[Ffmpeg-devel-irc] ffmpeg.log.20131103

[burek]

[02:30] <KnightHawk3> Could someone tell me if I could offset the subtitles when creating hardsubs (-vf subtitles=sub.srt)?
[02:53] <KnightHawk3> Ah I have solved it by just using a tool to automatically offset the subtitles in the srt file then having ffmpeg use it.
[03:18] <max2me> Hello everyone :) I'm converting bunch of photos into a timelapse video using x264 codec & following command: "-y -r 30 -i %04d.jpg -b:v 3000k -vcodec libx264 -g 24 output.mpg". Then I upload it to the server and when I try to open it from my phone, it has to download entire video before playing. How can I have it start playing as soon as few frames are downloaded? (kinda like youtube & all other video hostings do)
[03:19] <max2me> Do I need to modify something in the way I create video or do I need to setup streaming server for that?
[03:24] <KnightHawk3> max2me: Probably need to setup a streaming server.
[03:25] Action: KnightHawk3 would like to point out he isn't very smart and is just guessing.
[03:27] <max2me> Got it, that's what I've suspected as well. But I wanted to avoid paying for extra-bandwidth between cloud storage & streaming server.
[05:32] <DrBenway> im having this problem where i'm trying to decode using hardware acceleration. i have a test where i  capture from camera and encode in h264 to then feed into my decoder and then render the frame. it works perfectly in software but as soon as i enable hardware decoding... i start getting log about "missing reference picture" and it looks like only the upper left quarter of my frame is being decoded
[05:33] <DrBenway> this same code seems to work when reading from a file using avformat.
[05:33] <DrBenway> could it be the decoder context that is badly intialized or something/
[05:35] <DrBenway> or should i ask that stuff on the devel channel?
[10:33] <strk> yasm/nasm not found or too old. Use --disable-yasm for a crippled build. <--- how crippled ?
[10:34] <strk> happens on a CentOS release 5.8
[11:13] <snooky> good morging
[11:13] <snooky> how i calculate the bitrate with an target size?
[11:21] <JEEB> since bit rate is kb per second, you look at how long your content is second-wise, then solve (bit_rate_video + bit_rate_audio [say, 192kbps for audio?]) * time = size for bit_rate_video, then give it some leeway for the container by removing a bit from the video bit rate and that should be it :P
[11:24] <snooky> i want calculate the target size of my mkv file
[11:24] <snooky> the target size is 4000MB
[11:24] <snooky> the audiofile is 143mb big
[11:24] <snooky> and 448k
[11:44] <snooky> is that nen bitrate? 12081656
[12:05] <snooky> ???
[12:16] <snooky> 78.328 FPS
[12:16] <snooky> what was wrong?
[12:17] <snooky> bitrate 16552kbp/s ?
[12:35] <sacarasc> snooky: How long in seconds is your file?
[12:36] <snooky> 2684
[12:36] <snooky> the original file has a bitrate of 3633kbp/s ans 23.976 FPS
[12:39] <sacarasc> So, X + 448k * 2684 = 4000MB, this would make X (the video bitrate) around 1042k.
[12:39] <sacarasc> I think.
[12:40] <snooky> hhmmm...
[12:40] <snooky> target_bytes = target_size * 1024 * 1024   audio_bytes = ((abr * 1000) / 8) * time   return (target_bytes - audio_bytes) / time * 8
[12:40] <snooky> there i calculate the bitrate
[12:41] <sacarasc> Wait.
[12:41] <sacarasc> I got that wrong, because I forgot the bit/byte conversion.
[12:42] <GRMrGecko> Is it just me or can none make FFmpeg eat the CPU in Mavericks?
[12:43] <GRMrGecko> Max I've seen it was 380% CPU. not the max
[12:43] <GRMrGecko> I normally get 11FPS, but now only 6FPS
[12:46] <knc119023> Hey - what codec should I be using to decode WMA files?
[12:47] <knc119023> I tried with AV_CODEC_ID_WMAV1 and AV_CODEC_ID_WMAV2 but I get a "Could not open codec" error...
[12:47] <knc119023> Basically, `avcodec_find_decoder(AV_CODEC_ID_WMAV1)` fails
[14:24] <zvn3x> does anyone know, can i use this tool to batch convert mp3s to wav format?
[14:28] <zap0> zvn3x, ffmpeg doesn't 'batch' anything.  it converts.  the batch bit will have to be your own shell script / code.
[14:29] <zvn3x> i know i can convert files one by one, i am trying to construct a command to automate the process
[14:29] <klaxa> you can do something like: for i in $(ls *.mp3); do ffmpeg -i "$i" -c:a pcm_s16le "$i".wav; done
[14:30] <klaxa> will turn "Track 01.mp3" to "Track 01.mp3.wav"
[14:30] <klaxa> if you are on a shell that supports that of course
[14:30] <klaxa> like bash or something
[14:30] <zvn3x> thats exactly what i need, in a bash script, perserving quality and filename just changing to be 16bit pcm wav files instead of mp3
[14:32] <zvn3x> so this command should work in bash, for i in $(ls *.mp3); do ffmpeg -i "$i" -c:a pcm_s16le "$i".wav; done
[14:33] <zap0> tias
[14:33] <sacarasc> You shouldn't do that, zvn3x, try "for i in *.mp3", works better in most situations.
[14:35] <klaxa> oh right, i kinda thought there was some reason that wouldn't work
[14:35] <klaxa> but that was just me
[14:35] <juanmabc> *cough* adpcm_ima_amv encoder *cough* ;P
[14:35] <klaxa> confused it with using "while read line; do something; done < *.mp3
[14:35] <klaxa> "
[14:35] <c_14> zvn3x: You can also use "${i/\.mp3/.wav}" for the output if you don't want the .mp3.wav
[14:35] <klaxa> because that one doesn't work because *.mp3 is not a file/stream/whatever, i don't really know
[14:36] <zvn3x> yes i recently converted about 400 flv to mp4 using a similar command, but im not familiar with the options and i dont know bash
[14:36] <juanmabc> for file in *.mp3; do echo "<$file>"; done
[14:37] <juanmabc> #bash
[14:37] <juanmabc> for the novice linux user, not just programmers ;D
[14:38] <zvn3x> this part -c:a pcm_s16le is standard wave format?
[14:40] <zvn3x> for i in *.mp3; do name=`echo $i | cut -d'.' -f1`; echo $name; ffmpeg -i $i $name.mp4;
[14:44] <juanmabc> for file in *.mp3; do ffmpeg -i "$file" "${file%.*}.mp4"; done
[14:45] <juanmabc> usually test before running what would run
[14:45] <juanmabc> for file in *.mp3; do echo ffmpeg -i "$file" "${file%.*}.mp4"; done
[14:45] <juanmabc> you can't do that so out of the box on gui ;P
[14:46] <zvn3x> sorry i wanted the output to wav, so just change that mp4 to be wav, is there any options i need to pass to achieve this
[14:47] <juanmabc> ffmpeg takes care according to output file format, though the defaults perhaps are too low
[14:47] <juanmabc> like rate, freq, ...
[14:48] <juanmabc> -ac 2 -ar 44100 (2 channels, 44100 Hz)
[14:48] <juanmabc> -ab 128K (for mp3,ogg, ...)
[14:48] <juanmabc> you can see to what it converts in the process, it outputs that before conversion
[14:49] <zvn3x> -ac 1 (for mono?)
[14:49] <juanmabc> sure
[14:50] <juanmabc> (a_udio c_hannels, a_udio r_ate, ...
[14:56] <zvn3x> hey is this right? for file in *.mp3; do ffmpeg -i "$file" -ac 2 -ar 44100 "${file%.*}.mp4";
[14:57] <sacarasc> Except for missing "done" at the end, pretty much. You might wanna add -b:a too, though.
[14:58] <juanmabc> and replace .mp4 for .wav :?
[14:59] <zvn3x> for file in *mp3; do ffmpeg -i "$file" -ac 1 -ar 44100 "${file%.*}.wav"; done
[14:59] <zvn3x> should i put that -b:a in there?
[15:00] <sacarasc> No.
[15:00] <sacarasc> It should default to 1440k
[15:02] <zvn3x> what does that option do, i cant find this in the man.. i just want 44100, since it should be the standard quality
[15:04] <sacarasc> -b is bitrate, :a means it's for audio.
[15:04] <juanmabc> it's -ab 128K, the syntax is changing to -b:a 128K, currently both are supported
[15:13] <zvn3x> i copied the mp3s to a test directory and ran the script, it seems to have worked out so far. im going to try to create an analysis
[15:15] <zvn3x> has anyone used ffprobe
[15:19] <sacarasc> What do you need from it?
[15:21] <zvn3x> it would be helpful to print an analysis of every wav in the directory and see if iv achieved the right quality
[15:21] <klaxa> pcm is uncompressed
[15:21] <klaxa> there is no way i cannot have achieved the right quality
[15:21] <klaxa> the limiting factor here is bandwidth and the quality of your mp3 files
[15:22] <zvn3x> what about the way to see every detail of the newly converted files?
[15:22] <klaxa> you can run ffprobe somefile.wav
[15:22] <klaxa> but that will probably only show the bitrate being like 1536 kbps
[15:24] <zvn3x> i want to check that all files are 44.1kHz 16-bit mono
[15:25] <zvn3x> it would be cool to print all this info into a file
[15:33] <zap0> batch ffmpeg -i
[17:24] <someAlex> Hi! I'm trying to use ffmpeg+ffserver to re-translate incoming high-quality stream to low-quality to make it possible to watch it at my second, more slower PC.
[17:24] <someAlex> Incoming stream is http://localhost:8902/stream
[17:24] <someAlex> What I do:
[17:24] <someAlex> ffserver -f ~/ffs.conf &
[17:24] <someAlex> # Contents: http://pastebin.com/xZtzHBN6
[17:24] <someAlex> ffmpeg -i http://localhost:8902/stream -strict -2 -preset medium http://localhost:8090/feed1.ffm
[17:24] <someAlex> But the result is: http://pastebin.com/ZxGtUcGB
[17:24] <someAlex> Looks like both AAC and h264 codecs are available: http://pastebin.com/6wfJjJkq (lines 84 and 219)
[17:24] <someAlex> Any advice? I never worked with ffmpeg directly before, but for now I have no ideas, what to do with this situation.
[19:34] <klemax> hi
[19:34] <klemax> php: symbol lookup error: /usr/local/lib/php/extensions/no-debug-non-zts-20100525/ffmpeg.so: undefined symbol: avcodec_init
[19:34] <klemax> is there a way to  fix this error?
[00:00] --- Mon Nov  4 2013