p flags:0x4 [pad @ 01ae1770] w:864 h:720 -> w:864 h:720 x:864 y:720 color:0x108080FF[yuva] [pad @ 01ae1770] Input area 864:720:1728:1440 not within the padded area 0:0:864 :720 or zero-sized Error opening filters! If I put:-vf pad = 0:0:0:0: black ... it works !, why ? Here the operation under the default version of FFmpeg in Linux Mint 11 RC2: toto at toto-Samsung-NF310 ~ $ ffmpeg -i "/home/toto/Documents/ekd_tests/video/entree/machines_sous_escalier_divx.avi" -s 1296x1080 -aspect 16:9 -padleft 312 -padright 312 -padbottom 0 -padtop 0 -padcolor 808080 -sameq -y "/home/toto/Documents/ekd_tests/video/sortie/essai_filtre_video_bandes_exterieures.avi" FFmpeg version 0.6.2-4:0.6.2-1ubuntu1, Copyright (c) 2000-2010 the Libav developers built on Mar 22 2011 15:35:22 with gcc 4.5.2 configuration: --extra-version=4:0.6.2-1ubuntu1 --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-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static WARNING: library configuration mismatch libavutil configuration: --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac --enable-nonfree --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavcodec configuration: --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac --enable-nonfree --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavformat configuration: --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac --enable-nonfree --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavdevice configuration: --extra-version=4:0.6.2-1ubuntu1 --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-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libavfilter configuration: --extra-version=4:0.6.2-1ubuntu1 --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-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libswscale configuration: --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac --enable-nonfree --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay libpostproc configuration: --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libdirac --enable-libgsm --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx --disable-stripping --enable-runtime-cpudetect --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac --enable-nonfree --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared --disable-static --disable-ffmpeg --disable-ffplay 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 '/home/toto/Documents/ekd_tests/video/entree/machines_sous_escalier_divx.avi': Duration: 00:00:06.00, start: 0.000000, bitrate: 936 kb/s Stream #0.0: Video: msmpeg4, yuv420p, 384x320, 25 tbr, 25 tbn, 25 tbc Output #0, avi, to '/home/toto/Documents/ekd_tests/video/sortie/essai_filtre_video_bandes_exterieures.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0: Video: mpeg4, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 150 fps= 4 q=0.0 Lsize= 10931kB time=6.00 bitrate=14924.1kbits/s video:10922kB audio:0kB global headers:0kB muxing overhead 0.084284% Syntax:-vf pad ... Is now the syntax by default in all newer versions of FFmpeg ? I am looking for a very recent version of ffmpeg for Windows (ffmpeg.exe) with which the -padlef n -padright n ... -padcolor hexadecimal ... working properly, can you help me ? 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 luj125 at gmail.com Sat Aug 13 07:53:41 2011 From: luj125 at gmail.com (James Lu) Date: Sat, 13 Aug 2011 01:53:41 -0400 Subject: [FFmpeg-user] Syntax problems with different versions of FFmpeg In-Reply-To: References: Message-ID: On Fri, Aug 12, 2011 at 6:44 PM, PyProg PyProg wrote: > Hi, > > I try to apply a command with FFmpeg but I have a problem with > different versions of FFmpeg. The problem comes from the -padleft ... > -padright ... and a version of FFmpeg. > It is reported that the syntax is no longer valid ... but in a more > recent version of FFmpeg syntax in question works, and I'm wondering > ... > > That's the problem: > > A version of FFmpeg under windows: > > C:\Documents and Settings\toto1>ffmpeg -i > "E:\videos_pour_ekd\machines_sous_esc > lier_divx.avi" -s 1280x720 -aspect 16:9 -padleft 312 -padright 312 > -padbottom > -padtop 0 -padcolor 808080 -sameq -y "C:\Documents and Settings\toto1\Mes > docu > ents\ekd_tests\vid?o\08_08_11_video_filtre_bandes_predef.avi" > FFmpeg version git-c9e16a9-Sherpya, Copyright (c) 2000-2011 the FFmpeg > develope > s > built on Feb 4 2011 07:04:01 with gcc 4.2.5 20090330 (prerelease) > [Sherpya] > libavutil 50. 37. 0 / 50. 37. 0 > libavcore 0. 16. 1 / 0. 16. 1 > libavcodec 52.109. 0 / 52.109. 0 > libavformat 52. 95. 0 / 52. 95. 0 > libavdevice 52. 2. 3 / 52. 2. 3 > libavfilter 1. 74. 0 / 1. 74. 0 > libswscale 0. 12. 0 / 0. 12. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > Input #0, avi, from 'E:\videos_pour_ekd\machines_sous_escalier_divx.avi': > Duration: 00:00:06.00, start: 0.000000, bitrate: 936 kb/s > Stream #0.0: Video: msmpeg4, yuv420p, 384x320, 25 tbr, 25 tbn, 25 tbc > Option 'padleft' has been removed, use the pad filter instead > ffmpeg: failed to set value '312' for option 'padleft' > > I do not understand all that this request syntax as values (???): > > C:\Documents and Settings\toto1>ffmpeg -i > "E:\videos_pour_ekd\machines_sous_esca > lier_divx.avi" -s 864x720 -aspect 16:9 -vf pad=0:0:864:720:black -sameq -y > "C:\ > Documents and Settings\toto1\Mes > documents\ekd_tests\vid?o\08_08_11_video_filtre > _bandes_predef.avi" > FFmpeg version git-c9e16a9-Sherpya, Copyright (c) 2000-2011 the FFmpeg > developer > s > built on Feb 4 2011 07:04:01 with gcc 4.2.5 20090330 (prerelease) > [Sherpya] > libavutil 50. 37. 0 / 50. 37. 0 > libavcore 0. 16. 1 / 0. 16. 1 > libavcodec 52.109. 0 / 52.109. 0 > libavformat 52. 95. 0 / 52. 95. 0 > libavdevice 52. 2. 3 / 52. 2. 3 > libavfilter 1. 74. 0 / 1. 74. 0 > libswscale 0. 12. 0 / 0. 12. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > Input #0, avi, from 'E:/videos_pour_ekd\machines_sous_escalier_divx.avi': > Duration: 00:00:06.00, start: 0.000000, bitrate: 936 kb/s > Stream #0.0: Video: msmpeg4, yuv420p, 384x320, 25 tbr, 25 tbn, 25 tbc > [buffer @ 01ae1370] w:384 h:320 pixfmt:yuv420p > [scale @ 01ae1530] w:384 h:320 fmt:yuv420p -> w:864 h:720 fmt:yuv420p > flags:0x4 > [pad @ 01ae1770] w:864 h:720 -> w:864 h:720 x:864 y:720 > color:0x108080FF[yuva] > [pad @ 01ae1770] Input area 864:720:1728:1440 not within the padded area > 0:0:864 > :720 or zero-sized > Error opening filters! > > If I put:-vf pad = 0:0:0:0: black ... it works !, why ? > > Here the operation under the default version of FFmpeg in Linux Mint 11 > RC2: > > toto at toto-Samsung-NF310 ~ $ ffmpeg -i > > "/home/toto/Documents/ekd_tests/video/entree/machines_sous_escalier_divx.avi" > -s 1296x1080 -aspect 16:9 -padleft 312 -padright 312 -padbottom 0 > -padtop 0 -padcolor 808080 -sameq -y > > "/home/toto/Documents/ekd_tests/video/sortie/essai_filtre_video_bandes_exterieures.avi" > FFmpeg version 0.6.2-4:0.6.2-1ubuntu1, Copyright (c) 2000-2010 the > Libav developers > built on Mar 22 2011 15:35:22 with gcc 4.5.2 > configuration: --extra-version=4:0.6.2-1ubuntu1 --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-gpl --enable-postproc --enable-swscale > --enable-x11grab --enable-libdc1394 --enable-shared --disable-static > WARNING: library configuration mismatch > libavutil configuration: > --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr > --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib > --enable-libdirac --enable-libgsm --enable-libopenjpeg > --enable-libschroedinger --enable-libspeex --enable-libtheora > --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx > --disable-stripping --enable-runtime-cpudetect > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac > --enable-nonfree --enable-gpl --enable-postproc --enable-swscale > --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad > --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid > --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 > --enable-shared --disable-static --disable-ffmpeg --disable-ffplay > libavcodec configuration: > --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr > --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib > --enable-libdirac --enable-libgsm --enable-libopenjpeg > --enable-libschroedinger --enable-libspeex --enable-libtheora > --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx > --disable-stripping --enable-runtime-cpudetect > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac > --enable-nonfree --enable-gpl --enable-postproc --enable-swscale > --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad > --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid > --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 > --enable-shared --disable-static --disable-ffmpeg --disable-ffplay > libavformat configuration: > --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr > --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib > --enable-libdirac --enable-libgsm --enable-libopenjpeg > --enable-libschroedinger --enable-libspeex --enable-libtheora > --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx > --disable-stripping --enable-runtime-cpudetect > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac > --enable-nonfree --enable-gpl --enable-postproc --enable-swscale > --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad > --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid > --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 > --enable-shared --disable-static --disable-ffmpeg --disable-ffplay > libavdevice configuration: --extra-version=4:0.6.2-1ubuntu1 > --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-gpl > --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 > --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared > --disable-static --disable-ffmpeg --disable-ffplay > libavfilter configuration: --extra-version=4:0.6.2-1ubuntu1 > --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-gpl > --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 > --shlibdir=/usr/lib/i686/cmov --cpu=i686 --enable-shared > --disable-static --disable-ffmpeg --disable-ffplay > libswscale configuration: > --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr > --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib > --enable-libdirac --enable-libgsm --enable-libopenjpeg > --enable-libschroedinger --enable-libspeex --enable-libtheora > --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx > --disable-stripping --enable-runtime-cpudetect > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac > --enable-nonfree --enable-gpl --enable-postproc --enable-swscale > --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad > --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid > --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 > --enable-shared --disable-static --disable-ffmpeg --disable-ffplay > libpostproc configuration: > --extra-version=4:0.6.2-1ubuntu2+medibuntu1 --prefix=/usr > --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib > --enable-libdirac --enable-libgsm --enable-libopenjpeg > --enable-libschroedinger --enable-libspeex --enable-libtheora > --enable-libvorbis --enable-pthreads --enable-zlib --enable-libvpx > --disable-stripping --enable-runtime-cpudetect > --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-version3 --enable-vaapi --enable-libopenjpeg --enable-libfaac > --enable-nonfree --enable-gpl --enable-postproc --enable-swscale > --enable-x11grab --enable-libfaad --enable-libdirac --enable-libfaad > --enable-libmp3lame --enable-librtmp --enable-libx264 --enable-libxvid > --enable-libdc1394 --shlibdir=/usr/lib/i686/cmov --cpu=i686 > --enable-shared --disable-static --disable-ffmpeg --disable-ffplay > 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 > > '/home/toto/Documents/ekd_tests/video/entree/machines_sous_escalier_divx.avi': > Duration: 00:00:06.00, start: 0.000000, bitrate: 936 kb/s > Stream #0.0: Video: msmpeg4, yuv420p, 384x320, 25 tbr, 25 tbn, 25 tbc > Output #0, avi, to > > '/home/toto/Documents/ekd_tests/video/sortie/essai_filtre_video_bandes_exterieures.avi': > Metadata: > ISFT : Lavf52.64.2 > Stream #0.0: Video: mpeg4, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], > q=2-31, 200 kb/s, 25 tbn, 25 tbc > Stream mapping: > Stream #0.0 -> #0.0 > Press [q] to stop encoding > frame= 150 fps= 4 q=0.0 Lsize= 10931kB time=6.00 bitrate=14924.1kbits/s > video:10922kB audio:0kB global headers:0kB muxing overhead 0.084284% > > Syntax:-vf pad ... Is now the syntax by default in all newer versions > of FFmpeg ? > > I am looking for a very recent version of ffmpeg for Windows > (ffmpeg.exe) with which the -padlef n -padright n ... -padcolor > hexadecimal ... working properly, can you help me ? > > 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 > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Hi Pyprog -padleft etc. options are deprecated Use -vf pad=width:height:x:y:color as in libav doc found here: http://ffmpeg.org/libavfilter.html#SEC39 Check the documentation; yes it's not maintained well but even it says that the -padetc. options are deprecated :P Hope this helps, ~James From kev at xithing.com Sat Aug 13 11:07:59 2011 From: kev at xithing.com (Kevin) Date: Sat, 13 Aug 2011 11:07:59 +0200 Subject: [FFmpeg-user] Text in upper-right-corner of a .flv In-Reply-To: References: Message-ID: -----Original Message----- From: "Dave Bevan" To: "FFmpeg user questions and RTFMs" Date: Fri, 12 Aug 2011 14:50:40 +0100 Subject: Re: [FFmpeg-user] Text in upper-right-corner of a .flv > >> > Is it possible to add a text, for example: (c) THISCLIPISMINE.XYZ, > in > >> the > >> >upper-right-corner of the entire .flv-output-file? > >> > >> Consider using the DRAWTEXT filter - it will do /exactly/ what you > >> want. > >> > >> --D. > >> > > >Ok, that's what I'was looking for. > > > >So, I just have to add the next line into the existing exec( ) > commandline? > > > >drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf:te > xt='(c)THISCLIPISMINE.XYZ' " > > > > > >And ofcourse I have to check on the server where FFMPEG is running: > > > >1) if the libfreetype is enabled: (--enable libfree type ) > >and > >2) if the the desired font is avaible. > > > > > >Kevin > > Yes (re libfreetype etc). > > And consult the documentation - > http://www.ffmpeg.org/libavfilter.html#SEC20 > > -- Dave. > I still have to install the libfreetype, but that's for next week. I'm busy with other things this weekend. :) Thanks for your support! Kind regards, Kevin From pyprog05 at gmail.com Sat Aug 13 14:33:05 2011 From: pyprog05 at gmail.com (PyProg PyProg) Date: Sat, 13 Aug 2011 14:33:05 +0200 Subject: [FFmpeg-user] Syntax problems with different versions of FFmpeg In-Reply-To: References: Message-ID: 2011/8/13 James Lu : > Hi Pyprog Hi James, > -padleft etc. options are deprecated > Use -vf pad=width:height:x:y:color as in libav doc found here: > http://ffmpeg.org/libavfilter.html#SEC39 > > Check the documentation; yes it's not maintained well but even it says that > the -padetc. options are deprecated :P > > Hope this helps, Thanks for your answer, it helped me. I found a part of the solution (but I have another question, see below). That's what I do: in fact the source video is 384x320 (bands will be 208 pixels on the left side and 208 pixels on the right side, I want the video final in 1280x720 [with the bands], the video will be resized to 864x720 before): ... -s 864x720 -aspect 16:9 -vf pad=1280:720:208:0:blue ... ... but how to ensure that the color bands is defined by hex color or rgb color ? (with the syntax: -padcolor hexColor it was nice !). In fact it's a feature I added in EKD (EnKoDeur Mixer) software that I created in late 2004 and I develop. In relation to this, see this (sorry it's in French): http://ekdm.wordpress.com/2011/08/07/possibilites-etendues-pour-le-filtre-video-ajout-de-bandes-exterieures ... the interface is defined by spin colors ... that's why the old syntax with a hexadecimal color was convenient. > ~James 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 stefano.sabatini-lala at poste.it Sat Aug 13 14:47:58 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sat, 13 Aug 2011 14:47:58 +0200 Subject: [FFmpeg-user] Syntax problems with different versions of FFmpeg In-Reply-To: References: Message-ID: <20110813124758.GA27269@geppetto> On date Saturday 2011-08-13 14:33:05 +0200, PyProg PyProg encoded: > 2011/8/13 James Lu : [...] > ... -s 864x720 -aspect 16:9 -vf pad=1280:720:208:0:blue ... > > ... but how to ensure that the color bands is defined by hex color or > rgb color ? (with the syntax: -padcolor hexColor it was nice !). > > In fact it's a feature I added in EKD (EnKoDeur Mixer) software that I > created in late 2004 and I develop. > > In relation to this, see this (sorry it's in French): > > http://ekdm.wordpress.com/2011/08/07/possibilites-etendues-pour-le-filtre-video-ajout-de-bandes-exterieures > > ... the interface is defined by spin colors ... that's why the old > syntax with a hexadecimal color was convenient. >From the fine docs: |color | Specify the color of the padded area, it can be the name of a | color (case insensitive match) or a 0xRRGGBB[AA] sequence. | The default value of color is "black". -- ffmpeg-user random tip #9 One minute of audio silence with ffmpeg: ffmpeg -ar 48000 -t 60 -f s16le -acodec pcm_s16le -i /dev/zero \ -ab 64K -f mp2 -acodec mp2 -y silence.mp2 From pyprog05 at gmail.com Sat Aug 13 14:58:16 2011 From: pyprog05 at gmail.com (PyProg PyProg) Date: Sat, 13 Aug 2011 14:58:16 +0200 Subject: [FFmpeg-user] Syntax problems with different versions of FFmpeg In-Reply-To: <20110813124758.GA27269@geppetto> References: <20110813124758.GA27269@geppetto> Message-ID: 2011/8/13 Stefano Sabatini : > From the fine docs: > > |color > | ? ? ? ?Specify the color of the padded area, it can be the name of a > | ? ? ? ?color (case insensitive match) or a 0xRRGGBB[AA] sequence. > | ? ? ? ?The default value of color is "black". Sorry, I misread again. All that is fine. I just tried and it works well. 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 evan.m.scanlan at gmail.com Sat Aug 13 19:00:00 2011 From: evan.m.scanlan at gmail.com (Evan Scanlan) Date: Sat, 13 Aug 2011 13:00:00 -0400 Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even when -sn switch used Message-ID: Hi, Apologies if this has been posted/solved elsewhere? I searched and did not find this particular issue. Also, I tried to post this last night and it did not seem to go through -- sorry if this is a double-post. I am using FFMPEG to encode from an Matroska video (mkv) to a Quicktime mov file. Problem is that mov file includes subtitle stream even when -sn switch is used. This occurs with both the Fedora 12 current packaged binary (ffmpeg-0.7-45_rc1.fc12.x86_64) and with a build from GIT source (about one month old). It also happens when using the MacPorts FFMPEG. This does not seem to be a problem when converting from other file formats to mov (e.g. m2ts to mov) Here is a little example using the precompiled binary Fedora package. The second time I run ffmpeg the subtitle stream appears even though conversion was run with -sn the first time. Any help or suggestions would be much appreciated. Thanks! Evan [evan at linux]$ ffmpeg -t 5 -i test.mkv -map 0:0 -map 0:1 -vcodec copy -acodec libmp3lame -ac 2 -async 2 -ar 48000 -ab 448k -sn -f mov test.mov ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth --enable-libopencv --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping 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. 77. 0 / 1. 77. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 [matroska,webm @ 0x1fa3f40] max_analyze_duration reached [matroska,webm @ 0x1fa3f40] Estimating duration from bitrate, this may be inaccurate Input #0, matroska,webm, from 'test.mkv': Duration: 00:51:10.04, start: 0.000000, bitrate: 448 kb/s Chapter #0.0: start 0.000000, end 531.739533 Metadata: title : Chapter 00 Chapter #0.1: start 531.739533, end 1150.858022 Metadata: title : Chapter 01 Chapter #0.2: start 1150.858022, end 1862.610733 Metadata: title : Chapter 02 Chapter #0.3: start 1862.610733, end 2475.014200 Metadata: title : Chapter 03 Chapter #0.4: start 2475.014200, end 2475.014200 Metadata: title : Chapter 04 Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 23.98 tbc (default) Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s (default) Metadata: title : 3/2+1 Stream #0.2(eng): Subtitle: pgssub (default) Output #0, mov, to 'test.mov': Metadata: encoder : Lavf52.108.0 Chapter #0.0: start 0.000000, end 5.000000 Metadata: title : Chapter 00 Stream #0.0(eng): Video: vc-1 / 0x312D6376, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], q=2-31, 24k tbn, 23.98 tbc (default) Stream #0.1(eng): Audio: libmp3lame, 48000 Hz, 2 channels, s16, 448 kb/s (default) Metadata: title : 3/2+1 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 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 [mov @ 0x1fdc3a0] pts has no value Last message repeated 54 times frame= 120 fps= 0 q=-1.0 Lsize= 651kB time=5.00 bitrate=1064.9kbits/s video:449kB audio:198kB global headers:0kB muxing overhead 0.572616% [evan at linux]$ ffmpeg -i test.mov ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth --enable-libopencv --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping 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. 77. 0 / 1. 77. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt creation_time : 1970-01-01 00:00:00 encoder : Lavf52.108.0 Duration: 00:00:05.06, start: 0.000000, bitrate: 1052 kb/s Chapter #0.0: start 0.000000, end 5.000000 Metadata: title : Chapter 00 Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 734 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 23.98 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Audio: mp3, 48000 Hz, 2 channels, s16, 320 kb/s Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.2(eng): Subtitle: text / 0x74786574, 0 kb/s Metadata: creation_time : 1970-01-01 00:00:00 At least one output file must be specified From bouke at editb.nl Sat Aug 13 19:20:38 2011 From: bouke at editb.nl (bouke) Date: Sat, 13 Aug 2011 19:20:38 +0200 Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even when -snswitch used References: Message-ID: <01b101cc59dd$55546320$4301a8c0@hpkantoor> ----- Original Message ----- From: "Evan Scanlan" To: Sent: Saturday, August 13, 2011 7:00 PM Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even when -snswitch used > Hi, > > Apologies if this has been posted/solved elsewhere? I searched and did > not find this particular issue. Also, I tried to post this last night > and it did not seem to go through -- sorry if this is a double-post. > > I am using FFMPEG to encode from an Matroska video (mkv) to a > Quicktime mov file. Problem is that mov file includes subtitle stream > even when -sn switch is used. This occurs with both the Fedora 12 > current packaged binary (ffmpeg-0.7-45_rc1.fc12.x86_64) and with a > build from GIT source (about one month old). It also happens when > using the MacPorts FFMPEG. This does not seem to be a problem when > converting from other file formats to mov (e.g. m2ts to mov) > > Here is a little example using the precompiled binary Fedora package. > The second time I run ffmpeg the subtitle stream appears even though > conversion was run with -sn the first time. > > Any help or suggestions would be much appreciated. Not sure what and why, but does the metadata (aka text aka subtitle stream) bother you? (i mean, does it show on playback?) If so, can't you get rid of it in the second pass with mapping? (it appears as a seperate stream...) (or perhaps map the first pass, or pipe the second pass...) Bouke > Thanks! > Evan > > > > [evan at linux]$ ffmpeg -t 5 -i test.mkv -map 0:0 -map 0:1 -vcodec copy > -acodec libmp3lame -ac 2 -async 2 -ar 48000 -ab 448k -sn -f mov > test.mov > ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers > built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) > configuration: --prefix=/usr --libdir=/usr/lib64 > --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared > --enable-runtime-cpudetect --enable-gpl --enable-version3 > --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab > --enable-vdpau --disable-avisynth --enable-libopencv > --enable-libdc1394 --enable-libdirac --enable-libgsm > --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp > --enable-libschroedinger --enable-libspeex --enable-libtheora > --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid > --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 > -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 > -mtune=generic -fPIC' --disable-stripping > 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. 77. 0 / 1. 77. 0 > libswscale 0. 13. 0 / 0. 13. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [matroska,webm @ 0x1fa3f40] max_analyze_duration reached > [matroska,webm @ 0x1fa3f40] Estimating duration from bitrate, this may > be inaccurate > Input #0, matroska,webm, from 'test.mkv': > Duration: 00:51:10.04, start: 0.000000, bitrate: 448 kb/s > Chapter #0.0: start 0.000000, end 531.739533 > Metadata: > title : Chapter 00 > Chapter #0.1: start 531.739533, end 1150.858022 > Metadata: > title : Chapter 01 > Chapter #0.2: start 1150.858022, end 1862.610733 > Metadata: > title : Chapter 02 > Chapter #0.3: start 1862.610733, end 2475.014200 > Metadata: > title : Chapter 03 > Chapter #0.4: start 2475.014200, end 2475.014200 > Metadata: > title : Chapter 04 > Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR > 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 23.98 tbc (default) > Stream #0.1(eng): Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s (default) > Metadata: > title : 3/2+1 > Stream #0.2(eng): Subtitle: pgssub (default) > Output #0, mov, to 'test.mov': > Metadata: > encoder : Lavf52.108.0 > Chapter #0.0: start 0.000000, end 5.000000 > Metadata: > title : Chapter 00 > Stream #0.0(eng): Video: vc-1 / 0x312D6376, yuv420p, 1920x1080 > [PAR 1:1 DAR 16:9], q=2-31, 24k tbn, 23.98 tbc (default) > Stream #0.1(eng): Audio: libmp3lame, 48000 Hz, 2 channels, s16, > 448 kb/s (default) > Metadata: > title : 3/2+1 > Stream mapping: > Stream #0.0 -> #0.0 > Stream #0.1 -> #0.1 > 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 > [mov @ 0x1fdc3a0] pts has no value > Last message repeated 54 times > frame= 120 fps= 0 q=-1.0 Lsize= 651kB time=5.00 > bitrate=1064.9kbits/s > video:449kB audio:198kB global headers:0kB muxing overhead 0.572616% > > > [evan at linux]$ ffmpeg -i test.mov > ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers > built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) > configuration: --prefix=/usr --libdir=/usr/lib64 > --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared > --enable-runtime-cpudetect --enable-gpl --enable-version3 > --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab > --enable-vdpau --disable-avisynth --enable-libopencv > --enable-libdc1394 --enable-libdirac --enable-libgsm > --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb > --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp > --enable-libschroedinger --enable-libspeex --enable-libtheora > --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid > --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 > -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 > -mtune=generic -fPIC' --disable-stripping > 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. 77. 0 / 1. 77. 0 > libswscale 0. 13. 0 / 0. 13. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov': > Metadata: > major_brand : qt > minor_version : 512 > compatible_brands: qt > creation_time : 1970-01-01 00:00:00 > encoder : Lavf52.108.0 > Duration: 00:00:05.06, start: 0.000000, bitrate: 1052 kb/s > Chapter #0.0: start 0.000000, end 5.000000 > Metadata: > title : Chapter 00 > Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR > 1:1 DAR 16:9], 734 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 23.98 tbc > Metadata: > creation_time : 1970-01-01 00:00:00 > Stream #0.1(eng): Audio: mp3, 48000 Hz, 2 channels, s16, 320 kb/s > Metadata: > creation_time : 1970-01-01 00:00:00 > Stream #0.2(eng): Subtitle: text / 0x74786574, 0 kb/s > Metadata: > creation_time : 1970-01-01 00:00:00 > At least one output file must be specified > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From evan.m.scanlan at gmail.com Sat Aug 13 20:20:49 2011 From: evan.m.scanlan at gmail.com (Evan Scanlan) Date: Sat, 13 Aug 2011 14:20:49 -0400 Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even when -snswitch used In-Reply-To: <01b101cc59dd$55546320$4301a8c0@hpkantoor> References: <01b101cc59dd$55546320$4301a8c0@hpkantoor> Message-ID: On Sat, Aug 13, 2011 at 1:20 PM, bouke wrote: > ----- Original Message ----- > From: "Evan Scanlan" > To: > Sent: Saturday, August 13, 2011 7:00 PM > Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even > when -snswitch used > > >> Hi, >> >> Apologies if this has been posted/solved elsewhere? I searched and did >> not find this particular issue. ?Also, I tried to post this last night >> and it did not seem to go through -- sorry if this is a double-post. >> >> I am using FFMPEG to encode from an Matroska video (mkv) to a >> Quicktime mov file. ?Problem is that mov file includes subtitle stream >> even when -sn switch is used. ?This occurs with both the Fedora 12 >> current packaged binary (ffmpeg-0.7-45_rc1.fc12.x86_64) and with a >> build from GIT source (about one month old). ?It also happens when >> using the MacPorts FFMPEG. ?This does not seem to be a problem when >> converting from other file formats to mov (e.g. m2ts to mov) >> >> Here is a little example using the precompiled binary Fedora package. >> The second time I run ffmpeg the subtitle stream appears even though >> conversion was run with -sn the first time. >> >> Any help or suggestions would be much appreciated. > > Not sure what and why, but does the metadata (aka text aka subtitle stream) > bother you? Thanks for the response! It mainly bothers me because it makes it so that Quicktime cannot manipulate the file. So I cannot save the file from Quicktime nor can I use other programs like iFlicks (which I think uses Quicktime API) when the subtitle stream is present. On other converted mov containers (without subtitle stream) iFlicks and Quicktime work fine. > (i mean, does it show on playback?) > If so, can't you get rid of it in the second pass with mapping? (it appears > as a seperate stream...) > (or perhaps map the first pass, or pipe the second pass...) I tried that... running on the mov file again with -vocdec copy -acodec copy -sn -f mov ... does not get rid of the subtitle stream. It stays there and I cannot get rid of it no matter what I have tried. Any suggestions on how to trick ffmpeg into removing the stream would be great! From bouke at editb.nl Sat Aug 13 20:27:00 2011 From: bouke at editb.nl (bouke) Date: Sat, 13 Aug 2011 20:27:00 +0200 Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even when -snswitch used References: <01b101cc59dd$55546320$4301a8c0@hpkantoor> Message-ID: <01e701cc59e6$9ac2a760$4301a8c0@hpkantoor> ----- Original Message ----- From: "Evan Scanlan" To: "FFmpeg user questions and RTFMs" Sent: Saturday, August 13, 2011 8:20 PM Subject: Re: [FFmpeg-user] Subtitle stream added to QT MOV format even when -snswitch used > On Sat, Aug 13, 2011 at 1:20 PM, bouke wrote: >> ----- Original Message ----- >> From: "Evan Scanlan" >> To: >> Sent: Saturday, August 13, 2011 7:00 PM >> Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even >> when -snswitch used >> >> >>> Hi, >>> >>> Apologies if this has been posted/solved elsewhere? I searched and did >>> not find this particular issue. Also, I tried to post this last night >>> and it did not seem to go through -- sorry if this is a double-post. >>> >>> I am using FFMPEG to encode from an Matroska video (mkv) to a >>> Quicktime mov file. Problem is that mov file includes subtitle stream >>> even when -sn switch is used. This occurs with both the Fedora 12 >>> current packaged binary (ffmpeg-0.7-45_rc1.fc12.x86_64) and with a >>> build from GIT source (about one month old). It also happens when >>> using the MacPorts FFMPEG. This does not seem to be a problem when >>> converting from other file formats to mov (e.g. m2ts to mov) >>> >>> Here is a little example using the precompiled binary Fedora package. >>> The second time I run ffmpeg the subtitle stream appears even though >>> conversion was run with -sn the first time. >>> >>> Any help or suggestions would be much appreciated. >> >> Not sure what and why, but does the metadata (aka text aka subtitle >> stream) >> bother you? > > > Thanks for the response! > > It mainly bothers me because it makes it so that Quicktime cannot > manipulate the file. So I cannot save the file from Quicktime nor can > I use other programs like iFlicks (which I think uses Quicktime API) > when the subtitle stream is present. On other converted mov > containers (without subtitle stream) iFlicks and Quicktime work fine. This i don't understand. A subtitle track should be a custom formatted text file pasted in QT, has nothing to do with Quicktime being able to manipulate the file or not. (actually, QT player can be used to strip the track...) No idea about QT on Linux though, but this is the case on Win and Mac. >> (i mean, does it show on playback?) >> If so, can't you get rid of it in the second pass with mapping? (it >> appears >> as a seperate stream...) >> (or perhaps map the first pass, or pipe the second pass...) > > I tried that... running on the mov file again with -vocdec copy > -acodec copy -sn -f mov ... does not get rid of the subtitle stream. > It stays there and I cannot get rid of it no matter what I have tried. > Any suggestions on how to trick ffmpeg into removing the stream would > be great! Well, did you try -map instead of -sn ? Bouke > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From evan.m.scanlan at gmail.com Sun Aug 14 02:03:15 2011 From: evan.m.scanlan at gmail.com (Evan Scanlan) Date: Sat, 13 Aug 2011 20:03:15 -0400 Subject: [FFmpeg-user] Subtitle stream added to QT MOV format even when -snswitch used In-Reply-To: <01e701cc59e6$9ac2a760$4301a8c0@hpkantoor> References: <01b101cc59dd$55546320$4301a8c0@hpkantoor> <01e701cc59e6$9ac2a760$4301a8c0@hpkantoor> Message-ID: >> >> Thanks for the response! >> >> It mainly bothers me because it makes it so that Quicktime cannot >> manipulate the file. ?So I cannot save the file from Quicktime nor can >> I use other programs like iFlicks (which I think uses Quicktime API) >> when the subtitle stream is present. ?On other converted mov >> containers (without subtitle stream) iFlicks and Quicktime work fine. > > This i don't understand. > A subtitle track should be a custom formatted text file pasted in QT, > has nothing to do with Quicktime being able to manipulate the file or not. > (actually, QT player can be used to strip the track...) > No idea about QT on Linux though, but this is the case on Win and Mac. > Thanks again for helping with this. If I had to guess (and this is a 100% guess) I would say that there is not actually a subtitle track there at all -- instead maybe something in the mov container header (is there such a thing) that incorrectly marks an additional subtitle track when converting from mkv file? Then maybe when QT reads the file it sees the incorrect header and does not manipulate file correctly -- this is just a pure guess though -- I am curious if you do a conversion from mkv to mov whether you get this same outcome? >>> (i mean, does it show on playback?) No - does not show on playback (grayed out in QT Player) >>> If so, can't you get rid of it in the second pass with mapping? (it >>> appears >>> as a seperate stream...) >>> (or perhaps map the first pass, or pipe the second pass...) >> >> I tried that... running on the mov file again with -vocdec copy >> -acodec copy -sn -f mov ... does not get rid of the subtitle stream. >> It stays there and I cannot get rid of it no matter what I have tried. >> Any suggestions on how to trick ffmpeg into removing the stream would >> be great! > > Well, did you try -map instead of -sn ? Yes - happens equally if I use -map 0:0 -map 0:1 ... here is the result of running on the test.mov file that I used in my original post -- second run shows that subtitle track is still there... [evan at linux]$ ffmpeg -i test.mov -map 0:0 -map 0:1 -vcodec copy -acodec copy -sn -f mov test2.mov ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth --enable-libopencv --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping 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. 77. 0 / 1. 77. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt creation_time : 1970-01-01 00:00:00 encoder : Lavf52.108.0 Duration: 00:00:05.06, start: 0.000000, bitrate: 1052 kb/s Chapter #0.0: start 0.000000, end 5.000000 Metadata: title : Chapter 00 Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 734 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 23.98 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Audio: mp3, 48000 Hz, 2 channels, s16, 320 kb/s Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.2(eng): Subtitle: text / 0x74786574, 0 kb/s Metadata: creation_time : 1970-01-01 00:00:00 Output #0, mov, to 'test2.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt creation_time : 1970-01-01 00:00:00 encoder : Lavf52.108.0 Chapter #0.0: start 0.000000, end 5.000000 Metadata: title : Chapter 00 Stream #0.0(eng): Video: vc-1 / 0x312D6376, yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], q=2-31, 734 kb/s, 24k tbn, 23.98 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Audio: libmp3lame, 48000 Hz, 2 channels, 320 kb/s Metadata: creation_time : 1970-01-01 00:00:00 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 120 fps= 0 q=-1.0 Lsize= 651kB time=5.00 bitrate=1064.9kbits/s video:449kB audio:198kB global headers:0kB muxing overhead 0.621834% [evan at linux]$ ffmpeg -i test2.mov ffmpeg version 0.7-rc1, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 11 2011 21:15:16 with gcc 4.4.4 20100630 (Red Hat 4.4.4-10) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --enable-shared --enable-runtime-cpudetect --enable-gpl --enable-version3 --enable-postproc --enable-avfilter --enable-pthreads --enable-x11grab --enable-vdpau --disable-avisynth --enable-libopencv --enable-libdc1394 --enable-libdirac --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxavs --enable-libxvid --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --disable-stripping 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. 77. 0 / 1. 77. 0 libswscale 0. 13. 0 / 0. 13. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test2.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt creation_time : 1970-01-01 00:00:00 encoder : Lavf52.108.0 Duration: 00:00:05.06, start: 0.000000, bitrate: 1052 kb/s Chapter #0.0: start 0.000000, end 5.000000 Metadata: title : Chapter 00 Stream #0.0(eng): Video: vc1 (Advanced), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 734 kb/s, 23.98 fps, 23.98 tbr, 24k tbn, 23.98 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Audio: mp3, 48000 Hz, 2 channels, s16, 320 kb/s Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.2(eng): Subtitle: text / 0x74786574, 0 kb/s Metadata: creation_time : 1970-01-01 00:00:00 At least one output file must be specified [evan at linux]$ From rickcorteza at gmail.com Mon Aug 15 03:03:22 2011 From: rickcorteza at gmail.com (Rick C.) Date: Mon, 15 Aug 2011 09:03:22 +0800 Subject: [FFmpeg-user] aac update question Message-ID: Hi, I've been using the experimental aac encoder and it gives pretty good results most of the time, but there are still times where it produces a "tinny" sound or creates background noise. So my question would be what is the latest regarding development of the aac encoder? Is it still ongoing and should I expect that with each FFmpeg release it's getting better? Just curious where it stands thanks! rc From dr.fsilva246 at gmail.com Mon Aug 15 04:49:31 2011 From: dr.fsilva246 at gmail.com (Fernando Silva) Date: Mon, 15 Aug 2011 02:49:31 +0000 (UTC) Subject: [FFmpeg-user] Differences between ffmpeg mpeg2 codec and the official mpeg2 tm5 codec Message-ID: i'm comparing ffmpeg with official mpeg2 tm5 implementations and i?m getting some differences in PSNR measurement (tm5 indicades 4 to 5 dB less than ffmpeg). Is there some explanation for this? A post or pre-processing perhaps? Any thoughts would be helpful. From m.woywod at hff-potsdam.de Mon Aug 15 15:45:03 2011 From: m.woywod at hff-potsdam.de (Michael Woywod) Date: Mon, 15 Aug 2011 15:45:03 +0200 Subject: [FFmpeg-user] extracting a subclip, -ss and -t options do not work as expected Message-ID: <4E4922DF.9090109@hff-potsdam.de> Hello, I want to trim a video via ffmpeg cli. First obstacle, I detected was the order of parameters. Now I place -ss and -t before the -i option and get a valid output at least. This output still is not the subclip, i want to extract from the input file. I got a recent build of ffmpeg -> ffmpeg -version: ffmpeg N-31774-g6c4e9ca libavutil 51. 11. 1 / 51. 11. 1 libavcodec 53. 9. 1 / 53. 9. 1 libavformat 53. 6. 0 / 53. 6. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 28. 0 / 2. 28. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 The input is a 10 seconds video h264, created with ffmpeg via: ffmpeg -i original.mov -y -r 25 -vcodec libx264 -s 640x360 -sameq -maxrate 1500k -bufsize 500k first_transcode.mp4 following calls produce results: ffmpeg.exe -ss 5 -t 2 -i first_transcode.mp4 -y -vcodec copy -acodec copy first_trimmed_5_2.mp4 -> 7 seconds subclip, starting from 00:00:00 of the input-clip ffmpeg.exe -ss 00:00:05.000 -t 00:00:02.000 -i first_transcode.mp4 -y -vcodec copy -acodec copy first_trimmed_5_2.mp4 -> same result When playing around with numbers, the result seems to be more like: -ss 0 -t (ss+t). This even works, if the values for -ss are negative (-ss -6 -t 8 returns a 2sec clip, starting from 0 of the input file) Without -ss, the -t option seems to work fine, but i have no option to trim the first part of the clip. I'm looking for a sollution to precisely trim my clips via ffmpeg CLI for a while now but don't get it working. Any help would be greatly apreciated. I didn't find information regarding known bugs or useful examples for CLI usage yet. Best Regards Michael From belcampo at zonnet.nl Mon Aug 15 16:05:06 2011 From: belcampo at zonnet.nl (belcampo) Date: Mon, 15 Aug 2011 16:05:06 +0200 Subject: [FFmpeg-user] extracting a subclip, -ss and -t options do not work as expected In-Reply-To: <4E4922DF.9090109@hff-potsdam.de> References: <4E4922DF.9090109@hff-potsdam.de> Message-ID: <4E492792.4070401@zonnet.nl> On 08/15/2011 03:45 PM, Michael Woywod wrote: > Hello, > > I want to trim a video via ffmpeg cli. > First obstacle, I detected was the order of parameters. Now I place -ss > and -t before the -i option and get a valid output at least. This output > still is not the subclip, i want to extract from the input file. > > I got a recent build of ffmpeg -> ffmpeg -version: > ffmpeg N-31774-g6c4e9ca > libavutil 51. 11. 1 / 51. 11. 1 > libavcodec 53. 9. 1 / 53. 9. 1 > libavformat 53. 6. 0 / 53. 6. 0 > libavdevice 53. 2. 0 / 53. 2. 0 > libavfilter 2. 28. 0 / 2. 28. 0 > libswscale 2. 0. 0 / 2. 0. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > > The input is a 10 seconds video h264, created with ffmpeg via: > ffmpeg -i original.mov -y -r 25 -vcodec libx264 -s 640x360 -sameq > -maxrate 1500k -bufsize 500k first_transcode.mp4 > > following calls produce results: > > ffmpeg.exe -ss 5 -t 2 -i first_transcode.mp4 -y -vcodec copy -acodec > copy first_trimmed_5_2.mp4 > -> 7 seconds subclip, starting from 00:00:00 of the input-clip > ffmpeg.exe -ss 00:00:05.000 -t 00:00:02.000 -i first_transcode.mp4 -y > -vcodec copy -acodec copy first_trimmed_5_2.mp4 > -> same result > When playing around with numbers, the result seems to be more like: -ss > 0 -t (ss+t). This even works, if the values for -ss are negative (-ss -6 > -t 8 returns a 2sec clip, starting from 0 of the input file) > > Without -ss, the -t option seems to work fine, but i have no option to > trim the first part of the clip. I'm looking for a sollution to > precisely trim my clips via ffmpeg CLI for a while now but don't get it > working. Any help would be greatly apreciated. I didn't find information > regarding known bugs or useful examples for CLI usage yet. Trimming is done at key-frame/I-frame point. The standard parameter for distance between key-frames/GOP-size is 250 if used with x264 as encoding codec. So if the source is 10 seconds at a framerate of 25 per sec. then there will be only 1 key-frame, the 1st frame. If you want frame exact trimming, your source has to be a key-frame-only file, gop-size=1 > > Best Regards > Michael > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From rowntreerob at gmail.com Tue Aug 16 00:40:04 2011 From: rowntreerob at gmail.com (Robert Rowntree) Date: Mon, 15 Aug 2011 15:40:04 -0700 Subject: [FFmpeg-user] optimal slide encoding for low bandwidth at youtube Message-ID: i want to combine 1 slide ( vid ) and a 3gpp ( audio ) in output that is suitable for youtube upload. i have a CLI expression working for ffmpeg , but its video bandwidth is too high considering that i am muxing just a single jpeg. i have used the following , but think that i can do better on the bandwidth of the VIDEO track... *./ffmpeg -y -loop_input -f image2 -shortest -r 1/2 -i 1459.JPG -i 1459.3gpp -s 640x480 -b 100k -bt 240k -vcodec msmpeg4 -acodec libmp3lame -ab 64k -ar 22050 out.wmv* Once the output from the above is uploaded to youtube, youtube exposes 2 RTSP URI's - a high and low bandwidth version of the video. the low bandwidth youtube version still has 39K of video bandwidth and that is what i want to reduce.... its just a single slide in the video v=0 o=GoogleStreamer 98636128 107718649 IN IP4 74.125.213.247 s=Video c=IN IP4 0.0.0.0 b=AS:51 t=0 0 a=control:* a=range:npt=0-63.800000 m=video 0 RTP/AVP 98 *b=AS:39 <-- VIDEO bandwidth* a=rtpmap:98 H263-2000/90000 a=control:trackID=0 a=cliprect:0,0,144,176 a=framesize:98 176-144 a=fmtp:98 profile=0;level=10 m=audio 0 RTP/AVP 99 b=AS:12 a=rtpmap:99 AMR/8000/1 a=control:trackID=1 a=fmtp:99 octet-align * Connection #0 to host v8.cache8.c.youtube.com left intact for comparison purpose, the following URI on youtube uses just 8K for its video bandwidth and i want to try to get down near that. http://www.youtube.com/watch?v=FX9ccqQuzO4 is a typical sound track hosted on youtube having the SDP below from the rtsp feed: v=0 o=GoogleStreamer 515160982 1824503727 IN IP4 74.125.213.247 s=Video c=IN IP4 0.0.0.0 b=AS:20 t=0 0 a=control:* a=range:npt=0-190.200000 m=video 0 RTP/AVP 98 *b=AS:8 <--- low video bandwidth * a=rtpmap:98 H263-2000/90000 a=control:trackID=0 a=cliprect:0,0,144,176 a=framesize:98 176-144 a=fmtp:98 profile=0;level=10 m=audio 0 RTP/AVP 99 b=AS:12 a=rtpmap:99 AMR/8000/1 a=control:trackID=1 a=fmtp:99 octet-align From m.woywod at hff-potsdam.de Mon Aug 15 15:36:12 2011 From: m.woywod at hff-potsdam.de (Michael Woywod) Date: Mon, 15 Aug 2011 15:36:12 +0200 Subject: [FFmpeg-user] extracting a subclip, -ss and -t options do not work as expected Message-ID: <4E4920CC.1050208@hff-potsdam.de> Hello, I want to trim a video via ffmpeg cli. First obstacle, I detected was the order of parameters. Now I place -ss and -t before the -i option and get a valid output at least. This output still is not the subclip, i want to extract from the input file. I got a recent build of ffmpeg -> ffmpeg -version: ffmpeg N-31774-g6c4e9ca libavutil 51. 11. 1 / 51. 11. 1 libavcodec 53. 9. 1 / 53. 9. 1 libavformat 53. 6. 0 / 53. 6. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 28. 0 / 2. 28. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 The input is a 10 seconds video h264, created with ffmpeg via: ffmpeg -i original.mov -y -r 25 -vcodec libx264 -s 640x360 -sameq -maxrate 1500k -bufsize 500k first_transcode.mp4 following calls produce results: ffmpeg.exe -ss 5 -t 2 -i first_transcode.mp4 -y -vcodec copy -acodec copy first_trimmed_5_2.mp4 -> 7 seconds subclip, starting from 00:00:00 of the input-clip ffmpeg.exe -ss 00:00:05.000 -t 00:00:02.000 -i first_transcode.mp4 -y -vcodec copy -acodec copy first_trimmed_5_2.mp4 -> same result When playing around with numbers, the result seems to be more like: -ss 0 -t (ss+t). This even works, if the values for -ss are negative (-ss -6 -t 8 returns a 2sec clip, starting from 0 of the input file) Without -ss, the -t option seems to work fine, but i have no option to trim the first part of the clip. I'm looking for a sollution to precisely trim my clips via ffmpeg CLI for a while now but don't get it working. Any help would be greatly apreciated. I didn't find information regarding known bugs or useful examples for CLI usage yet. Best Regards Michael From jasv at acolyte-color.com Tue Aug 16 06:09:52 2011 From: jasv at acolyte-color.com (J A Stephen Viggiano) Date: Tue, 16 Aug 2011 00:09:52 -0400 Subject: [FFmpeg-user] Quantizer Matrix Message-ID: <4E49ED90.3030208@acolyte-color.com> Sorry if this is answered somewhere; I checked the man page, all the online documentation I could, and this list back to the Big Scism of 2011.... How do I get ffmpeg to use the MPEG quantizer? When use mediainfo to examine a video encoded using the xvid video coder (-vcodec libxvid) or the asp default (-acodec mpeg4), it tells me (among many other things) that the h.263 "matrix" was used; see below (sorry if this should go into pastebin instead). Assuming the mediainfo tool is correct, is there any way to specify the quantization method, such as to employ the MPEG matrices? Thank you! ================================ > ffmpeg -i ed_hd.ogv -vcodec mpeg4 -qscale 4 -acodec copy -t 10 sample.avi ffmpeg version 0.8.1, Copyright (c) 2000-2011 the FFmpeg developers built on Jul 28 2011 23:08:02 with gcc 4.5.0 20100604 [gcc-4_5-branch revision 160292] configuration: --shlibdir=/usr/lib --prefix=/usr --mandir=/usr/share/man --libdir=/usr/lib --enable-shared --disable-static --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libspeex --enable-libxvid --enable-postproc --enable-gpl --enable-x11grab --extra-cflags='-fomit-frame-pointer -fmessage-length=0 -O2 -Wall -D_FORTIFY_SOURCE=2 -fstack-protector -funwind-tables -fasynchronous-unwind-tables -g -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/include/gsm' --enable-debug --disable-stripping --enable-libgsm --enable-libschroedinger --enable-libdirac --enable-avfilter --enable-libvpx --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libdc1394 --enable-pthreads --enable-librtmp libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [theora @ 0x807da60] 7 bits left in packet 82 Input #0, ogg, from 'ed_hd.ogv': Duration: 00:10:53.78, start: 0.000000, bitrate: 558 kb/s Stream #0.0: Video: theora, yuv420p, 544x304, 24 fps, 24 tbr, 24 tbn, 24 tbc Stream #0.1: Audio: vorbis, 44100 Hz, stereo, s16 Metadata: TITLE : Elephants Dream ARTIST : Bassam Kurdali LICENSE : http://creativecommons.org/licenses/by/3.0/us/ DATE : 2006 ORGANIZATION : Orange Open Movie Project Studio LOCATION : http://www.archive.org/details/ElephantsDream [buffer @ 0x821c600] w:544 h:304 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [theora @ 0x807da60] 7 bits left in packet 82 Output #0, avi, to 'sample.avi': Metadata: ISFT : Lavf53.4.0 Stream #0.0: Video: mpeg4, yuv420p, 544x304, q=2-31, 200 kb/s, 24 tbn, 24 tbc Stream #0.1: Audio: libvorbis, 44100 Hz, stereo Metadata: TITLE : Elephants Dream ARTIST : Bassam Kurdali LICENSE : http://creativecommons.org/licenses/by/3.0/us/ DATE : 2006 ORGANIZATION : Orange Open Movie Project Studio LOCATION : http://www.archive.org/details/ElephantsDream Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help frame= 240 fps=214 q=4.0 Lsize= 631kB time=00:00:10.00 bitrate= 516.6kbits/s video:552kB audio:39kB global headers:0kB muxing overhead 6.610288% > mediainfo --Inform='Video;%Format_Settings_Matrix/String%' sample.avi Default (H.263) From hyunya77 at gmail.com Tue Aug 16 07:52:40 2011 From: hyunya77 at gmail.com (Hyun Park) Date: Tue, 16 Aug 2011 14:52:40 +0900 Subject: [FFmpeg-user] cross compile build error (gcc ver. 3.4.4) Message-ID: Hi. I'm newbie for ffmpeg. Recently, I'm trying to build the latest ffmpeg(0.8.2) using arm cross compile tool chain. Unfortunately, I met the below build error. (The result was same with another version too. e.g. 0.6.x) I used below configure option for build. (arm 720t processor) --enable-shared --disable-static --disable-avdevice --disable-ffmpeg --disable-ffplay --disable-ffprobe --disable-ffserver --disable-debug --enable-pthreads --arch=arm --disable-armv6 --disable-armv5te --disable-armv6t2 --target-os=linux --enable-cross-compile --cross-prefix=$(TOOLCHAINPATH)/$(CC_PREFIX)- --prefix=$(CROSSCOMP_ROOT)/usr --extra-cflags='$(INC_DIR) $(CPU_ARCH)' --extra-ldflags='$(CPU_ARCH)' --disable-static --enable-shared --enable-pic --disable-altivec --disable-amd3dnow --disable-amd3dnowext --disable-mmx --disable-mmx2 --disable-sse --disable-ssse3 --enable-armv5te --disable-armv6 --disable-armv6t2 --disable-armvfp --disable-iwmmxt --disable-mmi --disable-neon --disable-vis --disable-yasm I got below errors when compiling some assembly files. ... AS libavcodec/arm/dsputil_arm.o libavcodec/arm/dsputil_arm.S: Assembler messages: libavcodec/arm/dsputil_arm.S:25: Error: unknown pseudo-op: `.eabi_attribute' libavcodec/arm/dsputil_arm.S:110: Error: bad instruction `push {r4-r11,lr}' libavcodec/arm/dsputil_arm.S:113: Error: bad instruction `ldm r1,{r4-r7}' libavcodec/arm/dsputil_arm.S:115: Error: bad instruction `stm r0,{r4-r7}' libavcodec/arm/dsputil_arm.S:120: Error: bad instruction `pop {r4-r11,pc}' libavcodec/arm/dsputil_arm.S:123: Error: bad instruction `ldm r1,{r4-r8}' libavcodec/arm/dsputil_arm.S:128: Error: bad instruction `stm r0,{r9-r12}' libavcodec/arm/dsputil_arm.S:131: Error: bad instruction `pop {r4-r11,pc}' libavcodec/arm/dsputil_arm.S:134: Error: bad instruction `ldm r1,{r4-r8}' libavcodec/arm/dsputil_arm.S:139: Error: bad instruction `stm r0,{r9-r12}' libavcodec/arm/dsputil_arm.S:142: Error: bad instruction `pop {r4-r11,pc}' It seems like that gcc can't recognize some arm instruction. Is it right? BR From tfoucu at gmail.com Tue Aug 16 08:15:15 2011 From: tfoucu at gmail.com (Thierry Foucu) Date: Mon, 15 Aug 2011 23:15:15 -0700 Subject: [FFmpeg-user] Quantizer Matrix In-Reply-To: <4E49ED90.3030208@acolyte-color.com> References: <4E49ED90.3030208@acolyte-color.com> Message-ID: On Mon, Aug 15, 2011 at 9:09 PM, J A Stephen Viggiano < jasv at acolyte-color.com> wrote: > Sorry if this is answered somewhere; I checked the man page, all the online > documentation I could, and this list back to the Big Scism of 2011.... > > How do I get ffmpeg to use the MPEG quantizer? When use mediainfo to > examine a video encoded using the xvid video coder (-vcodec libxvid) or the > asp default (-acodec mpeg4), it tells me (among many other things) that the > h.263 "matrix" was used; see below (sorry if this should go into pastebin > instead). > > Assuming the mediainfo tool is correct, is there any way to specify the > quantization method, such as to employ the MPEG matrices? > > Thank you! > > try adding -mpeg_quant in the command line ==============================**== > > ffmpeg -i ed_hd.ogv -vcodec mpeg4 -qscale 4 -acodec copy -t 10 sample.avi > ffmpeg version 0.8.1, Copyright (c) 2000-2011 the FFmpeg developers > built on Jul 28 2011 23:08:02 with gcc 4.5.0 20100604 [gcc-4_5-branch > revision 160292] > configuration: --shlibdir=/usr/lib --prefix=/usr --mandir=/usr/share/man > --libdir=/usr/lib --enable-shared --disable-static --enable-libmp3lame > --enable-libvorbis --enable-libtheora --enable-libspeex --enable-libxvid > --enable-postproc --enable-gpl --enable-x11grab > --extra-cflags='-fomit-frame-**pointer -fmessage-length=0 -O2 -Wall > -D_FORTIFY_SOURCE=2 -fstack-protector -funwind-tables > -fasynchronous-unwind-tables -g -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 > -I/usr/include/gsm' --enable-debug --disable-stripping --enable-libgsm > --enable-libschroedinger --enable-libdirac --enable-avfilter --enable-libvpx > --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb > --enable-libx264 --enable-libdc1394 --enable-pthreads --enable-librtmp > libavutil 51. 9. 1 / 51. 9. 1 > libavcodec 53. 7. 0 / 53. 7. 0 > libavformat 53. 4. 0 / 53. 4. 0 > libavdevice 53. 1. 1 / 53. 1. 1 > libavfilter 2. 23. 0 / 2. 23. 0 > libswscale 2. 0. 0 / 2. 0. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > [theora @ 0x807da60] 7 bits left in packet 82 > Input #0, ogg, from 'ed_hd.ogv': > Duration: 00:10:53.78, start: 0.000000, bitrate: 558 kb/s > Stream #0.0: Video: theora, yuv420p, 544x304, 24 fps, 24 tbr, 24 tbn, 24 > tbc > Stream #0.1: Audio: vorbis, 44100 Hz, stereo, s16 > Metadata: > TITLE : Elephants Dream > ARTIST : Bassam Kurdali > LICENSE : http://creativecommons.org/**licenses/by/3.0/us/ > DATE : 2006 > ORGANIZATION : Orange Open Movie Project Studio > LOCATION : http://www.archive.org/**details/ElephantsDream > [buffer @ 0x821c600] w:544 h:304 pixfmt:yuv420p tb:1/1000000 sar:0/1 > sws_param: > [theora @ 0x807da60] 7 bits left in packet 82 > Output #0, avi, to 'sample.avi': > Metadata: > ISFT : Lavf53.4.0 > Stream #0.0: Video: mpeg4, yuv420p, 544x304, q=2-31, 200 kb/s, 24 tbn, > 24 tbc > Stream #0.1: Audio: libvorbis, 44100 Hz, stereo > Metadata: > TITLE : Elephants Dream > ARTIST : Bassam Kurdali > LICENSE : http://creativecommons.org/**licenses/by/3.0/us/ > DATE : 2006 > ORGANIZATION : Orange Open Movie Project Studio > LOCATION : http://www.archive.org/**details/ElephantsDream > Stream mapping: > Stream #0.0 -> #0.0 > Stream #0.1 -> #0.1 > Press [q] to stop, [?] for help > frame= 240 fps=214 q=4.0 Lsize= 631kB time=00:00:10.00 bitrate= > 516.6kbits/s > video:552kB audio:39kB global headers:0kB muxing overhead 6.610288% > > > mediainfo --Inform='Video;%Format_**Settings_Matrix/String%' sample.avi > Default (H.263) > > ______________________________**_________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/**listinfo/ffmpeg-user > From sreemnpy at gmail.com Tue Aug 16 08:31:59 2011 From: sreemnpy at gmail.com (sreerag) Date: Mon, 15 Aug 2011 23:31:59 -0700 (PDT) Subject: [FFmpeg-user] Error in compilation of the newer version of ffmpeg-0.8.2 Message-ID: <1313476319390-3746443.post@n4.nabble.com> Hi I compiled the the latest version of FFMPEG (ffmpeg-0.8.2). In the configuration file i enabled the 'avisynth'(--enable-avisynth). But when i issued the 'make' command the output showed like the following libavformat/avisynth.c:124: undefined reference to 'ff_codec_tags'. libavformat/avisynth.c:124: undefined reference to 'ff_wav_c_get_id'. make: ***[libavformat/avformat-53.dill] Error 1 What is the solution for this probelm? if anyone knows Kindly help me... Thanks & Regards sreerag -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Error-in-compilation-of-the-newer-version-of-ffmpeg-0-8-2-tp3746443p3746443.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From blacktrash at gmx.net Tue Aug 16 12:56:24 2011 From: blacktrash at gmx.net (Christian Ebert) Date: Tue, 16 Aug 2011 12:56:24 +0200 Subject: [FFmpeg-user] libavfilter: setsar works consistently with libx264! Message-ID: <20110816105624.GQ653@krille.blacktrash.org> Hi, Just a heads up that as of ffmpeg N-31898-g95b5b52 (or a few commits before) libavfilter's setsar with scale and even crop seems to work consistently when transcoding e.g. from pal sources. Before, the behaviour differed haphazardly from other target codecs - i.e. you needed setdar too, or couldn't get a correct result at all. So, do not change it ;-) c -- theatre - books - texts - movies Black Trash Productions at home: http://www.blacktrash.org Black Trash Productions on Facebook: http://www.facebook.com/blacktrashproductions From luj125 at gmail.com Tue Aug 16 14:52:25 2011 From: luj125 at gmail.com (James Lu) Date: Tue, 16 Aug 2011 08:52:25 -0400 Subject: [FFmpeg-user] optimal slide encoding for low bandwidth at youtube In-Reply-To: References: Message-ID: On Mon, Aug 15, 2011 at 6:40 PM, Robert Rowntree wrote: > i want to combine 1 slide ( vid ) and a 3gpp ( audio ) in output that is > suitable for youtube upload. i have a CLI expression working for ffmpeg , > but its video bandwidth is too high considering that i am muxing just a > single jpeg. > > i have used the following , but think that i can do better on the bandwidth > of the VIDEO track... > > *./ffmpeg -y -loop_input -f image2 -shortest -r 1/2 -i 1459.JPG -i > 1459.3gpp > -s 640x480 -b 100k -bt 240k -vcodec msmpeg4 -acodec libmp3lame -ab 64k -ar > 22050 out.wmv* > > Once the output from the above is uploaded to youtube, youtube exposes 2 > RTSP URI's - a high and low bandwidth version of the video. > > the low bandwidth youtube version still has 39K of video bandwidth and that > is what i want to reduce.... its just a single slide in the video > > v=0 > o=GoogleStreamer 98636128 107718649 IN IP4 74.125.213.247 > s=Video > c=IN IP4 0.0.0.0 > b=AS:51 > t=0 0 > a=control:* > a=range:npt=0-63.800000 > m=video 0 RTP/AVP 98 > *b=AS:39 <-- VIDEO bandwidth* > a=rtpmap:98 H263-2000/90000 > a=control:trackID=0 > a=cliprect:0,0,144,176 > a=framesize:98 176-144 > a=fmtp:98 profile=0;level=10 > m=audio 0 RTP/AVP 99 > b=AS:12 > a=rtpmap:99 AMR/8000/1 > a=control:trackID=1 > a=fmtp:99 octet-align > * Connection #0 to host v8.cache8.c.youtube.com left intact > > for comparison purpose, the following URI on youtube uses just 8K for its > video bandwidth and i want to try to get down near that. > > http://www.youtube.com/watch?v=FX9ccqQuzO4 is a typical sound track > hosted on youtube having the SDP below from the rtsp feed: > > v=0 > o=GoogleStreamer 515160982 1824503727 IN IP4 74.125.213.247 > s=Video > c=IN IP4 0.0.0.0 > b=AS:20 > t=0 0 > a=control:* > a=range:npt=0-190.200000 > m=video 0 RTP/AVP 98 > *b=AS:8 <--- low video bandwidth * > a=rtpmap:98 H263-2000/90000 > a=control:trackID=0 > a=cliprect:0,0,144,176 > a=framesize:98 176-144 > a=fmtp:98 profile=0;level=10 > m=audio 0 RTP/AVP 99 > b=AS:12 > a=rtpmap:99 AMR/8000/1 > a=control:trackID=1 > a=fmtp:99 octet-align > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Hey Robert, A few suggestions: 1) what may be killing you are the intra frames. What size is your jpg? Also look into making your GOP size as large as possible maybe? this is done by using option -g 'gop_size' 2) is there a specific reason you are using wmv? In my usage I've found libx264 to compress the most efficiently, and i'm fairly certain that youtube accepts .mp4 upload. (-vcodec libx264 -acodec aac -strict experimental) 3) not sure how low frame rate can go, but try dropping it more? maybe 1/(song_length_in_seconds) would work? 4) your -b option is at 100k.... try dropping to your real target? Same with -bt None of these are tested, just ideas. Hope one of them helps. ~James From eric.hollis at gmail.com Tue Aug 16 15:10:55 2011 From: eric.hollis at gmail.com (Eric Hollis) Date: Tue, 16 Aug 2011 09:10:55 -0400 Subject: [FFmpeg-user] Newbie question: Can FFMPEG grab frames from webcam? Message-ID: Hello. Looking for some guidance. Thanks in advance for your time... I currently use ffmpeg to assemble still images into movies. I have developed a 16mm film scanner, based on a projector, that gathers frames, similar to this project, which was the inspiration to my machine. ( http://hackaday.com/2011/01/15/converting-8mm-film-to-digital/) I am using a webcam (Microsoft LifeCam) in front of the projector lens. As each frame advances, the projector "clicks" an embedded mouse, taking a high res jpg image, utilizing the (gag) Microsoft lifecam software that the camera came with. I then use irfanview to crop/flip/renumber the images, and ffmpeg to assemble them into the movie. The trouble is...the Microsoft Lifecam Software has a memory leak, and after about 5000 frames (about 2 hours), the software chokes. So, I'm rethinking the workflow. Can FFMPEG be used to grab frames from a webcam with a mouse click? Thanks for any hints you can provide. From luj125 at gmail.com Tue Aug 16 17:37:12 2011 From: luj125 at gmail.com (James Lu) Date: Tue, 16 Aug 2011 11:37:12 -0400 Subject: [FFmpeg-user] Newbie question: Can FFMPEG grab frames from webcam? In-Reply-To: References: Message-ID: On Tue, Aug 16, 2011 at 9:10 AM, Eric Hollis wrote: > Hello. Looking for some guidance. Thanks in advance for your time... > > I currently use ffmpeg to assemble still images into movies. I have > developed a 16mm film scanner, based on a projector, that gathers frames, > similar to this project, which was the inspiration to my machine. ( > http://hackaday.com/2011/01/15/converting-8mm-film-to-digital/) I am using > a > webcam (Microsoft LifeCam) in front of the projector lens. As each frame > advances, the projector "clicks" an embedded mouse, taking a high res jpg > image, utilizing the (gag) Microsoft lifecam software that the camera came > with. I then use irfanview to crop/flip/renumber the images, and ffmpeg to > assemble them into the movie. The trouble is...the Microsoft Lifecam > Software has a memory leak, and after about 5000 frames (about 2 hours), > the > software chokes. So, I'm rethinking the workflow. Can FFMPEG be used to > grab frames from a webcam with a mouse click? > > Thanks for any hints you can provide. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Hey Eric, Assuming video for windows drivers, some research turns up a few bits of commands that may work. I have no experience in this, but here's something I would try running: ffmpeg -f vfwcap -i 0 -vcodec mjpeg -vf 1 -an output.jpg 0 is the default device, captured from vfw driver. I think this should give you a jpg of the image currently seen by the webcam. -vf 1 will make sure ffmpeg will only process for 1 frame. As for "running on a click" you can set up a batch script that increments the file name in a loop, and have a pause after each image capture that is released into the loop again when you 'press any key to continue' Hope this helps. ~James From ashrub at yandex.ru Tue Aug 16 19:12:20 2011 From: ashrub at yandex.ru (Alexey Shrub) Date: Tue, 16 Aug 2011 10:12:20 -0700 (PDT) Subject: [FFmpeg-user] Hint tracks support for MP4 In-Reply-To: <20091014210433.GA29082@geppetto> References: <77938bc20910141154o43e78f48ve4482371533057de@mail.gmail.com> <20091014210433.GA29082@geppetto> Message-ID: <1313514740885-3747811.post@n4.nabble.com> Hi, >> Any idea if FFMPEG supports / will be supporting adding hint tracks to >> MP4 files for RTSP streaming? > Patches are welcome :). Any changes from 2009 ? -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Hint-tracks-support-for-MP4-tp942732p3747811.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From glau.stuff at ridiculousprods.com Tue Aug 16 19:39:23 2011 From: glau.stuff at ridiculousprods.com (Glau Stuff) Date: Tue, 16 Aug 2011 10:39:23 -0700 Subject: [FFmpeg-user] converting Quicktime files with multiple audio tracks. Message-ID: Hi, occasionally I get a file that has multiple audio sound tracks in it. (not to be confused with channels). It might be a quicktime file with two separate stereo tracks. (Track 1 L/R; Track 2 L/R) or a file with two mono aac tracks. Some Skype phone recorders in particular seem to generate files that are one distinct mono track for each end of the conversation instead of one stereo track split L/R Let's assume I have the following: Quicktime MOV file with * Track 1 - Mono * Track 2 - Mono. How do I export this to mp3 such that both tracks are represented on the MP3 file. Either Track 1 to Left Chanel, Track 2 to Right channel; or Track 1 & 2 merged into a mono MP3 file I suspect it has to do with the -map feature, but a concrete example would help a lot. Thanks. From luj125 at gmail.com Tue Aug 16 19:46:21 2011 From: luj125 at gmail.com (James Lu) Date: Tue, 16 Aug 2011 13:46:21 -0400 Subject: [FFmpeg-user] converting Quicktime files with multiple audio tracks. In-Reply-To: References: Message-ID: On Tue, Aug 16, 2011 at 1:39 PM, Glau Stuff wrote: > Hi, occasionally I get a file that has multiple audio sound tracks in it. > (not to be confused with channels). > > It might be a quicktime file with two separate stereo tracks. (Track 1 > L/R; Track 2 L/R) or a file with two mono aac tracks. > > Some Skype phone recorders in particular seem to generate files that are > one distinct mono track for each end of the conversation instead of one > stereo track split L/R > > Let's assume I have the following: > > Quicktime MOV file with > > * Track 1 - Mono > * Track 2 - Mono. > > > How do I export this to mp3 such that both tracks are represented on the > MP3 file. Either Track 1 to Left Chanel, Track 2 to Right channel; or > Track 1 & 2 merged into a mono MP3 file > > I suspect it has to do with the -map feature, but a concrete example would > help a lot. > > Thanks. > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Hey Glau, I don't believe ffmpeg has the capability to map to channels. AFAIK, -map only maps different streams. I expect that when libavfilter gets audio filters that this will be among its most basic features. Open programs I suggest to do what you are looking for: Sox: http://sox.sourceforge.net/ Similar to ffmpeg in command line complexity Audacity: http://audacity.sourceforge.net/ GUI audio manipulation tool Hope this helps, ~James From pavel.taran at gmail.com Tue Aug 16 16:49:11 2011 From: pavel.taran at gmail.com (taran2L) Date: Tue, 16 Aug 2011 07:49:11 -0700 (PDT) Subject: [FFmpeg-user] ffmpeg rtsp + audio from other device Message-ID: <1313506151851-3747446.post@n4.nabble.com> Hello! How I can get video from ip camera (RTSP) and audio from computer microphone It's not work! ffmpeg -an -i rtsp://link -f alsa -i hw:0,0 -f flv "rtmp://link" Only one microphone work! ffmpeg -f alsa -i hw:0,0 -f flv "rtmp://link" HELP!!1 -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-rtsp-audio-from-other-device-tp3747446p3747446.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From luj125 at gmail.com Tue Aug 16 20:17:22 2011 From: luj125 at gmail.com (James Lu) Date: Tue, 16 Aug 2011 14:17:22 -0400 Subject: [FFmpeg-user] ffmpeg rtsp + audio from other device In-Reply-To: <1313506151851-3747446.post@n4.nabble.com> References: <1313506151851-3747446.post@n4.nabble.com> Message-ID: Pavel, Have you tried ffmpeg -f rstp -i rstp://link -f alsa -i hw:0,0 -f flv "rtmp://link" Adding more options on the output such as the vcodec and acodec may help. Hope this helps. ~James On Tue, Aug 16, 2011 at 10:49 AM, taran2L wrote: > Hello! > > How I can get video from ip camera (RTSP) and audio from computer > microphone > > It's not work! > ffmpeg -an -i rtsp://link -f alsa -i hw:0,0 -f flv "rtmp://link" > > Only one microphone work! > ffmpeg -f alsa -i hw:0,0 -f flv "rtmp://link" > > HELP!!1 > > -- > View this message in context: > http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-rtsp-audio-from-other-device-tp3747446p3747446.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 luj125 at gmail.com Tue Aug 16 20:17:59 2011 From: luj125 at gmail.com (James Lu) Date: Tue, 16 Aug 2011 14:17:59 -0400 Subject: [FFmpeg-user] ffmpeg rtsp + audio from other device In-Reply-To: References: <1313506151851-3747446.post@n4.nabble.com> Message-ID: On Tue, Aug 16, 2011 at 2:17 PM, James Lu wrote: > Pavel, > > Have you tried ffmpeg -f rstp -i rstp://link -f alsa -i hw:0,0 -f flv > "rtmp://link" > Adding more options on the output such as the vcodec and acodec may help. > > Hope this helps. > > ~James > > > On Tue, Aug 16, 2011 at 10:49 AM, taran2L wrote: > >> Hello! >> >> How I can get video from ip camera (RTSP) and audio from computer >> microphone >> >> It's not work! >> ffmpeg -an -i rtsp://link -f alsa -i hw:0,0 -f flv "rtmp://link" >> >> Only one microphone work! >> ffmpeg -f alsa -i hw:0,0 -f flv "rtmp://link" >> >> HELP!!1 >> >> -- >> View this message in context: >> http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-rtsp-audio-from-other-device-tp3747446p3747446.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 >> > > Oops, sorry I meant *rtsp From glau.stuff at ridiculousprods.com Tue Aug 16 21:01:32 2011 From: glau.stuff at ridiculousprods.com (Glau Stuff) Date: Tue, 16 Aug 2011 12:01:32 -0700 Subject: [FFmpeg-user] converting Quicktime files with multiple audio tracks. In-Reply-To: Message-ID: On 8/16/11 10:46 AM, "James Lu" wrote: >On Tue, Aug 16, 2011 at 1:39 PM, Glau Stuff >wrote: > >> >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> > >Hey Glau, > >I don't believe ffmpeg has the capability to map to channels. AFAIK, -map >only maps different streams. I expect that when libavfilter gets audio >filters that this will be among its most basic features. > >Open programs I suggest to do what you are looking for: >Sox: http://sox.sourceforge.net/ >Similar to ffmpeg in command line complexity >Audacity: http://audacity.sourceforge.net/ >GUI audio manipulation tool > >Hope this helps, >~James SOX doesn't seem to handle MOV files as a source input. Error Message: sox FAIL formats: no handler for file extension `mov' We had been getting around this problem with the exception of the multi-track issue by using the following chain on the command line. (only command line options are usable as this is part of a server solution) * ffmpeg -i original.mov temp.wav (This converts our source file to generic WAV) * sox 'temp.wav' 'sox_temp.wav' highpass -2 150 (roll off some bass on all files) * ffmpeg -y -i 'sox_temp.wav' -vn -acodec libmpelame -ac 2 -ar 22050 -ab 80Kk 'final.mp3' (make the final MP3) Since ffmpeg recognizes and can open more formats than sox, specifically MOV and AAC files, we use that as the first step to create a generic WAV. >From there we manipulate the WAV in sox and finally create our final low bandwidth MP3. This works fine except with the multi-track issues. ffmpeg (using the commands I'm familiar with) will only convert the first "Track" it sees and ignores any secondary tracks. Here's the output from a sample file just using "ffmpeg -i sample.mov". I've omitted the non-relevant output. > ffmpeg -i sample.mov libpostproc 51. 2. 0 / 51. 2. 0 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x80ea850]multiple edit list entries, a/v desync might occur, patch welcome Last message repeated 1 times Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'sample.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt title-eng : Decator #1 2010-06-02 at 10.02 title : Decator #1 2010-06-02 at 10.02 Duration: 00:09:26.65, start: 0.000000, bitrate: 94 kb/s Stream #0.0(eng): Audio: aac, 44100 Hz, mono, s16, 31 kb/s Stream #0.1(eng): Audio: aac, 44100 Hz, mono, s16, 31 kb/s At least one output file must be specified > As you can see from the output, ffmpeg has found two "Tracks": Stream #0.0 and #0.1 When using our chain listed above, we end up with a MP3 that only contains the audio for Stream #0.0 Right now we're having to download this edge case files and convert them on the desktop and reupload them to the server. Not a super great solution... :( From luj125 at gmail.com Tue Aug 16 21:22:14 2011 From: luj125 at gmail.com (James Lu) Date: Tue, 16 Aug 2011 15:22:14 -0400 Subject: [FFmpeg-user] converting Quicktime files with multiple audio tracks. In-Reply-To: References: Message-ID: > > SOX doesn't seem to handle MOV files as a source input. Error Message: > > sox FAIL formats: no handler for file extension `mov' > > We had been getting around this problem with the exception of the > multi-track issue by using the following chain on the command line. (only > command line options are usable as this is part of a server solution) > > * ffmpeg -i original.mov temp.wav (This converts our > source file to > generic WAV) > * sox 'temp.wav' 'sox_temp.wav' highpass -2 150 (roll off some bass > on > all files) > * ffmpeg -y -i 'sox_temp.wav' -vn -acodec libmpelame -ac 2 -ar 22050 -ab > 80Kk 'final.mp3' (make the final MP3) > > Since ffmpeg recognizes and can open more formats than sox, specifically > MOV and AAC files, we use that as the first step to create a generic WAV. > From there we manipulate the WAV in sox and finally create our final low > bandwidth MP3. This works fine except with the multi-track issues. > > > Hm, I think you can extract 2 different streams as 2 different files. namely: * ffmpeg -i original.mov -map 0:1 -vn -ac 1 temp1.wav * ffmpeg -i original.mov -map 0:2 -vn -ac 1 temp2.wav * sox -m 'temp1.wav' 'temp2.wav' -c 2 'sox_temp.wav' highpass -2 15 * ffmpeg -y -i 'sox_temp.wav' -vn -acodec libmpelame -ac 2 -ar 22050 -ab 80Kk 'final.mp3' Never used sox before, so no guarantee this works at all haha, maybe I'll get lucky? Basically what I changed is that instead of keeping the 2 streams in one file, I mapped the streams and then mixed them in sox using -m (-M *should* produce something different but I don't know how different in this case). Hope this helps, ~James From jasv at acolyte-color.com Tue Aug 16 21:38:16 2011 From: jasv at acolyte-color.com (J A Stephen Viggiano) Date: Tue, 16 Aug 2011 15:38:16 -0400 Subject: [FFmpeg-user] Quantizer Matrix Message-ID: <4E4AC728.7000906@acolyte-color.com> Thierry Foucu wrote: > try adding -mpeg_quant in the command line That did the trick -- thank you! From eric.hollis at gmail.com Wed Aug 17 15:23:24 2011 From: eric.hollis at gmail.com (Eric Hollis) Date: Wed, 17 Aug 2011 09:23:24 -0400 Subject: [FFmpeg-user] How to change -vcodec tiff resolution? Message-ID: Hi. First, thanks for the help yesterday re how to grab images from video (vfwcap) for a 16mm film archiving project. This works for me: ffmpeg.exe -r 1 -t 1 -rtbufsize 10000000 -f vfwcap -s hd720 -i 0 -vframes 1 -an -vcodec tiff test%d.tif However, I have one last problem with this. The tif image that results is 72dpi, and I want it to be 96dpi or higher. Could someone point me in the right direction? From devastapalle at gmail.com Wed Aug 17 02:23:43 2011 From: devastapalle at gmail.com (Devasta Palle) Date: Wed, 17 Aug 2011 02:23:43 +0200 Subject: [FFmpeg-user] frustration Message-ID: Why that stupid "cheese" works and ffmpeg makes me want to kill myself? From djleehaha at googlemail.com Wed Aug 17 09:40:20 2011 From: djleehaha at googlemail.com (Lee Smith) Date: Wed, 17 Aug 2011 08:40:20 +0100 Subject: [FFmpeg-user] rtsp error - intermittent Message-ID: Hi everyone, First time posting on this list. The short version: #ffmpeg -i rtsp://www.address.com/Auto%20DJ%20Channel%202?tcp test.wav ... [rtsp @ 003AA3E0] Failed to fix invalid RTSP-MS/ASF min_pktsize The long version: I am setting up an application to switch between different incoming audio streams for a single output. As part of this project I am using ffmpeg to decode the incoming data to wav format. I have it connecting to the Auto DJ over rtsp however it sometimes fails with the below error (and the freeze) and sometimes appears to freeze with no error at all: [rtsp @ 003AA3E0] Failed to fix invalid RTSP-MS/ASF min_pktsize No data is written to the output. The Auto DJ server is basically just a Windows media services 9.6 Server with a play list. Each time you connect it picks a (random?) file from its playlist and streams it to the client (ffmpeg). The odd thing is we have 2 such servers with different play lists. One will fail 1 time in 10, the other more like 9 times in 10 (these are guesstimates). Connecting to the same server using winamp (over mms, although I belive this is the same thing?) works first time every time. I am thinking this must be some kind of incompatibility between ffmpeg and Windows media services 9.6 ? I get the exact same behaviour whether I try it on Windows 7 or SUSE 11.4 on a different PC (the Hex number is different every time though) Full transcript of a failed run below if it helps Can anyone shed any light? Many Thanks Lee C:\dev\dj-djl.com\radioswitcher2>ffmpeg -i rtsp://www.address.com/Auto%20DJ%20Channel1?tcp test.wav ffmpeg version N-31100-g9251942, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 30 2011 21:17:59 with gcc 4.5.3 configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable- runtime-cpudetect --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 libavutil 51. 11. 0 / 51. 11. 0 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 24. 0 / 2. 24. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [rtsp @ 00329FC0] Failed to fix invalid RTSP-MS/ASF min_pktsize <> -- *DJL - XWAX And Vinyl DJ http://www.dj-djl.com Wednesday 8-10PM CH1 http://www.housefreaks.co.uk From hoot893 at gmail.com Wed Aug 17 17:14:51 2011 From: hoot893 at gmail.com (Andrew Stevanus) Date: Wed, 17 Aug 2011 11:14:51 -0400 Subject: [FFmpeg-user] Can't combine 2 pulse input streams in screencast Message-ID: Hi. I'm trying to record a screencast with ffmpeg and record my pc audio, and my microphone. Recording only one stream works fine, but when I try to record with two pulse inputs, I get no audio in the resulting video (I changed one of the pulse inputs with pavucontrol to use the pc audio instead of the microphone). Here's the command that I use: ffmpeg -f x11grab -r 15 -s 1366x768 -i :0.0 -f alsa -ac 2 -i pulse -f alsa -ac 1 -i pulse -vcodec libx264 -acodec pcm_s16le -vpre lossless_ultrafast -threads 2 output.mp4 I should also say that whenever I use the input hw:0,0 or something similar, when I play back the resulting video, the audio seems to get gradually out of sync. Can anyone help? From h.reindl at thelounge.net Wed Aug 17 19:04:50 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Wed, 17 Aug 2011 19:04:50 +0200 Subject: [FFmpeg-user] frustration In-Reply-To: References: Message-ID: <4E4BF4B2.10508@thelounge.net> Am 17.08.2011 02:23, schrieb Devasta Palle: > Why that stupid "cheese" works and ffmpeg makes me want to kill myself? what will you tell us with this idiotic post? describe your problems or consider not to post at all! -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From Cecil at decebal.nl Wed Aug 17 19:28:05 2011 From: Cecil at decebal.nl (Cecil Westerhof) Date: Wed, 17 Aug 2011 19:28:05 +0200 Subject: [FFmpeg-user] Converting for YouTube Message-ID: <87r54kkkwq.fsf@Compaq.site> I want to convert my video's for YouTube. At the moment I am using: ffmpeg -i input.MOV -ar 22050 -acodec libmp3lame -ab 32k -r 25 -vcodec flv -s vga -qscale 2.5 output.flv This seems to work, but still gives quit big files. 35 MB for 1:48. Input file is 89 MB. So I was wondering: is there a better way? -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof From bouke at editb.nl Wed Aug 17 20:42:40 2011 From: bouke at editb.nl (bouke) Date: Wed, 17 Aug 2011 20:42:40 +0200 Subject: [FFmpeg-user] Converting for YouTube References: <87r54kkkwq.fsf@Compaq.site> Message-ID: <003601cc5d0d$75137830$4301a8c0@hpkantoor> ----- Original Message ----- From: "Cecil Westerhof" To: Sent: Wednesday, August 17, 2011 7:28 PM Subject: [FFmpeg-user] Converting for YouTube >I want to convert my video's for YouTube. At the moment I am using: > ffmpeg -i input.MOV -ar 22050 -acodec libmp3lame -ab 32k -r 25 -vcodec > flv -s vga -qscale 2.5 output.flv > > This seems to work, but still gives quit big files. 35 MB for 1:48. > Input file is 89 MB. > > So I was wondering: is there a better way? For youtube one normally only preprocess. De-interlace, crop and set correct aspect ratio. Youtube will always recompress, so introducing compression artefacts is a big no-no. IOW, upload the biggest file your bandwith allows. Bouke > -- > 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 rogerdpack2 at gmail.com Thu Aug 18 02:15:50 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 17 Aug 2011 18:15:50 -0600 Subject: [FFmpeg-user] Newbie question: Can FFMPEG grab frames from webcam? In-Reply-To: References: Message-ID: > taking a high res jpg > image, utilizing the (gag) Microsoft lifecam software that the camera came > with. I believe ffmpeg can take input from directshow capture devices $ ffmpeg -f dshow -i video="USB2.0_Camera":audio="Microphone (USB Audio Device)" output.mkv http://betterlogic.com/roger/2011/08/ffmpeg-directshow From hoot893 at gmail.com Thu Aug 18 03:59:15 2011 From: hoot893 at gmail.com (Andrew Stevanus) Date: Wed, 17 Aug 2011 21:59:15 -0400 Subject: [FFmpeg-user] Can't combine 2 pulse input streams in screencast Message-ID: <201108172159.15427.hoot893@gmail.com> Hi. I'm trying to record a screencast with ffmpeg and record my pc audio, and my microphone. Recording only one stream works fine, but when I try to record with two pulse inputs, I get no audio in the resulting video (I changed one of the pulse inputs with pavucontrol to use the pc audio instead of the microphone). Here's the command that I use: ffmpeg -f x11grab -r 15 -s 1366x768 -i :0.0 -f alsa -ac 2 -i pulse -f alsa -ac 1 -i pulse -vcodec libx264 -acodec pcm_s16le -vpre lossless_ultrafast -threads 2 output.mp4 I should also say that whenever I use the input hw:0,0 or something similar, when I play back the resulting video, the audio seems to get gradually out of sync. Can anyone help? From hardik.sharma22 at yahoo.com Thu Aug 18 04:45:42 2011 From: hardik.sharma22 at yahoo.com (Hardik Sharma) Date: Wed, 17 Aug 2011 19:45:42 -0700 (PDT) Subject: [FFmpeg-user] how to drop slices in ffmpeg/x264 Message-ID: <1313635542.38593.YahooMailNeo@web46211.mail.sp1.yahoo.com> Hi? I am doing some experiment to check how ffmpeg/x264 react to packet or slice drop in h264 video streams. I already experimented with several sets of parameters to compare ffmpeg with JM. Can anyone tell me or help me to find out that how I can drop slices in ffmpeg/x264? I really appreciate any kind of help. Thanks From anuj_k_shah at indiatimes.com Thu Aug 18 08:58:42 2011 From: anuj_k_shah at indiatimes.com (anuj shah) Date: Thu, 18 Aug 2011 12:28:42 +0530 (IST) Subject: [FFmpeg-user] how to convert HD AVI to FLV Message-ID: <1119464697.795311313650722430.JavaMail.root@tilmb17.indiatimes.com> Hi Sir, i am using ffmpeg ,i want to convert HD Avi vedio to FLV in minimum size like You tube ,kindly send me the commandline command to change.. Regards Anuj K Shah From anuj_k_shah at indiatimes.com Thu Aug 18 09:00:09 2011 From: anuj_k_shah at indiatimes.com (anuj shah) Date: Thu, 18 Aug 2011 12:30:09 +0530 (IST) Subject: [FFmpeg-user] how to convert HD AVI to FLV Message-ID: <1345285762.795551313650809184.JavaMail.root@tilmb17.indiatimes.com> Hi Sir, i am using ffmpeg ,i want to convert HD Avi vedio to FLV in minimum size like You tube ,kindly send me the commandline command to change.. Regards Anuj K Shah From sreemnpy at gmail.com Thu Aug 18 12:17:45 2011 From: sreemnpy at gmail.com (sreerag r) Date: Thu, 18 Aug 2011 15:47:45 +0530 Subject: [FFmpeg-user] How can we enable dshow format in ffmpeg when compiling on windows? In-Reply-To: References: <1313147406257-3738949.post@n4.nabble.com> Message-ID: Hi, I tried out VLC directshow capture mode. It is working fine. But using VLC's dirctshow mode we can't do RTMP streaming. I need to stream webcam video to a Flash media server using dshow. If have any idea please reply me.. Thanks in advance.. On Sat, Aug 13, 2011 at 2:06 AM, James Lu wrote: > On Fri, Aug 12, 2011 at 4:22 PM, Roger Pack wrote: > > > > How can we enable dshow format in ffmpeg? > > According to > > http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=3&t=27&start=10 > > (last post) it should "just work" though I haven't tried it myself, > > and apparently it isn't easy... > > > > Make sure you are using the most recent libav* libraries in compiling > It seems you also should try --enable-avisynth ? Not sure... ref: > http://ffmpeg.org/faq.html#SEC26 > > If you're looking for an alternative, VLC has a directshow capture mode > that > works moderately well. Seeing as how VLC is built on the same libraries as > ffmpeg, I'm assuming it does the same job as if you got the -f dshow to > work > on the command line tool. > > Also, vlc works as a command line tool itself, try using it's syntax found > here: http://wiki.videolan.org/VLC_command-line_help if you can't get > ffmpeg > working? > > Hope one of these solutions works out for you, > ~James > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- with regards Sreerag R Research Assistant Amrita E-learning Research Lab Amritapuri ?...Fate Determines Who Comes Into Our Lives, But Heart Determines Who Stays...? From rhswdev at gmail.com Thu Aug 18 12:58:18 2011 From: rhswdev at gmail.com (Karoly Horvath) Date: Thu, 18 Aug 2011 11:58:18 +0100 Subject: [FFmpeg-user] TS slice Message-ID: I transcode with ffmpeg to H264 wrapped in TS (transport stream). ffmpeg doesn't output any splice flag and splice countdown. Is there a way to fix this? Thx -- Karoly Horvath From rowntreerob at gmail.com Thu Aug 18 21:34:11 2011 From: rowntreerob at gmail.com (Robert Rowntree) Date: Thu, 18 Aug 2011 12:34:11 -0700 Subject: [FFmpeg-user] optimal slide encoding for low bandwidth at youtube In-Reply-To: References: Message-ID: thanks for how to instructions on libx264. i got good result on mp4 produced by ffmpeg with very low video bandwidth using : ./ffmpeg -y -loop_input -shortest -i Optimized-DSC01448.JPG -i rec_1448.3gpp -pass 1 -vframes 500 -vcodec libx264 -r 1/100 -vf 'scale=320:180' -b 2k -g 250 -b_strategy 0 -coder 1 -qmin 10 -qmax 55 -sc_threshold 40 -flags -loop -subq 1 -i_qfactor 0.71 -qcomp 0.6 -qdiff 4 -flags2 +dct8x8+wpred +bpyramid+mixed_refs -trellis 1 -partitions +parti8x8+parti4x4+partp8x8+partp4x4+partb8x8 -acodec libmp3lame -ac 1 -ar 22050 -ab 22k _my.mp4 i think that the issue is at youtube because i was able to use the instructions for libx264 to create an mp4 combining a single slide and a voice track in which , mplayer reports the bandwidth on the video track as 1.8 kbps. After uploading that mp4 to youtube and after the resultant transcoding and processing of the youtube distribution platform, the Video track bandwidth had risen from 1.8kbps to 24 kbps. details on youtube process ( mp4 stats on mplayer, SDP for the youtube RTSP feed ... ) see: http://goo.gl/W0Dg8 not alot that i can do about that .. thank you for all the help On Tue, Aug 16, 2011 at 5:52 AM, James Lu wrote: > On Mon, Aug 15, 2011 at 6:40 PM, Robert Rowntree >wrote: > > > i want to combine 1 slide ( vid ) and a 3gpp ( audio ) in output that is > > suitable for youtube upload. i have a CLI expression working for ffmpeg , > > but its video bandwidth is too high considering that i am muxing just a > > single jpeg. > > > > i have used the following , but think that i can do better on the > bandwidth > > of the VIDEO track... > > > > *./ffmpeg -y -loop_input -f image2 -shortest -r 1/2 -i 1459.JPG -i > > 1459.3gpp > > -s 640x480 -b 100k -bt 240k -vcodec msmpeg4 -acodec libmp3lame -ab 64k > -ar > > 22050 out.wmv* > > > > Once the output from the above is uploaded to youtube, youtube exposes 2 > > RTSP URI's - a high and low bandwidth version of the video. > > > > the low bandwidth youtube version still has 39K of video bandwidth and > that > > is what i want to reduce.... its just a single slide in the video > > > > v=0 > > o=GoogleStreamer 98636128 107718649 IN IP4 74.125.213.247 > > s=Video > > c=IN IP4 0.0.0.0 > > b=AS:51 > > t=0 0 > > a=control:* > > a=range:npt=0-63.800000 > > m=video 0 RTP/AVP 98 > > *b=AS:39 <-- VIDEO bandwidth* > > a=rtpmap:98 H263-2000/90000 > > a=control:trackID=0 > > a=cliprect:0,0,144,176 > > a=framesize:98 176-144 > > a=fmtp:98 profile=0;level=10 > > m=audio 0 RTP/AVP 99 > > b=AS:12 > > a=rtpmap:99 AMR/8000/1 > > a=control:trackID=1 > > a=fmtp:99 octet-align > > * Connection #0 to host v8.cache8.c.youtube.com left intact > > > > for comparison purpose, the following URI on youtube uses just 8K for its > > video bandwidth and i want to try to get down near that. > > > > http://www.youtube.com/watch?v=FX9ccqQuzO4 is a typical sound track > > hosted on youtube having the SDP below from the rtsp feed: > > > > v=0 > > o=GoogleStreamer 515160982 1824503727 IN IP4 74.125.213.247 > > s=Video > > c=IN IP4 0.0.0.0 > > b=AS:20 > > t=0 0 > > a=control:* > > a=range:npt=0-190.200000 > > m=video 0 RTP/AVP 98 > > *b=AS:8 <--- low video bandwidth * > > a=rtpmap:98 H263-2000/90000 > > a=control:trackID=0 > > a=cliprect:0,0,144,176 > > a=framesize:98 176-144 > > a=fmtp:98 profile=0;level=10 > > m=audio 0 RTP/AVP 99 > > b=AS:12 > > a=rtpmap:99 AMR/8000/1 > > a=control:trackID=1 > > a=fmtp:99 octet-align > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > Hey Robert, > > A few suggestions: > 1) what may be killing you are the intra frames. What size is your jpg? > Also > look into making your GOP size as large as possible maybe? this is done by > using option -g 'gop_size' > 2) is there a specific reason you are using wmv? In my usage I've found > libx264 to compress the most efficiently, and i'm fairly certain that > youtube accepts .mp4 upload. (-vcodec libx264 -acodec aac -strict > experimental) > 3) not sure how low frame rate can go, but try dropping it more? maybe > 1/(song_length_in_seconds) would work? > 4) your -b option is at 100k.... try dropping to your real target? Same > with > -bt > > None of these are tested, just ideas. Hope one of them helps. > > ~James > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From seandarcy2 at gmail.com Thu Aug 18 23:25:41 2011 From: seandarcy2 at gmail.com (Sean Darcy) Date: Thu, 18 Aug 2011 17:25:41 -0400 Subject: [FFmpeg-user] Where are the presets for x264? Message-ID: Using the latest git, I only find these presets: /usr/share/ffmpeg/libvpx-1080p.ffpreset /usr/share/ffmpeg/libvpx-1080p50_60.ffpreset /usr/share/ffmpeg/libvpx-360p.ffpreset /usr/share/ffmpeg/libvpx-720p.ffpreset /usr/share/ffmpeg/libvpx-720p50_60.ffpreset /usr/share/ffmpeg/libx264-ipod320.ffpreset /usr/share/ffmpeg/libx264-ipod640.ffpreset /usr/share/ffmpeg/libx264-lossless_fast.ffpreset /usr/share/ffmpeg/libx264-lossless_max.ffpreset /usr/share/ffmpeg/libx264-lossless_medium.ffpreset /usr/share/ffmpeg/libx264-lossless_slow.ffpreset /usr/share/ffmpeg/libx264-lossless_slower.ffpreset /usr/share/ffmpeg/libx264-lossless_ultrafast.ffpreset i'm doing 2-pass encoding with x264. There used to be a firstpass preset. Is it just gone? What should I use instead for the first pass? sean From jeisom at gmail.com Thu Aug 18 23:30:36 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Thu, 18 Aug 2011 16:30:36 -0500 Subject: [FFmpeg-user] Where are the presets for x264? In-Reply-To: References: Message-ID: On Thu, Aug 18, 2011 at 4:25 PM, Sean Darcy wrote: > Using the latest git, I only find these presets: > > > /usr/share/ffmpeg/libvpx-1080p.ffpreset > /usr/share/ffmpeg/libvpx-1080p50_60.ffpreset > /usr/share/ffmpeg/libvpx-360p.ffpreset > /usr/share/ffmpeg/libvpx-720p.ffpreset > /usr/share/ffmpeg/libvpx-720p50_60.ffpreset > /usr/share/ffmpeg/libx264-ipod320.ffpreset > /usr/share/ffmpeg/libx264-ipod640.ffpreset > /usr/share/ffmpeg/libx264-lossless_fast.ffpreset > /usr/share/ffmpeg/libx264-lossless_max.ffpreset > /usr/share/ffmpeg/libx264-lossless_medium.ffpreset > /usr/share/ffmpeg/libx264-lossless_slow.ffpreset > /usr/share/ffmpeg/libx264-lossless_slower.ffpreset > /usr/share/ffmpeg/libx264-lossless_ultrafast.ffpreset > > i'm doing 2-pass encoding with x264. There used to be a firstpass preset. Is > it just gone? What should I use instead for the first pass? > Removed in April git has now added 3 new options for libx264 encoding: -preset -tune -profile These options are directly mapped to internal tune and presets within libx264. The old presets files have been removed, please use the new options. Such as: -profile baseline See "x264 --help" to see available presets, tunes, and profiles > sean > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rogerdpack2 at gmail.com Fri Aug 19 00:03:27 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 18 Aug 2011 16:03:27 -0600 Subject: [FFmpeg-user] How can we enable dshow format in ffmpeg when compiling on windows? In-Reply-To: References: <1313147406257-3738949.post@n4.nabble.com> Message-ID: > I tried out VLC directshow capture mode. It is working fine. But using VLC's > dirctshow mode we can't do RTMP streaming. Maybe ask the VLC people? Did you look at this? does it work? According to http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=3&t=27&start=10 (last post) it should "just work" though I haven't tried it myself, and apparently it isn't easy... From rogerdpack2 at gmail.com Fri Aug 19 00:04:59 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Thu, 18 Aug 2011 16:04:59 -0600 Subject: [FFmpeg-user] Can't combine 2 pulse input streams in screencast In-Reply-To: References: Message-ID: > I should also say that whenever I use the input hw:0,0 or something similar, > when I play back the resulting video, the audio seems to get gradually out > of sync. Can anyone help? Can you disable all cacheing? That might help... From hoot893 at gmail.com Fri Aug 19 00:47:02 2011 From: hoot893 at gmail.com (Andrew Stevanus) Date: Thu, 18 Aug 2011 18:47:02 -0400 Subject: [FFmpeg-user] Can't combine 2 pulse input streams in screencast In-Reply-To: References: Message-ID: <201108181847.02360.hoot893@gmail.com> On August 18, 2011 6:04:59 PM Roger Pack wrote: > > I should also say that whenever I use the input hw:0,0 or something > > similar, when I play back the resulting video, the audio seems to get > > gradually out of sync. Can anyone help? > > Can you disable all cacheing? That might help... > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user How exactly would I do that? From andycivil at gmail.com Fri Aug 19 03:21:18 2011 From: andycivil at gmail.com (Andy Civil) Date: Thu, 18 Aug 2011 21:21:18 -0400 Subject: [FFmpeg-user] optimal slide encoding for low bandwidth at youtube In-Reply-To: References: Message-ID: <4E4DBA8E.3020304@gmail.com> On 2011-08-15 6:40 PM, Robert Rowntree wrote: > i want to combine 1 slide ( vid ) and a 3gpp ( audio ) in output that is > suitable for youtube upload. i have a CLI expression working for ffmpeg , > but its video bandwidth is too high considering that i am muxing just a > single jpeg. > You may be interested in some experiments I did when I was trying to make a video slideshow with FFmpeg. I was offended by the obvious wasted data when using other methods. I understand this is somewhat different from your issue, because you only have one slide. I posted my experiences here: Basically, I use MPEG-2 as an intermediate format; it's not efficient, but it has the outstanding advantage that I can concatenate the video slides in a command window, and then convert and add a soundtrack afterwards. I also recently discovered that I can convert to H.264 and it stays pretty efficient, although the file is larger. I find that if I upload the video to youtube, it is accepted, but when I download it again it's a bit bigger. I think the issue is that they have standards for rewinding resolution, so their GOP is lower than I actually had in my file; therefore, they probably duplicated the i-frame, perhaps once per second or something. -- Andy From seandarcy2 at gmail.com Fri Aug 19 04:00:45 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Thu, 18 Aug 2011 22:00:45 -0400 Subject: [FFmpeg-user] Where are the presets for x264? In-Reply-To: References: Message-ID: On 08/18/2011 05:30 PM, Jonathan Isom wrote: > On Thu, Aug 18, 2011 at 4:25 PM, Sean Darcy wrote: >> Using the latest git, I only find these presets: >> >> >> /usr/share/ffmpeg/libvpx-1080p.ffpreset >> /usr/share/ffmpeg/libvpx-1080p50_60.ffpreset >> /usr/share/ffmpeg/libvpx-360p.ffpreset >> /usr/share/ffmpeg/libvpx-720p.ffpreset >> /usr/share/ffmpeg/libvpx-720p50_60.ffpreset >> /usr/share/ffmpeg/libx264-ipod320.ffpreset >> /usr/share/ffmpeg/libx264-ipod640.ffpreset >> /usr/share/ffmpeg/libx264-lossless_fast.ffpreset >> /usr/share/ffmpeg/libx264-lossless_max.ffpreset >> /usr/share/ffmpeg/libx264-lossless_medium.ffpreset >> /usr/share/ffmpeg/libx264-lossless_slow.ffpreset >> /usr/share/ffmpeg/libx264-lossless_slower.ffpreset >> /usr/share/ffmpeg/libx264-lossless_ultrafast.ffpreset >> >> i'm doing 2-pass encoding with x264. There used to be a firstpass preset. Is >> it just gone? What should I use instead for the first pass? >> > > Removed in April git has now added 3 new options for libx264 encoding: > -preset > -tune > -profile > > These options are directly mapped to internal tune and presets within > libx264. The old presets files have been removed, please use the new > options. > > Such as: > -profile baseline > > See "x264 --help" to see available presets, tunes, and profiles > > > > >> sean >> >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user >> OK. So what should I use for pass1? sean From pixelpartner at me.com Wed Aug 17 19:57:00 2011 From: pixelpartner at me.com (Thomas Kumlehn) Date: Wed, 17 Aug 2011 19:57:00 +0200 Subject: [FFmpeg-user] How to change -vcodec tiff resolution? In-Reply-To: References: Message-ID: <10C2861B-B0F0-485E-AE60-DC741E688833@me.com> The dpi header value just suggests a default presentation size and should not bother you at all. 72dpi means : spread 720px to 10 inches screen width. 96dpi just results in 7.5". But both images are 720 pixel wide. Use Irfanview or other tools to alter just this header detail later. Best wishes Thomas Kumlehn PIXEL PARTNER (R) gesendet von meinem iPad 3-D http://www.pixelpartner.de/ipad3d Am 17.08.2011 um 15:23 schrieb Eric Hollis : > Hi. First, thanks for the help yesterday re how to grab images from video > (vfwcap) for a 16mm film archiving project. This works for me: > > ffmpeg.exe -r 1 -t 1 -rtbufsize 10000000 -f vfwcap -s hd720 -i 0 -vframes 1 > -an -vcodec tiff test%d.tif > > However, I have one last problem with this. The tif image that results is > 72dpi, and I want it to be 96dpi or higher. Could someone point me in the > right direction? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From tonybaqain at gmail.com Wed Aug 17 23:10:47 2011 From: tonybaqain at gmail.com (Antoine Baqain) Date: Thu, 18 Aug 2011 00:10:47 +0300 Subject: [FFmpeg-user] iTunes aspect ratio Message-ID: Hello, Hope my words find you all well. I need your help in a matter that I can't find a solution through searching on fixes for it. I'm on an iMac 3.4GHz i3, 4GB of RAM, and x86_64 arch, and I'm trying to convert an AVI (DVDRIP as they call it), apparently a torrent movie :), and I'm using the following command (includes all info about source and output) : ffmpeg -y -i "Baby Einstein Baby Da Vinci 9 mos+.avi" -f mp4 -vcodec mpeg4 -b 40000 -acodec libfaac -ab 192k *-aspect 16:9* output_file.mov FFmpeg version 0.6.2, Copyright (c) 2000-2010 the FFmpeg developers built on May 20 2011 15:00:07 with gcc 4.2.1 (Apple Inc. build 5646) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-avfilter-lavf --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libfaac --enable-libfaad --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --enable-nonfree --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.2 --arch=x86_64 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 1.11. 0 / 1.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 29.98 (65535/2186) -> 29.97 (30000/1001) Input #0, avi, from 'Baby Einstein Baby Da Vinci 9 mos+.avi': Metadata: ISFT : VirtualDubMod 1.5.4.1 (build 2178/release) Duration: 00:32:36.88, start: 0.000000, bitrate: 2998 kb/s Stream #0.0: Video: mpeg4, yuv420p, 720x544 [PAR 1:1 DAR 45:34], 29.97 tbr, 29.97 tbn, 29.98 tbc Stream #0.1: Audio: mp3, 48000 Hz, 2 channels, s16, 128 kb/s Output #0, mp4, to 'output_file.mov': Metadata: encoder : Lavf52.64.2 Stream #0.0: Video: mpeg4, yuv420p, 720x544 [PAR 227:169 DAR 10215:5746], q=2-31, 40 kb/s, 30k tbn, 29.97 tbc Stream #0.1: Audio: libfaac, 48000 Hz, 2 channels, s16, 192 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 3467 fps=263 q=24.8 Lsize= 8835kB time=121.89 bitrate= 593.8kbits/s video:6289kB audio:2471kB global headers:0kB muxing overhead 0.848983% I bolded the aspect that I needed since from source it is 4:3, and I have a LED screen with 16:9 with apple TV 2. My problem is that, opening the output file through VLC, it opens with 16:9 aspect ratio, but in iTunes (which reflects Apple TV 2) it opens with 4:3 aspect ratio, please tell me if I'm missing any other option for ffmpeg and recognized by iTunes / Apple TV 2 that would convert the video with 16:9 aspect ratio. Regards, Tony. P.S: please don't tell me to use any application such as hand break or something, they all use ffmpeg and they output the same as my ffmpeg CMD, but much slower (UI complicated also when you get into advanced options). From maheshraja8 at gmail.com Thu Aug 18 07:54:40 2011 From: maheshraja8 at gmail.com (maheshraja) Date: Wed, 17 Aug 2011 22:54:40 -0700 (PDT) Subject: [FFmpeg-user] Howto compile ffplay.c in seperate directory with my own svn-version of ffmpeg In-Reply-To: <49F87FAC.7080805@netmaster.dk> References: <49F87FAC.7080805@netmaster.dk> Message-ID: <1313646880517-3751830.post@n4.nabble.com> hi , try this we can surely compile ffplay independently... gcc -o sample ffplay.c cmdutils.c -I../ffmpeg -L../ffmpeg/libavcodec -L../ffmpeg/libavformt -L../ffmpeg/libswscale -L../ffmpeg/libavutil -L../ffmpeg/libavfilter -L../ffmpeg/libavdevice -L../ffmpeg/libpostproc -lpostproc -lavdevice -lavfilter -lavformat -lswscale -lavutil -lmp3lame -lx264 `sdl-config --cflags --libs` -lz -lbz2 -lavcodec thanks mahesh raja -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Howto-compile-ffplay-c-in-seperate-directory-with-my-own-svn-version-of-ffmpeg-tp940857p3751830.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From maheshraja8 at gmail.com Thu Aug 18 11:01:00 2011 From: maheshraja8 at gmail.com (maheshraja) Date: Thu, 18 Aug 2011 02:01:00 -0700 (PDT) Subject: [FFmpeg-user] starting video from desired location not through command line Message-ID: <1313658060036-3752097.post@n4.nabble.com> hi, All can any one help me for playing video from required positon not through cmd-line option. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/starting-video-from-desired-location-not-through-command-line-tp3752097p3752097.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From lou at lrcd.com Fri Aug 19 07:51:20 2011 From: lou at lrcd.com (Lou) Date: Thu, 18 Aug 2011 21:51:20 -0800 Subject: [FFmpeg-user] Where are the presets for x264? In-Reply-To: References: Message-ID: <20110818215120.09b16905@lrcd.com> On Thu, 18 Aug 2011 22:00:45 -0400 sean darcy wrote: > On 08/18/2011 05:30 PM, Jonathan Isom wrote: > > On Thu, Aug 18, 2011 at 4:25 PM, Sean Darcy > > wrote: > >> Using the latest git, I only find these presets: > >> > >> > >> /usr/share/ffmpeg/libvpx-1080p.ffpreset > >> /usr/share/ffmpeg/libvpx-1080p50_60.ffpreset > >> /usr/share/ffmpeg/libvpx-360p.ffpreset > >> /usr/share/ffmpeg/libvpx-720p.ffpreset > >> /usr/share/ffmpeg/libvpx-720p50_60.ffpreset > >> /usr/share/ffmpeg/libx264-ipod320.ffpreset > >> /usr/share/ffmpeg/libx264-ipod640.ffpreset > >> /usr/share/ffmpeg/libx264-lossless_fast.ffpreset > >> /usr/share/ffmpeg/libx264-lossless_max.ffpreset > >> /usr/share/ffmpeg/libx264-lossless_medium.ffpreset > >> /usr/share/ffmpeg/libx264-lossless_slow.ffpreset > >> /usr/share/ffmpeg/libx264-lossless_slower.ffpreset > >> /usr/share/ffmpeg/libx264-lossless_ultrafast.ffpreset > >> > >> i'm doing 2-pass encoding with x264. There used to be a firstpass > >> preset. Is it just gone? What should I use instead for the first > >> pass? > >> > > > > Removed in April git has now added 3 new options for libx264 > > encoding: -preset > > -tune > > -profile > > > > These options are directly mapped to internal tune and presets > > within libx264. The old presets files have been removed, please use > > the new options. > > > > Such as: > > -profile baseline > > > > See "x264 --help" to see available presets, tunes, and profiles > > > > > > > > > >> sean > >> > OK. So what should I use for pass1? > > sean Now you just use the same preset for both passes for most cases. Example: ffmpeg -i input -pass 1 -vcodec libx264 -preset fast -b 512k \ -threads 0 -f mp4 -an -y /dev/null && \ ffmpeg -i input -pass 2 -vcodec libx264 -preset fast -b 512k \ -threads 0 -acodec libfaac -ab 128k -ac 2 output.mp4 From maheshraja8 at gmail.com Fri Aug 19 08:51:31 2011 From: maheshraja8 at gmail.com (maheshraja) Date: Thu, 18 Aug 2011 23:51:31 -0700 (PDT) Subject: [FFmpeg-user] audio and video not synchronizing Message-ID: <1313736691701-3754469.post@n4.nabble.com> hi i used stream_seek to seek video.. video is seeking but audio not!.. how to sync those two .. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/audio-and-video-not-synchronizing-tp3754469p3754469.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From maheshraja8 at gmail.com Fri Aug 19 08:51:30 2011 From: maheshraja8 at gmail.com (maheshraja) Date: Thu, 18 Aug 2011 23:51:30 -0700 (PDT) Subject: [FFmpeg-user] audio and video not synchronizing Message-ID: <1313736690035-3754468.post@n4.nabble.com> hi i used stream_seek to seek video.. video is seeking but audio not!.. how to sync those two .. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/audio-and-video-not-synchronizing-tp3754468p3754468.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From sreemnpy at gmail.com Fri Aug 19 08:56:15 2011 From: sreemnpy at gmail.com (sreerag) Date: Thu, 18 Aug 2011 23:56:15 -0700 (PDT) Subject: [FFmpeg-user] How can we enable dshow format in ffmpeg when compiling on windows? In-Reply-To: References: <1313147406257-3738949.post@n4.nabble.com> Message-ID: <1313736975522-3754481.post@n4.nabble.com> Thanks for the Reply, ya....I tried out that. In my compiled version of ffmpeg there is no dshow format. So that example didn't worked. Thanks and Regards Sreerag -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/How-can-we-enable-dshow-format-in-ffmpeg-when-compiling-on-windows-tp3738949p3754481.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From joolzg at btinternet.com Fri Aug 19 11:51:51 2011 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Fri, 19 Aug 2011 10:51:51 +0100 (BST) Subject: [FFmpeg-user] cant encode dvd vob file Message-ID: <1313747511.54867.YahooMailNeo@web86406.mail.ird.yahoo.com> ok im trying to encode a .vob file, but it stops at the same point every time, now i get 4 errors so dont know which one is the killer bit Input stream #0.1 frame changed from rate:48000 fmt:s16 ch:6 to rate:48000 fmt:s16 ch:2 [mpeg2video @ 0x269d540] mpeg_decode_postinit() failure Error while decoding stream #0.0 [mpeg2video @ 0x269d540] 00 motion_type at 5 24=00:15:45.16 bitrate= 503.0kbits/s dup=17 drop=0??? [mpeg2video @ 0x269d540] ac-tex damaged at 28 22 [mpeg2video @ 0x269d540] Warning MVs not available [mpeg2video @ 0x269d540] concealing 585 DC, 585 AC, 585 MV errors [ac3 @ 0x26af440] incomplete frame frame=23649 fps=352 q=31.0 Lsize=?? 58094kB time=00:15:45.64 bitrate= 503.3kbits/s dup=17 drop=0??? video:44566kB audio:7388kB global headers:0kB muxing overhead 11.818985% Now the file contains 5 audio streams and ive tried all of them with no luck whats the plan to get it to the right person to look at the error. joolz From kai at vizrt.com Fri Aug 19 13:57:50 2011 From: kai at vizrt.com (=?ISO-8859-1?Q?Kai-Mikael_J=E4=E4-Aro?=) Date: Fri, 19 Aug 2011 13:57:50 +0200 Subject: [FFmpeg-user] Several issues encoding DV* video Message-ID: <4E4E4FBE.6020303@vizrt.com> I'm trying to generate video in DV25, DVCPro25, DVCPro50 (and ideally also DVCPro 100, but I know this is not possible). I'd like to have them as both raw DV files and wrapped in MXF, but have problems getting the correct encoders. In short I do not know how to get DV25 with audio in DV and MXF wrappers and DVCPro25 in MXF wrapper. These are the results I get when testing different combinations: DV25 ffmpeg -i input.y4m -s 720x576 -r 25 -pix_fmt yuv420p output.dv generates a DV file. No problem. But then I try to add the audio tracks: ffmpeg -i input.y4m -i left.flac -i right.flac -s 720x576 -r 25 -pix_fmt yuv420p output.dv generates what mediainfo reports as a DVCPRO file, though the chroma is 4:2:0. The same thing happens if I wrap as MXF: ffmpeg -i input.y4m -s 720x576 -r 25 -pix_fmt yuv420p -target dv output.mxf generates DV and ffmpeg -i input.y4m -i left.flac -i right.flac -s 720x576 -r 25 -pix_fmt yuv420p -target dv output.mxf generates DVCPRO. DVCPro25 ffmpeg -i input.y4m -s 720x576 -r 25 -pix_fmt yuv411p output.dv generates a DVCPro25 file. No problem. ffmpeg -i input.y4m -s 720x576 -r 25 -pix_fmt yuv411p output.mxf generates MPEG-2 in MXF wrapper, whereas ffmpeg -i input.y4m -s 720x576 -r 25 -pix_fmt yuv411p -target dv output.mxf generates DV with 4:2:0 subsampling. Not what I wanted. Now, trying to add the audio tracks: ffmpeg -i input.y4m -i left.flac -i right.flac -s 720x576 -r 25 -pix_fmt yuv411p output.dv gives the error message "[dv @ 0x28e3960] Can't initialize DV format! Make sure that you supply exactly two streams: video: 25fps or 29.97fps, audio: 2ch/48kHz/PCM (50Mbps allows an optional second audio stream)" How come ffmpeg managed to take the two audio files earlier and combine them in one stereo stream in the output? I need to map the audio channels; this cannot be done in ffmpeg, as far as I have been able to figure out, but ffmbc lets me do this: ffmbc -i input.y4m -i left.flac -i right.flac -s 720x576 -r 25 -pix_fmt yuv411p output.dv -map_audio_channel 1:0:0:0:1:0 -map_audio_channel 2:0:0:0:1:1 This works and gives me DVCPro25 output. But, now attempting to wrap in MXF: ffmbc -i input.y4m -i left.flac -i right.flac -s 720x576 -r 25 -pix_fmt yuv411p -target dv output.mxf -map_audio_channel 1:0:0:0:1:0 -map_audio_channel 2:0:0:0:1:1 generates something marked as DVCPRO, but with 4:2:0 subsampling. DVCPro50 ffmpeg -i input.y4m -i left.flac -i right.flac -s 720x576 -r 25 -pix_fmt yuv422p output.dv generates the same error as before. ffmpeg -i input.y4m -i left.flac -i right.flac -s 720x576 -r 25 -pix_fmt yuv422p -target dv50 output.dv generates DVCPro50 output, with two stereo streams in the output file. ffmpeg -i input.y4m -i left.flac -i right.flac -s 720x576 -r 25 -pix_fmt yuv422p -target dv50 output.mxf also generates the expected output. From 103730258b at gmail.com Fri Aug 19 19:05:46 2011 From: 103730258b at gmail.com (maujhsn) Date: Fri, 19 Aug 2011 10:05:46 -0700 Subject: [FFmpeg-user] Newbie question: Can FFMPEG grab frames from webcam? In-Reply-To: References: Message-ID: Hi Eric; I really enjoyed reading your post! there are so many things that ffmpeg can do that many of us who use it, take for granted. I have used this command for turning single frames to movies: ffmpeg -f image2 -i img%03d.jpeg /tmp/a.mpg However the concept that you have written about & demonsrated is very interesting! Thanks, MJ From eric.hollis at gmail.com Fri Aug 19 19:10:09 2011 From: eric.hollis at gmail.com (Eric Hollis) Date: Fri, 19 Aug 2011 13:10:09 -0400 Subject: [FFmpeg-user] Newbie question: Can FFMPEG grab frames from webcam? In-Reply-To: References: Message-ID: Thanks...I'm going to blog the whole project. I've gotten lots of email like yours. surprising the level of intrest. I'll post a link to the users group when I'm done. Thanks for the tip, and the encouragment Eric On Fri, Aug 19, 2011 at 1:05 PM, maujhsn <103730258b at gmail.com> wrote: > Hi Eric; > > I really enjoyed reading your post! there are so many things that > ffmpeg can do that many of us who use it, take for granted. > > I have used this command for turning single frames to movies: > > ffmpeg -f image2 -i img%03d.jpeg /tmp/a.mpg > > However the concept that you have written about & demonsrated is very > interesting! > > Thanks, > > MJ > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From lou at lrcd.com Fri Aug 19 19:40:57 2011 From: lou at lrcd.com (Lou) Date: Fri, 19 Aug 2011 09:40:57 -0800 Subject: [FFmpeg-user] cant encode dvd vob file In-Reply-To: <1313747511.54867.YahooMailNeo@web86406.mail.ird.yahoo.com> References: <1313747511.54867.YahooMailNeo@web86406.mail.ird.yahoo.com> Message-ID: <20110819094057.3baef2ea@lrcd.com> On Fri, 19 Aug 2011 10:51:51 +0100 (BST) JULIAN GARDNER wrote: > ok im trying to encode a .vob file, but it stops at the same point > every time, now i get 4 errors so dont know which one is the killer > bit > > Input stream #0.1 frame changed from rate:48000 fmt:s16 ch:6 to > rate:48000 fmt:s16 ch:2 [mpeg2video @ 0x269d540] > mpeg_decode_postinit() failure Error while decoding stream #0.0 > [mpeg2video @ 0x269d540] 00 motion_type at 5 24=00:15:45.16 bitrate= > 503.0kbits/s dup=17 drop=0 [mpeg2video @ 0x269d540] ac-tex damaged at > 28 22 [mpeg2video @ 0x269d540] Warning MVs not available > [mpeg2video @ 0x269d540] concealing 585 DC, 585 AC, 585 MV errors > [ac3 @ 0x26af440] incomplete frame > frame=23649 fps=352 q=31.0 Lsize=?? 58094kB time=00:15:45.64 bitrate= > 503.3kbits/s dup=17 drop=0 video:44566kB audio:7388kB global > headers:0kB muxing overhead 11.818985% > > > Now the file contains 5 audio streams and ive tried all of them with > no luck > > whats the plan to get it to the right person to look at the error. > > joolz I would guess that this is a damaged input file. What is your ffmpeg command? Are you using a recent ffmpeg from Git or some ancient version? From eric.hollis at gmail.com Sat Aug 20 02:01:30 2011 From: eric.hollis at gmail.com (Eric Hollis) Date: Fri, 19 Aug 2011 20:01:30 -0400 Subject: [FFmpeg-user] Is there a way to speed up this ffmpeg command execution time? Message-ID: Hello. Looking for some additional guidance. Thanks in advance for looking... The command line below, when run on my Win 7 Pro 32 bit, Core2 Duo E8400 @3.0Ghz, 3G RAM machine, takes 3.86 seconds to write a pic to the disk. I have written a script that calls this command line each time a frame advances on my film scanner. I was hoping for an execute time of 1 to 2 seconds, since I am digitizing hundreds of thousands (probably millions) of frames of film. Any suggestions on how this could be made to run faster? ffmpeg.exe -r 1 -rtbufsize 100000000 -f vfwcap -s hd720 -i 0 -vframes 1 -vf crop=943:686:180:3,hflip test.tif At this point, I'm willing to entertain switching platforms to Linux, and dedicating a machine to this process. Ideas welcome. Thanks again for the help I've received from this group over the past couple of days! Eric (Short description of project) 8/17/11 I currently use ffmpeg to assemble still images into movies. I have developed a 16mm film scanner, based on a projector, that gathers frames, similar to this project, which was the inspiration to my machine. ( http://hackaday.com/2011/01/15/converting-8mm-film-to-digital/) I am using a webcam (Microsoft LifeCam) in front of the projector lens. As each frame advances, the projector "clicks" an embedded mouse, taking a high res jpg image, utilizing the (gag) Microsoft lifecam software that the camera came with. I then use irfanview to crop/flip/renumber the images, and ffmpeg to assemble them into the movie. The trouble is...the Microsoft Lifecam Software has a memory leak, and after about 5000 frames (about 2 hours), the software chokes. So, I'm rethinking the workflow. 8/19/11 I have now gotten the command line above to grab the frame and save it as a much-preferred uncompressed tif, replacing the Microsoft LifeCam software and the Irfanview crop/flip stage. Now all I need to do is finish the script that will invoke this command each time the projector sends a "left click". The projector can be slowed to ~ 2.5 seconds between frames, but ideally, it should run in the 1 second per frame range. From rogerdpack2 at gmail.com Sat Aug 20 02:42:53 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Fri, 19 Aug 2011 18:42:53 -0600 Subject: [FFmpeg-user] Is there a way to speed up this ffmpeg command execution time? In-Reply-To: References: Message-ID: > At this point, I'm willing to entertain switching platforms to Linux, and > dedicating a machine to this process. Ideas welcome. SSD? Linux? From rogerdpack2 at gmail.com Sat Aug 20 02:43:40 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Fri, 19 Aug 2011 18:43:40 -0600 Subject: [FFmpeg-user] cant encode dvd vob file In-Reply-To: <1313747511.54867.YahooMailNeo@web86406.mail.ird.yahoo.com> References: <1313747511.54867.YahooMailNeo@web86406.mail.ird.yahoo.com> Message-ID: > ok im trying to encode a .vob file, but it stops at the same point every time, now i get 4 errors so dont know which one is the killer bit What command line? From djleehaha at googlemail.com Sat Aug 20 14:11:54 2011 From: djleehaha at googlemail.com (Lee Smith) Date: Sat, 20 Aug 2011 13:11:54 +0100 Subject: [FFmpeg-user] Fwd: rtsp error - intermittent In-Reply-To: References: Message-ID: Hi everyone, First time posting on this list. The short version: #ffmpeg -i rtsp://www.address.com/Auto%20DJ%20Channel%202?tcp test.wav ... [rtsp @ 003AA3E0] Failed to fix invalid RTSP-MS/ASF min_pktsize The long version: I am setting up an application to switch between different incoming audio streams for a single output. As part of this project I am using ffmpeg to decode the incoming data to wav format. I have it connecting to the Auto DJ over rtsp however it sometimes fails with the below error (and the freeze) and sometimes appears to freeze with no error at all: [rtsp @ 003AA3E0] Failed to fix invalid RTSP-MS/ASF min_pktsize No data is written to the output. The Auto DJ server is basically just a Windows media services 9.6 Server with a play list. Each time you connect it picks a (random?) file from its playlist and streams it to the client (ffmpeg). The odd thing is we have 2 such servers with different play lists. One will fail 1 time in 10, the other more like 9 times in 10 (these are guesstimates). Connecting to the same server using winamp (over mms, although I belive this is the same thing?) works first time every time. I am thinking this must be some kind of incompatibility between ffmpeg and Windows media services 9.6 ? I get the exact same behaviour whether I try it on Windows 7 or SUSE 11.4 on a different PC (the Hex number is different every time though) Full transcript of a failed run below if it helps Can anyone shed any light? Many Thanks Lee C:\dev\dj-djl.com\radioswitcher2>ffmpeg -i rtsp://www.address.com/Auto%20DJ%20Channel1?tcp test.wav ffmpeg version N-31100-g9251942, Copyright (c) 2000-2011 the FFmpeg developers ?built on Jun 30 2011 21:17:59 with gcc 4.5.3 ?configuration: --enable-gpl --enable-version3 --enable-memalign-hack --enable- runtime-cpudetect --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 ?libavutil ? ?51. 11. 0 / 51. 11. 0 ?libavcodec ? 53. ?7. 0 / 53. ?7. 0 ?libavformat ?53. ?4. 0 / 53. ?4. 0 ?libavdevice ?53. ?2. 0 / 53. ?2. 0 ?libavfilter ? 2. 24. 0 / ?2. 24. 0 ?libswscale ? ?2. ?0. 0 / ?2. ?0. 0 ?libpostproc ?51. ?2. 0 / 51. ?2. 0 [rtsp @ 00329FC0] Failed to fix invalid RTSP-MS/ASF min_pktsize <> -- *DJL - XWAX And Vinyl DJ http://www.dj-djl.com Wednesday 8-10PM CH1 http://www.housefreaks.co.uk From Cecil at decebal.nl Sat Aug 20 16:41:37 2011 From: Cecil at decebal.nl (Cecil Westerhof) Date: Sat, 20 Aug 2011 16:41:37 +0200 Subject: [FFmpeg-user] Pasting a picture into a video Message-ID: <87wre887ry.fsf@Compaq.site> I have a video in which I would like to paste a photo, but without loosing the audio. So for example from second 23 to second 27 the video is replaced by my photo, but the original audio is still played. How should I do this? -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof From Cecil at decebal.nl Sat Aug 20 19:43:27 2011 From: Cecil at decebal.nl (Cecil Westerhof) Date: Sat, 20 Aug 2011 19:43:27 +0200 Subject: [FFmpeg-user] Reduce audio noise Message-ID: <87pqk07zcw.fsf@Compaq.site> My camera is not the best there is and I can not connect a microphone to it. So the audio is recorded with the internal microphone. It is not bad, but it gives some audio noise. How do I filter this noise away? -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof From stefano.sabatini-lala at poste.it Sat Aug 20 22:03:22 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sat, 20 Aug 2011 22:03:22 +0200 Subject: [FFmpeg-user] Pasting a picture into a video In-Reply-To: <87wre887ry.fsf@Compaq.site> References: <87wre887ry.fsf@Compaq.site> Message-ID: <20110820200322.GA13947@geppetto> On date Saturday 2011-08-20 16:41:37 +0200, Cecil Westerhof encoded: > I have a video in which I would like to paste a photo, but without > loosing the audio. So for example from second 23 to second 27 the > video is replaced by my photo, but the original audio is still played. > How should I do this? Currently you can overlay a video on top of another, in a static position. I suppose overlay could be extended by making x/y depending on time, but currently not yet possible. Patches/feature requests are welcome. From stefano.sabatini-lala at poste.it Sat Aug 20 22:04:25 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sat, 20 Aug 2011 22:04:25 +0200 Subject: [FFmpeg-user] Reduce audio noise In-Reply-To: <87pqk07zcw.fsf@Compaq.site> References: <87pqk07zcw.fsf@Compaq.site> Message-ID: <20110820200425.GB13947@geppetto> On date Saturday 2011-08-20 19:43:27 +0200, Cecil Westerhof encoded: > My camera is not the best there is and I can not connect a microphone > to it. So the audio is recorded with the internal microphone. It is > not bad, but it gives some audio noise. How do I filter this noise > away? Audio filtering is a work in progress, and will hopefully land soon, right now not yet possible. From Cecil at decebal.nl Sat Aug 20 23:49:59 2011 From: Cecil at decebal.nl (Cecil Westerhof) Date: Sat, 20 Aug 2011 23:49:59 +0200 Subject: [FFmpeg-user] Reduce audio noise In-Reply-To: <20110820200425.GB13947@geppetto> (Stefano Sabatini's message of "Sat, 20 Aug 2011 22:04:25 +0200") References: <87pqk07zcw.fsf@Compaq.site> <20110820200425.GB13947@geppetto> Message-ID: <87liun92ig.fsf@Compaq.site> Op zaterdag 20 aug 2011 22:04 CEST schreef Stefano Sabatini: > On date Saturday 2011-08-20 19:43:27 +0200, Cecil Westerhof encoded: >> My camera is not the best there is and I can not connect a microphone >> to it. So the audio is recorded with the internal microphone. It is >> not bad, but it gives some audio noise. How do I filter this noise >> away? > > Audio filtering is a work in progress, and will hopefully land soon, > right now not yet possible. Okay, thanks. -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof From Cecil at decebal.nl Sat Aug 20 23:54:53 2011 From: Cecil at decebal.nl (Cecil Westerhof) Date: Sat, 20 Aug 2011 23:54:53 +0200 Subject: [FFmpeg-user] Converting for YouTube In-Reply-To: <003601cc5d0d$75137830$4301a8c0@hpkantoor> (bouke@editb.nl's message of "Wed, 17 Aug 2011 20:42:40 +0200") References: <87r54kkkwq.fsf@Compaq.site> <003601cc5d0d$75137830$4301a8c0@hpkantoor> Message-ID: <87hb5b92aa.fsf@Compaq.site> Op woensdag 17 aug 2011 20:42 CEST schreef bouke at editb.nl: >> I want to convert my video's for YouTube. At the moment I am using: >> ffmpeg -i input.MOV -ar 22050 -acodec libmp3lame -ab 32k -r 25 -vcodec >> flv -s vga -qscale 2.5 output.flv >> >> This seems to work, but still gives quit big files. 35 MB for 1:48. >> Input file is 89 MB. >> >> So I was wondering: is there a better way? > > For youtube one normally only preprocess. > De-interlace, crop and set correct aspect ratio. > Youtube will always recompress, so introducing compression artefacts is a > big no-no. > IOW, upload the biggest file your bandwith allows. I understood that when you preprocess you do not have the risk that YouTube converts wrongly. Today I made a 3? minute video. This was 169 MB big and took almost an hour to upload. The converted (24 MB) took 8 minutes. So I think I'll keep converting. It looked okay, but if someone has better parameters? -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof From h.reindl at thelounge.net Sat Aug 20 23:59:09 2011 From: h.reindl at thelounge.net (Reindl Harald) Date: Sat, 20 Aug 2011 23:59:09 +0200 Subject: [FFmpeg-user] Converting for YouTube In-Reply-To: <87hb5b92aa.fsf@Compaq.site> References: <87r54kkkwq.fsf@Compaq.site> <003601cc5d0d$75137830$4301a8c0@hpkantoor> <87hb5b92aa.fsf@Compaq.site> Message-ID: <4E502E2D.8030500@thelounge.net> Am 20.08.2011 23:54, schrieb Cecil Westerhof: > Op woensdag 17 aug 2011 20:42 CEST schreef bouke at editb.nl: > >>> I want to convert my video's for YouTube. At the moment I am using: >>> ffmpeg -i input.MOV -ar 22050 -acodec libmp3lame -ab 32k -r 25 -vcodec >>> flv -s vga -qscale 2.5 output.flv >>> >>> This seems to work, but still gives quit big files. 35 MB for 1:48. >>> Input file is 89 MB. >>> >>> So I was wondering: is there a better way? >> >> For youtube one normally only preprocess. >> De-interlace, crop and set correct aspect ratio. >> Youtube will always recompress, so introducing compression artefacts is a >> big no-no. >> IOW, upload the biggest file your bandwith allows. > > I understood that when you preprocess you do not have the risk that > YouTube converts wrongly. > > Today I made a 3? minute video. This was 169 MB big and took almost an > hour to upload. The converted (24 MB) took 8 minutes. So I think I'll > keep converting. It looked okay, but if someone has better parameters where are you living that 169 MB upload takes an hour? :-) 10 MBit up are priced with < 100,- per month abd valueable if needed on the other hand - who interests that hour ONCE for upload while quality is degraded for thousands of users if you crap the input file before the upload? -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature URL: From Cecil at decebal.nl Sun Aug 21 00:18:15 2011 From: Cecil at decebal.nl (Cecil Westerhof) Date: Sun, 21 Aug 2011 00:18:15 +0200 Subject: [FFmpeg-user] Converting for YouTube In-Reply-To: <4E502E2D.8030500@thelounge.net> (Reindl Harald's message of "Sat, 20 Aug 2011 23:59:09 +0200") References: <87r54kkkwq.fsf@Compaq.site> <003601cc5d0d$75137830$4301a8c0@hpkantoor> <87hb5b92aa.fsf@Compaq.site> <4E502E2D.8030500@thelounge.net> Message-ID: <87d3fz917c.fsf@Compaq.site> Op zaterdag 20 aug 2011 23:59 CEST schreef Reindl Harald: > Am 20.08.2011 23:54, schrieb Cecil Westerhof: >> Op woensdag 17 aug 2011 20:42 CEST schreef bouke at editb.nl: >> >>>> I want to convert my video's for YouTube. At the moment I am using: >>>> ffmpeg -i input.MOV -ar 22050 -acodec libmp3lame -ab 32k -r 25 -vcodec >>>> flv -s vga -qscale 2.5 output.flv >>>> >>>> This seems to work, but still gives quit big files. 35 MB for 1:48. >>>> Input file is 89 MB. >>>> >>>> So I was wondering: is there a better way? >>> >>> For youtube one normally only preprocess. >>> De-interlace, crop and set correct aspect ratio. >>> Youtube will always recompress, so introducing compression artefacts is a >>> big no-no. >>> IOW, upload the biggest file your bandwith allows. >> >> I understood that when you preprocess you do not have the risk that >> YouTube converts wrongly. >> >> Today I made a 3? minute video. This was 169 MB big and took almost an >> hour to upload. The converted (24 MB) took 8 minutes. So I think I'll >> keep converting. It looked okay, but if someone has better parameters > > where are you living that 169 MB upload takes an hour? :-) > 10 MBit up are priced with < 100,- per month abd valueable if needed The Netherlands, but with ADSL your download is fast, but your upload less. > on the other hand - who interests that hour ONCE for upload > while quality is degraded for thousands of users if you crap the > input file before the upload? I do not see a difference between the full upload and the converted one. -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof From bouke at editb.nl Sun Aug 21 01:11:59 2011 From: bouke at editb.nl (bouke) Date: Sun, 21 Aug 2011 01:11:59 +0200 Subject: [FFmpeg-user] Converting for YouTube References: <87r54kkkwq.fsf@Compaq.site><003601cc5d0d$75137830$4301a8c0@hpkantoor> <87hb5b92aa.fsf@Compaq.site><4E502E2D.8030500@thelounge.net> <87d3fz917c.fsf@Compaq.site> Message-ID: <009e01cc5f8e$9394f7b0$4301a8c0@hpkantoor> ----- Original Message ----- From: "Cecil Westerhof" >> where are you living that 169 MB upload takes an hour? :-) >> 10 MBit up are priced with < 100,- per month abd valueable if needed > > The Netherlands, but with ADSL your download is fast, but your upload > less. > > >> on the other hand - who interests that hour ONCE for upload >> while quality is degraded for thousands of users if you crap the >> input file before the upload? > > I do not see a difference between the full upload and the converted one. But perhaps others do... (it's an art to spot these things...) Nice trick to test, rip both, apply a difference matte on the two clips. If you see something else than pure black, there sure is a difference, and those are the spots you need to carefully check. Bouke > 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 Cecil at decebal.nl Sun Aug 21 12:51:11 2011 From: Cecil at decebal.nl (Cecil Westerhof) Date: Sun, 21 Aug 2011 12:51:11 +0200 Subject: [FFmpeg-user] Converting for YouTube In-Reply-To: <009e01cc5f8e$9394f7b0$4301a8c0@hpkantoor> (bouke@editb.nl's message of "Sun, 21 Aug 2011 01:11:59 +0200") References: <87r54kkkwq.fsf@Compaq.site> <003601cc5d0d$75137830$4301a8c0@hpkantoor> <87hb5b92aa.fsf@Compaq.site> <4E502E2D.8030500@thelounge.net> <87d3fz917c.fsf@Compaq.site> <009e01cc5f8e$9394f7b0$4301a8c0@hpkantoor> Message-ID: <878vqn82cg.fsf@Compaq.site> Op zondag 21 aug 2011 01:11 CEST schreef bouke at editb.nl: >> I do not see a difference between the full upload and the converted one. > > But perhaps others do... > (it's an art to spot these things...) You have a point there. > Nice trick to test, rip both, apply a difference matte on the two clips. > If you see something else than pure black, there sure is a difference, and > those are the spots you need to carefully check. I have been googling, but I can not find how to do that (the difference matte, ripping is not a problem). How to do that? -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof From rhkramer at gmail.com Sun Aug 21 13:01:32 2011 From: rhkramer at gmail.com (Randy Kramer) Date: Sun, 21 Aug 2011 07:01:32 -0400 Subject: [FFmpeg-user] Converting for YouTube In-Reply-To: <009e01cc5f8e$9394f7b0$4301a8c0@hpkantoor> References: <87r54kkkwq.fsf@Compaq.site> <87d3fz917c.fsf@Compaq.site> <009e01cc5f8e$9394f7b0$4301a8c0@hpkantoor> Message-ID: <201108210701.32309.rhkramer@gmail.com> On Saturday 20 August 2011 07:11:59 pm bouke wrote: > Nice trick to test, rip both, apply a difference matte on the two clips. > If you see something else than pure black, there sure is a difference, and > those are the spots you need to carefully check. I'm not the OP, but I'm interested but not familiar with the terminology. Can you give a few more clues as to what you mean by: * rip (what do you rip, what command could you use) * apply a difference matte (what command could you use) Thanks! Randy Kramer From bouke at editb.nl Sun Aug 21 13:08:53 2011 From: bouke at editb.nl (bouke) Date: Sun, 21 Aug 2011 13:08:53 +0200 Subject: [FFmpeg-user] Converting for YouTube References: <87r54kkkwq.fsf@Compaq.site> <87d3fz917c.fsf@Compaq.site><009e01cc5f8e$9394f7b0$4301a8c0@hpkantoor> <201108210701.32309.rhkramer@gmail.com> Message-ID: <007501cc5ff2$b6e99ef0$4301a8c0@hpkantoor> ----- Original Message ----- From: "Randy Kramer" To: "FFmpeg user questions and RTFMs" Sent: Sunday, August 21, 2011 1:01 PM Subject: Re: [FFmpeg-user] Converting for YouTube > On Saturday 20 August 2011 07:11:59 pm bouke wrote: >> Nice trick to test, rip both, apply a difference matte on the two clips. >> If you see something else than pure black, there sure is a difference, >> and >> those are the spots you need to carefully check. > > I'm not the OP, but I'm interested but not familiar with the terminology. > Can > you give a few more clues as to what you mean by: > * rip (what do you rip, what command could you use) > * apply a difference matte (what command could you use) Several tools around to get the video from youtube in it's orignal form. Do a google search on 'rip youtube' a Matte is a key signal. Normally black and white. A Difference matte is comparing two streams by subtracting each pixel's value. If the pixels are the same, the result is of course pure black. So it generates a stream where differences between the two streams is shown in a black and white image, used for special effects, or, in this case, quality control. Probably there is an Avisynth filter, but most NLE's / special fx packages have something like that. (after effects for sure). So it depends on the tools you have.... Bouke VideoToolShed van Oldenbarneveltstraat 33 6512 AS NIJMEGEN The Netherlands +31 24 3553311 www.videotoolshed.com For large files: http://dropbox.yousendit.com/BoukeVahl998172 > Thanks! > > Randy Kramer > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From damian.ml at frey.co.nz Sun Aug 21 21:24:58 2011 From: damian.ml at frey.co.nz (Damian Stewart) Date: Sun, 21 Aug 2011 21:24:58 +0200 Subject: [FFmpeg-user] AVCHD from a Canon HF G10: 'non monotonically increasing dts' + dts mess Message-ID: Hi all, I'm trying to convert a number of AVCHD 1080i 50fps videos from a Canon HF G10 to mp4/mov container format, but ffmpeg is failing with a 'non monotonically increasing dts' error message at the start of each one. I'm on Mac OSX 10.6, I've tried both ffmpeg 0.8.0 (compiled by Homebrew) and 0.8.2 (compiled by ./configure && make). Here's the commandline. The file is here: http://frey.co.nz/share/00167.MTS (2.2MB) $ ffmpeg -i 00167.MTS -vcodec copy -acodec copy out/00167.mp4 Output fails with: Input #0, mpegts, from '00167.MTS': Duration: 00:00:01.44, start: 0.483578, bitrate: 12967 kb/s Program 1 Stream #0.0[0x1011]: Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc Stream #0.1[0x1100]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s ... [mp4 @ 0x101071200] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 1 >= 1 If I run ffprobe, the timestamps seem to be scrambled: $ ffprobe -show_packets 00167.MTS | grep dts ... dts=47119 dts_time=0.523544 dts=49999 dts_time=0.555544 dts=52879 dts_time=0.587544 dts=55759 dts_time=0.619544 dts=43519 dts_time=0.483544 dts=45319 dts_time=0.503544 dts=47119 dts_time=0.523544 dts=48919 dts_time=0.543544 dts=50719 dts_time=0.563544 dts=58639 dts_time=0.651544 dts=52519 dts_time=0.583544 ... dts=75919 dts_time=0.843544 dts=75919 dts_time=0.843544 dts=78799 dts_time=0.875544 dts=77719 dts_time=0.863544 dts=79519 dts_time=0.883544 dts=81679 dts_time=0.907544 dts=81319 dts_time=0.903544 dts=83119 dts_time=0.9 What's going on here? Is my camera really putting out such scrambled timestamps? Is there anything I can do about it? Any help appreciated. Cheers, Damian -- damian stewart . @damian0815 . damian at frey.co.nz frey . contemporary art . http://www.frey.co.nz From akshar_tank at yahoo.com Sun Aug 21 21:50:40 2011 From: akshar_tank at yahoo.com (tank pranav) Date: Sun, 21 Aug 2011 12:50:40 -0700 (PDT) Subject: [FFmpeg-user] A doubt in generation of frames out of a ts file. Message-ID: <1313956240.16097.YahooMailNeo@web122504.mail.ne1.yahoo.com> I am using ffmpeg. I am generating images out of a ts video file. But my observation says that ffmpeg gives some gray color images initially and after some gray images , actual video images are getting generated. How can I give command to ffmpeg that it directly generate actual video images. I will be very much thankful to you if you kindly answer for my above queries. Thanks. From joolzg at btinternet.com Mon Aug 22 12:26:09 2011 From: joolzg at btinternet.com (JULIAN GARDNER) Date: Mon, 22 Aug 2011 11:26:09 +0100 (BST) Subject: [FFmpeg-user] HLS Segmenter Message-ID: <1314008769.35553.YahooMailNeo@web86401.mail.ird.yahoo.com> Output stream is H264, AAC using x264 and faac Im playing with a segmenter and cutting the streams on an AV_PKT_FLAG_KEY, but what im seeing is a cut where the 1st frame does not start with sps/pps, How do i find the correct key frame to split the stream?. Is there a flag returned i can use. joolz From rhkramer at gmail.com Mon Aug 22 14:53:49 2011 From: rhkramer at gmail.com (Randy Kramer) Date: Mon, 22 Aug 2011 08:53:49 -0400 Subject: [FFmpeg-user] Converting for YouTube In-Reply-To: <007501cc5ff2$b6e99ef0$4301a8c0@hpkantoor> References: <87r54kkkwq.fsf@Compaq.site> <201108210701.32309.rhkramer@gmail.com> <007501cc5ff2$b6e99ef0$4301a8c0@hpkantoor> Message-ID: <201108220853.49422.rhkramer@gmail.com> Bouke, Thanks! Randy Kramer On Sunday 21 August 2011 07:08:53 am bouke wrote: > Several tools around to get the video from youtube in it's orignal form. > Do a google search on 'rip youtube' > a Matte is a key signal. Normally black and white. > A Difference matte is comparing two streams by subtracting each pixel's > value. > If the pixels are the same, the result is of course pure black. > So it generates a stream where differences between the two streams is shown > in a black and white image, > used for special effects, or, in this case, quality control. > > Probably there is an Avisynth filter, but most NLE's / special fx packages > have something like that. > (after effects for sure). > So it depends on the tools you have.... From tundra5280 at gmail.com Mon Aug 22 17:09:43 2011 From: tundra5280 at gmail.com (tundra5280) Date: Mon, 22 Aug 2011 10:09:43 -0500 Subject: [FFmpeg-user] ac3 invalid bit rate error Message-ID: Have been converting dts audio to ac3 successfully with ffmpeg for almost a year. Built a new ffmpeg executable today (ffmpeg version N-32060-g124deea) on Mac Pro running 10.7.1 and now get the following error when attempting to convert: [ac3 @ 0x7fbe9b02f400] invalid bit rate Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height I am also seeing: [mpegts @ 0x7fbe9b008800] Continuity Check Failed The conversion process does not seem to be hanging up on the continuity check but is hanging up on the ac3 error but maybe they are related. Configure: ./configure --prefix=${TARGET} --cc=clang --enable-nonfree --enable-gpl --enable-version3 --enable-libx264 --enable-pthreads --enable-libfaac --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libgsm --enable-libopenjpeg --enable-libxvid --enable-libschroedinger --enable-libdirac --enable-libxavs --enable-librtmp --enable-libfreetype --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-swscale --enable-avfilter --enable-filters --enable-postproc --target-os=darwin --arch=x86_64 --enable-runtime-cpudetect My conversion command line has not changed. The audio portion of command line: -acodec ac3 -ab 448k -ar 48000 448k is a valid bit rate!!! Get the same error wether the audio source is dts or ac3. Any work arounds for this error? Thanks in advance for help. From jeisom at gmail.com Mon Aug 22 17:46:01 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Mon, 22 Aug 2011 10:46:01 -0500 Subject: [FFmpeg-user] x264 and profile flag no longer working Message-ID: Using the -profile flag not longer works. Any changes to the command that I'm unaware or is this a bug? command : ffmpeg -i /tmp/tmp.avi -y -vcodec libx264 -profile baseline -preset medium -threads 2 -b 900000 -acodec libfaac -ac 2 -ab 128000 -f mp4 -s 848x480 /tmp/tmp.mp4 Error: [NULL @ 0xf46020] [Eval @ 0x7fffa7c6c2f0] Undefined constant or missing '(' in 'baseline' [NULL @ 0xf46020] Unable to parse option value "baseline" [NULL @ 0xf46020] Error setting option profile to value baseline. Full Output: ffmpeg version N-32033-gff96098, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 21 2011 08:45:16 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-encoder=mp2 --enable-encoder=ac3 --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. 13. 0 / 51. 13. 0 libavcodec 53. 11. 0 / 53. 11. 0 libavformat 53. 9. 0 / 53. 9. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 34. 1 / 2. 34. 1 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mpeg2video @ 0xc5f280] allocate dummy last picture for field based first keyframe [mpeg2video @ 0xc5f280] ac-tex damaged at 0 16 Input #0, avi, from '/tmp.test.avi': Metadata: encoder : MEncoder 33951-4.5.2 Duration: 00:25:32.83, start: 0.000000, bitrate: 2755 kb/s Stream #0.0: Video: mpeg2video (Main), yuv420p, 720x480 [SAR 32:27 DAR 16:9], 8500 kb/s, 29.97 fps, 29.97 tbr, 29.97 tbn, 59.94 tbc Stream #0.1: Audio: ac3, 48000 Hz, stereo, s16, 224 kb/s [buffer @ 0xd52b70] w:720 h:480 pixfmt:yuv420p tb:1/1000000 sar:32/27 sws_param: [scale @ 0xc5f0b0] w:720 h:480 fmt:yuv420p -> w:854 h:480 fmt:yuv420p flags:0x4 [libx264 @ 0xc5cdd0] using SAR=32/27 [libx264 @ 0xc5cdd0] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 0xc5cdd0] profile Constrained Baseline, level 3.1 [libx264 @ 0xc5cdd0] 264 - core 116 r2057 0ba8a9c - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 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=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=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=900 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 [NULL @ 0xc9f020] [Eval @ 0x7ffffb7df610] Undefined constant or missing '(' in 'baseline' [NULL @ 0xc9f020] Unable to parse option value "baseline" [NULL @ 0xc9f020] Error setting option profile to value baseline. Output #0, mp4, to '/tmp/tmp.mp4': Stream #0.0: Video: libx264, yuv420p, 854x480 [SAR 32:27 DAR 854:405], q=2-31, 900 kb/s, 90k tbn, 29.97 tbc Stream #0.1: Audio: libfaac, 48000 Hz, 2 channels, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height From jeisom at gmail.com Mon Aug 22 17:50:08 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Mon, 22 Aug 2011 10:50:08 -0500 Subject: [FFmpeg-user] ac3 invalid bit rate error In-Reply-To: References: Message-ID: On Mon, Aug 22, 2011 at 10:09 AM, tundra5280 wrote: > Have been converting dts audio to ac3 successfully with ffmpeg for almost a year. ?Built a new ffmpeg executable today (ffmpeg version N-32060-g124deea) on Mac Pro running 10.7.1 and now get the following error when attempting to convert: > > [ac3 @ 0x7fbe9b02f400] invalid bit rate > > Error while opening encoder for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height > > I am also seeing: > [mpegts @ 0x7fbe9b008800] Continuity Check Failed > > The conversion process does not seem to be hanging up on the continuity check but is hanging up on the ac3 error but maybe they are related. > > Configure: > ./configure --prefix=${TARGET} --cc=clang --enable-nonfree --enable-gpl --enable-version3 --enable-libx264 --enable-pthreads --enable-libfaac --enable-libspeex --enable-libvpx --disable-decoder=libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-libgsm --enable-libopenjpeg --enable-libxvid --enable-libschroedinger --enable-libdirac --enable-libxavs --enable-librtmp --enable-libfreetype --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-swscale --enable-avfilter --enable-filters --enable-postproc --target-os=darwin --arch=x86_64 --enable-runtime-cpudetect > > My conversion command line has not changed. ?The audio portion of command line: > -acodec ac3 -ab 448k -ar 48000 > > 448k is a valid bit rate!!! You may try changing it to 448000 HTH Jonathan > Get the same error wether the audio source is dts or ac3. > > Any work arounds for this error? ?Thanks in advance for help. > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rogerdpack2 at gmail.com Mon Aug 22 18:14:07 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 22 Aug 2011 10:14:07 -0600 Subject: [FFmpeg-user] A doubt in generation of frames out of a ts file. In-Reply-To: <1313956240.16097.YahooMailNeo@web122504.mail.ne1.yahoo.com> References: <1313956240.16097.YahooMailNeo@web122504.mail.ne1.yahoo.com> Message-ID: > I am using ffmpeg. I am generating images out of a ts video file. But my observation says that ffmpeg gives some gray color images initially and after some gray images , actual video images are getting generated. I have noticed green frames when playing things sometimes with ffmpeg, only at the beginning. Maybe it's waiting for an i-frame and your stream doesn't have one for a while? -r From rectalogic at rectalogic.com Mon Aug 22 18:57:43 2011 From: rectalogic at rectalogic.com (Andrew Wason) Date: Mon, 22 Aug 2011 12:57:43 -0400 Subject: [FFmpeg-user] x264 and profile flag no longer working In-Reply-To: References: Message-ID: On Mon, Aug 22, 2011 at 11:46 AM, Jonathan Isom wrote: > Using the -profile flag not longer works. ?Any changes to the command > that I'm unaware or is this a bug? Use -vprofile. https://ffmpeg.org/trac/ffmpeg/ticket/387 Andrew From jeisom at gmail.com Mon Aug 22 19:00:35 2011 From: jeisom at gmail.com (Jonathan Isom) Date: Mon, 22 Aug 2011 12:00:35 -0500 Subject: [FFmpeg-user] x264 and profile flag no longer working In-Reply-To: References: Message-ID: On Mon, Aug 22, 2011 at 11:57 AM, Andrew Wason wrote: > On Mon, Aug 22, 2011 at 11:46 AM, Jonathan Isom wrote: >> Using the -profile flag not longer works. ?Any changes to the command >> that I'm unaware or is this a bug? > > Use -vprofile. > That did the trick. Thanks Jonathan > https://ffmpeg.org/trac/ffmpeg/ticket/387 > > Andrew > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From mimeini at gmail.com Mon Aug 22 20:27:10 2011 From: mimeini at gmail.com (mikkel meinike) Date: Mon, 22 Aug 2011 20:27:10 +0200 Subject: [FFmpeg-user] Reduce audio noise In-Reply-To: <87liun92ig.fsf@Compaq.site> References: <87pqk07zcw.fsf@Compaq.site> <20110820200425.GB13947@geppetto> <87liun92ig.fsf@Compaq.site> Message-ID: I have experimented a lot with it. One can for example take the sound out something like it ffmpeg-i sameq myvideo.avi -acodec copy mysound.wav so you can run a noise filter in sox or audacity. And then take the sound back again something like this. ffmpeg-i sameq myvideo.avi -vcodec copy -mynewsound.wav -acodec copy mynewvideo.avi I will not go into details of what they do in sox and audacity because I myself have been very disappointed with the result but if you'd like to try, I will come here to explain it. I sit right now and is about to do it in a movie where there has been much wind into the microphone. Mikkel On Sat, Aug 20, 2011 at 11:49 PM, Cecil Westerhof wrote: > Op zaterdag 20 aug 2011 22:04 CEST schreef Stefano Sabatini: > >> On date Saturday 2011-08-20 19:43:27 +0200, Cecil Westerhof encoded: >>> My camera is not the best there is and I can not connect a microphone >>> to it. So the audio is recorded with the internal microphone. It is >>> not bad, but it gives some audio noise. How do I filter this noise >>> away? >> >> Audio filtering is a work in progress, and will hopefully land soon, >> right now not yet possible. > > Okay, thanks. > > -- > 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 shenguyoulan1986 at 126.com Mon Aug 22 03:49:04 2011 From: shenguyoulan1986 at 126.com (guchunlan) Date: Sun, 21 Aug 2011 18:49:04 -0700 (PDT) Subject: [FFmpeg-user] DNxHD / MOV: what is the extra data in stsd? In-Reply-To: References: Message-ID: <1313977744617-3759247.post@n4.nabble.com> Can i ask you a question ?When decoding a DNXHD .MOV file,How do you figure out the format of dnxhd,for example ,185 or 185X,which has different bitcount? Thanks a lot! -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/DNxHD-MOV-what-is-the-extra-data-in-stsd-tp3561664p3759247.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From ramandumcs at gmail.com Mon Aug 22 14:11:16 2011 From: ramandumcs at gmail.com (raman gupta) Date: Mon, 22 Aug 2011 17:41:16 +0530 Subject: [FFmpeg-user] FFMPEG support for Audio\video codec in MP4 container Message-ID: Hi All, I am using FFMPEG for transcoding of MP4 files. Could any one pls tell me what all video\audio codec combination are supported by ffmpeg in MP4 container. MP4 by definition does not specify the supported audio\video codec list. Any pointer orhelp on this will help me in moving forward. Thx in advance. Regards, Raman Gupta From luj125 at gmail.com Mon Aug 22 21:08:24 2011 From: luj125 at gmail.com (James Lu) Date: Mon, 22 Aug 2011 15:08:24 -0400 Subject: [FFmpeg-user] FFMPEG support for Audio\video codec in MP4 container In-Reply-To: References: Message-ID: Hi Raman, For mp4, I like to use -vcodec libx264 -acodec aac -strict experimental Generally mp4 accepts most codecs, and will play on any player that has the correct decoders. Wikipedia is your friend : http://en.wikipedia.org/wiki/MPEG-4_Part_14 ~James On Mon, Aug 22, 2011 at 8:11 AM, raman gupta wrote: > Hi All, > > I am using FFMPEG for transcoding of MP4 files. > Could any one pls tell me what all video\audio codec combination are > supported by ffmpeg in MP4 container. > MP4 by definition does not specify the supported audio\video codec list. > > Any pointer orhelp on this will help me in moving forward. > > Thx in advance. > > Regards, > Raman Gupta > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From Cecil at decebal.nl Mon Aug 22 22:37:07 2011 From: Cecil at decebal.nl (Cecil Westerhof) Date: Mon, 22 Aug 2011 22:37:07 +0200 Subject: [FFmpeg-user] Reduce audio noise In-Reply-To: (mikkel meinike's message of "Mon, 22 Aug 2011 20:27:10 +0200") References: <87pqk07zcw.fsf@Compaq.site> <20110820200425.GB13947@geppetto> <87liun92ig.fsf@Compaq.site> Message-ID: <87zkj1kwss.fsf@Compaq.site> Op maandag 22 aug 2011 20:27 CEST schreef mikkel meinike: > I will not go into details of what they do in sox and audacity because > I myself have been very disappointed with the result but if you'd like > to try, I will come here to explain it. I sit right now and is about > to do it in a movie where there has been much wind into the > microphone. I would like to try. By the way alert.sh produces an annoying sound. ;-} -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof From mimeini at gmail.com Mon Aug 22 23:18:41 2011 From: mimeini at gmail.com (mikkel meinike) Date: Mon, 22 Aug 2011 23:18:41 +0200 Subject: [FFmpeg-user] Reduce audio noise In-Reply-To: <87zkj1kwss.fsf@Compaq.site> References: <87pqk07zcw.fsf@Compaq.site> <20110820200425.GB13947@geppetto> <87liun92ig.fsf@Compaq.site> <87zkj1kwss.fsf@Compaq.site> Message-ID: Ok I have just done it so I'll go through the steps I just did it meself. You must have audacity installed to do this. I isolate the sound ffmpeg -sameq -i shoe12.mpg sound12.wav In audacity I open audacity and drag the audio file into Audacity. I listen and find a place on the sound file where there is wind noise. I mark this place. So I go in Effects -> Noise removal Then comes a dialogue and here I press button "Get Nois profile" Then I go back to the track and now I mark the entire track. Now again i do. Effects -> Noise removal dialog appears again but this time simply press ok at the bottom of the box. Then it remove the noise. Now you need to save. File -> Export save (export) as wav Back to ffmpeg Now I take the sound of the original video ffmpeg -i ude.mpg -an -vcodec copy udeuden.mpg And now I put the new noise reducing sound in instad ffmpeg -i udeuden.mpg -vcodec copy -i soundreduce.wav new_out.mpg In this case, I was really pleased with the outcome That was it. Hope you can follow me, otherwise ask. The video I used this for, I just put on youtube it lies here. Be aware that only a small fraction of the total video is noise is reduced. Only that which takes place outside. http://www.youtube.com/watch?v=HPKcvcvxB3s good luck :-) Mikkel From seandarcy2 at gmail.com Tue Aug 23 01:20:11 2011 From: seandarcy2 at gmail.com (sean darcy) Date: Mon, 22 Aug 2011 19:20:11 -0400 Subject: [FFmpeg-user] AVCHD from a Canon HF G10: 'non monotonically increasing dts' + dts mess In-Reply-To: References: Message-ID: On 08/21/2011 03:24 PM, Damian Stewart wrote: > Hi all, > > I'm trying to convert a number of AVCHD 1080i 50fps videos from a Canon HF G10 to mp4/mov container format, but ffmpeg is failing with a 'non monotonically increasing dts' error message at the start of each one. I'm on Mac OSX 10.6, I've tried both ffmpeg 0.8.0 (compiled by Homebrew) and 0.8.2 (compiled by ./configure&& make). > > Here's the commandline. The file is here: http://frey.co.nz/share/00167.MTS (2.2MB) > > $ ffmpeg -i 00167.MTS -vcodec copy -acodec copy out/00167.mp4 > > Output fails with: > > Input #0, mpegts, from '00167.MTS': > Duration: 00:00:01.44, start: 0.483578, bitrate: 12967 kb/s > Program 1 > Stream #0.0[0x1011]: Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc > Stream #0.1[0x1100]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s > ... > [mp4 @ 0x101071200] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 1>= 1 > > If I run ffprobe, the timestamps seem to be scrambled: > > $ ffprobe -show_packets 00167.MTS | grep dts > > ... > dts=47119 > dts_time=0.523544 > dts=49999 > dts_time=0.555544 > dts=52879 > dts_time=0.587544 > dts=55759 > dts_time=0.619544 > dts=43519 > dts_time=0.483544 > dts=45319 > dts_time=0.503544 > dts=47119 > dts_time=0.523544 > dts=48919 > dts_time=0.543544 > dts=50719 > dts_time=0.563544 > dts=58639 > dts_time=0.651544 > dts=52519 > dts_time=0.583544 > ... > dts=75919 > dts_time=0.843544 > dts=75919 > dts_time=0.843544 > dts=78799 > dts_time=0.875544 > dts=77719 > dts_time=0.863544 > dts=79519 > dts_time=0.883544 > dts=81679 > dts_time=0.907544 > dts=81319 > dts_time=0.903544 > dts=83119 > dts_time=0.9 > > > What's going on here? Is my camera really putting out such scrambled timestamps? Is there anything I can do about it? > > Any help appreciated. > > Cheers, > Damian > > -- > damian stewart . @damian0815 . damian at frey.co.nz > frey . contemporary art . http://www.frey.co.nz This is a common problem for me. Sometimes this works when ffmpeg fails: mplayer -nosound -benchmark -vo yuv4mpeg:file=>(x264 --demuxer y4m --crf 20 --threads auto --output output.264 - 2>x264.log) input.mts See: https://sites.google.com/site/linuxencoding/x264-encoding-guide I can't imagine why mplayer (based on ffmpeg) sometimes works when ffmpeg doesn't. Sadly, if mplayer doesn't work you're probably out of luck. At least in the open-source world. And yes, it's puzzling why so much AVCHD doesn't work with the open source tools. good luck. sean From glau.stuff at ridiculousprods.com Tue Aug 23 02:49:56 2011 From: glau.stuff at ridiculousprods.com (Glau Stuff) Date: Mon, 22 Aug 2011 17:49:56 -0700 Subject: [FFmpeg-user] RAW audio to MP3 via ffmpeg Message-ID: I've got a .raw audio file I'd like to convert to MP3 using ffmpeg. I know that I could first convert it using sox with something like: sox -r 44100 -e unsigned -b 8 -c1 test_file.raw test_file.wav And then use ffmpeg to convert the WAV to MP3. But I want to skip a sox pass if I can and go straight to MP3. Is that possible? I saw mention of raw formats in the documentation but not a set of commands specific for audio (unless I missed it) -- Phil From luj125 at gmail.com Tue Aug 23 03:04:30 2011 From: luj125 at gmail.com (James Lu) Date: Mon, 22 Aug 2011 21:04:30 -0400 Subject: [FFmpeg-user] RAW audio to MP3 via ffmpeg In-Reply-To: References: Message-ID: On Mon, Aug 22, 2011 at 8:49 PM, Glau Stuff wrote: > I've got a .raw audio file I'd like to convert to MP3 using ffmpeg. I know > that I could first convert it using sox with something like: > > sox -r 44100 -e unsigned -b 8 -c1 test_file.raw test_file.wav > > And then use ffmpeg to convert the WAV to MP3. > > But I want to skip a sox pass if I can and go straight to MP3. Is that > possible? I saw mention of raw formats in the documentation but not a set > of > commands specific for audio (unless I missed it) > -- > Phil > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Hi Phil, ffmpeg is really awesome, it should do any encoding you want. just go ahead and do what you think you should: ffmpeg -i input.raw -acodec libmp3 output.mp3 With encoding, you want to put in as many options as you can, so I suggest using -ab. Check the documentation for more commands Some tips: 1) use ffmpeg -formats and ffmpeg -codecs to check for which formats and codecs are available for your build of ffmpeg 2) for audio manipulations, sox and audacity are usually better alternatives, but ffmpeg has a much larger array of formats available. Hope this helps, ~James From lou at lrcd.com Tue Aug 23 03:22:49 2011 From: lou at lrcd.com (Lou) Date: Mon, 22 Aug 2011 17:22:49 -0800 Subject: [FFmpeg-user] RAW audio to MP3 via ffmpeg In-Reply-To: References: Message-ID: <20110822172249.3b2621ec@lrcd.com> On Mon, 22 Aug 2011 21:04:30 -0400 James Lu wrote: > On Mon, Aug 22, 2011 at 8:49 PM, Glau Stuff > wrote: > > > I've got a .raw audio file I'd like to convert to MP3 using ffmpeg. > > I know that I could first convert it using sox with something like: > > > > sox -r 44100 -e unsigned -b 8 -c1 test_file.raw test_file.wav > > > > And then use ffmpeg to convert the WAV to MP3. > > > > But I want to skip a sox pass if I can and go straight to MP3. Is > > that possible? I saw mention of raw formats in the documentation > > but not a set of > > commands specific for audio (unless I missed it) > > -- > > Phil > > Hi Phil, > > ffmpeg is really awesome, it should do any encoding you want. just go > ahead and do what you think you should: > > ffmpeg -i input.raw -acodec libmp3 output.mp3 -acodec libmp3lame You may have to tell ffmpeg more about the raw input such as: ffmpeg -f u16le -ar 44100 -ac 1 -i input.raw ... > With encoding, you want to put in as many options as you can, so I > suggest using -ab. Check the documentation for more commands Often overlooked is -aq instead of -ab. Appropriate values for this option depend on the encoder. With libmp3lame, this option is equivalent to the -V option in lame. For a nice chart see: > Some tips: > 1) use ffmpeg -formats and ffmpeg -codecs to check for which formats > and codecs are available for your build of ffmpeg > 2) for audio manipulations, sox and audacity are usually better > alternatives, but ffmpeg has a much larger array of formats available. SoX can be compiled to support MP3 encoding too. > Hope this helps, > ~James From david at davidfavor.com Tue Aug 23 05:04:23 2011 From: david at davidfavor.com (David Favor) Date: Mon, 22 Aug 2011 22:04:23 -0500 Subject: [FFmpeg-user] Pad filter seems to have no effect Message-ID: <4E5318B7.80707@davidfavor.com> Input footage is a Skype recording with two video tracks which require to be positioned side by side in the final video. Looks like the pad + overlay filter is the best way to do this. David-Favor-iMac> ffmpeg -filters | egrep 'pad|overlay' ffmpeg version 0.7.1, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 15 2011 19:40:46 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 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). Input footage is... David-Favor-iMac> ffmpeg -i test.skype.mov ffmpeg version 0.7.1, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 15 2011 19:40:46 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x122054400] multiple edit list entries, a/v desync might occur, patch welcome Last message repeated 1 times [mov,mp4,m4a,3gp,3g2,mj2 @ 0x122054400] max_analyze_duration 5000000 reached at 5015510 Seems stream 2 codec frame rate differs from container frame rate: 2000.00 (2000/1) -> 1000.00 (2000/2) Seems stream 3 codec frame rate differs from container frame rate: 2000.00 (2000/1) -> 1000.00 (2000/2) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.skype.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-08-18 14:38:00 Duration: 00:15:32.39, start: 0.000000, bitrate: 818 kb/s Stream #0.0(eng): Audio: aac, 44100 Hz, mono, s16, 64 kb/s Metadata: creation_time : 2011-08-18 14:38:00 Stream #0.1(eng): Audio: aac, 44100 Hz, stereo, s16, 97 kb/s Metadata: creation_time : 2011-08-18 14:38:00 Stream #0.2(eng): Video: h264 (Main), yuv420p, 640x480, 343 kb/s, 14.53 fps, 1k tbr, 1k tbn, 2k tbc Metadata: creation_time : 2011-08-18 14:38:00 Stream #0.3(eng): Video: h264 (Main), yuv420p, 640x480, 297 kb/s, 15.02 fps, 1k tbr, 1k tbn, 2k tbc Metadata: creation_time : 2011-08-18 14:38:00 At least one output file must be specified I've tried pad with all sorts of variations of arguments and codecs, with no effect. The command which appears to be correct is: ffmpeg -y -i test.skype.mov -an -vf pad=640:480:640:0:white -map 0.2 -vcodec copy test.video.mov My goal is to just create an output video right padding one video track by 640 pixels. Someone let me know where I've gone wrong. Thanks. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! From akshar_tank at yahoo.com Tue Aug 23 07:36:04 2011 From: akshar_tank at yahoo.com (tank pranav) Date: Mon, 22 Aug 2011 22:36:04 -0700 (PDT) Subject: [FFmpeg-user] A doubt in generation of frames out of a ts file. In-Reply-To: References: <1313956240.16097.YahooMailNeo@web122504.mail.ne1.yahoo.com> Message-ID: <1314077764.1168.YahooMailNeo@web122515.mail.ne1.yahoo.com> But how we can give command to ffmpeg not to generate those green/gray images ?? Do u have any idea for that ??? pranav. ________________________________ From: Roger Pack To: FFmpeg user questions and RTFMs Sent: Monday, August 22, 2011 9:44 PM Subject: Re: [FFmpeg-user] A doubt in generation of frames out of a ts file. > I am using ffmpeg. I am generating images out of a ts video file. But my observation says that ffmpeg gives some gray color images initially and after some gray images , actual video images are getting generated. I have noticed green frames when playing things sometimes with ffmpeg, only at the beginning.? Maybe it's waiting for an i-frame and your stream doesn't have one for a while? -r _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From krueger at signal7.de Tue Aug 23 09:21:25 2011 From: krueger at signal7.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Tue, 23 Aug 2011 09:21:25 +0200 Subject: [FFmpeg-user] AVCHD from a Canon HF G10: 'non monotonically increasing dts' + dts mess In-Reply-To: References: Message-ID: Damian, On Aug 21, 2011, at 21:24 , Damian Stewart wrote: > Hi all, > > I'm trying to convert a number of AVCHD 1080i 50fps videos from a Canon HF G10 to mp4/mov container format, but ffmpeg is failing with a 'non monotonically increasing dts' error message at the start of each one. I'm on Mac OSX 10.6, I've tried both ffmpeg 0.8.0 (compiled by Homebrew) and 0.8.2 (compiled by ./configure && make). > > Here's the commandline. The file is here: http://frey.co.nz/share/00167.MTS (2.2MB) > > $ ffmpeg -i 00167.MTS -vcodec copy -acodec copy out/00167.mp4 > > Output fails with: > > Input #0, mpegts, from '00167.MTS': > Duration: 00:00:01.44, start: 0.483578, bitrate: 12967 kb/s > Program 1 > Stream #0.0[0x1011]: Video: h264 (High), yuv420p, 1920x1080 [PAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc > Stream #0.1[0x1100]: Audio: ac3, 48000 Hz, stereo, s16, 256 kb/s > ... > [mp4 @ 0x101071200] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 1 >= 1 > > If I run ffprobe, the timestamps seem to be scrambled: > > $ ffprobe -show_packets 00167.MTS | grep dts > > ... > dts=47119 > dts_time=0.523544 > dts=49999 > dts_time=0.555544 > dts=52879 > dts_time=0.587544 > dts=55759 > dts_time=0.619544 > dts=43519 > dts_time=0.483544 > dts=45319 > dts_time=0.503544 > dts=47119 > dts_time=0.523544 > dts=48919 > dts_time=0.543544 > dts=50719 > dts_time=0.563544 > dts=58639 > dts_time=0.651544 > dts=52519 > dts_time=0.583544 > ... > dts=75919 > dts_time=0.843544 > dts=75919 > dts_time=0.843544 > dts=78799 > dts_time=0.875544 > dts=77719 > dts_time=0.863544 > dts=79519 > dts_time=0.883544 > dts=81679 > dts_time=0.907544 > dts=81319 > dts_time=0.903544 > dts=83119 > dts_time=0.9 > > > What's going on here? Is my camera really putting out such scrambled timestamps? Is there anything I can do about it? > > Any help appreciated. > > Cheers, > Damian > could you please file a bug report in the bug tracker and supply a very short sample clip with it? I've had a similar problem in the past and it is buried somewhere in the old roundup bugtracker that I don't know if anyone is using it or if the information is essentially lost. I managed to get this working in my case with a hack in the code by replacing a ">"-Check with ">=" where the error occurred but it might be something else in your case and of course that was not a proper fix but if you want to get your material transcoded, you probably don't really care about that (I didn't). In any case, having this in the bug tracker with a small sample increases chances that someone can/will fix it. Please provide the camera model and recording mode in your bug report so others find this when they run into the same problem. Cheers, Robert From damian.ml at frey.co.nz Tue Aug 23 10:56:50 2011 From: damian.ml at frey.co.nz (Damian Stewart) Date: Tue, 23 Aug 2011 10:56:50 +0200 Subject: [FFmpeg-user] AVCHD from a Canon HF G10: 'non monotonically increasing dts' + dts mess In-Reply-To: References: Message-ID: <3504BFD0-D2DF-466A-BAC5-0BD5E1D0F8BC@frey.co.nz> On 23 Aug 2011, at 09:21, Robert Kr?ger wrote: > Damian, > could you please file a bug report in the bug tracker and supply a very short sample clip with it? Done: https://ffmpeg.org/trac/ffmpeg/ticket/415 Could be related: http://ffmpeg.org/trac/ffmpeg/ticket/177 FWIW, writing the mp4 stream to an mkv file seems to be working fine, but this is not ideal as the video editing software I'd like to use doesn't natively support mkv. -- damian stewart . @damian0815 . damian at frey.co.nz frey . contemporary art . http://www.frey.co.nz From ffmpeg-user at adruna.org Tue Aug 23 11:22:48 2011 From: ffmpeg-user at adruna.org (Adruna) Date: Tue, 23 Aug 2011 11:22:48 +0200 Subject: [FFmpeg-user] AVCHD from a Canon HF G10: 'non monotonically increasing dts' + dts mess In-Reply-To: References: Message-ID: <4E537168.3000904@adruna.org> Hi, On 23-8-2011 9:21, Robert Kr?ger wrote: > could you please file a bug report in the bug tracker and supply a very short sample clip with it? > > I've had a similar problem in the past and it is buried somewhere in the old roundup bugtracker that I don't know if anyone is using it or if the information is essentially lost. I managed to get this working in my case with a hack in the code by replacing a ">"-Check with ">=" where the error occurred but it might be something else in your case and of course that was not a proper fix but if you want to get your material transcoded, you probably don't really care about that (I didn't). In any case, having this in the bug tracker with a small sample increases chances that someone can/will fix it. Please provide the camera model and recording mode in your bug report so others find this when they run into the same problem. I have the same problem with the "JVC Everio GZ-HM650". The problem only occurs when copying the original streams to the new container. So I get the error (build: N-32023-gf138c7f) with the command from Damian: ffmpeg -i 00167.MTS -vcodec copy -acodec copy out/00167.mp4 but when transcoding like: ffmpeg -i 00167.MTS -vcodec libx264 -acodec copy out/00167.mp4 it works. Please let me know the bug submission, so I can add a sample file, Recording device info and this note if needed. Regards, Adruna From rhodri at kynesim.co.uk Tue Aug 23 12:35:17 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Tue, 23 Aug 2011 11:35:17 +0100 Subject: [FFmpeg-user] A doubt in generation of frames out of a ts file. In-Reply-To: <1314077764.1168.YahooMailNeo@web122515.mail.ne1.yahoo.com> References: <1313956240.16097.YahooMailNeo@web122504.mail.ne1.yahoo.com> <1314077764.1168.YahooMailNeo@web122515.mail.ne1.yahoo.com> Message-ID: On Tue, 23 Aug 2011 06:36:04 +0100, tank pranav wrote: > But how we can give command to ffmpeg not to generate those green/gray > images ?? Do u have any idea for that ?? Do you mean that you want the output to start from the point where you get an I-frame? That could potentially be a long time in the future -- I've had to deal with streams that went several minutes between I-frames. -- Rhodri James Kynesim Ltd From akshar_tank at yahoo.com Tue Aug 23 12:40:05 2011 From: akshar_tank at yahoo.com (tank pranav) Date: Tue, 23 Aug 2011 03:40:05 -0700 (PDT) Subject: [FFmpeg-user] A doubt in generation of frames out of a ts file. In-Reply-To: References: <1313956240.16097.YahooMailNeo@web122504.mail.ne1.yahoo.com> <1314077764.1168.YahooMailNeo@web122515.mail.ne1.yahoo.com> Message-ID: <1314096005.66942.YahooMailNeo@web122505.mail.ne1.yahoo.com> Yes, I would like ffmpeg should generate frame when it finds first I frame. If it gives initial gray color frame then it definately means that it did not find I-frame thati s why ffmpeg keeps on inserting dummy frames. I dont have any idea which command does this removing gray color frames. We need to change something in coding. Where should we change that I am still figuring out. If you have any idea how we should keep on drop the frame till we get I frame then it may solve my purpose. pranav. ________________________________ From: Rhodri James co To: FFmpeg user questions and RTFMs Sent: Tuesday, August 23, 2011 4:05 PM Subject: Re: [FFmpeg-user] A doubt in generation of frames out of a ts file. On Tue, 23 Aug 2011 06:36:04 +0100, tank pranav wrote: > But how we can give command to ffmpeg not to generate those green/gray images ?? Do u have any idea for that ?? Do you mean that you want the output to start from the point where you get an I-frame?? That could potentially be a long time in the future -- I've had to deal with streams that went several minutes between I-frames. --Rhodri James Kynesim Ltd _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From stefano.sabatini-lala at poste.it Tue Aug 23 13:54:31 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Tue, 23 Aug 2011 13:54:31 +0200 Subject: [FFmpeg-user] Pad filter seems to have no effect In-Reply-To: <4E5318B7.80707@davidfavor.com> References: <4E5318B7.80707@davidfavor.com> Message-ID: <20110823115431.GA21218@geppetto> On date Monday 2011-08-22 22:04:23 -0500, David Favor encoded: > Input footage is a Skype recording with two video > tracks which require to be positioned side by side > in the final video. > > Looks like the pad + overlay filter is the best way > to do this. [...] > David-Favor-iMac> ffmpeg -i test.skype.mov [...] > Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.skype.mov': > Metadata: > major_brand : qt > minor_version : 537199360 > compatible_brands: qt > creation_time : 2011-08-18 14:38:00 > Duration: 00:15:32.39, start: 0.000000, bitrate: 818 kb/s > Stream #0.0(eng): Audio: aac, 44100 Hz, mono, s16, 64 kb/s > Metadata: > creation_time : 2011-08-18 14:38:00 > Stream #0.1(eng): Audio: aac, 44100 Hz, stereo, s16, 97 kb/s > Metadata: > creation_time : 2011-08-18 14:38:00 > Stream #0.2(eng): Video: h264 (Main), yuv420p, 640x480, 343 kb/s, 14.53 fps, 1k tbr, 1k tbn, 2k tbc > Metadata: > creation_time : 2011-08-18 14:38:00 > Stream #0.3(eng): Video: h264 (Main), yuv420p, 640x480, 297 kb/s, 15.02 fps, 1k tbr, 1k tbn, 2k tbc > Metadata: > creation_time : 2011-08-18 14:38:00 > At least one output file must be specified > > I've tried pad with all sorts of variations of arguments and codecs, > with no effect. > > The command which appears to be correct is: > > ffmpeg -y -i test.skype.mov -an -vf pad=640:480:640:0:white -map 0.2 -vcodec copy test.video.mov pad=width:height[:x:y[:color]] the padded area (specified by width:height) must contain the input video placed at position 640:0, so it should be: pad=640+640:480:640:0 From damian.ml at frey.co.nz Tue Aug 23 14:22:05 2011 From: damian.ml at frey.co.nz (Damian Stewart) Date: Tue, 23 Aug 2011 14:22:05 +0200 Subject: [FFmpeg-user] AVCHD from a Canon HF G10: 'non monotonically increasing dts' + dts mess In-Reply-To: <4E537168.3000904@adruna.org> References: <4E537168.3000904@adruna.org> Message-ID: I've discovered that ffmbc http://code.google.com/p/ffmbc/ can do this out of the box: ffmbc -i 00167.MTS -vcodec copy -acodec aac -ab 256k 00167.mp4 produces an .mp4 that Quicktime is perfectly happy to play. I wonder how easy/difficult it would be to backport ffmbc patches to ffmpeg? On 23 Aug 2011, at 11:22, Adruna wrote: > Hi, > > On 23-8-2011 9:21, Robert Kr?ger wrote: >> could you please file a bug report in the bug tracker and supply a very short sample clip with it? >> >> I've had a similar problem in the past and it is buried somewhere in the old roundup bugtracker that I don't know if anyone is using it or if the information is essentially lost. I managed to get this working in my case with a hack in the code by replacing a ">"-Check with ">=" where the error occurred but it might be something else in your case and of course that was not a proper fix but if you want to get your material transcoded, you probably don't really care about that (I didn't). In any case, having this in the bug tracker with a small sample increases chances that someone can/will fix it. Please provide the camera model and recording mode in your bug report so others find this when they run into the same problem. > > I have the same problem with the "JVC Everio GZ-HM650". The problem only > occurs when copying the original streams to the new container. > > So I get the error (build: N-32023-gf138c7f) with the command from Damian: > > ffmpeg -i 00167.MTS -vcodec copy -acodec copy out/00167.mp4 > > but when transcoding like: > > ffmpeg -i 00167.MTS -vcodec libx264 -acodec copy out/00167.mp4 > > it works. Please let me know the bug submission, so I can add a sample > file, Recording device info and this note if needed. > > Regards, > > Adruna > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- damian stewart . @damian0815 . damian at frey.co.nz frey . contemporary art . http://www.frey.co.nz From krueger at signal7.de Tue Aug 23 14:55:39 2011 From: krueger at signal7.de (=?iso-8859-1?Q?Robert_Kr=FCger?=) Date: Tue, 23 Aug 2011 14:55:39 +0200 Subject: [FFmpeg-user] AVCHD from a Canon HF G10: 'non monotonically increasing dts' + dts mess In-Reply-To: References: <4E537168.3000904@adruna.org> Message-ID: <2C1EDAE3-6E54-4233-92C6-2D61AE96D690@signal7.de> On Aug 23, 2011, at 14:22 , Damian Stewart wrote: > I've discovered that ffmbc http://code.google.com/p/ffmbc/ can do this out of the box: > > ffmbc -i 00167.MTS -vcodec copy -acodec aac -ab 256k 00167.mp4 > > produces an .mp4 that Quicktime is perfectly happy to play. I wonder how easy/difficult it would be to backport ffmbc patches to ffmpeg? > > On 23 Aug 2011, at 11:22, Adruna wrote: > >> Hi, >> >> On 23-8-2011 9:21, Robert Kr?ger wrote: >>> could you please file a bug report in the bug tracker and supply a very short sample clip with it? >>> >>> I've had a similar problem in the past and it is buried somewhere in the old roundup bugtracker that I don't know if anyone is using it or if the information is essentially lost. I managed to get this working in my case with a hack in the code by replacing a ">"-Check with ">=" where the error occurred but it might be something else in your case and of course that was not a proper fix but if you want to get your material transcoded, you probably don't really care about that (I didn't). In any case, having this in the bug tracker with a small sample increases chances that someone can/will fix it. Please provide the camera model and recording mode in your bug report so others find this when they run into the same problem. >> >> I have the same problem with the "JVC Everio GZ-HM650". The problem only >> occurs when copying the original streams to the new container. >> >> So I get the error (build: N-32023-gf138c7f) with the command from Damian: >> >> ffmpeg -i 00167.MTS -vcodec copy -acodec copy out/00167.mp4 >> >> but when transcoding like: >> >> ffmpeg -i 00167.MTS -vcodec libx264 -acodec copy out/00167.mp4 >> >> it works. Please let me know the bug submission, so I can add a sample >> file, Recording device info and this note if needed. >> >> Regards, >> >> Adruna >> _______________________________________________ >> ffmpeg-user mailing list >> ffmpeg-user at ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user technically this would probably work but ffmbc is GPL whereas the corresponding code in ffmpeg is LGPL, i.e. backporting without the explicit consent of ffmbc's author baptiste to relicense to LGPL would make the ffmpeg code GPL which would be a problem for many people. From damian.ml at frey.co.nz Tue Aug 23 15:04:25 2011 From: damian.ml at frey.co.nz (Damian Stewart) Date: Tue, 23 Aug 2011 15:04:25 +0200 Subject: [FFmpeg-user] AVCHD from a Canon HF G10: 'non monotonically increasing dts' + dts mess In-Reply-To: <2C1EDAE3-6E54-4233-92C6-2D61AE96D690@signal7.de> References: <4E537168.3000904@adruna.org> <2C1EDAE3-6E54-4233-92C6-2D61AE96D690@signal7.de> Message-ID: <43F02636-3602-4245-9154-CECC240D5EF1@frey.co.nz> On 23 Aug 2011, at 14:55, Robert Kr?ger wrote: > On Aug 23, 2011, at 14:22 , Damian Stewart wrote: > >> I wonder how easy/difficult it would be to backport ffmbc patches to ffmpeg? > > technically this would probably work but ffmbc is GPL whereas the corresponding code in ffmpeg is LGPL, i.e. backporting without the explicit consent of ffmbc's author baptiste to relicense to LGPL would make the ffmpeg code GPL which would be a problem for many people. Aah, that'd be true.. -- damian stewart . @damian0815 . damian at frey.co.nz frey . contemporary art . http://www.frey.co.nz From david at davidfavor.com Tue Aug 23 15:50:23 2011 From: david at davidfavor.com (David Favor) Date: Tue, 23 Aug 2011 08:50:23 -0500 Subject: [FFmpeg-user] Pad filter seems to have no effect In-Reply-To: <20110823115431.GA21218@geppetto> References: <4E5318B7.80707@davidfavor.com> <20110823115431.GA21218@geppetto> Message-ID: <4E53B01F.3050301@davidfavor.com> >> The command which appears to be correct is: >> >> ffmpeg -y -i test.skype.mov -an -vf pad=640:480:640:0:white -map 0.2 -vcodec copy test.video.mov > > pad=width:height[:x:y[:color]] > > the padded area (specified by width:height) must contain the input > video placed at position 640:0, so it should be: > pad=640+640:480:640:0 Already tried your suggestion... This produces no effect: ffmpeg -y -i test.skype.mov -an -vf pad=1280:480:640:0:white -map 0.2 -vcodec copy test.video.mov Maybe I best bump up to latest, which is probably 0.7.2 or 0.7.3 and retest. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! From nicolas.george at normalesup.org Tue Aug 23 16:32:08 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Tue, 23 Aug 2011 16:32:08 +0200 Subject: [FFmpeg-user] Pad filter seems to have no effect In-Reply-To: <4E53B01F.3050301@davidfavor.com> References: <4E5318B7.80707@davidfavor.com> <20110823115431.GA21218@geppetto> <4E53B01F.3050301@davidfavor.com> Message-ID: <20110823143208.GA17422@phare.normalesup.org> Le sextidi 6 fructidor, an CCXIX, David Favor a ?crit?: > ffmpeg -y -i test.skype.mov -an -vf pad=1280:480:640:0:white -map 0.2 -vcodec copy test.video.mov Obviously, you can only apply video filters to the decoded video, but your command tells ffmpeg not to decode the video. It may be theoretically possible to add borders without reencoding, provided the size of the borders is a multiple of the block size of the codec, but ffmpeg does not have any feature of the sort, and will probably never have. 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 david at davidfavor.com Tue Aug 23 16:40:44 2011 From: david at davidfavor.com (David Favor) Date: Tue, 23 Aug 2011 09:40:44 -0500 Subject: [FFmpeg-user] Pad filter seems to have no effect In-Reply-To: <4E53B01F.3050301@davidfavor.com> References: <4E5318B7.80707@davidfavor.com> <20110823115431.GA21218@geppetto> <4E53B01F.3050301@davidfavor.com> Message-ID: <4E53BBEC.2070106@davidfavor.com> David Favor wrote: >>> The command which appears to be correct is: >>> >>> ffmpeg -y -i test.skype.mov -an -vf pad=640:480:640:0:white -map >>> 0.2 -vcodec copy test.video.mov >> >> pad=width:height[:x:y[:color]] >> >> the padded area (specified by width:height) must contain the input >> video placed at position 640:0, so it should be: >> pad=640+640:480:640:0 > > Already tried your suggestion... > > This produces no effect: > > ffmpeg -y -i test.skype.mov -an -vf pad=1280:480:640:0:white -map 0.2 > -vcodec copy test.video.mov > > Maybe I best bump up to latest, which is probably 0.7.2 or 0.7.3 and > retest. > Updated to ffmpeg-0.7.3 and still no effect. Neither of these produce any effect: ffmpeg -y -i test.skype.mov -an -vf scale=1280:480 -map 0.2 -vcodec copy test.video.mov ffmpeg -y -i test.skype.mov -an -vf scale=1280:480,pad=1280:480:640:0:white -map 0.2 -vcodec copy test.video.mov Maybe the problem is -vcodec copy... although I've tried various re-encoding schemes with no effect either. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! From david at davidfavor.com Tue Aug 23 17:04:59 2011 From: david at davidfavor.com (David Favor) Date: Tue, 23 Aug 2011 10:04:59 -0500 Subject: [FFmpeg-user] Pad filter seems to have no effect In-Reply-To: <4E53BBEC.2070106@davidfavor.com> References: <4E5318B7.80707@davidfavor.com> <20110823115431.GA21218@geppetto> <4E53B01F.3050301@davidfavor.com> <4E53BBEC.2070106@davidfavor.com> Message-ID: <4E53C19B.8010702@davidfavor.com> David Favor wrote: > David Favor wrote: >>>> The command which appears to be correct is: >>>> >>>> ffmpeg -y -i test.skype.mov -an -vf pad=640:480:640:0:white -map >>>> 0.2 -vcodec copy test.video.mov >>> >>> pad=width:height[:x:y[:color]] >>> >>> the padded area (specified by width:height) must contain the input >>> video placed at position 640:0, so it should be: >>> pad=640+640:480:640:0 >> >> Already tried your suggestion... >> >> This produces no effect: >> >> ffmpeg -y -i test.skype.mov -an -vf pad=1280:480:640:0:white -map >> 0.2 -vcodec copy test.video.mov >> >> Maybe I best bump up to latest, which is probably 0.7.2 or 0.7.3 and >> retest. >> > > Updated to ffmpeg-0.7.3 and still no effect. > > Neither of these produce any effect: > > ffmpeg -y -i test.skype.mov -an -vf scale=1280:480 -map 0.2 -vcodec > copy test.video.mov > > ffmpeg -y -i test.skype.mov -an -vf > scale=1280:480,pad=1280:480:640:0:white -map 0.2 -vcodec copy > test.video.mov > > Maybe the problem is -vcodec copy... although I've tried various > re-encoding schemes with no effect either. > This works... ffmpeg -benchmark -y -i 30sec-test.skype.mov -an -vf pad=1280:480:640:0 -map 0.2 \ -vcodec libx264 -threads 0 -vpre lossless_fast -crf 25 30sec-test.video.mov And takes a huge amount of time. Best estimate on an 8x core machine is around 3 minutes encoding time for every 1 minute of video. Ouch. Got to be a better way to extract the video streams intact. I'll start a new thread about the problems with simple mapping to copy the video streams intact. Thanks. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! From mbp at netomia.dk Mon Aug 22 21:06:55 2011 From: mbp at netomia.dk (Martin Bay (Netomia)) Date: Mon, 22 Aug 2011 21:06:55 +0200 Subject: [FFmpeg-user] x264 encoding of certain (small) videos take hours Message-ID: <4E52A8CF.3070106@netomia.dk> Whenever I try to encode a video file form my Android phone with x264, it takes several hours - as compared to seconds with other videos of the same length and size. Does anyone know of a fix? ffmpeg -y -i -ab 96k -acodec libfaac -ac 2 -vcodec libx264 -vprofile baseline -threads 4 .mp4 ffmpeg version N-31979-gedae3db, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 18 2011 20:59:51 with gcc 4.4.5 configuration: --enable-gpl --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-nonfree --enable-postproc --enable-version3 --enable-x11grab libavutil 51. 12. 0 / 51. 12. 0 libavcodec 53. 11. 0 / 53. 11. 0 libavformat 53. 8. 0 / 53. 8. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 32. 0 / 2. 32. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x32b3460] multiple edit list entries, a/v desync might occur, patch welcome Metadata: major_brand : 3gp4 minor_version : 0 compatible_brands: isom3gp4 creation_time : 1945-07-08 03:28:22 Duration: 00:00:50.24, start: 0.000000, bitrate: 3142 kb/s Stream #0.0(eng): Audio: aac, 44100 Hz, mono, s16, 63 kb/s Metadata: creation_time : 1945-07-08 03:28:22 Stream #0.1(eng): Video: h264 (Baseline), yuv420p, 480x800, 2960 kb/s, SAR 65536:65536 DAR 3:5, 22.40 fps, 90k tbr, 90k tbn, 180k tbc Metadata: creation_time : 1945-07-08 03:28:22 [mp4 @ 0x32b4920] Frame rate very high for a muxer not effciciently supporting it. Please consider specifiying a lower framerate, a different muxer or -vsync 2 [buffer @ 0x32adc40] w:480 h:800 pixfmt:yuv420p tb:1/1000000 sar:65536/65536 sws_param: [libx264 @ 0x32ad780] Default settings detected, using medium profile [libx264 @ 0x32ad780] using SAR=1/1 [libx264 @ 0x32ad780] MB rate (135000000) > level limit (983040) [libx264 @ 0x32ad780] using cpu capabilities: MMX2 SSE2Fast SSSE3 Cache64 [libx264 @ 0x32ad780] profile Constrained Baseline, level 5.1 [libx264 @ 0x32ad780] 264 - core 116 r2057 0ba8a9c - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=0 ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 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=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 -- Regards, Martin Bay Pedersen From david at davidfavor.com Wed Aug 24 01:32:21 2011 From: david at davidfavor.com (David Favor) Date: Tue, 23 Aug 2011 18:32:21 -0500 Subject: [FFmpeg-user] ffmpeg-0.7.3 vcodec copy problem with multiple video streams Message-ID: <4E543885.1000809@davidfavor.com> :::: Starting with a Skype recording... David-Favor-iMac> ffmpeg 30sec-test.skype.mov ffmpeg version 0.7.3, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 23 2011 08:57:36 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Incompatible sample format '(null)' for codec 'aac', auto-selecting format 's16' File '30sec-test.skype.mov' already exists. Overwrite ? [y/N] ^C David-Favor-iMac> ffmpeg -i 30sec-test.skype.mov ffmpeg version 0.7.3, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 23 2011 08:57:36 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 2 codec frame rate differs from container frame rate: 2000.00 (2000/1) -> 15.08 (181/12) Seems stream 3 codec frame rate differs from container frame rate: 2000.00 (2000/1) -> 15.08 (181/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '30sec-test.skype.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2011-02-25 17:04:25 Duration: 00:00:31.99, start: 0.000000, bitrate: 5209 kb/s Stream #0.0(eng): Audio: pcm_f32le, 44100 Hz, 1 channels, flt, 1411 kb/s Metadata: creation_time : 2011-02-25 17:04:25 Stream #0.1(eng): Audio: pcm_f32le, 44100 Hz, 2 channels, flt, 2822 kb/s Metadata: creation_time : 2011-02-25 17:04:25 Stream #0.2(eng): Video: h264 (Main), yuv420p, 640x480, 294 kb/s, 14.97 fps, 15.08 tbr, 1k tbn, 2k tbc Metadata: creation_time : 2011-02-25 17:04:25 Stream #0.3(eng): Video: h264 (Main), yuv420p, 640x480, 678 kb/s, 15 fps, 15.08 tbr, 1k tbn, 2k tbc Metadata: creation_time : 2011-02-25 17:04:25 :::: My goal is to copy both video streams without any change. This command comes close: ffmpeg -y -i 30sec-test.skype.mov -an -map 0.2 -map 0.3 \ -vcodec copy -vcodec copy 30sec-test.video.mov -newvideo Except the second stream in the output video has changed codecs... David-Favor-iMac> ffmpeg -i 30sec-test.video.mov ffmpeg version 0.7.3, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 23 2011 08:57:36 with gcc 4.2.1 (Apple Inc. build 5666) (dot 3) configuration: --prefix=/opt/local --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libxvid --enable-libx264 --enable-libvpx --enable-libspeex --mandir=/opt/local/share/man --enable-shared --enable-pthreads --disable-indevs --cc=/usr/bin/gcc-4.2 --arch=x86_64 --enable-yasm libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 2000.00 (2000/1) -> 15.08 (181/12) Seems stream 1 codec frame rate differs from container frame rate: 181.00 (181/1) -> 15.08 (181/12) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '30sec-test.video.mov': Metadata: major_brand : qt minor_version : 512 compatible_brands: qt creation_time : 1970-01-01 00:00:00 encoder : Lavf52.110.0 Duration: 00:00:31.99, start: 0.000000, bitrate: 962 kb/s Stream #0.0(eng): Video: h264 (Main), yuv420p, 640x480, 678 kb/s, 15 fps, 15.08 tbr, 1k tbn, 2k tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Video: mpeg4, yuv420p, 640x480 [PAR 1:1 DAR 4:3], 282 kb/s, 15.11 fps, 15.08 tbr, 181 tbn, 181 tbc Metadata: creation_time : 1970-01-01 00:00:00 :::: Maybe there's additional syntax for -map to do the copy correctly. Pass along suggestions. Thanks. -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! From office at motorrad-austria.at Tue Aug 23 21:56:45 2011 From: office at motorrad-austria.at (=?iso-8859-1?Q?Ingmar_Erd=F6s?=) Date: Tue, 23 Aug 2011 21:56:45 +0200 Subject: [FFmpeg-user] ERROR: libx264 version must be >= 0.99 Message-ID: <8CA26C1D558A4F4987394B31C194E46F69FD74@server.Globalcom.intern> Hi, getting x264 from git and ffmpeg by svn - both newest versions. By starting ./configure with -enable-libx264 I only get following error: ERROR: libx264 version must be >= 0.99. Is there any work around out there to solve this problem? From lou at lrcd.com Wed Aug 24 03:22:01 2011 From: lou at lrcd.com (Lou) Date: Tue, 23 Aug 2011 17:22:01 -0800 Subject: [FFmpeg-user] ERROR: libx264 version must be >= 0.99 In-Reply-To: <8CA26C1D558A4F4987394B31C194E46F69FD74@server.Globalcom.intern> References: <8CA26C1D558A4F4987394B31C194E46F69FD74@server.Globalcom.intern> Message-ID: <20110823172201.5de6912a@lrcd.com> On Tue, 23 Aug 2011 21:56:45 +0200 Ingmar Erd?s wrote: > Hi, getting x264 from git and ffmpeg by svn - both newest versions. > By starting ./configure with -enable-libx264 I only get following > error: > > > > ERROR: libx264 version must be >= 0.99. > > > > Is there any work around out there to solve this problem? FFmpeg uses Git instead of SVN now. The last revision to the FFmpeg SVN repository was on 2011-01-19, so try using more recent source from Git. From rickcorteza at gmail.com Wed Aug 24 04:34:11 2011 From: rickcorteza at gmail.com (Rick C.) Date: Wed, 24 Aug 2011 10:34:11 +0800 Subject: [FFmpeg-user] subtitles again please Message-ID: Hi again, I have asked in the past about hard-coding subtitles, but if I just want to add a subtitle track as a stream would someone mind giving me a working example? I'm aware of the documentation which says this: ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt And in FFmpeg -formats it says it supports decoding/encoding of .srt files. So how would I take a file without a subtitle stream and add a new one? Something like: ffmpeg -i mov.avi -i sub.srt output.avi Sorry to ask again but I'm assuming since it's there in the documentation it works but I really can't figure it out. Thanks! rc From dmitry at interhost.co.il Wed Aug 24 11:58:30 2011 From: dmitry at interhost.co.il (Dmitry Sherman) Date: Wed, 24 Aug 2011 12:58:30 +0300 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 Message-ID: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> Hello, I receive sometimes an error when I try to transcode some mp4 files to flv using this command: ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv I receive this error: Output #0, flv, to 'a.flv': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42isomavc1 creation_time : 2011-07-05 16:50:32 encoder : Lavf53.9.0 Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 416x224 [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc Metadata: creation_time : 2011-07-05 16:50:32 Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, stereo, s16, 64 kb/s Metadata: creation_time : 2011-07-05 16:50:32 Stream mapping: Stream #0.0 -> #0.0: copy Stream #0.1 -> #0.1: aac -> libfaac Press [q] to stop, [?] for help [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 av_interleaved_write_frame(): Invalid argument what can I do to solve this problem? Thank you. Dmitry Sherman dmitry at interhost.co.il Interhost Networks Ltd t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ http://facebook.com/InterhostIL From david at davidfavor.com Wed Aug 24 14:03:57 2011 From: david at davidfavor.com (David Favor) Date: Wed, 24 Aug 2011 07:03:57 -0500 Subject: [FFmpeg-user] FFmpeg vcodec copy problem with multiple video streams In-Reply-To: References: Message-ID: <4E54E8AD.8000007@davidfavor.com> Carl Eugen Hoyos wrote: > Hi! > > You have to place the second -vcodec copy right before -newvideo: > ffmpeg -y -i 30sec-test.skype.mov -an -map 0.2 -map 0.3 \ > -vcodec copy 30sec-test.video.mov -vcodec copy -newvideo > > Please always provide complete, uncut output of the *failing* command! > > Any reason you are not using latest git head master? > (Please note that there is not much "stable" about FFmpeg releases, > distributions unfortunately need them. 0.7 has the positive side-effect > that its shared libraries can be used for older third-party applications > but this should affect the binaries.) > > Carl Eugen Flarg... Could have sworn I tried this. Works like a charm. Thanks! -- Love feeling your best ever, all day, every day? Click http://RadicalHealth.com for the easy way! From tonybaqain at gmail.com Wed Aug 24 19:25:12 2011 From: tonybaqain at gmail.com (Antoine Baqain) Date: Wed, 24 Aug 2011 20:25:12 +0300 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 In-Reply-To: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> Message-ID: Does this http://ffmpeg.org/trac/ffmpeg/ticket/16 help you ? On Wed, Aug 24, 2011 at 12:58 PM, Dmitry Sherman wrote: > Hello, > I receive sometimes an error when I try to transcode some mp4 files to flv > using this command: > ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv > > I receive this error: > Output #0, flv, to 'a.flv': > Metadata: > major_brand : mp42 > minor_version : 0 > compatible_brands: mp42isomavc1 > creation_time : 2011-07-05 16:50:32 > encoder : Lavf53.9.0 > Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 416x224 > [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc > Metadata: > creation_time : 2011-07-05 16:50:32 > Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, stereo, > s16, 64 kb/s > Metadata: > creation_time : 2011-07-05 16:50:32 > Stream mapping: > Stream #0.0 -> #0.0: copy > Stream #0.1 -> #0.1: aac -> libfaac > Press [q] to stop, [?] for help > [flv @ 0x2bf0da0] Application provided invalid, non monotonically > increasing dts to muxer in stream 0: 160 >= 160 > av_interleaved_write_frame(): Invalid argument > > what can I do to solve this problem? > > Thank you. > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 > http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Antoine Baqain System Administration Lead Maktoob.com Inc. From stefano.sabatini-lala at poste.it Wed Aug 24 20:00:36 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Wed, 24 Aug 2011 20:00:36 +0200 Subject: [FFmpeg-user] subtitles again please In-Reply-To: References: Message-ID: <20110824180036.GB8530@geppetto> On date Wednesday 2011-08-24 10:34:11 +0800, Rick C. encoded: > Hi again, > > I have asked in the past about hard-coding subtitles, but if I just > want to add a subtitle track as a stream would someone mind giving > me a working example? I'm aware of the documentation which says > this: > > ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt > > And in FFmpeg -formats it says it supports decoding/encoding of .srt > files. So how would I take a file without a subtitle stream and add > a new one? Something like: > > ffmpeg -i mov.avi -i sub.srt output.avi > > Sorry to ask again but I'm assuming since it's there in the > documentation it works but I really can't figure it out. Thanks! Not very experienced about subtitles and all, AFAIK subtitles support in FFmpeg is still a bit... rudimentary. But if you want subtitles in output you need a format which supports them, I know matroska (.mkv) should do. As for hardcoding subtitles right onto the video, I'm afraid we still miss that facility (subtitles filtering + video overlaying seems the right solution), correct if I'm wrong. -- ffmpeg-user random tip #20 VHOOK has been removed, check out libavfilter: http://wiki.multimedia.cx/index.php?title=Libavfilter From dieterknopf at googlemail.com Wed Aug 24 08:36:55 2011 From: dieterknopf at googlemail.com (Dieter Knopf) Date: Wed, 24 Aug 2011 08:36:55 +0200 Subject: [FFmpeg-user] Time-lapse with ffmpeg, frame rate and duration Message-ID: Hello, i have a problem with ffmpeg and time-lapse movies. I want to create a time-lapse movie out of 80k jpg-pictures with a duration of ~5 minutes. 250 pictures/sec would be enough for 5,3 minutes. But how? I tried something like that: ffmpeg -r 250 -i %06d.jpg -vcodec libx264 -crf 22 -threads 0 -vf scale=1920:-1 -r 15 -metadata title="foo" output.mp4 This works fine, but the video stops exactly after 46 seconds. Here the log: # ffmpeg -r 250 -i %06d.jpg -vcodec libx264 -crf 22 -threads 0 -vf scale=1920:-1 -r 15 -metadata title="foo" output.mp4 ffmpeg version 0.7.3, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 24 2011 06:05:52 with gcc 4.4.5 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-network --disable-vaapi --enable-libmp3lame --enable-libvo-aacenc --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libfaac --enable-nonfree --disable-indev=v4l --disable-indev=v4l2 --disable-indev=oss --disable-indev=jack --enable-x11grab --disable-outdev=oss --enable-libfreetype --enable-pthreads --enable-libopenjpeg --disable-altivec --disable-avx --cpu=core2 --enable-hardcoded-tables libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.122. 0 / 52.122. 0 libavformat 52.110. 0 / 52.110. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 Seems stream 0 codec frame rate differs from container frame rate: 250.00 (250/1) -> 0.08 (1/12) Input #0, image2, from '%06d.jpg': Duration: 00:04:52.18, start: 0.000000, bitrate: N/A Stream #0.0: Video: mjpeg, yuvj420p, 640x480 [PAR 1:1 DAR 4:3], 250 fps, 0.08 tbr, 250 tbn, 250 tbc [buffer @ 0x2490170] w:640 h:480 pixfmt:yuvj420p tb:1/1000000 sar:1/1 sws_param: [scale @ 0x248f050] w:640 h:480 fmt:yuvj420p -> w:1920 h:1440 fmt:yuvj420p flags:0x4 [libx264 @ 0x248b2a0] Default settings detected, using medium profile [libx264 @ 0x248b2a0] using SAR=1/1 [libx264 @ 0x248b2a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x248b2a0] profile High, level 5.0 [libx264 @ 0x248b2a0] 264 - core 115 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=15 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'output.mp4': Metadata: title : foo encoder : Lavf52.110.0 Stream #0.0: Video: libx264, yuvj420p, 1920x1440 [PAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 15 tbn, 15 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop, [?] for help [mjpeg @ 0x248bd90] overread 535 56452kB time=00:00:33.86 bitrate=13655.0kbits/s dup=0 drop=8740 frame= 704 fps= 5 q=-1.0 Lsize= 78174kB time=00:00:46.80 bitrate=13683.7kbits/s dup=0 drop=11008 video:78164kB audio:0kB global headers:0kB muxing overhead 0.012308% frame I:11 Avg QP:22.90 size:236916 [libx264 @ 0x248b2a0] frame P:510 Avg QP:24.86 size:121248 [libx264 @ 0x248b2a0] frame B:183 Avg QP:25.38 size: 85228 [libx264 @ 0x248b2a0] consecutive B-frames: 48.7% 49.7% 0.4% 1.1% [libx264 @ 0x248b2a0] mb I I16..4: 4.3% 86.0% 9.7% [libx264 @ 0x248b2a0] mb P I16..4: 5.4% 35.5% 2.0% P16..4: 29.9% 17.2% 7.5% 0.0% 0.0% skip: 2.6% [libx264 @ 0x248b2a0] mb B I16..4: 1.8% 8.6% 0.4% B16..8: 30.0% 14.5% 5.1% direct:23.5% skip:16.2% L0:50.5% L1:33.2% BI:16.3% [libx264 @ 0x248b2a0] 8x8 transform intra:82.6% inter:81.3% [libx264 @ 0x248b2a0] coded y,uvDC,uvAC intra: 71.8% 57.7% 13.5% inter: 70.2% 67.9% 4.0% [libx264 @ 0x248b2a0] i16 v,h,dc,p: 53% 23% 4% 20% [libx264 @ 0x248b2a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 11% 28% 19% 6% 7% 5% 8% 5% 10% [libx264 @ 0x248b2a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 12% 30% 10% 8% 11% 7% 10% 6% 8% [libx264 @ 0x248b2a0] i8c dc,h,v,p: 49% 32% 13% 6% [libx264 @ 0x248b2a0] Weighted P-Frames: Y:46.7% UV:41.6% [libx264 @ 0x248b2a0] ref P L0: 39.4% 17.1% 21.1% 16.0% 6.5% [libx264 @ 0x248b2a0] ref B L0: 59.0% 41.0% [libx264 @ 0x248b2a0] ref B L1: 100.0% 0.0% [libx264 @ 0x248b2a0] kb/s:13643.05 Thanks. From ffmpeg at neoprimitive.net Wed Aug 24 21:44:42 2011 From: ffmpeg at neoprimitive.net (Jesse) Date: Wed, 24 Aug 2011 15:44:42 -0400 Subject: [FFmpeg-user] Avid DNxHD QTs dropped/doubled frame problem Message-ID: <20110824194441.GW7923@neoprimitive.net> Hi, all. I've run in to an issue when importing DNxHD Quicktime movies made with ffmpeg in to Avid Media Composer. The source material are image sequences rendered at 24 FPS. The output format is DNxHD 36 Mb/s, 23.976 FPS. I am doing my encoding with ffmpeg SVN-r20372 packaged with Fedora Core 12. All output movies from ffmpeg play back with the correct number of frames in mplayer on Linux, and in Quicktime on a Mac. However, when the movies are imported in to an Avid Media Composer project set to DNxHD36, 23.976 FPS, movies with frame lengths ending in 0, 3, and 6 (e.g. 10, 13, 16, 20, 23, ...), have a doubled first frame, and a missing last frame. All DNxHD36 Quicktimes created with Quicktime on a Mac maintain the correct order and number of frames when imported in to the Avid, which is why I think my problem originates with ffmpeg. This is an example of my ffmpeg command on Linux: ffmpeg -f image2 -r 24 -i ./1-20/test.%04d.tif -vcodec dnxhd -b 36Mb -r 23.976 -s 1920x1080 ./test.1-20.mov This is the complete terminal output of the run. FFmpeg version SVN-r20372, Copyright (c) 2000-2009 Fabrice Bellard, et al. built on Nov 7 2009 10:57:27 with gcc 4.4.2 20091027 (Red Hat 4.4.2-7) 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='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --extra-version=rpmfusion --enable-bzlib --enable-libdc1394 --enable-libdirac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avfilter-lavf --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 50. 3. 0 / 50. 3. 0 libavcodec 52.37. 1 / 52.37. 1 libavformat 52.39. 2 / 52.39. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1. 4. 1 / 1. 4. 1 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 [image2 @ 0x9735e0]MAX_READ_SIZE:5000000 reached Input #0, image2, from '1-20/test.%04d.tif': Duration: 00:00:00.83, start: 0.000000, bitrate: N/A Stream #0.0: Video: tiff, rgb24, 1920x1080, 24 tbr, 24 tbn, 24 tbc Output #0, mov, to './test.1-20.mov': Stream #0.0: Video: dnxhd, yuv422p, 1920x1080, q=2-31, 36000 kb/s, 2997 tbn, 23.98 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 20 fps= 8 q=1.0 Lsize= 3681kB time=0.83 bitrate=36148.6kbits/s video:3680kB audio:0kB global headers:0kB muxing overhead 0.024573% Does anyone see anything conspicuously wrong with my command line that would cause the behavior I described above? Is it normal to see "2997 tbn, 23.98 tbc" when writing DNxHD Quicktimes? Thanks for reading. -Jesse From tberg at vivox.com Wed Aug 24 22:00:50 2011 From: tberg at vivox.com (Ted Berg) Date: Wed, 24 Aug 2011 16:00:50 -0400 Subject: [FFmpeg-user] Failure building ffmpeg 3.6.0 for i386 on OSX Message-ID: <4E555872.4030003@vivox.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am trying to build ffmpeg 6.3.0 with VP8 support on OSX 10.6.6 with Xcode 3.2.5 for the i386 architecture. The ffmpeg configure phase fails with the error "ERROR: libvpx not found", but the real issue is that libvpx.a is built for i386 only. The ffmpeg build does not appear to support the i386 architecture, instead building only x86_64. Is this the case, or am I doing something wrong? If it is the case, is there a workaround? Ted VPX sources http://code.google.com/p/webm/downloads/detail?name=libvpx-v0.9.7-p1.tar.bz2&can=2&q= VPX/ffmpeg patch http://code.google.com/p/webm/downloads/detail?name=ffmpeg-0.6.3_libvpx-v0.9.5-135-gc28b10a-3.diff.gz&can=2&q= ffmpeg sources http://ffmpeg.org/releases/ffmpeg-0.6.3.tar.bz2 - -- ID: 0x9AAE10A5 Keyserver: pool.sks-keyservers.net Fingerprint: E79C 8FB2 D41D FCA3 410D 3D11 B5BD 5130 9AAE 10A5 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJOVVhtAAoJELW9UTCarhClvqsH/A/G+4m19s9459fPVS2qYpDZ REGa1yYymnlUu3fKS/QPZnOnFmIkm9HeegnF9pmE36OWKW9uodi9pIfLgOMWTRNm fMaiLW6vT+j9qY4EdU8RhoZzjY5MG4i1WOs2p4Z2mLEab6EhxhRg7w4XdWKdyDrq y7mktUOmgFX6jxvYiZGGeSBaYTHpKDFuT9i+CYz9o8fKUD/hO6tF67amnu059j9j 1mD6yjXkd/r+DzUg21azZ+hLHy1ulqcINd706C0nsjy/qP8AgINxtjNeKXtCANsE GQ9uiSkntTEHE7NLXlhJGb2IIgIVBZ9MC8HBOwWF8AcPEQYx/CDToCTaIXAL0hU= =BGoY -----END PGP SIGNATURE----- -------------- next part -------------- A non-text attachment was scrubbed... Name: tberg.vcf Type: text/x-vcard Size: 110 bytes Desc: not available URL: From miker at tippett.com Wed Aug 24 22:35:02 2011 From: miker at tippett.com (Michael Root) Date: Wed, 24 Aug 2011 13:35:02 -0700 Subject: [FFmpeg-user] Avid DNxHD QTs dropped/doubled frame problem In-Reply-To: <20110824194441.GW7923@neoprimitive.net> References: <20110824194441.GW7923@neoprimitive.net> Message-ID: <4E556076.9040606@tippett.com> We had the same problem. This fixed it: --- libavformat/movenc.c 2011-08-24 13:33:37.000000000 -0700 +++ libavformat/movenc.c.orig 2011-08-24 13:33:23.000000000 -0700 @@ -1392,10 +1392,8 @@ avio_wb32(pb, 0); /* size */ ffio_wfourcc(pb, "trak"); mov_write_tkhd_tag(pb, track, st); - if (track->mode == MODE_PSP - || track->flags & MOV_TRACK_CTTS - || track->enc->codec_id == CODEC_ID_DNXHD) - mov_write_edts_tag(pb, track); // DNXHD and PSP Movies require edts box + if (track->mode == MODE_PSP || track->flags & MOV_TRACK_CTTS || track->cluster[0].dts) + mov_write_edts_tag(pb, track); // PSP Movies require edts box if (track->tref_tag) mov_write_tref_tag(pb, track); mov_write_mdia_tag(pb, track); There are still (relatively minor) colorspace problems, though. JFYI. -miker On 08/24/2011 12:44 PM, Jesse wrote: > Hi, all. I've run in to an issue when importing DNxHD Quicktime > movies made with ffmpeg in to Avid Media Composer. > The source material are image sequences rendered at 24 FPS. > > The output format is DNxHD 36 Mb/s, 23.976 FPS. > > I am doing my encoding with ffmpeg SVN-r20372 packaged with Fedora > Core 12. > > All output movies from ffmpeg play back with the correct number of > frames in mplayer on Linux, and in Quicktime on a Mac. > > However, when the movies are imported in to an Avid Media Composer > project set to DNxHD36, 23.976 FPS, movies with frame lengths ending > in 0, 3, and 6 (e.g. 10, 13, 16, 20, 23, ...), have a doubled first > frame, and a missing last frame. > > All DNxHD36 Quicktimes created with Quicktime on a Mac maintain the > correct order and number of frames when imported in to the Avid, which > is why I think my problem originates with ffmpeg. > > > This is an example of my ffmpeg command on Linux: > > ffmpeg -f image2 -r 24 -i ./1-20/test.%04d.tif -vcodec dnxhd -b 36Mb -r 23.976 -s 1920x1080 ./test.1-20.mov > > > This is the complete terminal output of the run. > > FFmpeg version SVN-r20372, Copyright (c) 2000-2009 Fabrice Bellard, et al. > built on Nov 7 2009 10:57:27 with gcc 4.4.2 20091027 (Red Hat 4.4.2-7) > 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='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --extra-version=rpmfusion --enable-bzlib --enable-libdc1394 --enable-libdirac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avfilter-lavf --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect > libavutil 50. 3. 0 / 50. 3. 0 > libavcodec 52.37. 1 / 52.37. 1 > libavformat 52.39. 2 / 52.39. 2 > libavdevice 52. 2. 0 / 52. 2. 0 > libavfilter 1. 4. 1 / 1. 4. 1 > libswscale 0. 7. 1 / 0. 7. 1 > libpostproc 51. 2. 0 / 51. 2. 0 > [image2 @ 0x9735e0]MAX_READ_SIZE:5000000 reached > Input #0, image2, from '1-20/test.%04d.tif': > Duration: 00:00:00.83, start: 0.000000, bitrate: N/A > Stream #0.0: Video: tiff, rgb24, 1920x1080, 24 tbr, 24 tbn, 24 tbc > Output #0, mov, to './test.1-20.mov': > Stream #0.0: Video: dnxhd, yuv422p, 1920x1080, q=2-31, 36000 kb/s, 2997 tbn, 23.98 tbc > Stream mapping: > Stream #0.0 -> #0.0 > Press [q] to stop encoding > frame= 20 fps= 8 q=1.0 Lsize= 3681kB time=0.83 bitrate=36148.6kbits/s > video:3680kB audio:0kB global headers:0kB muxing overhead 0.024573% > > > Does anyone see anything conspicuously wrong with my command line that > would cause the behavior I described above? Is it normal to see > "2997 tbn, 23.98 tbc" when writing DNxHD Quicktimes? > > Thanks for reading. > > -Jesse > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From bouke at editb.nl Wed Aug 24 22:51:37 2011 From: bouke at editb.nl (bouke) Date: Wed, 24 Aug 2011 22:51:37 +0200 Subject: [FFmpeg-user] Avid DNxHD QTs dropped/doubled frame problem References: <20110824194441.GW7923@neoprimitive.net> Message-ID: <066b01cc629f$a1651460$4301a8c0@hpkantoor> ----- Original Message ----- From: "Jesse" To: Sent: Wednesday, August 24, 2011 9:44 PM Subject: [FFmpeg-user] Avid DNxHD QTs dropped/doubled frame problem > Hi, all. I've run in to an issue when importing DNxHD Quicktime > movies made with ffmpeg in to Avid Media Composer. > The source material are image sequences rendered at 24 FPS. > > The output format is DNxHD 36 Mb/s, 23.976 FPS. > > I am doing my encoding with ffmpeg SVN-r20372 packaged with Fedora > Core 12. > > All output movies from ffmpeg play back with the correct number of > frames in mplayer on Linux, and in Quicktime on a Mac. > > However, when the movies are imported in to an Avid Media Composer > project set to DNxHD36, 23.976 FPS, movies with frame lengths ending > in 0, 3, and 6 (e.g. 10, 13, 16, 20, 23, ...), have a doubled first > frame, and a missing last frame. Did you try 'ignoreQTrate True' in the Console in Avid? And/or AMA in instead of import? Might solve something.. (but i use WriteAvidMXF and mediatool the material in, way faster) Bouke From tomfinegan at google.com Thu Aug 25 00:17:59 2011 From: tomfinegan at google.com (Tom Finegan) Date: Wed, 24 Aug 2011 18:17:59 -0400 Subject: [FFmpeg-user] Failure building ffmpeg 3.6.0 for i386 on OSX In-Reply-To: <4E555872.4030003@vivox.com> References: <4E555872.4030003@vivox.com> Message-ID: On Wed, Aug 24, 2011 at 16:00, Ted Berg wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > I am trying to build ffmpeg 6.3.0 with VP8 support on OSX 10.6.6 with > Xcode 3.2.5 for the i386 architecture. > > The ffmpeg configure phase fails with the error "ERROR: libvpx not > found", but the real issue is that libvpx.a is built for i386 only. > The ffmpeg build does not appear to support the i386 architecture, > instead building only x86_64. > > Is this the case, or am I doing something wrong? If it is the case, > is there a workaround? > > Ted > > VPX sources > > http://code.google.com/p/webm/downloads/detail?name=libvpx-v0.9.7-p1.tar.bz2&can=2&q= > > If you really need i386 this isn't going to help... Is there any reason why you cannot rebuild libvpx from sources? I think libvpx will default to the x86_64 target you need if configure is run without arguments. If not, you probably just need: /path/to/libvpx/configure --target=x86_64-darwin10-gcc Then the usual make/sudo make install will work (or tell ffmpeg where to look at configure time, if you don't want to install). Tom > VPX/ffmpeg patch > > http://code.google.com/p/webm/downloads/detail?name=ffmpeg-0.6.3_libvpx-v0.9.5-135-gc28b10a-3.diff.gz&can=2&q= > > ffmpeg sources > http://ffmpeg.org/releases/ffmpeg-0.6.3.tar.bz2 > > - -- > ID: 0x9AAE10A5 Keyserver: pool.sks-keyservers.net > Fingerprint: E79C 8FB2 D41D FCA3 410D 3D11 B5BD 5130 9AAE 10A5 > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.11 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > iQEcBAEBAgAGBQJOVVhtAAoJELW9UTCarhClvqsH/A/G+4m19s9459fPVS2qYpDZ > REGa1yYymnlUu3fKS/QPZnOnFmIkm9HeegnF9pmE36OWKW9uodi9pIfLgOMWTRNm > fMaiLW6vT+j9qY4EdU8RhoZzjY5MG4i1WOs2p4Z2mLEab6EhxhRg7w4XdWKdyDrq > y7mktUOmgFX6jxvYiZGGeSBaYTHpKDFuT9i+CYz9o8fKUD/hO6tF67amnu059j9j > 1mD6yjXkd/r+DzUg21azZ+hLHy1ulqcINd706C0nsjy/qP8AgINxtjNeKXtCANsE > GQ9uiSkntTEHE7NLXlhJGb2IIgIVBZ9MC8HBOwWF8AcPEQYx/CDToCTaIXAL0hU= > =BGoY > -----END PGP SIGNATURE----- > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > From eric.hollis at gmail.com Thu Aug 25 01:13:41 2011 From: eric.hollis at gmail.com (Eric Hollis) Date: Wed, 24 Aug 2011 19:13:41 -0400 Subject: [FFmpeg-user] How to set video size when using -f dshow? Message-ID: Hi. When running the command line below, I get "Option video_size not found." I've tried -video_size hd720, with no effect. How do you set the video size when using dshow? Thanks for any help you can provide. ffmpeg -rtbufsize 100000000 -f dshow -s hd720 -i video="Microsoft LifeCam Cinema" -vframes 1 test%06d.tif Eric From rogerdpack2 at gmail.com Thu Aug 25 01:26:55 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 24 Aug 2011 17:26:55 -0600 Subject: [FFmpeg-user] How to set video size when using -f dshow? In-Reply-To: References: Message-ID: > When running the command line below, I get "Option video_size not found." > I've tried -video_size hd720, with no effect. > > How do you set the video size when using dshow? Thanks for any help you can > provide. > > ffmpeg -rtbufsize 100000000 -f dshow -s hd720 -i video="Microsoft LifeCam > Cinema" -vframes 1 test%06d.tif You can't yet it's not implemented, I believe (just looking at it, I think maybe it's using DirectConnect to connect the pins, which just always choose some default). My current work around is to (are you ready for this?) create a graphviz file with one box as your LifeCam, then add a "Tee" box and connect the two. Then right click on your output pin coming from the LifeCam, and set the resolution. Save the graphviz file. Now create an avisynth file like: DirectShowSource("tee.grf", fps=2.0, audio=False, framecount=1000000) (from https://github.com/rdp/dirt-simple-usb-surveillance/blob/master/tee.avs ) Then use ffmpeg on that file as input. (you have to install avisynth as well, obviously). Phew! I'd be willing to offer a bounty for somebody to implement it so that you can select the size and the fps though, that would be sweet and avoid this work around. -roger- From rogerdpack2 at gmail.com Thu Aug 25 01:27:31 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 24 Aug 2011 17:27:31 -0600 Subject: [FFmpeg-user] Failure building ffmpeg 3.6.0 for i386 on OSX In-Reply-To: <4E555872.4030003@vivox.com> References: <4E555872.4030003@vivox.com> Message-ID: > Is this the case, or am I doing something wrong? ?If it is the case, > is there a workaround? macports? -roger- From rickcorteza at gmail.com Thu Aug 25 03:26:53 2011 From: rickcorteza at gmail.com (Rick C.) Date: Thu, 25 Aug 2011 09:26:53 +0800 Subject: [FFmpeg-user] subtitles again please In-Reply-To: <20110824180036.GB8530@geppetto> References: <20110824180036.GB8530@geppetto> Message-ID: <14C1EFC6-D278-406C-B58D-78F8818C84EA@gmail.com> On Aug 25, 2011, at 2:00 AM, Stefano Sabatini wrote: > On date Wednesday 2011-08-24 10:34:11 +0800, Rick C. encoded: >> Hi again, >> >> I have asked in the past about hard-coding subtitles, but if I just >> want to add a subtitle track as a stream would someone mind giving >> me a working example? I'm aware of the documentation which says >> this: >> >> ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt >> >> And in FFmpeg -formats it says it supports decoding/encoding of .srt >> files. So how would I take a file without a subtitle stream and add >> a new one? Something like: >> >> ffmpeg -i mov.avi -i sub.srt output.avi >> >> Sorry to ask again but I'm assuming since it's there in the >> documentation it works but I really can't figure it out. Thanks! > > Not very experienced about subtitles and all, AFAIK subtitles support > in FFmpeg is still a bit... rudimentary. > > But if you want subtitles in output you need a format which supports > them, I know matroska (.mkv) should do. > > As for hardcoding subtitles right onto the video, I'm afraid we still > miss that facility (subtitles filtering + video overlaying seems the > right solution), correct if I'm wrong. Thanks for the reply. So what would be a working command line if the output was to .mkv for example? From sadams at iii.com Thu Aug 25 00:43:29 2011 From: sadams at iii.com (Stephanie Adams) Date: Wed, 24 Aug 2011 15:43:29 -0700 Subject: [FFmpeg-user] problem configuring FFmpeg for solaris Message-ID: <010301cc62af$51586c50$f40944f0$@com> Hi I am getting the following errors: iii at iiics (solaris10-r2009B_1.0) 45 > !22 bash ./configure --prefix=/opt/ffmpeg-SVN-r15797 --extra-cflags="-fPIC" --disable-mmx --disable-protocol=udp --disable-encoder=nellymoser ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ./configure: line 2012: 5185 Broken Pipe $cc -v 2>&1 5186 Killed | grep -q '^gcc.*LLVM' ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ./configure: line 2012: 5187 Exit 126 $cc -v 2>&1 5188 Killed | grep -qi ^gcc ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ./configure: line 2012: 5189 Exit 126 $cc --version 2>/dev/null 5190 Killed | grep -q Intel ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ./configure: line 2012: 5191 Exit 126 $cc -v 2>&1 5192 Killed | grep -q xlc ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ./configure: line 2012: 5193 Exit 126 $cc -V 2>/dev/null 5194 Killed | grep -q Compaq ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ./configure: line 2012: 5195 Exit 126 $cc --vsn 2>/dev/null 5196 Killed | grep -q "ARM C/C++ Compiler" ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ./configure: line 2012: 5197 Exit 126 $cc -version 2>/dev/null 5198 Killed | grep -q TMS470 ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory ./configure: line 2012: 5199 Exit 126 $cc -v 2>&1 5200 Killed | grep -q clang Can you help ? Do I need to place ld.so and libintl.so somewhere? If so, where? Thank you for your help --Stephanie Adams-- Release Engineer From dmitry at interhost.co.il Thu Aug 25 08:50:33 2011 From: dmitry at interhost.co.il (Dmitry Sherman) Date: Thu, 25 Aug 2011 09:50:33 +0300 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 In-Reply-To: References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> Message-ID: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC140@exchange.exchange.interhost.co.il> Hi, thanks for answer, I am using latest ffmpeg git, I cannot patch, I get an error: patching file '.\ffmpeg-r26400-swscale-r32676_fixed\libavformat\rtsp.c' Hunk #1 FAILED at 706. 1 out of 1 hunk FAILED -- saving rejects to file '.\ffmpeg-r26400-swscale-r32676_fixed\libavformat\rtsp.c.rej' Thanks. Dmitry Sherman dmitry at interhost.co.il Interhost Networks Ltd t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ http://facebook.com/InterhostIL -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain Sent: Wednesday, August 24, 2011 8:25 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 Does this http://ffmpeg.org/trac/ffmpeg/ticket/16 help you ? On Wed, Aug 24, 2011 at 12:58 PM, Dmitry Sherman wrote: > Hello, > I receive sometimes an error when I try to transcode some mp4 files to > flv using this command: > ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv > > I receive this error: > Output #0, flv, to 'a.flv': > Metadata: > major_brand : mp42 > minor_version : 0 > compatible_brands: mp42isomavc1 > creation_time : 2011-07-05 16:50:32 > encoder : Lavf53.9.0 > Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, > 416x224 [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc > Metadata: > creation_time : 2011-07-05 16:50:32 > Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, > stereo, s16, 64 kb/s > Metadata: > creation_time : 2011-07-05 16:50:32 > Stream mapping: > Stream #0.0 -> #0.0: copy > Stream #0.1 -> #0.1: aac -> libfaac > Press [q] to stop, [?] for help > [flv @ 0x2bf0da0] Application provided invalid, non monotonically > increasing dts to muxer in stream 0: 160 >= 160 > av_interleaved_write_frame(): Invalid argument > > what can I do to solve this problem? > > Thank you. > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Antoine Baqain System Administration Lead Maktoob.com Inc. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. From dmitry at interhost.co.il Thu Aug 25 08:52:43 2011 From: dmitry at interhost.co.il (Dmitry Sherman) Date: Thu, 25 Aug 2011 09:52:43 +0300 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 In-Reply-To: References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> Message-ID: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC141@exchange.exchange.interhost.co.il> I found that I have already this patch applied. Dmitry Sherman dmitry at interhost.co.il Interhost Networks Ltd t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ http://facebook.com/InterhostIL -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain Sent: Wednesday, August 24, 2011 8:25 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 Does this http://ffmpeg.org/trac/ffmpeg/ticket/16 help you ? On Wed, Aug 24, 2011 at 12:58 PM, Dmitry Sherman wrote: > Hello, > I receive sometimes an error when I try to transcode some mp4 files to > flv using this command: > ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv > > I receive this error: > Output #0, flv, to 'a.flv': > Metadata: > major_brand : mp42 > minor_version : 0 > compatible_brands: mp42isomavc1 > creation_time : 2011-07-05 16:50:32 > encoder : Lavf53.9.0 > Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, > 416x224 [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc > Metadata: > creation_time : 2011-07-05 16:50:32 > Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, > stereo, s16, 64 kb/s > Metadata: > creation_time : 2011-07-05 16:50:32 > Stream mapping: > Stream #0.0 -> #0.0: copy > Stream #0.1 -> #0.1: aac -> libfaac > Press [q] to stop, [?] for help > [flv @ 0x2bf0da0] Application provided invalid, non monotonically > increasing dts to muxer in stream 0: 160 >= 160 > av_interleaved_write_frame(): Invalid argument > > what can I do to solve this problem? > > Thank you. > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Antoine Baqain System Administration Lead Maktoob.com Inc. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. From stefano.sabatini-lala at poste.it Thu Aug 25 13:55:32 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Thu, 25 Aug 2011 13:55:32 +0200 Subject: [FFmpeg-user] subtitles again please In-Reply-To: <14C1EFC6-D278-406C-B58D-78F8818C84EA@gmail.com> References: <20110824180036.GB8530@geppetto> <14C1EFC6-D278-406C-B58D-78F8818C84EA@gmail.com> Message-ID: <20110825115532.GA7493@geppetto> On date Thursday 2011-08-25 09:26:53 +0800, Rick C. encoded: > > On Aug 25, 2011, at 2:00 AM, Stefano Sabatini wrote: > > > On date Wednesday 2011-08-24 10:34:11 +0800, Rick C. encoded: > >> Hi again, > >> > >> I have asked in the past about hard-coding subtitles, but if I just > >> want to add a subtitle track as a stream would someone mind giving > >> me a working example? I'm aware of the documentation which says > >> this: > >> > >> ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt > >> > >> And in FFmpeg -formats it says it supports decoding/encoding of .srt > >> files. So how would I take a file without a subtitle stream and add > >> a new one? Something like: > >> > >> ffmpeg -i mov.avi -i sub.srt output.avi > >> > >> Sorry to ask again but I'm assuming since it's there in the > >> documentation it works but I really can't figure it out. Thanks! > > > > Not very experienced about subtitles and all, AFAIK subtitles support > > in FFmpeg is still a bit... rudimentary. > > > > But if you want subtitles in output you need a format which supports > > them, I know matroska (.mkv) should do. > > > > As for hardcoding subtitles right onto the video, I'm afraid we still > > miss that facility (subtitles filtering + video overlaying seems the > > right solution), correct if I'm wrong. > > > Thanks for the reply. So what would be a working command line if the output was to .mkv for example? $ ffmpeg -i INPUT -i SUBTITLE_FILE -y OUTPUT then you should see something like this: Stream #0.0 -> #0.0: mpeg4 -> libx264 Stream #0.1 -> #0.1: mp3 -> libvorbis Stream #1.0 -> #0.2: srt -> ass You can set the subtitle output format to srt by using -scodec srt, or forcing the copy with -scodec copy. -- ffmpeg-user random tip #5 FFmpeg documentation: http://www.ffmpeg.org/documentation.html From stephen.martin.wilson at gmail.com Thu Aug 25 16:04:17 2011 From: stephen.martin.wilson at gmail.com (Stephen Wilson) Date: Thu, 25 Aug 2011 15:04:17 +0100 Subject: [FFmpeg-user] Bits per pixel encoding Message-ID: Is it possible to use FFMpeg to encode a video at a different bits per pixel level than the original? Thanks Stephen From luj125 at gmail.com Thu Aug 25 16:22:53 2011 From: luj125 at gmail.com (James Lu) Date: Thu, 25 Aug 2011 10:22:53 -0400 Subject: [FFmpeg-user] Bits per pixel encoding In-Reply-To: References: Message-ID: On Thu, Aug 25, 2011 at 10:04 AM, Stephen Wilson < stephen.martin.wilson at gmail.com> wrote: > Is it possible to use FFMpeg to encode a video at a different bits per > pixel > level than the original? > > Thanks > > Stephen > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Hi Stephen, The only information I have is regarding colorspace conversion such as from YUV444 to YUV420p, this would reduce bits per pixel by reducing chroma subsampling. This is possible through -vf format=yuv420p I believe yuv420p is 1.5 bytes per pixel If you're talking about going from truecolor color to other palettes such as 256-color, I don't see anything about that initially in libavfilter, but would suggest looking for something like that in frei0r filters? Either way these options will only work with a codec that supports different colorspaces and color palettes, which, I'm not sure exactly which ones do. Hope this helps, ~James From dmitry at interhost.co.il Thu Aug 25 16:29:05 2011 From: dmitry at interhost.co.il (Dmitry Sherman) Date: Thu, 25 Aug 2011 17:29:05 +0300 Subject: [FFmpeg-user] [buffer @ 0x17db380] Invalid pixel format '-1' Message-ID: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC14E@exchange.exchange.interhost.co.il> Hello, I receive another new error when trying to extract images from video: ffmpeg -ss 00:01:01 -i 41167.flv -vframes 1 -f image2 4.jpg I get: [buffer @ 0x17db380] Invalid pixel format '-1' Error opening filters! Full output: [buffer @ 0x17db380] Invalid pixel format '-1' Error opening filters! Thanks. Dmitry Sherman dmitry at interhost.co.il Interhost Networks Ltd t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ http://facebook.com/InterhostIL From tonybaqain at gmail.com Thu Aug 25 16:40:34 2011 From: tonybaqain at gmail.com (Antoine Baqain) Date: Thu, 25 Aug 2011 17:40:34 +0300 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 In-Reply-To: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC141@exchange.exchange.interhost.co.il> References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC141@exchange.exchange.interhost.co.il> Message-ID: As far as I know, ffmpeg stopped using GIT, can you try SVN to head and compile ? seems like an outdated or unknowably buggy one you have. On Thu, Aug 25, 2011 at 9:52 AM, Dmitry Sherman wrote: > I found that I have already this patch applied. > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 > http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto: > ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain > Sent: Wednesday, August 24, 2011 8:25 PM > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, > non monotonically increasing dts to muxer in stream 0: 160 >= 160 > > Does this http://ffmpeg.org/trac/ffmpeg/ticket/16 help you ? > > On Wed, Aug 24, 2011 at 12:58 PM, Dmitry Sherman >wrote: > > > Hello, > > I receive sometimes an error when I try to transcode some mp4 files to > > flv using this command: > > ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv > > > > I receive this error: > > Output #0, flv, to 'a.flv': > > Metadata: > > major_brand : mp42 > > minor_version : 0 > > compatible_brands: mp42isomavc1 > > creation_time : 2011-07-05 16:50:32 > > encoder : Lavf53.9.0 > > Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, > > 416x224 [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc > > Metadata: > > creation_time : 2011-07-05 16:50:32 > > Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, > > stereo, s16, 64 kb/s > > Metadata: > > creation_time : 2011-07-05 16:50:32 > > Stream mapping: > > Stream #0.0 -> #0.0: copy > > Stream #0.1 -> #0.1: aac -> libfaac > > Press [q] to stop, [?] for help > > [flv @ 0x2bf0da0] Application provided invalid, non monotonically > > increasing dts to muxer in stream 0: 160 >= 160 > > av_interleaved_write_frame(): Invalid argument > > > > what can I do to solve this problem? > > > > Thank you. > > > > Dmitry Sherman > > dmitry at interhost.co.il > > Interhost Networks Ltd > > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > > http://facebook.com/InterhostIL > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > -- > Antoine Baqain > System Administration Lead > Maktoob.com Inc. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Antoine Baqain System Administration Lead Maktoob.com Inc. From dmitry at interhost.co.il Thu Aug 25 16:44:08 2011 From: dmitry at interhost.co.il (Dmitry Sherman) Date: Thu, 25 Aug 2011 17:44:08 +0300 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 In-Reply-To: References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC141@exchange.exchange.interhost.co.il> Message-ID: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC14F@exchange.exchange.interhost.co.il> Thanks, But this page states that ffmpeg developed with GIT: http://ffmpeg.org/download.html Dmitry Sherman dmitry at interhost.co.il Interhost Networks Ltd t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ http://facebook.com/InterhostIL -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain Sent: Thursday, August 25, 2011 5:41 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 As far as I know, ffmpeg stopped using GIT, can you try SVN to head and compile ? seems like an outdated or unknowably buggy one you have. On Thu, Aug 25, 2011 at 9:52 AM, Dmitry Sherman wrote: > I found that I have already this patch applied. > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto: > ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain > Sent: Wednesday, August 24, 2011 8:25 PM > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided > invalid, non monotonically increasing dts to muxer in stream 0: 160 >= > 160 > > Does this http://ffmpeg.org/trac/ffmpeg/ticket/16 help you ? > > On Wed, Aug 24, 2011 at 12:58 PM, Dmitry Sherman > >wrote: > > > Hello, > > I receive sometimes an error when I try to transcode some mp4 files > > to flv using this command: > > ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv > > > > I receive this error: > > Output #0, flv, to 'a.flv': > > Metadata: > > major_brand : mp42 > > minor_version : 0 > > compatible_brands: mp42isomavc1 > > creation_time : 2011-07-05 16:50:32 > > encoder : Lavf53.9.0 > > Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, > > 416x224 [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc > > Metadata: > > creation_time : 2011-07-05 16:50:32 > > Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, > > stereo, s16, 64 kb/s > > Metadata: > > creation_time : 2011-07-05 16:50:32 > > Stream mapping: > > Stream #0.0 -> #0.0: copy > > Stream #0.1 -> #0.1: aac -> libfaac Press [q] to stop, [?] for help > > [flv @ 0x2bf0da0] Application provided invalid, non monotonically > > increasing dts to muxer in stream 0: 160 >= 160 > > av_interleaved_write_frame(): Invalid argument > > > > what can I do to solve this problem? > > > > Thank you. > > > > Dmitry Sherman > > dmitry at interhost.co.il > > Interhost Networks Ltd > > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > > http://facebook.com/InterhostIL > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > -- > Antoine Baqain > System Administration Lead > Maktoob.com Inc. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Antoine Baqain System Administration Lead Maktoob.com Inc. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. From dmitry at interhost.co.il Thu Aug 25 16:44:31 2011 From: dmitry at interhost.co.il (Dmitry Sherman) Date: Thu, 25 Aug 2011 17:44:31 +0300 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 In-Reply-To: References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC141@exchange.exchange.interhost.co.il> Message-ID: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC150@exchange.exchange.interhost.co.il> This is what said on ffmpeg download page: FFmpeg development has moved to a Git repository, and the SVN repository is no longer updated. The last revision committed to SVN was r26402 on 2011-01-19. The SVN repository may be removed in a near future, so you're recommended to use the Git repository instead. Dmitry Sherman dmitry at interhost.co.il Interhost Networks Ltd t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ http://facebook.com/InterhostIL -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain Sent: Thursday, August 25, 2011 5:41 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 As far as I know, ffmpeg stopped using GIT, can you try SVN to head and compile ? seems like an outdated or unknowably buggy one you have. On Thu, Aug 25, 2011 at 9:52 AM, Dmitry Sherman wrote: > I found that I have already this patch applied. > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto: > ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain > Sent: Wednesday, August 24, 2011 8:25 PM > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided > invalid, non monotonically increasing dts to muxer in stream 0: 160 >= > 160 > > Does this http://ffmpeg.org/trac/ffmpeg/ticket/16 help you ? > > On Wed, Aug 24, 2011 at 12:58 PM, Dmitry Sherman > >wrote: > > > Hello, > > I receive sometimes an error when I try to transcode some mp4 files > > to flv using this command: > > ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv > > > > I receive this error: > > Output #0, flv, to 'a.flv': > > Metadata: > > major_brand : mp42 > > minor_version : 0 > > compatible_brands: mp42isomavc1 > > creation_time : 2011-07-05 16:50:32 > > encoder : Lavf53.9.0 > > Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, > > 416x224 [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc > > Metadata: > > creation_time : 2011-07-05 16:50:32 > > Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, > > stereo, s16, 64 kb/s > > Metadata: > > creation_time : 2011-07-05 16:50:32 > > Stream mapping: > > Stream #0.0 -> #0.0: copy > > Stream #0.1 -> #0.1: aac -> libfaac Press [q] to stop, [?] for help > > [flv @ 0x2bf0da0] Application provided invalid, non monotonically > > increasing dts to muxer in stream 0: 160 >= 160 > > av_interleaved_write_frame(): Invalid argument > > > > what can I do to solve this problem? > > > > Thank you. > > > > Dmitry Sherman > > dmitry at interhost.co.il > > Interhost Networks Ltd > > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > > http://facebook.com/InterhostIL > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > -- > Antoine Baqain > System Administration Lead > Maktoob.com Inc. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Antoine Baqain System Administration Lead Maktoob.com Inc. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. From tonybaqain at gmail.com Thu Aug 25 16:49:34 2011 From: tonybaqain at gmail.com (Antoine Baqain) Date: Thu, 25 Aug 2011 17:49:34 +0300 Subject: [FFmpeg-user] problem configuring FFmpeg for solaris In-Reply-To: <010301cc62af$51586c50$f40944f0$@com> References: <010301cc62af$51586c50$f40944f0$@com> Message-ID: This is a problem with your Solaris actually, you can do : 1. try searching for the libintl like *root at mytest: find / -name libintl.so.3 ?ls *and if you find any instance, link it with *ln -s* to * /usr/lib* and try again, Else 2. you have to install *libintl-3.4.0-sol8-sparc-local* for the missing libintl.so.3 On Thu, Aug 25, 2011 at 1:43 AM, Stephanie Adams wrote: > Hi > > > > I am getting the following errors: > > iii at iiics (solaris10-r2009B_1.0) 45 > !22 > > bash ./configure --prefix=/opt/ffmpeg-SVN-r15797 --extra-cflags="-fPIC" > --disable-mmx --disable-protocol=udp --disable-encoder=nellymoser > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ./configure: line 2012: 5185 Broken Pipe $cc -v 2>&1 > > 5186 Killed | grep -q '^gcc.*LLVM' > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ./configure: line 2012: 5187 Exit 126 $cc -v 2>&1 > > 5188 Killed | grep -qi ^gcc > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ./configure: line 2012: 5189 Exit 126 $cc --version > 2>/dev/null > > 5190 Killed | grep -q Intel > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ./configure: line 2012: 5191 Exit 126 $cc -v 2>&1 > > 5192 Killed | grep -q xlc > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ./configure: line 2012: 5193 Exit 126 $cc -V 2>/dev/null > > 5194 Killed | grep -q Compaq > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ./configure: line 2012: 5195 Exit 126 $cc --vsn 2>/dev/null > > 5196 Killed | grep -q "ARM C/C++ Compiler" > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ./configure: line 2012: 5197 Exit 126 $cc -version > 2>/dev/null > > 5198 Killed | grep -q TMS470 > > ld.so.1: grep: fatal: libintl.so.3: open failed: No such file or directory > > ./configure: line 2012: 5199 Exit 126 $cc -v 2>&1 > > 5200 Killed | grep -q clang > > > > Can you help ? Do I need to place ld.so and libintl.so somewhere? If > so, where? > > > > Thank you for your help > > > > --Stephanie Adams-- > > Release Engineer > > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Antoine Baqain System Administration Lead Maktoob.com Inc. From tonybaqain at gmail.com Thu Aug 25 16:52:53 2011 From: tonybaqain at gmail.com (Antoine Baqain) Date: Thu, 25 Aug 2011 17:52:53 +0300 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 In-Reply-To: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC150@exchange.exchange.interhost.co.il> References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC141@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC150@exchange.exchange.interhost.co.il> Message-ID: Sorry I wrote it in the opposite way. Was pushing forward an update from your side, but it seems you have the latest, I would say for my last help to try to test the command using the normal stable ffmpeg tgz you download, to see if that fixes the issue. Otherwise, I'm sorry, out of answers for your issue. On Thu, Aug 25, 2011 at 5:44 PM, Dmitry Sherman wrote: > This is what said on ffmpeg download page: > FFmpeg development has moved to a Git repository, and the SVN repository is > no longer updated. The last revision committed to SVN was r26402 on > 2011-01-19. The SVN repository may be removed in a near future, so you're > recommended to use the Git repository instead. > > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 > http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto: > ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain > Sent: Thursday, August 25, 2011 5:41 PM > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, > non monotonically increasing dts to muxer in stream 0: 160 >= 160 > > As far as I know, ffmpeg stopped using GIT, can you try SVN to head and > compile ? seems like an outdated or unknowably buggy one you have. > > On Thu, Aug 25, 2011 at 9:52 AM, Dmitry Sherman >wrote: > > > I found that I have already this patch applied. > > > > Dmitry Sherman > > dmitry at interhost.co.il > > Interhost Networks Ltd > > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > > http://facebook.com/InterhostIL > > > > > > -----Original Message----- > > From: ffmpeg-user-bounces at ffmpeg.org [mailto: > > ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain > > Sent: Wednesday, August 24, 2011 8:25 PM > > To: FFmpeg user questions and RTFMs > > Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided > > invalid, non monotonically increasing dts to muxer in stream 0: 160 >= > > 160 > > > > Does this http://ffmpeg.org/trac/ffmpeg/ticket/16 help you ? > > > > On Wed, Aug 24, 2011 at 12:58 PM, Dmitry Sherman > > > >wrote: > > > > > Hello, > > > I receive sometimes an error when I try to transcode some mp4 files > > > to flv using this command: > > > ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv > > > > > > I receive this error: > > > Output #0, flv, to 'a.flv': > > > Metadata: > > > major_brand : mp42 > > > minor_version : 0 > > > compatible_brands: mp42isomavc1 > > > creation_time : 2011-07-05 16:50:32 > > > encoder : Lavf53.9.0 > > > Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, > > > 416x224 [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc > > > Metadata: > > > creation_time : 2011-07-05 16:50:32 > > > Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, > > > stereo, s16, 64 kb/s > > > Metadata: > > > creation_time : 2011-07-05 16:50:32 > > > Stream mapping: > > > Stream #0.0 -> #0.0: copy > > > Stream #0.1 -> #0.1: aac -> libfaac Press [q] to stop, [?] for help > > > [flv @ 0x2bf0da0] Application provided invalid, non monotonically > > > increasing dts to muxer in stream 0: 160 >= 160 > > > av_interleaved_write_frame(): Invalid argument > > > > > > what can I do to solve this problem? > > > > > > Thank you. > > > > > > Dmitry Sherman > > > dmitry at interhost.co.il > > > Interhost Networks Ltd > > > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > > > http://facebook.com/InterhostIL > > > > > > _______________________________________________ > > > ffmpeg-user mailing list > > > ffmpeg-user at ffmpeg.org > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > > > > > > -- > > Antoine Baqain > > System Administration Lead > > Maktoob.com Inc. > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > -- > > This message has been scanned for viruses and dangerous content by > > MailScanner, and is believed to be clean. > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > -- > Antoine Baqain > System Administration Lead > Maktoob.com Inc. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Antoine Baqain System Administration Lead Maktoob.com Inc. From holger.bause at gmail.com Thu Aug 25 18:04:03 2011 From: holger.bause at gmail.com (Holger Bause) Date: Thu, 25 Aug 2011 11:04:03 -0500 Subject: [FFmpeg-user] read rtmp live stream delay Message-ID: Hey Guys, I'm trying to load a live stream from red5 and save it to a file. Using the following is resulting a very long delay (5 - 10 minutes) before it start to read the stream: ffmpeg -i "rtmp://192.168.1.88/test/newTest test.flv (I can see in the red5 logs that it connects and the stream starts) if I use the live option, it connects and but the stream never plays: ffmpeg -i "rtmp://192.168.1.88/test/newTest live=1" test.flv Loading a static stream from red5 works just dandy. Any suggestions would be appreciated. Thanks! HB From tim.nicholson at bbc.co.uk Thu Aug 25 18:28:25 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Thu, 25 Aug 2011 17:28:25 +0100 Subject: [FFmpeg-user] Static compile with embeded 3rd party libraries. Message-ID: <4E567829.1050508@bbc.co.uk> Having been building ffmpeg statically for a while to avoid conflicts with distro versions by using:- "... --enable-static --disable-shared ..." I want to go one step further and end up with a portable executable that also includes all the third party libs such as lame faac etc. Googling around suggests adding in:- "--extra-libs=-static --extra-cflags=--static" However when I do this configure fails with an error saying it cannot find the third party libs. "gcc -o /tmp/ffconf.kNbWfR0S /tmp/ffconf.htm52E9X.o -lfaac -lm -pthread -static /usr/lib64/gcc/x86_64-suse-linux/4.5/../../../../x86_64-suse-linux/bin/ld: cannot find -lfaac collect2: ld returned 1 exit status ERROR: libfaac not found" I am obviously missing a bit, as it links to them fine without the extra two options. Can anyone assist in pointing me in the right direction? -- 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 james.darnley at gmail.com Thu Aug 25 20:01:03 2011 From: james.darnley at gmail.com (James Darnley) Date: Thu, 25 Aug 2011 19:01:03 +0100 Subject: [FFmpeg-user] Static compile with embeded 3rd party libraries. In-Reply-To: <4E567829.1050508@bbc.co.uk> References: <4E567829.1050508@bbc.co.uk> Message-ID: On 25/08/2011, Tim Nicholson wrote: > Having been building ffmpeg statically for a while to avoid conflicts > with distro versions by using:- > > "... --enable-static --disable-shared ..." > > I want to go one step further and end up with a portable executable that > also includes all the third party libs such as lame faac etc. Googling > around suggests adding in:- > > "--extra-libs=-static --extra-cflags=--static" > > However when I do this configure fails with an error saying it cannot > find the third party libs. > > "gcc -o /tmp/ffconf.kNbWfR0S /tmp/ffconf.htm52E9X.o -lfaac -lm -pthread > -static > /usr/lib64/gcc/x86_64-suse-linux/4.5/../../../../x86_64-suse-linux/bin/ld: > cannot find -lfaac > collect2: ld returned 1 exit status > ERROR: libfaac not found" > > I am obviously missing a bit, as it links to them fine without the extra > two options. Can anyone assist in pointing me in the right direction? Do you have a static libfaac? From pgoldweic at northwestern.edu Thu Aug 25 20:34:21 2011 From: pgoldweic at northwestern.edu (Patricia N Goldweic) Date: Thu, 25 Aug 2011 18:34:21 +0000 Subject: [FFmpeg-user] problem compiling ffmpeg 0.8.2 Message-ID: <316ABB7EE42BC448BDBDE73B4323B09B1DFFF585@evcspmbx4.ads.northwestern.edu> Hi, I've been trying to compile ffmpeg 0.8.2 on linux (rel6), and after compiling all the third party libraries I needed, I am getting the following error: .... libavutil/samplefmt.c: In function ?av_samples_fill_arrays?: libavutil/samplefmt.c:88: warning: ?av_get_bits_per_sample_fmt? is deprecated (declared at libavutil/samplefmt.c:76) CC libavutil/sha.o CC libavutil/tree.o CC libavutil/utils.o CC libavutil/x86/cpu.o LD libavutil/libavutil.so.51 LD libavcodec/libavcodec.so.53 /usr/bin/ld: /usr/local/lib/libvpx.a(vpx_codec.c.o): relocation R_X86_64_32 against `.rodata.str1.1' can not be used when making a shared object; recompile with -fPIC /usr/local/lib/libvpx.a: could not read symbols: Bad value collect2: ld returned 1 exit status make: *** [libavcodec/libavcodec.so.53] Error 1 I've configured ffmpeg with the following command: sudo ./configure --enablelibvpx --enable-libfaac --enable-shared --enable-memalign-hack --enable-gpl --enable-libtheora --enable-libmp3lame --enable-libvorbis --enable-libx264 --enable-libxvid --enable-nonfree --enable-postproc --enable-avfilter --enable-swscale --enable-pthreads --enable-libfreetype --enable-libopenjpeg --arch=x86_64 (and I've installed version 0.9.7 of libvpx) Any ideas on what is happening or how to fix it? Thanks in advance, -Patricia From lou at lrcd.com Thu Aug 25 20:55:59 2011 From: lou at lrcd.com (Lou) Date: Thu, 25 Aug 2011 10:55:59 -0800 Subject: [FFmpeg-user] problem configuring FFmpeg for solaris In-Reply-To: <010301cc62af$51586c50$f40944f0$@com> References: <010301cc62af$51586c50$f40944f0$@com> Message-ID: <20110825105559.1b0eadc5@lrcd.com> On Wed, 24 Aug 2011 15:43:29 -0700 "Stephanie Adams" wrote: > Hi > > > > I am getting the following errors: > > iii at iiics (solaris10-r2009B_1.0) 45 > !22 > > bash ./configure --prefix=/opt/ffmpeg-SVN-r15797 Are you really using r15797? That's from 2008-11-09 and is considered absolutely ancient. Why not at least attempt to use a recent FFmpeg release (or Git master)? From lou at lrcd.com Thu Aug 25 21:10:23 2011 From: lou at lrcd.com (Lou) Date: Thu, 25 Aug 2011 11:10:23 -0800 Subject: [FFmpeg-user] problem compiling ffmpeg 0.8.2 In-Reply-To: <316ABB7EE42BC448BDBDE73B4323B09B1DFFF585@evcspmbx4.ads.northwestern.edu> References: <316ABB7EE42BC448BDBDE73B4323B09B1DFFF585@evcspmbx4.ads.northwestern.edu> Message-ID: <20110825111023.6cc6dbb5@lrcd.com> On Thu, 25 Aug 2011 18:34:21 +0000 Patricia N Goldweic wrote: > Hi, > I've been trying to compile ffmpeg 0.8.2 on linux (rel6), and after > compiling all the third party libraries I needed, I am getting the > following error: > > > .... > libavutil/samplefmt.c: In function ?av_samples_fill_arrays?: > libavutil/samplefmt.c:88: warning: ?av_get_bits_per_sample_fmt? is > deprecated (declared at libavutil/samplefmt.c:76) CC > libavutil/sha.o CC libavutil/tree.o > CC libavutil/utils.o > CC libavutil/x86/cpu.o > LD libavutil/libavutil.so.51 > LD libavcodec/libavcodec.so.53 > /usr/bin/ld: /usr/local/lib/libvpx.a(vpx_codec.c.o): relocation > R_X86_64_32 against `.rodata.str1.1' can not be used when making a > shared object; recompile with -fPIC /usr/local/lib/libvpx.a: could > not read symbols: Bad value collect2: ld returned 1 exit status make: > *** [libavcodec/libavcodec.so.53] Error 1 > > > > I've configured ffmpeg with the following command: > > sudo ./configure --enablelibvpx --enable-libfaac --enable-shared > --enable-memalign-hack --enable-gpl --enable-libtheora > --enable-libmp3lame --enable-libvorbis --enable-libx264 > --enable-libxvid --enable-nonfree --enable-postproc --enable-avfilter > --enable-swscale --enable-pthreads --enable-libfreetype > --enable-libopenjpeg --arch=x86_64 > > (and I've installed version 0.9.7 of libvpx) > > Any ideas on what is happening or how to fix it? Thanks in advance, > -Patricia What was your ./configure for libvpx? If you compile ffmpeg with --enable-shared, then you probably should also compile libvpx with --enable-shared. However, unless you know that you absolutely need shared, then I recommend omitting --enable-shared for ffmpeg and any required external libraries. Shared can be problematic as you've experienced. Navigate back to your ffmpeg source directory, run "make distclean" (you might need to use sudo with that because you used sudo with your ./configure for some reason) and try your ./configure again without --enable-shared and optionally the others I listed below: --enable-memalign-hack - I think this is a Windows specific option. --enable-pthreads - should now be autodetected. --enable-avfilter - looks like it's a default option. From bler at earthplanet.org Thu Aug 25 21:57:29 2011 From: bler at earthplanet.org (=?UTF-8?B?TWljaGHFgiBMZXNpYWs=?=) Date: Thu, 25 Aug 2011 21:57:29 +0200 Subject: [FFmpeg-user] Problem with -f video4linux2 thru all versions from stock Ubuntu 0.5.1 Message-ID: <4E56A929.8050901@earthplanet.org> Hello, I'm having problems with v4l2 indev on ffmpegs > 0.5.1. Ubuntu 10.04 LTS (fresh install), stock kernel 2.6.32-33-generic. cmd line: ffmpeg -f video4linux2 -i /dev/video0 1.avi v4l device as per dmesg: [39444.503290] A827 registered V4L2 device video0[video] [39444.503312] A827 registered V4L2 device vbi0[vbi] [39444.503329] A827 registered V4L2 device radio0[radio] [39444.503518] A827 registered ALSA sound card 1 [39444.503531] DVB: registering new adapter (A827[0] DVB-T) [39444.503533] A827[0] DVB-T registered DVB adapter 0 [39444.503733] DVB: registering adapter 0 frontend 0 (A827[0] DVB-T)... Everything works fine with ffmpeg found in this Ubuntu version's aptitude, which I believe is 0.5.1 (at least it works at all, im using it for streaming mpeg4 via ffserwer and it get clogged after a couple of hours, that's why im trying newer versions). I've tried releases 0.6.3, 0.7.3 and 0.8.2, also tried latest (a couple of minutes ago) git clone from git://git.videolan.org/ffmpeg.git. All of them give more or less the same output: ffmpeg version N-32098-gabe0b8e, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 25 2011 21:35:50 with gcc 4.4.3 configuration: libavutil 51. 13. 0 / 51. 13. 0 libavcodec 53. 12. 0 / 53. 12. 0 libavformat 53. 9. 0 / 53. 9. 0 libavdevice 53. 3. 0 / 53. 3. 0 libavfilter 2. 34. 2 / 2. 34. 2 libswscale 2. 0. 0 / 2. 0. 0 [video4linux2 @ 0x1888420] ioctl set time per frame(1/30) failed [video4linux2 @ 0x1888420] Could not find codec parameters (Unknown: none) [video4linux2 @ 0x1888420] Estimating duration from bitrate, this may be inaccurate /dev/video0: could not find codec parameters Is there anything I can do about it? I'd rather not go back to 0.5.1 as there cleary were some issues with mpeg4 or ffserver which I hope are fixed now. Best regards, ML From pgoldweic at northwestern.edu Thu Aug 25 22:07:15 2011 From: pgoldweic at northwestern.edu (Patricia N Goldweic) Date: Thu, 25 Aug 2011 20:07:15 +0000 Subject: [FFmpeg-user] problem compiling ffmpeg 0.8.2 In-Reply-To: <20110825111023.6cc6dbb5@lrcd.com> References: <316ABB7EE42BC448BDBDE73B4323B09B1DFFF585@evcspmbx4.ads.northwestern.edu> <20110825111023.6cc6dbb5@lrcd.com> Message-ID: <316ABB7EE42BC448BDBDE73B4323B09B1DFFF636@evcspmbx4.ads.northwestern.edu> Thanks. I'll try that out. I believe I probably did not use the --enable-shared option for libvpx. -Patricia -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Lou Sent: Thursday, August 25, 2011 2:10 PM To: ffmpeg-user at ffmpeg.org Subject: Re: [FFmpeg-user] problem compiling ffmpeg 0.8.2 On Thu, 25 Aug 2011 18:34:21 +0000 Patricia N Goldweic wrote: > Hi, > I've been trying to compile ffmpeg 0.8.2 on linux (rel6), and after > compiling all the third party libraries I needed, I am getting the > following error: > > > .... > libavutil/samplefmt.c: In function ?av_samples_fill_arrays?: > libavutil/samplefmt.c:88: warning: ?av_get_bits_per_sample_fmt? is > deprecated (declared at libavutil/samplefmt.c:76) CC > libavutil/sha.o CC libavutil/tree.o > CC libavutil/utils.o > CC libavutil/x86/cpu.o > LD libavutil/libavutil.so.51 > LD libavcodec/libavcodec.so.53 > /usr/bin/ld: /usr/local/lib/libvpx.a(vpx_codec.c.o): relocation > R_X86_64_32 against `.rodata.str1.1' can not be used when making a > shared object; recompile with -fPIC /usr/local/lib/libvpx.a: could not > read symbols: Bad value collect2: ld returned 1 exit status make: > *** [libavcodec/libavcodec.so.53] Error 1 > > > > I've configured ffmpeg with the following command: > > sudo ./configure --enablelibvpx --enable-libfaac --enable-shared > --enable-memalign-hack --enable-gpl --enable-libtheora > --enable-libmp3lame --enable-libvorbis --enable-libx264 > --enable-libxvid --enable-nonfree --enable-postproc --enable-avfilter > --enable-swscale --enable-pthreads --enable-libfreetype > --enable-libopenjpeg --arch=x86_64 > > (and I've installed version 0.9.7 of libvpx) > > Any ideas on what is happening or how to fix it? Thanks in advance, > -Patricia What was your ./configure for libvpx? If you compile ffmpeg with --enable-shared, then you probably should also compile libvpx with --enable-shared. However, unless you know that you absolutely need shared, then I recommend omitting --enable-shared for ffmpeg and any required external libraries. Shared can be problematic as you've experienced. Navigate back to your ffmpeg source directory, run "make distclean" (you might need to use sudo with that because you used sudo with your ./configure for some reason) and try your ./configure again without --enable-shared and optionally the others I listed below: --enable-memalign-hack - I think this is a Windows specific option. --enable-pthreads - should now be autodetected. --enable-avfilter - looks like it's a default option. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From luj125 at gmail.com Thu Aug 25 22:37:36 2011 From: luj125 at gmail.com (James Lu) Date: Thu, 25 Aug 2011 16:37:36 -0400 Subject: [FFmpeg-user] [buffer @ 0x17db380] Invalid pixel format '-1' In-Reply-To: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC14E@exchange.exchange.interhost.co.il> References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC14E@exchange.exchange.interhost.co.il> Message-ID: On Thu, Aug 25, 2011 at 10:29 AM, Dmitry Sherman wrote: > Hello, > I receive another new error when trying to extract images from video: > ffmpeg -ss 00:01:01 -i 41167.flv -vframes 1 -f image2 4.jpg > > I get: > [buffer @ 0x17db380] Invalid pixel format '-1' > Error opening filters! > > Full output: > [buffer @ 0x17db380] Invalid pixel format '-1' > Error opening filters! > > > Thanks. > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 > http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Can you give us full output from the flv input? I suspect it may be a colorspace issue, in which case you can use -vf format='pixel_format' to correct it. Also, I use -vcodec mjpeg to get jpeg still frames rather than -f image2. Perhaps that may work? Hope this helps, ~James From ffmpeg at neoprimitive.net Thu Aug 25 23:46:05 2011 From: ffmpeg at neoprimitive.net (Jesse Lucas) Date: Thu, 25 Aug 2011 17:46:05 -0400 Subject: [FFmpeg-user] Avid DNxHD QTs dropped/doubled frame problem In-Reply-To: <4E556076.9040606@tippett.com> References: <20110824194441.GW7923@neoprimitive.net> <4E556076.9040606@tippett.com> Message-ID: <20110825214604.GM7923@neoprimitive.net> On 24/08/11 13:35 -0700, Michael Root wrote: >> On 08/24/2011 12:44 PM, Jesse wrote: >> Hi, all. I've run in to an issue when importing DNxHD Quicktime >> movies made with ffmpeg in to Avid Media Composer. >> The source material are image sequences rendered at 24 FPS. >> >> The output format is DNxHD 36 Mb/s, 23.976 FPS. >> >> I am doing my encoding with ffmpeg SVN-r20372 packaged with Fedora >> Core 12. >> >> All output movies from ffmpeg play back with the correct number of >> frames in mplayer on Linux, and in Quicktime on a Mac. >> >> However, when the movies are imported in to an Avid Media Composer >> project set to DNxHD36, 23.976 FPS, movies with frame lengths ending >> in 0, 3, and 6 (e.g. 10, 13, 16, 20, 23, ...), have a doubled first >> frame, and a missing last frame. >> >> All DNxHD36 Quicktimes created with Quicktime on a Mac maintain the >> correct order and number of frames when imported in to the Avid, which >> is why I think my problem originates with ffmpeg. >> >> >> This is an example of my ffmpeg command on Linux: >> >> ffmpeg -f image2 -r 24 -i ./1-20/test.%04d.tif -vcodec dnxhd -b 36Mb -r 23.976 -s 1920x1080 ./test.1-20.mov >> >> >> This is the complete terminal output of the run. >> >> FFmpeg version SVN-r20372, Copyright (c) 2000-2009 Fabrice Bellard, et al. >> built on Nov 7 2009 10:57:27 with gcc 4.4.2 20091027 (Red Hat 4.4.2-7) >> 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='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --extra-version=rpmfusion --enable-bzlib --enable-libdc1394 --enable-libdirac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-avfilter-lavf --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect >> libavutil 50. 3. 0 / 50. 3. 0 >> libavcodec 52.37. 1 / 52.37. 1 >> libavformat 52.39. 2 / 52.39. 2 >> libavdevice 52. 2. 0 / 52. 2. 0 >> libavfilter 1. 4. 1 / 1. 4. 1 >> libswscale 0. 7. 1 / 0. 7. 1 >> libpostproc 51. 2. 0 / 51. 2. 0 >> [image2 @ 0x9735e0]MAX_READ_SIZE:5000000 reached >> Input #0, image2, from '1-20/test.%04d.tif': >> Duration: 00:00:00.83, start: 0.000000, bitrate: N/A >> Stream #0.0: Video: tiff, rgb24, 1920x1080, 24 tbr, 24 tbn, 24 tbc >> Output #0, mov, to './test.1-20.mov': >> Stream #0.0: Video: dnxhd, yuv422p, 1920x1080, q=2-31, 36000 kb/s, 2997 tbn, 23.98 tbc >> Stream mapping: >> Stream #0.0 -> #0.0 >> Press [q] to stop encoding >> frame= 20 fps= 8 q=1.0 Lsize= 3681kB time=0.83 bitrate=36148.6kbits/s >> video:3680kB audio:0kB global headers:0kB muxing overhead 0.024573% >> >> >> Does anyone see anything conspicuously wrong with my command line that >> would cause the behavior I described above? Is it normal to see >> "2997 tbn, 23.98 tbc" when writing DNxHD Quicktimes? >> >> Thanks for reading. >> >> -Jesse > > We had the same problem. This fixed it: > > --- libavformat/movenc.c 2011-08-24 13:33:37.000000000 -0700 > +++ libavformat/movenc.c.orig 2011-08-24 13:33:23.000000000 -0700 > @@ -1392,10 +1392,8 @@ > avio_wb32(pb, 0); /* size */ > ffio_wfourcc(pb, "trak"); > mov_write_tkhd_tag(pb, track, st); > - if (track->mode == MODE_PSP > - || track->flags & MOV_TRACK_CTTS > - || track->enc->codec_id == CODEC_ID_DNXHD) > - mov_write_edts_tag(pb, track); // DNXHD and PSP Movies require edts box > + if (track->mode == MODE_PSP || track->flags & MOV_TRACK_CTTS || > track->cluster[0].dts) > + mov_write_edts_tag(pb, track); // PSP Movies require edts box > if (track->tref_tag) > mov_write_tref_tag(pb, track); > mov_write_mdia_tag(pb, track); > > > There are still (relatively minor) colorspace problems, though. > JFYI. Thanks a lot, Mike. I applied that patch to ffmpeg from git as of an hour ago, compiled, and it did indeed produce a DNxHD36 Quicktime that imported in to our Avid without the problems I was seeing before. So, is this an ffmpeg bug, or an Avid bug? If the former, was this patch submitted to the ffmpeg-devel list? Any idea why it hasn't yet been merged in, or an equivalent fix made since the problem has been identified? Here is the patch again against current ffmpeg source. --- libavformat/movenc.c.orig 2011-08-25 15:19:22.000000000 -0400 +++ libavformat/movenc.c 2011-08-25 15:24:42.000000000 -0400 @@ -1392,7 +1392,10 @@ avio_wb32(pb, 0); /* size */ ffio_wfourcc(pb, "trak"); mov_write_tkhd_tag(pb, track, st); - if (track->mode == MODE_PSP || track->flags & MOV_TRACK_CTTS || track->cluster[0].dts) + if (track->mode == MODE_PSP || + track->flags & MOV_TRACK_CTTS || + track->cluster[0].dts || + track->enc->codec_id == CODEC_ID_DNXHD) mov_write_edts_tag(pb, track); // PSP Movies require edts box if (track->tref_tag) mov_write_tref_tag(pb, track); Thanks again. -J From ffmpeg at neoprimitive.net Thu Aug 25 23:53:15 2011 From: ffmpeg at neoprimitive.net (Jesse Lucas) Date: Thu, 25 Aug 2011 17:53:15 -0400 Subject: [FFmpeg-user] Avid DNxHD QTs dropped/doubled frame problem In-Reply-To: <066b01cc629f$a1651460$4301a8c0@hpkantoor> References: <20110824194441.GW7923@neoprimitive.net> <066b01cc629f$a1651460$4301a8c0@hpkantoor> Message-ID: <20110825215315.GN7923@neoprimitive.net> On 24/08/11 22:51 +0200, bouke wrote: > > ----- Original Message ----- > From: "Jesse" > To: > Sent: Wednesday, August 24, 2011 9:44 PM > Subject: [FFmpeg-user] Avid DNxHD QTs dropped/doubled frame problem > > > > Hi, all. I've run in to an issue when importing DNxHD Quicktime > > movies made with ffmpeg in to Avid Media Composer. > > The source material are image sequences rendered at 24 FPS. > > > > The output format is DNxHD 36 Mb/s, 23.976 FPS. > > > > I am doing my encoding with ffmpeg SVN-r20372 packaged with Fedora > > Core 12. > > > > All output movies from ffmpeg play back with the correct number of > > frames in mplayer on Linux, and in Quicktime on a Mac. > > > > However, when the movies are imported in to an Avid Media Composer > > project set to DNxHD36, 23.976 FPS, movies with frame lengths ending > > in 0, 3, and 6 (e.g. 10, 13, 16, 20, 23, ...), have a doubled first > > frame, and a missing last frame. > > > Did you try 'ignoreQTrate True' in the Console in Avid? And/or AMA in > instead of import? > Might solve something.. > (but i use WriteAvidMXF and mediatool the material in, way faster) Thanks, Bouke. Setting 'ignoreQTRate' to 'true' does seem to solve the problem on import for the time being. This gives me time to look in to the feasibility of deploying a new ffmpeg binary built with Mike Root's patch. Are there wider implications for our project if we just leave this option set to true in the Avid? This may be veering in to OT land (e.g. Avid support), so I guess, if you have any advice, reply off-list. Cheers, J From dev at rarevision.com Thu Aug 25 23:57:16 2011 From: dev at rarevision.com (Thomas Worth) Date: Thu, 25 Aug 2011 14:57:16 -0700 Subject: [FFmpeg-user] Static compile with embeded 3rd party libraries. In-Reply-To: <4E567829.1050508@bbc.co.uk> References: <4E567829.1050508@bbc.co.uk> Message-ID: On Thu, Aug 25, 2011 at 9:28 AM, Tim Nicholson wrote: > Having been building ffmpeg statically for a while to avoid conflicts with > distro versions by using:- > > "... --enable-static --disable-shared ..." > > I want to go one step further and end up with a portable executable that > also includes all the third party libs such as lame faac etc. Googling > around suggests adding in:- > > "--extra-libs=-static --extra-cflags=--static" > > However when I do this configure fails with an error saying it cannot find > the third party libs. I've found that I don't need --extra-libs=-static or --extra-cflags=--static if compiling third-party libs with --disable-shared. Since a .dylib (Mac OS X) is not generated, gcc by default compiles statically with the .a version of the library. I have to give the path to the library for this to work, which sometimes is put in a hidden directory. Here's an example: $ cd /libfaac $ ./configure --enable-static --enable-shared=no --without-mp4v2 --build=x86_64-apple-darwin10 $ make This compiles and puts libfaac.a in /libfaac/.libs/libfaac.a (yes, that's a dot in front of libs) Since I don't do "make install," I must specify the absolute path to the lib in ffmpeg's configure line: $ cd /ffmpeg $ ./configure $CONFIG_STUFF --extra-cflags=-I/libfaac/include --extra-ldflags=-L/libfaac/.libs $ make That should generate a static binary that includes libfaac. > > "gcc -o /tmp/ffconf.kNbWfR0S /tmp/ffconf.htm52E9X.o -lfaac -lm -pthread > -static > /usr/lib64/gcc/x86_64-suse-linux/4.5/../../../../x86_64-suse-linux/bin/ld: > cannot find -lfaac > collect2: ld returned 1 exit status > ERROR: libfaac not found" > > I am obviously missing a bit, as it links to them fine without the extra two > options. Can anyone assist in pointing me in the right direction? > > -- > 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. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From miker at tippett.com Fri Aug 26 01:19:29 2011 From: miker at tippett.com (Michael Root) Date: Thu, 25 Aug 2011 16:19:29 -0700 Subject: [FFmpeg-user] Avid DNxHD QTs dropped/doubled frame problem In-Reply-To: <20110825214604.GM7923@neoprimitive.net> References: <20110824194441.GW7923@neoprimitive.net> Message-ID: <4E56D881.2090403@tippett.com> > Thanks a lot, Mike. Happy to be of service. > I applied that patch to ffmpeg from git as of an hour ago, compiled, > and it did indeed produce a DNxHD36 Quicktime that imported in to our > Avid without the problems I was seeing before. > > So, is this an ffmpeg bug, or an Avid bug? As far as I'm concerned, it's a Quicktime bug--meaning, the entire format itself is an unholy mess of things that are undocumented, half implemented, or wrongly implemented. For what it's worth, I don't think it's an Avid problem--If I remember correctly, qtpro was doing the same thing. > If the former, was this patch submitted to the ffmpeg-devel list? Any > idea why it hasn't yet been merged in, or an equivalent fix made since > the problem has been identified? It hasn't been submitted to ffmpeg-devel. I only just got on the ffmpeg-user list the other day. Don't usually have the bandwidth... I'll add myself though. Got some "fun" questions about DNxHD colorspace. :) -miker > Here is the patch again against current ffmpeg source. > > > --- libavformat/movenc.c.orig 2011-08-25 15:19:22.000000000 -0400 > +++ libavformat/movenc.c 2011-08-25 15:24:42.000000000 -0400 > @@ -1392,7 +1392,10 @@ > avio_wb32(pb, 0); /* size */ > ffio_wfourcc(pb, "trak"); > mov_write_tkhd_tag(pb, track, st); > - if (track->mode == MODE_PSP || track->flags& MOV_TRACK_CTTS || track->cluster[0].dts) > + if (track->mode == MODE_PSP || > + track->flags& MOV_TRACK_CTTS || > + track->cluster[0].dts || > + track->enc->codec_id == CODEC_ID_DNXHD) > mov_write_edts_tag(pb, track); // PSP Movies require edts box > if (track->tref_tag) > mov_write_tref_tag(pb, track); > > > Thanks again. > > -J > > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From rickcorteza at gmail.com Fri Aug 26 01:51:32 2011 From: rickcorteza at gmail.com (Rick C.) Date: Fri, 26 Aug 2011 07:51:32 +0800 Subject: [FFmpeg-user] subtitles again please In-Reply-To: <20110825115532.GA7493@geppetto> References: <20110824180036.GB8530@geppetto> <14C1EFC6-D278-406C-B58D-78F8818C84EA@gmail.com> <20110825115532.GA7493@geppetto> Message-ID: On Aug 25, 2011, at 7:55 PM, Stefano Sabatini wrote: > On date Thursday 2011-08-25 09:26:53 +0800, Rick C. encoded: >> >> On Aug 25, 2011, at 2:00 AM, Stefano Sabatini wrote: >> >>> On date Wednesday 2011-08-24 10:34:11 +0800, Rick C. encoded: >>>> Hi again, >>>> >>>> I have asked in the past about hard-coding subtitles, but if I just >>>> want to add a subtitle track as a stream would someone mind giving >>>> me a working example? I'm aware of the documentation which says >>>> this: >>>> >>>> ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt >>>> >>>> And in FFmpeg -formats it says it supports decoding/encoding of .srt >>>> files. So how would I take a file without a subtitle stream and add >>>> a new one? Something like: >>>> >>>> ffmpeg -i mov.avi -i sub.srt output.avi >>>> >>>> Sorry to ask again but I'm assuming since it's there in the >>>> documentation it works but I really can't figure it out. Thanks! >>> >>> Not very experienced about subtitles and all, AFAIK subtitles support >>> in FFmpeg is still a bit... rudimentary. >>> >>> But if you want subtitles in output you need a format which supports >>> them, I know matroska (.mkv) should do. >>> >>> As for hardcoding subtitles right onto the video, I'm afraid we still >>> miss that facility (subtitles filtering + video overlaying seems the >>> right solution), correct if I'm wrong. >> >> >> Thanks for the reply. So what would be a working command line if the output was to .mkv for example? > > $ ffmpeg -i INPUT -i SUBTITLE_FILE -y OUTPUT > > then you should see something like this: > Stream #0.0 -> #0.0: mpeg4 -> libx264 > Stream #0.1 -> #0.1: mp3 -> libvorbis > Stream #1.0 -> #0.2: srt -> ass > > You can set the subtitle output format to srt by using -scodec srt, or > forcing the copy with -scodec copy. > -- Great I'll try this and let you know thanks! From tim.nicholson at bbc.co.uk Fri Aug 26 09:23:11 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Fri, 26 Aug 2011 08:23:11 +0100 Subject: [FFmpeg-user] Static compile with embeded 3rd party libraries. In-Reply-To: References: <4E567829.1050508@bbc.co.uk> Message-ID: <4E5749DF.8050209@bbc.co.uk> On 25/08/11 19:01, James Darnley wrote: > On 25/08/2011, Tim Nicholson wrote: >> Having been building ffmpeg statically for a while to avoid conflicts >> with distro versions by using:- >> >> "... --enable-static --disable-shared ..." >> >> I want to go one step further and end up with a portable executable that >> also includes all the third party libs such as lame faac etc. Googling >> around suggests adding in:- >> >> "--extra-libs=-static --extra-cflags=--static" >> >> However when I do this configure fails with an error saying it cannot >> find the third party libs. >> >> "gcc -o /tmp/ffconf.kNbWfR0S /tmp/ffconf.htm52E9X.o -lfaac -lm -pthread >> -static >> /usr/lib64/gcc/x86_64-suse-linux/4.5/../../../../x86_64-suse-linux/bin/ld: >> cannot find -lfaac >> collect2: ld returned 1 exit status >> ERROR: libfaac not found" >> >> I am obviously missing a bit, as it links to them fine without the extra >> two options. Can anyone assist in pointing me in the right direction? > > Do you have a static libfaac? According to yast, yes! libfaac-devel - Header files and static library for the faac library /usr/include/faac.h /usr/include/faaccfg.h /usr/lib64/libfaac.so -- 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 james.darnley at gmail.com Fri Aug 26 11:25:10 2011 From: james.darnley at gmail.com (James Darnley) Date: Fri, 26 Aug 2011 10:25:10 +0100 Subject: [FFmpeg-user] Static compile with embeded 3rd party libraries. In-Reply-To: <4E5749DF.8050209@bbc.co.uk> References: <4E567829.1050508@bbc.co.uk> <4E5749DF.8050209@bbc.co.uk> Message-ID: On 26/08/2011, Tim Nicholson wrote: > On 25/08/11 19:01, James Darnley wrote: >> Do you have a static libfaac? > > According to yast, yes! > > libfaac-devel - Header files and static library for the faac library > > /usr/include/faac.h > /usr/include/faaccfg.h > /usr/lib64/libfaac.so You might want to check that. I thought static libs were usually *.a and shared ones *.so. From gkinsey at ad-holdings.co.uk Fri Aug 26 11:52:13 2011 From: gkinsey at ad-holdings.co.uk (Gavin Kinsey) Date: Fri, 26 Aug 2011 10:52:13 +0100 Subject: [FFmpeg-user] read rtmp live stream delay In-Reply-To: References: Message-ID: <201108261052.13308.gkinsey@ad-holdings.co.uk> On Thursday 25 August 2011 17:04:03 Holger Bause wrote: > > I'm trying to load a live stream from red5 and save it to a file. > Using the following is resulting a very long delay (5 - 10 minutes) > before it start to read the stream: > > Any suggestions would be appreciated. Try setting a small probesize (-probesize 32768). I find this helps on network streams that don't provide duration info, such as live streams. -- Gavin Kinsey AD Holdings Plc Closed IPTV, the new safe and secure deterministic IP Video solution from Dedicated Micros, is now shipping through selected distributors. - Come and see this award winning plug and play, IP Video innovation at one of our Roadshows around the UK visit: https://www.dedicatedmicros.com/europe/ClosedIPTVRoadshow2011 - Contact our Customer Services Team for more information regarding how to enter the world of safe and secure hybrid IP and analogue video surveillance systems. From dev at rarevision.com Fri Aug 26 12:01:30 2011 From: dev at rarevision.com (Thomas Worth) Date: Fri, 26 Aug 2011 03:01:30 -0700 Subject: [FFmpeg-user] Static compile with embeded 3rd party libraries. In-Reply-To: References: <4E567829.1050508@bbc.co.uk> <4E5749DF.8050209@bbc.co.uk> Message-ID: On Fri, Aug 26, 2011 at 2:25 AM, James Darnley wrote: > On 26/08/2011, Tim Nicholson wrote: >> On 25/08/11 19:01, James Darnley wrote: >>> Do you have a static libfaac? >> >> According to yast, yes! >> >> libfaac-devel - Header files and static library for the faac library >> >> /usr/include/faac.h >> /usr/include/faaccfg.h >> /usr/lib64/libfaac.so > > You might want to check that. ?I thought static libs were usually *.a > and shared ones *.so. "so" means "shared object" as far as I know. That's a shared lib. Static is .a, as James pointed out. FYI, on Mac OS X shared libs have a .dylib extension. From tim.nicholson at bbc.co.uk Fri Aug 26 12:47:58 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Fri, 26 Aug 2011 11:47:58 +0100 Subject: [FFmpeg-user] Static compile with embeded 3rd party libraries. In-Reply-To: References: <4E567829.1050508@bbc.co.uk> <4E5749DF.8050209@bbc.co.uk> Message-ID: <4E5779DE.80607@bbc.co.uk> On 26/08/11 11:01, Thomas Worth wrote: > On Fri, Aug 26, 2011 at 2:25 AM, James Darnley wrote: >> On 26/08/2011, Tim Nicholson wrote: >>> On 25/08/11 19:01, James Darnley wrote: >>>> Do you have a static libfaac? >>> >>> According to yast, yes! >>> >>> libfaac-devel - Header files and static library for the faac library >>> >>> /usr/include/faac.h >>> /usr/include/faaccfg.h >>> /usr/lib64/libfaac.so >> >> You might want to check that. I thought static libs were usually *.a >> and shared ones *.so. > > "so" means "shared object" as far as I know. That's a shared lib. > Static is .a, as James pointed out. > Ahh. I'm more of a dynamic language man myself, C(++) not really my thing, so I'm still get used to the intricacies of compiler/linking etc. So I will need to build all the third party includes statically first then, as the distro only includes the shared.so and C headers.... > FYI, on Mac OS X shared libs have a .dylib extension. Useful info... Thanks for all this. -- 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 rickcorteza at gmail.com Fri Aug 26 13:33:50 2011 From: rickcorteza at gmail.com (Rick C.) Date: Fri, 26 Aug 2011 19:33:50 +0800 Subject: [FFmpeg-user] subtitles again please In-Reply-To: <20110825115532.GA7493@geppetto> References: <20110824180036.GB8530@geppetto> <14C1EFC6-D278-406C-B58D-78F8818C84EA@gmail.com> <20110825115532.GA7493@geppetto> Message-ID: <597D72FA-D56F-4420-8728-874D9480D0D8@gmail.com> On Aug 25, 2011, at 7:55 PM, Stefano Sabatini wrote: > On date Thursday 2011-08-25 09:26:53 +0800, Rick C. encoded: >> >> On Aug 25, 2011, at 2:00 AM, Stefano Sabatini wrote: >> >>> On date Wednesday 2011-08-24 10:34:11 +0800, Rick C. encoded: >>>> Hi again, >>>> >>>> I have asked in the past about hard-coding subtitles, but if I just >>>> want to add a subtitle track as a stream would someone mind giving >>>> me a working example? I'm aware of the documentation which says >>>> this: >>>> >>>> ffmpeg -i file.mov -an -vn -sbsf mov2textsub -scodec copy -f rawvideo sub.txt >>>> >>>> And in FFmpeg -formats it says it supports decoding/encoding of .srt >>>> files. So how would I take a file without a subtitle stream and add >>>> a new one? Something like: >>>> >>>> ffmpeg -i mov.avi -i sub.srt output.avi >>>> >>>> Sorry to ask again but I'm assuming since it's there in the >>>> documentation it works but I really can't figure it out. Thanks! >>> >>> Not very experienced about subtitles and all, AFAIK subtitles support >>> in FFmpeg is still a bit... rudimentary. >>> >>> But if you want subtitles in output you need a format which supports >>> them, I know matroska (.mkv) should do. >>> >>> As for hardcoding subtitles right onto the video, I'm afraid we still >>> miss that facility (subtitles filtering + video overlaying seems the >>> right solution), correct if I'm wrong. >> >> >> Thanks for the reply. So what would be a working command line if the output was to .mkv for example? > > $ ffmpeg -i INPUT -i SUBTITLE_FILE -y OUTPUT > > then you should see something like this: > Stream #0.0 -> #0.0: mpeg4 -> libx264 > Stream #0.1 -> #0.1: mp3 -> libvorbis > Stream #1.0 -> #0.2: srt -> ass > > You can set the subtitle output format to srt by using -scodec srt, or > forcing the copy with -scodec copy. Ok so I was able to get this to work thank you for that! It was successful going to .mkv but not to .avi or .mp4 (container doesn't support .srt it says). I guess that's what you meant when you said it was limited? Well hopefully we will see more from this in the future because it would be great! :-) This is also nicer than hard-coding the subs... From dev at rarevision.com Fri Aug 26 13:34:35 2011 From: dev at rarevision.com (Thomas Worth) Date: Fri, 26 Aug 2011 04:34:35 -0700 Subject: [FFmpeg-user] Static compile with embeded 3rd party libraries. In-Reply-To: <4E5779DE.80607@bbc.co.uk> References: <4E567829.1050508@bbc.co.uk> <4E5749DF.8050209@bbc.co.uk> <4E5779DE.80607@bbc.co.uk> Message-ID: On Fri, Aug 26, 2011 at 3:47 AM, Tim Nicholson wrote: > On 26/08/11 11:01, Thomas Worth wrote: >> >> On Fri, Aug 26, 2011 at 2:25 AM, James Darnley >> ?wrote: >>> >>> On 26/08/2011, Tim Nicholson ?wrote: >>>> >>>> On 25/08/11 19:01, James Darnley wrote: >>>>> >>>>> Do you have a static libfaac? >>>> >>>> According to yast, yes! >>>> >>>> libfaac-devel - Header files and static library for the faac library >>>> >>>> /usr/include/faac.h >>>> /usr/include/faaccfg.h >>>> /usr/lib64/libfaac.so >>> >>> You might want to check that. ?I thought static libs were usually *.a >>> and shared ones *.so. >> >> "so" means "shared object" as far as I know. That's a shared lib. >> Static is .a, as James pointed out. >> > > Ahh. I'm more of a dynamic language man myself, C(++) not really my thing, > so I'm still get used to the intricacies of compiler/linking etc. > > So I will need to build all the third party includes statically first then, > as the distro only includes the shared.so and C headers.... Yeah, you'll need to compile your third-party libs from source using --disable-shared on the configure line. You can probably do "make install," which will put the static libs in system location(s). FFmpeg's configure should then be able to find them. I don't like doing this though because I would rather not junk up my system with static libs that I only use to build static binaries of other programs. This is a problem though, because once you compile many third-party libs they put the libs in some hidden directory. So, you might have /libfaac/include with your headers, but no libfaac.a is obvious so you don't know what path to give to configure. That's because it's in /libfaac/.libs/libfaac.a, which is hidden. I discovered that several third-party sources do it this way. I'm not sure why, but it's annoying. If anyone has a better solution to this, I'd love to hear. From csillag.kristof at gmail.com Fri Aug 26 15:46:52 2011 From: csillag.kristof at gmail.com (Csillag Kristof) Date: Fri, 26 Aug 2011 15:46:52 +0200 Subject: [FFmpeg-user] syncing x11grab with a drawing application Message-ID: <4E57A3CC.5030200@gmail.com> Dear ffmpeg-users, I would like to use ffmpeg's x11grab input with a (legacy) X application in such a way that the captured frames are synchronized to the application's output. To be more precise, I would like to somehow notify ffmpeg every time when the application has finished drawing a frame, so that ffmpeg can capture it. When it has captured it, then the application would move on to draw the next frame. No audio is involved, so the only concern of synchronization is the application drawing the frames. Is something like this supported? (Obviously, I would need to add the special notification code to the application, but that's ok.) Thank you for your help: Kristof Csillag From dmitry at interhost.co.il Fri Aug 26 16:19:35 2011 From: dmitry at interhost.co.il (Dmitry Sherman) Date: Fri, 26 Aug 2011 17:19:35 +0300 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 In-Reply-To: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC150@exchange.exchange.interhost.co.il> References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC141@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC150@exchange.exchange.interhost.co.il> Message-ID: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC15E@exchange.exchange.interhost.co.il> The problem is persists only when muxing to flv container, I tried muxing into 3gp, avi, mp4 - no problems. ffmpeg -i /home/vmedia/public_html/viewstream/41167.mp4 -acodec copy -vcodec copy -sameq -f flv a.flv - fails... can I submit a bug for this issue? thank you. Dmitry Sherman dmitry at interhost.co.il Interhost Networks Ltd t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ http://facebook.com/InterhostIL -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Dmitry Sherman Sent: Thursday, August 25, 2011 5:45 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 This is what said on ffmpeg download page: FFmpeg development has moved to a Git repository, and the SVN repository is no longer updated. The last revision committed to SVN was r26402 on 2011-01-19. The SVN repository may be removed in a near future, so you're recommended to use the Git repository instead. Dmitry Sherman dmitry at interhost.co.il Interhost Networks Ltd t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ http://facebook.com/InterhostIL -----Original Message----- From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain Sent: Thursday, August 25, 2011 5:41 PM To: FFmpeg user questions and RTFMs Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 As far as I know, ffmpeg stopped using GIT, can you try SVN to head and compile ? seems like an outdated or unknowably buggy one you have. On Thu, Aug 25, 2011 at 9:52 AM, Dmitry Sherman wrote: > I found that I have already this patch applied. > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto: > ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain > Sent: Wednesday, August 24, 2011 8:25 PM > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided > invalid, non monotonically increasing dts to muxer in stream 0: 160 >= > 160 > > Does this http://ffmpeg.org/trac/ffmpeg/ticket/16 help you ? > > On Wed, Aug 24, 2011 at 12:58 PM, Dmitry Sherman > >wrote: > > > Hello, > > I receive sometimes an error when I try to transcode some mp4 files > > to flv using this command: > > ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv > > > > I receive this error: > > Output #0, flv, to 'a.flv': > > Metadata: > > major_brand : mp42 > > minor_version : 0 > > compatible_brands: mp42isomavc1 > > creation_time : 2011-07-05 16:50:32 > > encoder : Lavf53.9.0 > > Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, > > 416x224 [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc > > Metadata: > > creation_time : 2011-07-05 16:50:32 > > Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, > > stereo, s16, 64 kb/s > > Metadata: > > creation_time : 2011-07-05 16:50:32 > > Stream mapping: > > Stream #0.0 -> #0.0: copy > > Stream #0.1 -> #0.1: aac -> libfaac Press [q] to stop, [?] for help > > [flv @ 0x2bf0da0] Application provided invalid, non monotonically > > increasing dts to muxer in stream 0: 160 >= 160 > > av_interleaved_write_frame(): Invalid argument > > > > what can I do to solve this problem? > > > > Thank you. > > > > Dmitry Sherman > > dmitry at interhost.co.il > > Interhost Networks Ltd > > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > > http://facebook.com/InterhostIL > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > -- > Antoine Baqain > System Administration Lead > Maktoob.com Inc. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Antoine Baqain System Administration Lead Maktoob.com Inc. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. From mangesh.m at sigmainfo.net Fri Aug 26 17:09:00 2011 From: mangesh.m at sigmainfo.net (Mangesh M) Date: Fri, 26 Aug 2011 20:39:00 +0530 Subject: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 160 >= 160 In-Reply-To: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC15E@exchange.exchange.interhost.co.il> References: <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC119@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC141@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC150@exchange.exchange.interhost.co.il> <30A6CB623C6A374DB7B8879CA38FB3EA02C561DEC15E@exchange.exchange.interhost.co.il> Message-ID: Thanks please submit. On Fri, Aug 26, 2011 at 7:49 PM, Dmitry Sherman wrote: > The problem is persists only when muxing to flv container, > I tried muxing into 3gp, avi, mp4 - no problems. > > ffmpeg -i /home/vmedia/public_html/viewstream/41167.mp4 -acodec copy > -vcodec copy -sameq -f flv a.flv - fails... > > can I submit a bug for this issue? > thank you. > > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 > http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto: > ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Dmitry Sherman > Sent: Thursday, August 25, 2011 5:45 PM > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, > non monotonically increasing dts to muxer in stream 0: 160 >= 160 > > This is what said on ffmpeg download page: > FFmpeg development has moved to a Git repository, and the SVN repository is > no longer updated. The last revision committed to SVN was r26402 on > 2011-01-19. The SVN repository may be removed in a near future, so you're > recommended to use the Git repository instead. > > > Dmitry Sherman > dmitry at interhost.co.il > Interhost Networks Ltd > t: (+972)-543181182 f: (+972)-577976157 > http://www.interhost.co.il/ > http://facebook.com/InterhostIL > > > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto: > ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain > Sent: Thursday, August 25, 2011 5:41 PM > To: FFmpeg user questions and RTFMs > Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided invalid, > non monotonically increasing dts to muxer in stream 0: 160 >= 160 > > As far as I know, ffmpeg stopped using GIT, can you try SVN to head and > compile ? seems like an outdated or unknowably buggy one you have. > > On Thu, Aug 25, 2011 at 9:52 AM, Dmitry Sherman >wrote: > > > I found that I have already this patch applied. > > > > Dmitry Sherman > > dmitry at interhost.co.il > > Interhost Networks Ltd > > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > > http://facebook.com/InterhostIL > > > > > > -----Original Message----- > > From: ffmpeg-user-bounces at ffmpeg.org [mailto: > > ffmpeg-user-bounces at ffmpeg.org] On Behalf Of Antoine Baqain > > Sent: Wednesday, August 24, 2011 8:25 PM > > To: FFmpeg user questions and RTFMs > > Subject: Re: [FFmpeg-user] [flv @ 0x2bf0da0] Application provided > > invalid, non monotonically increasing dts to muxer in stream 0: 160 >= > > 160 > > > > Does this http://ffmpeg.org/trac/ffmpeg/ticket/16 help you ? > > > > On Wed, Aug 24, 2011 at 12:58 PM, Dmitry Sherman > > > >wrote: > > > > > Hello, > > > I receive sometimes an error when I try to transcode some mp4 files > > > to flv using this command: > > > ffmpeg -i 40735.mp4 -vcodec copy -acodec libfaa^C-f flv 40735.flv > > > > > > I receive this error: > > > Output #0, flv, to 'a.flv': > > > Metadata: > > > major_brand : mp42 > > > minor_version : 0 > > > compatible_brands: mp42isomavc1 > > > creation_time : 2011-07-05 16:50:32 > > > encoder : Lavf53.9.0 > > > Stream #0.0(und): Video: h264 ([7][0][0][0] / 0x0007), yuv420p, > > > 416x224 [SAR 38:39 DAR 38:21], q=2-31, 449 kb/s, 1k tbn, 90k tbc > > > Metadata: > > > creation_time : 2011-07-05 16:50:32 > > > Stream #0.1(und): Audio: aac ([10][0][0][0] / 0x000A), 48000 Hz, > > > stereo, s16, 64 kb/s > > > Metadata: > > > creation_time : 2011-07-05 16:50:32 > > > Stream mapping: > > > Stream #0.0 -> #0.0: copy > > > Stream #0.1 -> #0.1: aac -> libfaac Press [q] to stop, [?] for help > > > [flv @ 0x2bf0da0] Application provided invalid, non monotonically > > > increasing dts to muxer in stream 0: 160 >= 160 > > > av_interleaved_write_frame(): Invalid argument > > > > > > what can I do to solve this problem? > > > > > > Thank you. > > > > > > Dmitry Sherman > > > dmitry at interhost.co.il > > > Interhost Networks Ltd > > > t: (+972)-543181182 f: (+972)-577976157 http://www.interhost.co.il/ > > > http://facebook.com/InterhostIL > > > > > > _______________________________________________ > > > ffmpeg-user mailing list > > > ffmpeg-user at ffmpeg.org > > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > > > > > > -- > > Antoine Baqain > > System Administration Lead > > Maktoob.com Inc. > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > -- > > This message has been scanned for viruses and dangerous content by > > MailScanner, and is believed to be clean. > > > > _______________________________________________ > > ffmpeg-user mailing list > > ffmpeg-user at ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > > > -- > Antoine Baqain > System Administration Lead > Maktoob.com Inc. > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > -- > This message has been scanned for viruses and dangerous content by > MailScanner, and is believed to be clean. > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Thanks & Regards, ** ** ** ** *Mangesh Kagale* Senior Project Lead Sigma Infosolutions A TUV Certified ISO 9001:2008 Company mangesh.m at sigmainfo.net | Skype: mangeshsis1 www.sigmainfo.net | 1-888-861-7360 (Toll Free) | India: 080-40865100, Ext- 229 __________________________________________________________________________ *Please do not print this email unless it is absolutely necessary, save a tree. * This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to which they are addressed. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the named addressee you should not disseminate, distribute, copy this e-mail distributing or taking any action in reliance on the contents of this information is strictly prohibited. From ronag89 at gmail.com Fri Aug 26 22:39:12 2011 From: ronag89 at gmail.com (Robert Nagy) Date: Fri, 26 Aug 2011 22:39:12 +0200 Subject: [FFmpeg-user] request: vf_yadif alpha Message-ID: I would like to request that YUVA support would be added to vf_yadif. From omkiran.for.wiki at gmail.com Fri Aug 26 23:18:29 2011 From: omkiran.for.wiki at gmail.com (Omkiran Sharma) Date: Sat, 27 Aug 2011 02:48:29 +0530 Subject: [FFmpeg-user] Skip specific number of frames at start of mp4 file Message-ID: Hello all, Is there any way to skip a specific number of frames in the start of a mp4 stream? I know the number of frames I want to skip and am guaranteed that it will be a reference picture. But I do not know it in seconds. I guess if a stream has different frame rates it would mean I would have to manually convert from frames to time offset. Or is there a way on the command line? Cheers! From forum.amit.mangal at gmail.com Fri Aug 26 13:55:10 2011 From: forum.amit.mangal at gmail.com (Amit Mangal) Date: Fri, 26 Aug 2011 17:25:10 +0530 Subject: [FFmpeg-user] How to play asf file using ffmpeg ? Message-ID: Hi everyone, i want to play asf stream using ffmpeg is it possible ? anbody is having any clue. thanks From amitkumarmangal at gmail.com Fri Aug 26 14:05:22 2011 From: amitkumarmangal at gmail.com (Amit Mangal) Date: Fri, 26 Aug 2011 17:35:22 +0530 Subject: [FFmpeg-user] how to play asf stream using ffmpeg Message-ID: Hi everyone, i want to play asf stream using ffmpeg is it possible ? anybody is having any clue. thanks From steve at squaregoldfish.co.uk Sat Aug 27 13:19:35 2011 From: steve at squaregoldfish.co.uk (Steve Jones) Date: Sat, 27 Aug 2011 12:19:35 +0100 Subject: [FFmpeg-user] Compiling ffmpeg with libx264 Message-ID: <4E58D2C7.3030007@squaregoldfish.co.uk> Hi, I'm trying to compile ffmpeg with the libx264 library (both cloned from git today). libx264 installs fine, but when I try to compile ffmpeg I get the following error: /sources/ffmpeg/libavcodec/libx264.c:452: undefined reference to `x264_encoder_open_116' My Google-fu has failed me on this one. Can anyone help? Steve. From stefano.sabatini-lala at poste.it Sat Aug 27 13:27:50 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sat, 27 Aug 2011 13:27:50 +0200 Subject: [FFmpeg-user] Skip specific number of frames at start of mp4 file In-Reply-To: References: Message-ID: <20110827112750.GA6054@geppetto> On date Saturday 2011-08-27 02:48:29 +0530, Omkiran Sharma encoded: > Hello all, > > Is there any way to skip a specific number of frames in the start of a mp4 > stream? I know the number of frames I want to skip and am guaranteed that it > will be a reference picture. But I do not know it in seconds. I guess if a > stream has different frame rates it would mean I would have to manually > convert from frames to time offset. Or is there a way on the command line? You can try with the select filter, for example to skip the first 10 frames: -vf select="gte(n\, 10)" -- ffmpeg-user random tip #20 VHOOK has been removed, check out libavfilter: http://wiki.multimedia.cx/index.php?title=Libavfilter From nicolas.george at normalesup.org Sat Aug 27 13:41:04 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Sat, 27 Aug 2011 13:41:04 +0200 Subject: [FFmpeg-user] Compiling ffmpeg with libx264 In-Reply-To: <4E58D2C7.3030007@squaregoldfish.co.uk> References: <4E58D2C7.3030007@squaregoldfish.co.uk> Message-ID: <20110827114104.GA29805@phare.normalesup.org> Le decadi 10 fructidor, an CCXIX, Steve Jones a ?crit?: > I'm trying to compile ffmpeg with the libx264 library (both cloned > from git today). libx264 installs fine, but when I try to compile > ffmpeg I get the following error: > > /sources/ffmpeg/libavcodec/libx264.c:452: undefined reference to > `x264_encoder_open_116' It looks like the version you are using for compilation (.h files) is not the same as the version you are using for linking (.a or .so files). Look at the -I and -L options to the compiler. 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 ramandumcs at gmail.com Sat Aug 27 17:10:50 2011 From: ramandumcs at gmail.com (raman gupta) Date: Sat, 27 Aug 2011 20:40:50 +0530 Subject: [FFmpeg-user] FFMPEG support for Audio\video codec in MP4 container Message-ID: Hi All, I am using FFMPEG for transcoding of MP4 files. Could any one pls tell me what all video\audio codec combination are supported by ffmpeg in MP4 container. MP4 by definition does not specify the supported audio\video codec list. Any pointer orhelp on this will help me in moving forward. Thx in advance. Regards, Raman Gupta From stefano.sabatini-lala at poste.it Sat Aug 27 18:37:14 2011 From: stefano.sabatini-lala at poste.it (Stefano Sabatini) Date: Sat, 27 Aug 2011 18:37:14 +0200 Subject: [FFmpeg-user] request: vf_yadif alpha In-Reply-To: References: Message-ID: <20110827163714.GA23112@geppetto> On date Friday 2011-08-26 22:39:12 +0200, Robert Nagy encoded: > I would like to request that YUVA support would be added to vf_yadif. Does this work for you? -- 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 -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-vf_yadif-add-support-to-yuva420p.patch Type: text/x-diff Size: 1186 bytes Desc: not available URL: From sales at sevana.fi Sat Aug 27 18:50:54 2011 From: sales at sevana.fi (Sevana Oy) Date: Sat, 27 Aug 2011 20:50:54 +0400 Subject: [FFmpeg-user] Audio compression optimization Message-ID: <5D33B6A3E32746F58B26CD4C62D4F6C3@laptop> Hi, We would like to share an article on audio encoding optimization for podcasting. No need to think what bit rate to choose the batch file will automatically encode your source audio with the best bit rate to preserve the best quality and smallest file size. The approach was tested on MP3, Nero AAC and OGG. This paper will give you understanding on how one can achieve better compression ratio by bit rate optimization. The key point is that our approach describes a fully automated manner of choosing the bit rate that will preserve the audio quality you define. Read this paper through and find out how to save on size when encoding your podcasts, save on bandwidth when transmitting your audio streams in the network, make more audio tracks fit your memory stick when grabbed from a CD, or store more audio books on your mobile device. This paper will tell you how to save up to 50% on audio file size and up to 50% on the bit rate you encode your audio with still having a descent sound quality. Read full text of how to encode audio with lowest bit rate at highest quality in this blog post: http://blog.sevana.fi/optimize-bitrate-and-size-preserving-high-audio-quality-in-tracks-podcasts-tunes-with-aqua-wideband/ Best regards, Sevana Oy From steve at squaregoldfish.co.uk Sat Aug 27 22:32:20 2011 From: steve at squaregoldfish.co.uk (Steve Jones) Date: Sat, 27 Aug 2011 21:32:20 +0100 Subject: [FFmpeg-user] Compiling ffmpeg with libx264 In-Reply-To: <20110827114104.GA29805@phare.normalesup.org> References: <4E58D2C7.3030007@squaregoldfish.co.uk> <20110827114104.GA29805@phare.normalesup.org> Message-ID: <4E595454.10605@squaregoldfish.co.uk> On 08/27/2011 12:41 PM, Nicolas George wrote: > Le decadi 10 fructidor, an CCXIX, Steve Jones a ?crit : >> I'm trying to compile ffmpeg with the libx264 library (both cloned >> from git today). libx264 installs fine, but when I try to compile >> ffmpeg I get the following error: >> >> /sources/ffmpeg/libavcodec/libx264.c:452: undefined reference to >> `x264_encoder_open_116' > > It looks like the version you are using for compilation (.h files) is not > the same as the version you are using for linking (.a or .so files). Look at > the -I and -L options to the compiler. > > Regards, > I think you're right. I have two versions installed, one in /usr and one in /usr/local. For some reason the configure script is finding the version in /usr/local, but at compile time it's trying to use the one in /usr. I don't really know how configure and make work, so how can I work out why it's confused? Steve. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 554 bytes Desc: OpenPGP digital signature URL: From nicolas.george at normalesup.org Sat Aug 27 23:43:20 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Sat, 27 Aug 2011 23:43:20 +0200 Subject: [FFmpeg-user] Compiling ffmpeg with libx264 In-Reply-To: <4E595454.10605@squaregoldfish.co.uk> References: <4E58D2C7.3030007@squaregoldfish.co.uk> <20110827114104.GA29805@phare.normalesup.org> <4E595454.10605@squaregoldfish.co.uk> Message-ID: <20110827214320.GA15718@phare.normalesup.org> Le decadi 10 fructidor, an CCXIX, Steve Jones a ?crit?: > I think you're right. I have two versions installed, one in /usr and one > in /usr/local. For some reason the configure script is finding the > version in /usr/local, but at compile time it's trying to use the one in > /usr. I don't really know how configure and make work, so how can I work > out why it's confused? The configure for ffmpeg has the unusual behaviour that it does not try to look in /usr/local like most autoconf-produced configure. You can tell it to with the following options: --extra-cflags="-I/usr/local/include" --extra-ldflags="-L/usr/local/lib" But before that, you should look at the compile and link lines, with make V=1, and look in config.log to see where the strange flags come from. 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 ronag89 at gmail.com Sun Aug 28 00:34:41 2011 From: ronag89 at gmail.com (Robert Nagy) Date: Sun, 28 Aug 2011 00:34:41 +0200 Subject: [FFmpeg-user] request: vf_yadif alpha In-Reply-To: <20110827163714.GA23112@geppetto> References: <20110827163714.GA23112@geppetto> Message-ID: Yes, that should do it. On Sat, Aug 27, 2011 at 6:37 PM, Stefano Sabatini < stefano.sabatini-lala at poste.it> wrote: > On date Friday 2011-08-26 22:39:12 +0200, Robert Nagy encoded: > > I would like to request that YUVA support would be added to vf_yadif. > > Does this work for you? > -- > 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 steve at squaregoldfish.co.uk Sun Aug 28 12:48:47 2011 From: steve at squaregoldfish.co.uk (Steve Jones) Date: Sun, 28 Aug 2011 11:48:47 +0100 Subject: [FFmpeg-user] Compiling ffmpeg with libx264 In-Reply-To: <20110827214320.GA15718@phare.normalesup.org> References: <4E58D2C7.3030007@squaregoldfish.co.uk> <20110827114104.GA29805@phare.normalesup.org> <4E595454.10605@squaregoldfish.co.uk> <20110827214320.GA15718@phare.normalesup.org> Message-ID: <4E5A1D0F.5070106@squaregoldfish.co.uk> On 27/08/11 22:43, Nicolas George wrote: > Le decadi 10 fructidor, an CCXIX, Steve Jones a ?crit : >> I think you're right. I have two versions installed, one in /usr and one >> in /usr/local. For some reason the configure script is finding the >> version in /usr/local, but at compile time it's trying to use the one in >> /usr. I don't really know how configure and make work, so how can I work >> out why it's confused? > > The configure for ffmpeg has the unusual behaviour that it does not try to > look in /usr/local like most autoconf-produced configure. You can tell it to > with the following options: > > --extra-cflags="-I/usr/local/include" > --extra-ldflags="-L/usr/local/lib" > > But before that, you should look at the compile and link lines, with > make V=1, and look in config.log to see where the strange flags come from. > > Regards, > OK, so in config.log I'm seeing the following: check_lib x264.h x264_encoder_encode -lx264 check_header x264.h 1 #include /usr/local/include/x264.h:36:4: warning: #warning You must include stdint.h or inttypes.h before x264.h so the #include directive is picking up x264.h in /usr/local. This seems to go against what you say about it only looking in /usr/lib, although it could be that my build setup is overriding ffmpeg's wishes at this stage. The compile line that fails is: gcc -Llibavcodec -Llibavdevice -Llibavfilter -Llibavformat -Llibavutil -Llibpostproc -Llibswscale -Wl,--as-needed -Wl,--warn-common -Wl,-rpath-link=libpostproc:libswscale:libavfilter:libavdevice:libavformat:libavcodec:libavutil -o ffmpeg_g ffmpeg.o cmdutils.o -lavdevice -lavfilter -lavformat -lavcodec -lpostproc -lswscale -lavutil -ldl -lX11 -lXext -lXfixes -lxvidcore -lx264 -lvorbisenc -lvorbis -logg -ltheoraenc -ltheoradec -logg -lopencore-amrwb -lopencore-amrnb -lmp3lame -L/usr/lib/x86_64-linux-gnu -lfreetype -lfaac -lm -pthread -lz so there's no specific directive to look anywhere for libx264 (it's just -lx264). This must be why it's falling back to /usr/lib without the extra compile flags. Using the explicit flags you mentioned above works perfectly. Thanks for the help. Steve. PS If you want me to do more tests, just let me know. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 554 bytes Desc: OpenPGP digital signature URL: From nicolas.george at normalesup.org Sun Aug 28 12:58:34 2011 From: nicolas.george at normalesup.org (Nicolas George) Date: Sun, 28 Aug 2011 12:58:34 +0200 Subject: [FFmpeg-user] Compiling ffmpeg with libx264 In-Reply-To: <4E5A1D0F.5070106@squaregoldfish.co.uk> References: <4E58D2C7.3030007@squaregoldfish.co.uk> <20110827114104.GA29805@phare.normalesup.org> <4E595454.10605@squaregoldfish.co.uk> <20110827214320.GA15718@phare.normalesup.org> <4E5A1D0F.5070106@squaregoldfish.co.uk> Message-ID: <20110828105834.GA26111@phare.normalesup.org> Le primidi 11 fructidor, an CCXIX, Steve Jones a ?crit?: > so there's no specific directive to look anywhere for libx264 (it's just > -lx264). This must be why it's falling back to /usr/lib without the > extra compile flags. That's the normal behaviour. The strange part is that it finds x264.h in /usr/local. You may want to try to compile the following program: #include #include int main(void) { x264_encoder_open(0); return 0; } with just -lx264. If it compiles but fails to link, then there is something seriously inconsistent in your compiler's settings, and it will probably cause you trouble again later. If it fails to compile or success entirely, it is probably a problem in ffmpeg's configure. > Using the explicit flags you mentioned above works perfectly. Glad I could help. 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 ffmpeg2 at engene.se Sun Aug 28 21:43:40 2011 From: ffmpeg2 at engene.se (Marcus Engene) Date: Sun, 28 Aug 2011 21:43:40 +0200 Subject: [FFmpeg-user] Overlay, video ontop instead of below. Message-ID: <4E5A9A6C.5050404@engene.se> Hi, If I run this... ffmpeg -i apa.mov -y -vf 'movie=checkerboard640x480.png,crop=640:360 [ch];[ch][in] overlay=0:0:1 [out]' -acodec copy -vcodec png citron.mov ...it hangs (full cpu and file does not grow). Note [ch][in] instead of (working) [in][ch]. Is it possible to do what I want to accomplish somehow? I have vids with transparency I want to put on top of an image. Thanks, Marcus From ramandumcs at gmail.com Mon Aug 29 08:06:05 2011 From: ramandumcs at gmail.com (raman gupta) Date: Mon, 29 Aug 2011 11:36:05 +0530 Subject: [FFmpeg-user] FFMPEG support for Audio\video codec in MP4 container In-Reply-To: References: Message-ID: Just to add on some more details: Does FFMPEG supports H264 with NellyMoser in MP4 files.?? I have a mp4 file, which contains h264 video and Nellymoser Audio. And FFMPEG is throwing error when this file is used as an input. Its working fine when MP3 is used with H264 in MP4 file. Thanks in Advance. Regards, Raman Gupta On Sat, Aug 27, 2011 at 8:40 PM, raman gupta wrote: > Hi All, > > I am using FFMPEG for transcoding of ?MP4 files. > Could any one pls tell me what all video\audio codec combination are > supported by ffmpeg in MP4 container. > MP4 by definition does not specify the supported audio\video codec list. > > Any pointer or help on this will help me in moving forward. > > Thx in advance. > > Regards, > Raman Gupta > From ramandumcs at gmail.com Mon Aug 29 13:44:58 2011 From: ramandumcs at gmail.com (raman gupta) Date: Mon, 29 Aug 2011 17:14:58 +0530 Subject: [FFmpeg-user] Nellymoser in mp4 In-Reply-To: References: Message-ID: Hi Carl, Thanks for your response. I have uploaded the MP4 file at: http://www.datafilehost.com/download-10082e81.html The file can be streamed through Flash Media server and can be played in Flash Player. On using ffmpeg its throwing error that no decoder found for audio codec id. NellyMoser in flv format works fine with ffmpeg, but it throws error if NellyMoser in MP4 is used with ffmpeg Any help would be appreciated. Thanks, Raman Gupta On Mon, Aug 29, 2011 at 5:59 PM, Carl Eugen Hoyos wrote: > Hi! > > Could you upload a sample of Nellymoser in mp4 to > http://www.datafilehost.com/ ? > We will need a sample to fix it in FFmpeg. > > Is there an application that can play the file? > > Thank you, Carl Eugen > From marc at hallmarcwebsites.com Mon Aug 29 13:57:33 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Mon, 29 Aug 2011 07:57:33 -0400 Subject: [FFmpeg-user] Nellymoser in mp4 In-Reply-To: References: Message-ID: ---QUOTE--- > The file can be streamed through Flash Media server and can be played in > Flash Player. > On using ffmpeg its throwing error that no decoder found for audio codec id. > NellyMoser in flv format works fine with ffmpeg, but it throws error if > NellyMoser in MP4 is used with ffmpeg > > Any help would be appreciated. > Thanks, > Raman Gupta [>] ---END QUOTE--- This sounds like an incorrect audio codec has been used within the mp4 container; which codec did you use for the audio? From ramandumcs at gmail.com Mon Aug 29 14:55:05 2011 From: ramandumcs at gmail.com (raman gupta) Date: Mon, 29 Aug 2011 18:25:05 +0530 Subject: [FFmpeg-user] Nellymoser in mp4 In-Reply-To: References: Message-ID: I am using NellyMoser Audio Codec in Mp4 Container. Could you pls let me know the valid audio\video codecs which can be used in MP4 container for ffmpeg Thx, Raman Gupta On Mon, Aug 29, 2011 at 5:27 PM, HallMarc Websites wrote: > ---QUOTE--- >> The file can be streamed through Flash Media server and can be played in >> Flash Player. >> On using ffmpeg its throwing error that no decoder found for audio codec > id. >> NellyMoser in flv format works fine with ffmpeg, but it throws error if >> NellyMoser in MP4 ?is used with ffmpeg >> >> Any help would be appreciated. >> Thanks, >> Raman Gupta > [>] > ---END QUOTE--- > > This sounds like an incorrect audio codec has been used within the mp4 > container; which codec did you use for the audio? > > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From marc at hallmarcwebsites.com Mon Aug 29 15:41:29 2011 From: marc at hallmarcwebsites.com (HallMarc Websites) Date: Mon, 29 Aug 2011 09:41:29 -0400 Subject: [FFmpeg-user] Nellymoser in mp4 In-Reply-To: References: Message-ID: The CoDecs that I know of for mp4 are AAC, AMR and ALAC. There may be more. I personally use AAC-LC. You can Google search for more info and check out these 2 sites as well http://en.wikipedia.org/wiki/MPEG-4_Part_14 http://www.mp4ra.org/ Marc Hall HallMarc Websites > -----Original Message----- > From: ffmpeg-user-bounces at ffmpeg.org [mailto:ffmpeg-user- > bounces at ffmpeg.org] On Behalf Of raman gupta > Sent: Monday, August 29, 2011 8:55 AM > To: FFmpeg user questions and RTFMs > Cc: Carl Eugen Hoyos > Subject: Re: [FFmpeg-user] Nellymoser in mp4 > > I am using NellyMoser Audio Codec in Mp4 Container. > Could you pls let me know the valid audio\video codecs which can be used in > MP4 container for ffmpeg > > Thx, > Raman Gupta > > On Mon, Aug 29, 2011 at 5:27 PM, HallMarc Websites > wrote: > > ---QUOTE--- > >> The file can be streamed through Flash Media server and can be played > >> in Flash Player. > >> On using ffmpeg its throwing error that no decoder found for audio > >> codec > > id. > >> NellyMoser in flv format works fine with ffmpeg, but it throws error > >> if NellyMoser in MP4 ?is used with ffmpeg > >> > >> Any help would be appreciated. > >> Thanks, > >> Raman Gupta > > [>] > > ---END QUOTE--- > > > > This sounds like an incorrect audio codec has been used within the mp4 > > container; which codec did you use for the audio? > > > > _______________________________________________ > > 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 rogerdpack2 at gmail.com Mon Aug 29 18:09:34 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 29 Aug 2011 10:09:34 -0600 Subject: [FFmpeg-user] FFMPEG support for Audio\video codec in MP4 container In-Reply-To: References: Message-ID: > I am using FFMPEG for transcoding of ?MP4 files. > Could any one pls tell me what all video\audio codec combination are > supported by ffmpeg in MP4 container. > MP4 by definition does not specify the supported audio\video codec list. http://www.videolan.org/streaming-features.html might help as a matrix list of what is compatible et al. For VLC at least :P From rogerdpack2 at gmail.com Mon Aug 29 19:45:10 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 29 Aug 2011 11:45:10 -0600 Subject: [FFmpeg-user] Bounty: $250 make directshow capture inputs "user configurable" Message-ID: Hello all. I've decided to offer a small bounty ($250) for any enterprising developer interested in the following task: Background: ffmpeg can take dshow input, like $ ffmpeg -f dshow -i video="USB2.0_Camera":audio="Microphone (USB Audio Device)" output.mkv However you can't specify "1280x1024 7.5fps" (instead it chooses some poor default like 640x480 10 fps). So this would be a task to make this possible. Example open source programs that do this apparently already: VLC, iSpy. Extra credit if you can enumerate available dshow capture devices (video+audio): $50 Extra credit if you can enumerate the options of a given capture source. $50 Extra credit if you can set options for audio: $50 Related thread: >> When running the command line below, I get "Option video_size not found." >> I've tried -video_size hd720, with no effect. >> >> How do you set the video size when using dshow? Thanks for any help you can >> provide. >> >> ffmpeg -rtbufsize 100000000 -f dshow -s hd720 -i video="Microsoft LifeCam >> Cinema" -vframes 1 test%06d.tif > > You can't yet it's not implemented, I believe (just looking at it, I > think maybe it's using ConnectDirect to connect the pins, which just > always choose some default). From tom at tomkoole.com Mon Aug 29 22:41:47 2011 From: tom at tomkoole.com (tkoole) Date: Mon, 29 Aug 2011 13:41:47 -0700 (PDT) Subject: [FFmpeg-user] YUV 422P Conversion Clipping detection Message-ID: <1314650507833-3777344.post@n4.nabble.com> I'm trying to detect clipping when converting from YUV422p to RGB24. I've looked through some of the variables included when calling sws_scale() but didn't find any flags that stood out so I'm trying to convert manually using the descriptions on wikipedia. http://en.wikipedia.org/wiki/YCbCr I've tried several different methods of converting but I don't seem to be getting the proper rgb values. sws_scale converts the image properly however when I convert it I get values between 16-20; 'img' is just a small struct wrapping the frame 'frame' is the AVFrame containing the image int vc = 0, uc = 0; for(DWORD i = 0; i < img.width - 1; i += 2) { float y1 = img.frame->data[0][i]; float y2 = img.frame->data[0][i+1]; float u = img.frame->data[1][vc++]; float v = img.frame->data[2][uc++]; //int Cr = u; //int Cb = v; //float r = y1 + 1.402f * (Cr - 128); //float g = y1 - 0.344f * (Cb - 128) - 0.714 * (Cr - 128); //float b = y1 + 1.772f * (Cb - 128); //int r = y1 + Cr + (Cr >> 2) + (Cr >> 3) + (Cr >> 5); //int g = y1 - ((Cb >> 2) + (Cb >> 4) + (Cb >> 5)) - ((Cr >> 1) + (Cr >> 3) + (Cr >> 4) + (Cr >> 5)); //int b = y1 + Cb + (Cb >> 1) + (Cb >> 2) + (Cb >> 6); float r = 255.f / 219.f * (y1 - 0.f) + 255.f / 112.f * 0.701f * (u - 128.f); float g = 255.f / 219.f * (y1 - 0.f) - 255.f / 112.f * 0.886f * 0.114f / 0.587f * (v - 128.f) - 255.f / 112.f * 0.701f * 0.299f / 0.587f * (u - 128.f); float b = 255.f / 219.f * (y1 - 0.f) + 255.f / 112.f * 0.886f * (v - 128.f); rMax = (r > rMax) ? r : rMax; gMax = (g > gMax) ? g : gMax; bMax = (b > bMax) ? b : bMax; rMin = (r < rMin) ? r : rMin; gMin = (g < gMin) ? g : gMin; bMin = (b < bMin) ? b : bMin; // ..... repeat for y2 } Thanks for your time. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/YUV-422P-Conversion-Clipping-detection-tp3777344p3777344.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From rogerdpack2 at gmail.com Tue Aug 30 00:39:56 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Mon, 29 Aug 2011 16:39:56 -0600 Subject: [FFmpeg-user] YUV 422P Conversion Clipping detection In-Reply-To: <1314650507833-3777344.post@n4.nabble.com> References: <1314650507833-3777344.post@n4.nabble.com> Message-ID: > I'm trying to detect clipping when converting from YUV422p to RGB24. I've > looked through some of the variables included when calling sws_scale() but > didn't find any flags that stood out so I'm trying to convert manually using > the descriptions on wikipedia. http://en.wikipedia.org/wiki/YCbCr > > I've tried several different methods of converting but I don't seem to be > getting the proper rgb values. sws_scale converts the image properly however > when I convert it I get values between 16-20; Are you trying to fix this problem? http://ffmpeg.org/pipermail/ffmpeg-user/2011-August/001848.html (ffmpeg seems to convert from rgb wrong)? -r From phamsyquybk at gmail.com Tue Aug 30 10:48:37 2011 From: phamsyquybk at gmail.com (Quy Pham Sy) Date: Tue, 30 Aug 2011 17:48:37 +0900 Subject: [FFmpeg-user] Encoding video file with subtitles Message-ID: Hi all, I use the following command to encode video with a subtitle file (.srt) --------- $ ffmpeg -i hifi.avi -i hifi.srt -acodec libfaac -ar 48000 -ab 128k -ac 2 -vcodec libx264 -vpre ipod640 -s 480x240 -b 256k -scodec copy hifi.m4v -newsubtitle ---------- but it failed, and here is the output ---------- ffmpeg version 0.8.git, Copyright (c) 2000-2011 the FFmpeg developers built on Aug 4 2011 11:11:39 with gcc 4.5.2 configuration: --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib --disable-shared --enable-static --enable-gpl --enable-postproc --enable-pthreads --enable-ffplay --disable-ffserver --enable-memalign-hack --enable-nonfree --enable-libfaac --arch=x86 --enable-swscale --enable-libx264 --enable-avfilter --enable-debug=3 libavutil 51. 11. 1 / 51. 11. 1 libavcodec 53. 9. 1 / 53. 9. 1 libavformat 53. 6. 0 / 53. 6. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 27. 5 / 2. 27. 5 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, avi, from 'hifi.avi': Metadata: encoder : VirtualDubMod 1.5.4.1 (build 2178/release) IAS1 : English Duration: 01:49:02.20, start: 0.000000, bitrate: 897 kb/s Stream #0.0: Video: mpeg4 (Simple Profile), yuv420p, 544x304 [SAR 1:1 DAR 34:19], 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 32 kb/s [srt @ 0152c100] Estimating duration from bitrate, this may be inaccurate Input #1, srt, from 'hifi.srt': Duration: N/A, start: 56.080000, bitrate: N/A Stream #1.0: Subtitle: srt File 'hifi.m4v' already exists. Overwrite ? [y/N] y [buffer @ 02548920] w:544 h:304 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [scale @ 02558ee0] w:544 h:304 fmt:yuv420p -> w:480 h:240 fmt:yuv420p flags:0x4 [libx264 @ 02547f20] Default settings detected, using medium profile [libx264 @ 02547f20] using SAR=1/1 [libx264 @ 02547f20] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.1 Cache64 [libx264 @ 02547f20] profile High, level 3.0 [libx264 @ 02547f20] 264 - core 115 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=0 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=256 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, ipod, to 'hifi.m4v': Stream #0.0: Video: libx264, yuv420p, 480x240 [SAR 1:1 DAR 2:1], q=2-31, 256 kb/s, 90k tbn, 25 tbc Stream #0.1: Audio: libfaac, 48000 Hz, 2 channels, s16, 128 kb/s Stream #0.2: Subtitle: srt Stream #0.3: Subtitle: [0][0][0][0] / 0x0000, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Stream #1.0 -> #0.2 Stream #1.0 -> #0.3 Encoder (codec id 0) not found for output stream #0.3 ---------- Can anyone tell what i did miss in my command?? Thanks, Quy From tim.nicholson at bbc.co.uk Tue Aug 30 13:30:36 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 30 Aug 2011 12:30:36 +0100 Subject: [FFmpeg-user] mov->.dv rewrapping Message-ID: <4E5CC9DC.9050502@bbc.co.uk> Is it reasonable that :- "ffmpeg -i SYNCTEST.mov -vcodec copy -acodec copy out.dv" fails with:- [..] Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SYNCTEST.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2006-09-15 18:15:58 Duration: 00:02:00.00, start: 0.000000, bitrate: 30336 kb/s Stream #0.0(eng): Video: dvvideo (dvcp / 0x70637664), yuv420p, 720x576 [SAR 16:15 DAR 4:3], 28800 kb/s, 25 fps, 25 tbr, 25k tbn, 25 tbc Metadata: creation_time : 2006-09-15 18:15:59 Stream #0.1(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, 2 channels, s16, 1536 kb/s Metadata: creation_time : 2006-09-15 18:15:59 [dv @ 0x12a1f80] Can't initialize DV format! Make sure that you supply exactly two streams: video: 25fps or 29.97fps, audio: 2ch/48kHz/PCM (50Mbps allows an optional second audio stream) [...] which is a tad unhelpful since exactly two streams have been supplied of a compatible format... but:- "ffmpeg -i SYNCTEST.mov -target dv out.dv" works fine with:- [...] Assuming PAL for target. File '/mnt/pserver/avidnet/out.dv' already exists. Overwrite ? [y/N] y [buffer @ 0x12a1ec0] w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:16/15 sws_param: Output #0, dv, to '/mnt/pserver/avidnet/out.dv': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2006-09-15 18:15:58 encoder : Lavf53.10.0 Stream #0.0(eng): Video: dvvideo, yuv420p, 720x576 [SAR 16:15 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 25 tbc Metadata: creation_time : 2006-09-15 18:15:59 Stream #0.1(eng): Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s Metadata: creation_time : 2006-09-15 18:15:59 Stream mapping: Stream #0.0 -> #0.0: dvvideo -> dvvideo Stream #0.1 -> #0.1: pcm_s16be -> pcm_s16le [...] ...but I wonder if it has actually done an unnecessary recode.... -- 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 gregbartnick at gmail.com Tue Aug 30 17:04:55 2011 From: gregbartnick at gmail.com (Greg Bartnick) Date: Tue, 30 Aug 2011 10:04:55 -0500 Subject: [FFmpeg-user] Error with DNxHD 36 MXF files Message-ID: Hello List, Does anyone know why this error is occurring when running ffprobe or ffmpeg on a DNxHD 36 MXF file? [mxf @ 02a108c0] error, essence data could not be found Here are a couple of example files http://kcintrab.com/examples/21228257V01.4E5282E6.9364A0.mxf - 259 MB http://kcintrab.com/examples/reel_1ab_grd27.88919.mxf - 719 MB Does something need to be added to be able to support these files? I'd be willing to do some coding, but need some direction as to what needs to be done. thanks much, Greg From tim.nicholson at bbc.co.uk Tue Aug 30 17:23:50 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Tue, 30 Aug 2011 16:23:50 +0100 Subject: [FFmpeg-user] Error with DNxHD 36 MXF files In-Reply-To: References: Message-ID: <4E5D0086.7070907@bbc.co.uk> On 30/08/11 16:04, Greg Bartnick wrote: > Hello List, > > Does anyone know why this error is occurring when running ffprobe or ffmpeg > on a DNxHD 36 MXF file? > > [mxf @ 02a108c0] error, essence data could not be found > > Here are a couple of example files > http://kcintrab.com/examples/21228257V01.4E5282E6.9364A0.mxf - 259 MB > http://kcintrab.com/examples/reel_1ab_grd27.88919.mxf - 719 MB > Before I waste bandwidth downloading these, I notice that one of them has "...V01..." embedded in the filename suggesting it is a video only file (Avid media??). Could it therefore be that the file is an op atom mxf which ffmpeg does not support? > Does something need to be added to be able to support these files? I'd be > willing to do some coding, but need some direction as to what needs to be > done. > Answering the above may go some way to answering this... > thanks much, > Greg -- 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 gregbartnick at gmail.com Tue Aug 30 18:50:24 2011 From: gregbartnick at gmail.com (Greg Bartnick) Date: Tue, 30 Aug 2011 11:50:24 -0500 Subject: [FFmpeg-user] Error with DNxHD 36 MXF files In-Reply-To: <4E5D0086.7070907@bbc.co.uk> References: <4E5D0086.7070907@bbc.co.uk> Message-ID: On Tue, Aug 30, 2011 at 10:23 AM, Tim Nicholson wrote: > > Before I waste bandwidth downloading these, I notice that one of them has > "...V01..." embedded in the filename suggesting it is a video only file > (Avid media??). Could it therefore be that the file is an op atom mxf which > ffmpeg does not support? > > Yes that does look to be the case. The files are OP-Atom format. thanks for the reply, Greg From rogerdpack2 at gmail.com Tue Aug 30 19:19:16 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Tue, 30 Aug 2011 11:19:16 -0600 Subject: [FFmpeg-user] Bounty: $250 make directshow capture inputs "user configurable" In-Reply-To: References: Message-ID: > Hello all. > I've decided to offer a small bounty ($250) for any enterprising > developer interested in the following task [directshow improvements]: Looks like there have been a few interested parties so the bounty has been called for. Thanks for your responses! -roger- From koxaniy at mail.ru Tue Aug 30 22:43:15 2011 From: koxaniy at mail.ru (Tuuls) Date: Tue, 30 Aug 2011 13:43:15 -0700 (PDT) Subject: [FFmpeg-user] mov->.dv rewrapping In-Reply-To: <4E5CC9DC.9050502@bbc.co.uk> References: <4E5CC9DC.9050502@bbc.co.uk> Message-ID: <1314736995804-3779700.post@n4.nabble.com> Use : ffmpeg -i SYNCTEST.mov *-f dv* -vcodec copy -acodec copy out.dv -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/mov-dv-rewrapping-tp3778444p3779700.html Sent from the FFmpeg-users mailing list archive at Nabble.com. From koxaniy at mail.ru Tue Aug 30 22:55:49 2011 From: koxaniy at mail.ru (=?UTF-8?B?0J/QsNC/0LAg0JvRjtCx0Y/RidC40Lk=?=) Date: Wed, 31 Aug 2011 00:55:49 +0400 Subject: [FFmpeg-user] =?utf-8?q?mov-=3E=2Edv_rewrapping?= In-Reply-To: <4E5CC9DC.9050502@bbc.co.uk> References: <4E5CC9DC.9050502@bbc.co.uk> Message-ID: Use : ffmpeg -i SYNCTEST.mov -f dv -vcodec copy -acodec copy out.dv? 30 ??????? 2011, 15:30 ?? Tim Nicholson : Is it reasonable that :- "ffmpeg -i SYNCTEST.mov -vcodec copy -acodec copy out.dv" fails with:- [..] Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SYNCTEST.mov': ???Metadata: ?????major_brand : qt ?????minor_version : 537199360 ?????compatible_brands: qt ?????creation_time : 2006-09-15 18:15:58 ???Duration: 00:02:00.00, start: 0.000000, bitrate: 30336 kb/s ?????Stream #0.0(eng): Video: dvvideo (dvcp / 0x70637664), yuv420p, 720x576 [SAR 16:15 DAR 4:3], 28800 kb/s, 25 fps, 25 tbr, 25k tbn, 25 tbc ?????Metadata: ???????creation_time : 2006-09-15 18:15:59 ?????Stream #0.1(eng): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, 2 channels, s16, 1536 kb/s ?????Metadata: ???????creation_time : 2006-09-15 18:15:59 [dv @ 0x12a1f80] Can't initialize DV format! Make sure that you supply exactly two streams: ? ? ??????video: 25fps or 29.97fps, audio: 2ch/48kHz/PCM ? ? ??????(50Mbps allows an optional second audio stream) [...] which is a tad unhelpful since exactly two streams have been supplied of a compatible format... but:- "ffmpeg -i SYNCTEST.mov -target dv out.dv" works fine with:- [...] Assuming PAL for target. File '/mnt/pserver/avidnet/out.dv' already exists. Overwrite ? [y/N] y [buffer @ 0x12a1ec0] w:720 h:576 pixfmt:yuv420p tb:1/1000000 sar:16/15 sws_param: Output #0, dv, to '/mnt/pserver/avidnet/out.dv': ???Metadata: ?????major_brand : qt ?????minor_version : 537199360 ?????compatible_brands: qt ?????creation_time : 2006-09-15 18:15:58 ?????encoder : Lavf53.10.0 ?????Stream #0.0(eng): Video: dvvideo, yuv420p, 720x576 [SAR 16:15 DAR 4:3], q=2-31, 200 kb/s, 90k tbn, 25 tbc ?????Metadata: ???????creation_time : 2006-09-15 18:15:59 ?????Stream #0.1(eng): Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s ?????Metadata: ???????creation_time : 2006-09-15 18:15:59 Stream mapping: ???Stream #0.0 -> #0.0: dvvideo -> dvvideo ???Stream #0.1 -> #0.1: pcm_s16be -> pcm_s16le [...] ...but I wonder if it has actually done an unnecessary recode.... -- 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. _______________________________________________ ffmpeg-user mailing list ffmpeg-user at ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user From nanda at trinaytech.com Sat Aug 27 13:47:14 2011 From: nanda at trinaytech.com (Nanda) Date: Sat, 27 Aug 2011 17:17:14 +0530 Subject: [FFmpeg-user] Unable to remove ffmpeg Message-ID: <001e01cc64af$13df08a0$3b9d19e0$@com> Hi, We have installed ffmpeg on centos using make and make install, that was installed perfect. Now we want add speex codec in that, so we have tried uninstall the ffmpeg. We have used the following command Yum remove ffmpeg x264 faad2 faad2-devel also we have tried make uninstall The ffmpeg was not removed, so we have deleted the ffmpeg directory from the server. Again we have download and try to install at this time we are getting the following error. Makefile:1: config.mak: No such file or directory libavutil/Makefile:1: libavutil/../config.mak: No such file or directory libavutil/../subdir.mak:96: warning: overriding commands for target `libavutil/' libavutil/../subdir.mak:26: warning: ignoring old commands for target `libavutil/' libavutil/../subdir.mak:96: warning: overriding commands for target `libavutil/' libavutil/../subdir.mak:96: warning: ignoring old commands for target `libavutil/' Makefile:234: /tests/fate.mak: No such file or directory Makefile:235: /tests/fate2.mak: No such file or directory Makefile:237: /tests/fate/aac.mak: No such file or directory Makefile:238: /tests/fate/als.mak: No such file or directory Makefile:239: /tests/fate/fft.mak: No such file or directory Makefile:240: /tests/fate/h264.mak: No such file or directory Makefile:241: /tests/fate/mp3.mak: No such file or directory Makefile:242: /tests/fate/vorbis.mak: No such file or directory Makefile:243: /tests/fate/vp8.mak: No such file or directory make: *** No rule to make target `/tests/fate/vp8.mak'. Stop. So please help us how can we remove the ffmpeg completely and install again. Is there any way to include the speex codec without re-install? Please help us on this. Thanks Nanda From tim.nicholson at bbc.co.uk Wed Aug 31 09:27:04 2011 From: tim.nicholson at bbc.co.uk (Tim Nicholson) Date: Wed, 31 Aug 2011 08:27:04 +0100 Subject: [FFmpeg-user] mov->.dv rewrapping In-Reply-To: References: <4E5CC9DC.9050502@bbc.co.uk> Message-ID: <4E5DE248.50506@bbc.co.uk> On 30/08/11 21:55, ???? ??????? wrote: > Use : ffmpeg -i SYNCTEST.mov -f dv -vcodec copy -acodec copy out.dv? > According to the docs, "-f nnn" is used to force the format when it cannot otherwise be determined by the file extension. Since I have specified a file extension compatible with the required format I would not expect the -f to be necessary. As it happens the above does not work and fails in the same way. However I have found the cause of the issue and its an old gotcha I spotted when looking at my working version- Stream mapping: Stream #0.0 -> #0.0: dvvideo -> dvvideo Stream #0.1 -> #0.1: pcm_s16be -> pcm_s16le The DV mov has the audio be, but ffmpeg insists that for .dv streams the audio must be le, so the solution to my issue is:- ffmpeg -i SYNCTEST.mov -vcodec copy out.dv which gives:- Stream mapping: Stream #0.0 -> #0.0: copy Stream #0.1 -> #0.1: pcm_s16be -> pcm_s16le > > 30 ??????? 2011, 15:30 ?? Tim Nicholson: > > > > Is it reasonable that :- > "ffmpeg -i SYNCTEST.mov -vcodec copy -acodec copy out.dv" > > fails with:- Thanks for all your thoughts though...as trying your solutions and failing put me on the right track! -- 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 x2305andy2305x at yahoo.com Wed Aug 31 10:38:26 2011 From: x2305andy2305x at yahoo.com (Andy Andy) Date: Wed, 31 Aug 2011 01:38:26 -0700 (PDT) Subject: [FFmpeg-user] (no subject) Message-ID: <1314779906.39111.YahooMailNeo@web32505.mail.mud.yahoo.com> Hi guys, Here's what i'm trying to do: i'm trying to chain together two video filters: pad and watermark. I can't seem to do it properly. Here's what i'm doing: ffmpeg -y -i 00728.mts -f mp4 -vcodec libx264 -r 25 -b 5000000 -s 1440x1080? -vf "pad=1920:1080:240:0,movie=logo.png[watermark];[in][watermark]overlay=10:10[out]" -g 250 -threads 0 -pix_fmt yuvj420p -deinterlace -coder 1 -flags +loop -cmp +chroma -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -me_method hex -subq 2 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -bf 3 -refs 1 -directpred 1 -trellis 0 -flags2 +bpyramid-mixed_refs+wpred+dct8x8+fastpskip -wpredp 0 -rc_lookahead 10 -acodec libfaac -ar 44100 -ab 320000 -ac 2 img.mp4 ffmpeg version N-31743-g324b8ad, Copyright (c) 2000-2011 the FFmpeg developers ? built on Aug? 3 2011 15:13:54 with gcc 4.5.2 ? configuration: --enable-shared --disable-static --disable-doc --disable-ffplay --disable-ffserver --enable-avfilter --enable-postproc --enable-swscale --enable-gpl --enable-nonfree --enable-runtime-cpudetect --enable-pthreads --enable-bzlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-zlib --enable-version3 --enable-libopenjpeg ? libavutil??? 51. 11. 1 / 51. 11. 1 ? libavcodec?? 53.? 9. 1 / 53.? 9. 1 ? libavformat? 53.? 6. 0 / 53.? 6. 0 ? libavdevice? 53.? 2. 0 / 53.? 2. 0 ? libavfilter?? 2. 27. 5 /? 2. 27. 5 ? libswscale??? 2.? 0. 0 /? 2.? 0. 0 ? libpostproc? 51.? 2. 0 / 51.? 2. 0 Continuity Check Failed [mpegts @ 0xa33460] max_analyze_duration 5000000 reached at 5000000 Continuity Check Failed stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 50.00 (50/1) Input #0, mpegts, from '/home/alexandru-david/Desktop/00728.mts': ? Duration: 00:00:11.97, start: 1.000033, bitrate: 6829 kb/s ? Program 1 ??? Stream #0.0[0x1011]: Video: h264 (Main), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc ??? Stream #0.1[0x1100]: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s [buffer @ 0xa547a0] w:1440 h:1080 pixfmt:yuv420p tb:1/1000000 sar:4/3 sws_param: [movie @ 0xa54f00] seek_point:0 format_name:(null) file_name:/home/alexandru-david/Exporter/resources/logo_creaza_720.png stream_index:0 Too many inputs specified for the "movie" filter. Error opening filters! The reason why i try to do this, i need to pad the 4:3?aspect ratio 1080p input so that it is fullHD (1920x1080) , then apply a fullHD png logo watermark in top left corner. Can you guys see what i'm doing wrong? I even tried to put the pad filter after watermark to see if i get any changes, although that's not the right order in which i want this done, then message is Not enough inputs specified for the "pad" filter., with a command like ffmpeg -y -i 00728.mts -f mp4 -vcodec libx264 -r 25 -b 5000000 -s 1440x1080? -vf "movie=logo.png[watermark];[in][watermark]overlay=10:10[out],pad=1920:1080:240:0" -g 250 -threads 0 -pix_fmt yuvj420p -deinterlace -coder 1 -flags +loop -cmp +chroma -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -me_method hex -subq 2 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -bf 3 -refs 1 -directpred 1 -trellis 0 -flags2 +bpyramid-mixed_refs+wpred+dct8x8+fastpskip -wpredp 0 -rc_lookahead 10 -acodec libfaac -ar 44100 -ab 320000 -ac 2 img.mp4 Any help will be greatly appreciated. Regards, DAV From thomas at pixelpartner.de Wed Aug 31 11:51:02 2011 From: thomas at pixelpartner.de (Thomas Kumlehn) Date: Wed, 31 Aug 2011 11:51:02 +0200 Subject: [FFmpeg-user] (no subject) In-Reply-To: <1314779906.39111.YahooMailNeo@web32505.mail.mud.yahoo.com> References: <1314779906.39111.YahooMailNeo@web32505.mail.mud.yahoo.com> Message-ID: <21840DFC-A92A-4E0A-9480-C3B7EC2CAD97@pixelpartner.de> Try this > -vf "[in]pad=1920:1080:240:0[p];movie=logo.png[watermark];[p][watermark]overlay=10:10[out]" Any filter needs an input and if you don't specify one, it will take the last used/available. You cannot reuse an already used input. To work around this, the op split= was introduced, but in my opinion fails to work. I always get grey instead of content on the second output of split. best wishes, Thomas Kumlehn PIXEL PARTNER (R) Send from my iPad 3-D http://www.pixelpartner.de Am 31.08.2011 um 10:38 schrieb Andy Andy : > Hi guys, > > Here's what i'm trying to do: i'm trying to chain together two video filters: pad and watermark. I can't seem to do it properly. Here's what i'm doing: > > ffmpeg -y -i 00728.mts -f mp4 -vcodec libx264 -r 25 -b 5000000 -s 1440x1080 -vf "pad=1920:1080:240:0,movie=logo.png[watermark];[in][watermark]overlay=10:10[out]" -g 250 -threads 0 -pix_fmt yuvj420p -deinterlace -coder 1 -flags +loop -cmp +chroma -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -me_method hex -subq 2 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -bf 3 -refs 1 -directpred 1 -trellis 0 -flags2 +bpyramid-mixed_refs+wpred+dct8x8+fastpskip -wpredp 0 -rc_lookahead 10 -acodec libfaac -ar 44100 -ab 320000 -ac 2 img.mp4 > ffmpeg version N-31743-g324b8ad, Copyright (c) 2000-2011 the FFmpeg developers > built on Aug 3 2011 15:13:54 with gcc 4.5.2 > configuration: --enable-shared --disable-static --disable-doc --disable-ffplay --disable-ffserver --enable-avfilter --enable-postproc --enable-swscale --enable-gpl --enable-nonfree --enable-runtime-cpudetect --enable-pthreads --enable-bzlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-zlib --enable-version3 --enable-libopenjpeg > libavutil 51. 11. 1 / 51. 11. 1 > libavcodec 53. 9. 1 / 53. 9. 1 > libavformat 53. 6. 0 / 53. 6. 0 > libavdevice 53. 2. 0 / 53. 2. 0 > libavfilter 2. 27. 5 / 2. 27. 5 > libswscale 2. 0. 0 / 2. 0. 0 > libpostproc 51. 2. 0 / 51. 2. 0 > Continuity Check Failed > [mpegts @ 0xa33460] max_analyze_duration 5000000 reached at 5000000 > Continuity Check Failed stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 50.00 (50/1) > Input #0, mpegts, from '/home/alexandru-david/Desktop/00728.mts': > Duration: 00:00:11.97, start: 1.000033, bitrate: 6829 kb/s > Program 1 > Stream #0.0[0x1011]: Video: h264 (Main), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc > Stream #0.1[0x1100]: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s > [buffer @ 0xa547a0] w:1440 h:1080 pixfmt:yuv420p tb:1/1000000 sar:4/3 sws_param: > [movie @ 0xa54f00] seek_point:0 format_name:(null) file_name:/home/alexandru-david/Exporter/resources/logo_creaza_720.png stream_index:0 > Too many inputs specified for the "movie" filter. > Error opening filters! > > > The reason why i try to do this, i need to pad the 4:3 aspect ratio 1080p input so that it is fullHD (1920x1080) , then apply a fullHD png logo watermark in top left corner. > > Can you guys see what i'm doing wrong? I even tried to put the pad filter after watermark to see if i get any changes, although that's not the right order in which i want this done, then message is Not enough inputs specified for the "pad" filter., with a command like > > ffmpeg -y -i 00728.mts -f mp4 -vcodec libx264 -r 25 -b 5000000 -s 1440x1080 -vf "movie=logo.png[watermark];[in][watermark]overlay=10:10[out],pad=1920:1080:240:0" -g 250 -threads 0 -pix_fmt yuvj420p -deinterlace -coder 1 -flags +loop -cmp +chroma -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -me_method hex -subq 2 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -bf 3 -refs 1 -directpred 1 -trellis 0 -flags2 +bpyramid-mixed_refs+wpred+dct8x8+fastpskip -wpredp 0 -rc_lookahead 10 -acodec libfaac -ar 44100 -ab 320000 -ac 2 img.mp4 > > Any help will be greatly appreciated. > > Regards, > DAV > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user From x2305andy2305x at yahoo.com Wed Aug 31 12:15:59 2011 From: x2305andy2305x at yahoo.com (Andy Andy) Date: Wed, 31 Aug 2011 03:15:59 -0700 (PDT) Subject: [FFmpeg-user] (no subject) In-Reply-To: <21840DFC-A92A-4E0A-9480-C3B7EC2CAD97@pixelpartner.de> References: <1314779906.39111.YahooMailNeo@web32505.mail.mud.yahoo.com> <21840DFC-A92A-4E0A-9480-C3B7EC2CAD97@pixelpartner.de> Message-ID: <1314785759.95527.YahooMailNeo@web32505.mail.mud.yahoo.com> It definitely works so thank you very much for the solution. Unfortunately i don't really understand what you did. If you have the time, could you explain what the [p] is going on there? Thanks in advance. Regards, DAV ________________________________ From: Thomas Kumlehn To: FFmpeg user questions and RTFMs Sent: Wednesday, August 31, 2011 12:51 PM Subject: Re: [FFmpeg-user] (no subject) Try this > -vf "[in]pad=1920:1080:240:0[p];movie=logo.png[watermark];[p][watermark]overlay=10:10[out]" Any filter needs an input and if you don't specify one, it will take the last used/available. You cannot reuse an already used input. To work around this, the op split= was introduced, but in my opinion fails to work. I always get grey instead of content on the second output of split. best wishes, Thomas Kumlehn PIXEL PARTNER (R) Send from my iPad 3-D http://www.pixelpartner.de Am 31.08.2011 um 10:38 schrieb Andy Andy : > Hi guys, > > Here's what i'm trying to do: i'm trying to chain together two video filters: pad and watermark. I can't seem to do it properly. Here's what i'm doing: > > ffmpeg -y -i 00728.mts -f mp4 -vcodec libx264 -r 25 -b 5000000 -s 1440x1080? -vf "pad=1920:1080:240:0,movie=logo.png[watermark];[in][watermark]overlay=10:10[out]" -g 250 -threads 0 -pix_fmt yuvj420p -deinterlace -coder 1 -flags +loop -cmp +chroma -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -me_method hex -subq 2 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -bf 3 -refs 1 -directpred 1 -trellis 0 -flags2 +bpyramid-mixed_refs+wpred+dct8x8+fastpskip -wpredp 0 -rc_lookahead 10 -acodec libfaac -ar 44100 -ab 320000 -ac 2 img.mp4 > ffmpeg version N-31743-g324b8ad, Copyright (c) 2000-2011 the FFmpeg developers >? built on Aug? 3 2011 15:13:54 with gcc 4.5.2 >? configuration: --enable-shared --disable-static --disable-doc --disable-ffplay --disable-ffserver --enable-avfilter --enable-postproc --enable-swscale --enable-gpl --enable-nonfree --enable-runtime-cpudetect --enable-pthreads --enable-bzlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-zlib --enable-version3 --enable-libopenjpeg >? libavutil? ? 51. 11. 1 / 51. 11. 1 >? libavcodec? 53.? 9. 1 / 53.? 9. 1 >? libavformat? 53.? 6. 0 / 53.? 6. 0 >? libavdevice? 53.? 2. 0 / 53.? 2. 0 >? libavfilter? 2. 27. 5 /? 2. 27. 5 >? libswscale? ? 2.? 0. 0 /? 2.? 0. 0 >? libpostproc? 51.? 2. 0 / 51.? 2. 0 > Continuity Check Failed > [mpegts @ 0xa33460] max_analyze_duration 5000000 reached at 5000000 > Continuity Check Failed stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 50.00 (50/1) > Input #0, mpegts, from '/home/alexandru-david/Desktop/00728.mts': >? Duration: 00:00:11.97, start: 1.000033, bitrate: 6829 kb/s >? Program 1 >? ? Stream #0.0[0x1011]: Video: h264 (Main), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc >? ? Stream #0.1[0x1100]: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s > [buffer @ 0xa547a0] w:1440 h:1080 pixfmt:yuv420p tb:1/1000000 sar:4/3 sws_param: > [movie @ 0xa54f00] seek_point:0 format_name:(null) file_name:/home/alexandru-david/Exporter/resources/logo_creaza_720.png stream_index:0 > Too many inputs specified for the "movie" filter. > Error opening filters! > > > The reason why i try to do this, i need to pad the 4:3 aspect ratio 1080p input so that it is fullHD (1920x1080) , then apply a fullHD png logo watermark in top left corner. > > Can you guys see what i'm doing wrong? I even tried to put the pad filter after watermark to see if i get any changes, although that's not the right order in which i want this done, then message is Not enough inputs specified for the "pad" filter., with a command like > > ffmpeg -y -i 00728.mts -f mp4 -vcodec libx264 -r 25 -b 5000000 -s 1440x1080? -vf "movie=logo.png[watermark];[in][watermark]overlay=10:10[out],pad=1920:1080:240:0" -g 250 -threads 0 -pix_fmt yuvj420p -deinterlace -coder 1 -flags +loop -cmp +chroma -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -me_method hex -subq 2 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -bf 3 -refs 1 -directpred 1 -trellis 0 -flags2 +bpyramid-mixed_refs+wpred+dct8x8+fastpskip -wpredp 0 -rc_lookahead 10 -acodec libfaac -ar 44100 -ab 320000 -ac 2 img.mp4 > > Any help will be greatly appreciated. > > Regards, > DAV > _______________________________________________ > 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.amit.mangal at gmail.com Wed Aug 31 15:35:05 2011 From: forum.amit.mangal at gmail.com (Amit Mangal) Date: Wed, 31 Aug 2011 06:35:05 -0700 Subject: [FFmpeg-user] how to play asf stream using ffmpeg ? Message-ID: Hi Everyone, Anybody knows how to play asf stream using ffmpeg ? thanks From rogerdpack2 at gmail.com Wed Aug 31 17:50:30 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 31 Aug 2011 09:50:30 -0600 Subject: [FFmpeg-user] mac os x webcam capture Message-ID: Hello all. Is it possible for ffmpeg to accept input from say a webcam on os x? Thanks in advance. -roger- From rogerdpack2 at gmail.com Wed Aug 31 17:51:52 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 31 Aug 2011 09:51:52 -0600 Subject: [FFmpeg-user] Unable to remove ffmpeg In-Reply-To: <001e01cc64af$13df08a0$3b9d19e0$@com> References: <001e01cc64af$13df08a0$3b9d19e0$@com> Message-ID: > Makefile:1: config.mak: No such file or directory > Now we want add speex codec in that, so we have tried uninstall the ffmpeg. make clean? From rogerdpack2 at gmail.com Wed Aug 31 17:52:18 2011 From: rogerdpack2 at gmail.com (Roger Pack) Date: Wed, 31 Aug 2011 09:52:18 -0600 Subject: [FFmpeg-user] how to play asf stream using ffmpeg ? In-Reply-To: References: Message-ID: > Anybody knows how to play asf stream using ffmpeg ? ffplay? From forum.amit.mangal at gmail.com Wed Aug 31 17:58:26 2011 From: forum.amit.mangal at gmail.com (Amit Mangal) Date: Wed, 31 Aug 2011 08:58:26 -0700 Subject: [FFmpeg-user] how to play asf stream using ffmpeg ? In-Reply-To: References: Message-ID: what ffplay ? On Wed, Aug 31, 2011 at 8:52 AM, Roger Pack wrote: > > Anybody knows how to play asf stream using ffmpeg ? > > ffplay? > _______________________________________________ > ffmpeg-user mailing list > ffmpeg-user at ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > From rhodri at kynesim.co.uk Wed Aug 31 18:04:25 2011 From: rhodri at kynesim.co.uk (Rhodri James) Date: Wed, 31 Aug 2011 17:04:25 +0100 Subject: [FFmpeg-user] how to play asf stream using ffmpeg ? In-Reply-To: References: Message-ID: On Wed, 31 Aug 2011 16:58:26 +0100, Amit Mangal wrote: > On Wed, Aug 31, 2011 at 8:52 AM, Roger Pack > wrote: > >> > Anybody knows how to play asf stream using ffmpeg ? >> >> ffplay? > > what ffplay ? rhodri at sto-helit:~$ which ffplay /usr/bin/ffplay It's built as part of your ffmpeg installation. -- Rhodri James Kynesim Ltd From thomas at pixelpartner.de Wed Aug 31 17:44:42 2011 From: thomas at pixelpartner.de (Thomas Kumlehn) Date: Wed, 31 Aug 2011 17:44:42 +0200 Subject: [FFmpeg-user] (no subject) In-Reply-To: <1314785759.95527.YahooMailNeo@web32505.mail.mud.yahoo.com> References: <1314779906.39111.YahooMailNeo@web32505.mail.mud.yahoo.com> <21840DFC-A92A-4E0A-9480-C3B7EC2CAD97@pixelpartner.de> <1314785759.95527.YahooMailNeo@web32505.mail.mud.yahoo.com> Message-ID: <7D76607B-7DBC-4F8C-A0A5-7F0F5AEBB82B@pixelpartner.de> Sure, Andy! I first explain, why/how my solution works: The -vf parameter can be split into 3 independant parts, separated by semicolons 1 [in]pad=1920:1080:240:0[p] Takes the first and only input video with the portlabel [in], pads it and stores the result in port [p]. You could ommit [in] because the only availabel port to feed pad in this part of vf would be [in] and pad would take it. 2 movie=logo.png[watermark] Does not consume or need any input port, but feeds an output that is manually labeled [watermark] for later reference 3 [p][watermark]overlay=10:10[out] The overlay needs two inputs that we gave in the proper order. If the last output of the prior node is identical to the first input needed for the current, I could skip naming them and replace ; by a , so both nodes are chained together I don't know if separating node chains with ; and labeling all in/ouputs causes performance penelties, but it helped me a lot to understand what happens when. Thomas Kumlehn PIXEL PARTNER (R) Send from my iPad 3-D http://www.pixelpartner.de Am 31.08.2011 um 12:15 schrieb Andy Andy : > It definitely works so thank you very much for the solution. Unfortunately i don't really understand what you did. > > > If you have the time, could you explain what the [p] is going on there? > > Thanks in advance. > > Regards, > DAV > > > > ________________________________ > From: Thomas Kumlehn > To: FFmpeg user questions and RTFMs > Sent: Wednesday, August 31, 2011 12:51 PM > Subject: Re: [FFmpeg-user] (no subject) > > Try this >> -vf "[in]pad=1920:1080:240:0[p];movie=logo.png[watermark];[p][watermark]overlay=10:10[out]" > Any filter needs an input and if you don't specify one, it will take the last used/available. > You cannot reuse an already used input. To work around this, the op split= was introduced, but in my opinion fails to work. I always get grey instead of content on the second output of split. > > best wishes, > > Thomas Kumlehn > PIXEL PARTNER (R) > > Send from my iPad 3-D > http://www.pixelpartner.de > > Am 31.08.2011 um 10:38 schrieb Andy Andy : > >> Hi guys, >> >> Here's what i'm trying to do: i'm trying to chain together two video filters: pad and watermark. I can't seem to do it properly. Here's what i'm doing: >> >> ffmpeg -y -i 00728.mts -f mp4 -vcodec libx264 -r 25 -b 5000000 -s 1440x1080 -vf "pad=1920:1080:240:0,movie=logo.png[watermark];[in][watermark]overlay=10:10[out]" -g 250 -threads 0 -pix_fmt yuvj420p -deinterlace -coder 1 -flags +loop -cmp +chroma -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -me_method hex -subq 2 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -bf 3 -refs 1 -directpred 1 -trellis 0 -flags2 +bpyramid-mixed_refs+wpred+dct8x8+fastpskip -wpredp 0 -rc_lookahead 10 -acodec libfaac -ar 44100 -ab 320000 -ac 2 img.mp4 >> ffmpeg version N-31743-g324b8ad, Copyright (c) 2000-2011 the FFmpeg developers >> built on Aug 3 2011 15:13:54 with gcc 4.5.2 >> configuration: --enable-shared --disable-static --disable-doc --disable-ffplay --disable-ffserver --enable-avfilter --enable-postproc --enable-swscale --enable-gpl --enable-nonfree --enable-runtime-cpudetect --enable-pthreads --enable-bzlib --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfaac --enable-libgsm --enable-libmp3lame --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libx264 --enable-zlib --enable-version3 --enable-libopenjpeg >> libavutil 51. 11. 1 / 51. 11. 1 >> libavcodec 53. 9. 1 / 53. 9. 1 >> libavformat 53. 6. 0 / 53. 6. 0 >> libavdevice 53. 2. 0 / 53. 2. 0 >> libavfilter 2. 27. 5 / 2. 27. 5 >> libswscale 2. 0. 0 / 2. 0. 0 >> libpostproc 51. 2. 0 / 51. 2. 0 >> Continuity Check Failed >> [mpegts @ 0xa33460] max_analyze_duration 5000000 reached at 5000000 >> Continuity Check Failed stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) -> 50.00 (50/1) >> Input #0, mpegts, from '/home/alexandru-david/Desktop/00728.mts': >> Duration: 00:00:11.97, start: 1.000033, bitrate: 6829 kb/s >> Program 1 >> Stream #0.0[0x1011]: Video: h264 (Main), yuv420p, 1440x1080 [SAR 4:3 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 50 tbc >> Stream #0.1[0x1100]: Audio: ac3, 48000 Hz, 5.1, s16, 448 kb/s >> [buffer @ 0xa547a0] w:1440 h:1080 pixfmt:yuv420p tb:1/1000000 sar:4/3 sws_param: >> [movie @ 0xa54f00] seek_point:0 format_name:(null) file_name:/home/alexandru-david/Exporter/resources/logo_creaza_720.png stream_index:0 >> Too many inputs specified for the "movie" filter. >> Error opening filters! >> >> >> The reason why i try to do this, i need to pad the 4:3 aspect ratio 1080p input so that it is fullHD (1920x1080) , then apply a fullHD png logo watermark in top left corner. >> >> Can you guys see what i'm doing wrong? I even tried to put the pad filter after watermark to see if i get any changes, although that's not the right order in which i want this done, then message is Not enough inputs specified for the "pad" filter., with a command like >> >> ffmpeg -y -i 00728.mts -f mp4 -vcodec libx264 -r 25 -b 5000000 -s 1440x1080 -vf "movie=logo.png[watermark];[in][watermark]overlay=10:10[out],pad=1920:1080:240:0" -g 250 -threads 0 -pix_fmt yuvj420p -deinterlace -coder 1 -flags +loop -cmp +chroma -partitions +parti8x8+parti4x4+partp8x8+partb8x8 -me_method hex -subq 2 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -b_strategy 1 -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -bf 3 -refs 1 -directpred 1 -trellis 0 -flags2 +bpyramid-mixed_refs+wpred+dct8x8+fastpskip -wpredp 0 -rc_lookahead 10 -acodec libfaac -ar 44100 -ab 320000 -ac 2 img.mp4 >> >> Any help will be greatly appreciated. >> >> Regards, >> DAV >> _______________________________________________ >> 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