February 2008 - ffmpeg-user - lists.ffmpeg.org

Segmentation fault on latest revision
by evert＠rooftopsolutions.nl 05 Feb '08

05 Feb '08

Hi Guys, Here's my command line: ffmpeg -y -i zoe.mov -b 600000 zoe.flv And the output: FFmpeg version SVN-r11690, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-shared --enable-gpl --prefix=/home/guapo/flvdeamon/tools/ffmpeg/build/ --enable-libmp3lame --enable-libx264 --enable-libfaad --enable-debug libavutil version: 49.6.0 libavcodec version: 51.50.0 libavformat version: 52.7.0 libavdevice version: 52.0.0 built on Jan 30 2008 23:23:59, gcc: 4.1.2 (Ubuntu 4.1.2-0ubuntu4) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'zoe.mov': Duration: 00:00:13.2, start: 0.000000, bitrate: 15750 kb/s Stream #0.0(eng): Video: mjpeg, yuvj422p, 640x480 [PAR 0:1 DAR 0:1], 30.00 tb(r) Stream #0.1(eng): Audio: pcm_s16le, 44100 Hz, stereo, 1411 kb/s Output #0, flv, to 'zoe.flv': Stream #0.0(eng): Video: flv, yuv420p, 640x480 [PAR 0:1 DAR 0:1], q=2-31, 600 kb/s, 30.00 tb(c) Stream #0.1(eng): Audio: libmp3lame, 44100 Hz, stereo, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 [flv @ 0xb7e10710]removing common factors from framerate Press [q] to stop encoding Segmentation fault FFMpeg is running on Amazon's EC2.. Is there an older revision known to work or could the problem be somewhere else? Evert

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possible bug? timecode detection error with libx264
by nodaster＠gmail.com 05 Feb '08

05 Feb '08

Hi everyone! I'm unsure if this is the best place to report it, but I think I found a bug in ffmepg (I tried latest CVS version - it's still here). I'm trying to convert video into an iPod-compatible format (mp4 / h264 + aac). Here is the command used and the resulting output: ------- ffmpeg.exe -y -i "C:\Documents and Settings\My Documents\lost401.avi" -f mp4 -vcodec libx264 -level 30 -crf 19 -ab 128k -subq 4 -refs 2 -me hex -partitions +parti4x4+partp4x4+partp8x8+partb8x8 -s 480x320 -maxrate 1500k -bufsize 4M -g 250 -coder 0 -threads auto -acodec libfaac -ac 2 -ar 44100 "C:\Documents and Settings\My Documents\lost401.mp4" FFmpeg version Sherpya-r11050, Copyright (c) 2000-2007 Fabrice Bellard, et al. libavutil version: 49.5.0 libavcodec version: 51.48.0 libavformat version: 51.19.0 built on Nov 18 2007 09:00:58, gcc: 4.2.1 [Sherpya] Seems stream 0 codec frame rate differs from container frame rate: 23.98 (65535/2733) -> 23.98 (23976024/1000000) Input #0, avi, from 'C:\Documents and Settings\My Documents\lost401.avi': Duration: 00:41:50.7, start: 0.000000, bitrate: 1168 kb/s Stream #0.0: Video: mpeg4, yuv420p, 624x352 [PAR 1:1 DAR 39:22], 23.98 tb(r) Stream #0.1: Audio: mp3, 48000 Hz, stereo, 128 kb/s Output #0, mp4, to 'C:\Documents and Settings\My Documents\lost401 .mp4': Stream #0.0: Video: libx264, yuv420p, 480x320 [PAR 13:11 DAR 39:22], q=2-31, 200 kb/s, 23.98 tb(c) Stream #0.1: Audio: libfaac, 44100 Hz, stereo, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 [libx264 @ 00A50410]using SAR=13/11 [libx264 @ 00A50410]using cpu capabilities: MMX MMXEXT SSE SSE2 SSSE3 WARNING codec timebase is very high. If duration is too long, file may not be playable by quicktime. Specify a shorter timebase or choose different container. Press [q] to stop encoding video:244774kB audio:39224kB global headers:1kB muxing overhead 0.463570% [libx264 @ 00A50410]slice I:255 Avg QP:17.75 size: 13992 [libx264 @ 00A50410]slice P:59943 Avg QP:19.87 size: 4122 [libx264 @ 00A50410]mb I I16..4: 30.5% 0.0% 69.5% [libx264 @ 00A50410]mb P I16..4: 7.0% 0.0% 6.1% P16..4: 41.4% 17.9% 4.9% 0.9% 0.6% skip:21.1% [libx264 @ 00A50410]ref P 82.1% 17.9% [libx264 @ 00A50410]SSIM Mean Y:0.9850856 [libx264 @ 00A50410]kb/s:798.6 -------- The resulting video outputs an error in quicktime, and can't be added into iTunes. (plays fine with VLC, but VLC is ok with many wrong videos ;) The problem lies with this: >WARNING codec timebase is very high. If duration is too long, >file may not be playable by quicktime. Specify a shorter timebase >or choose different container. Even if the timebase seems to be detected correctly, it seems it's not set correctly when giving it to the x264 encoder. I looked at the sources, and this message appears if the timebase is > 100000 which should not be the case [as it's 23.98 (23976024/1000000)]. There is no problems with the same video, same parameters except using libxvid codec instead of libx264 (the previous error message is not here, and the video plays fine in quicktime and itunes), so my guess is that it comes from the way the timbase is given to the x264 encoder (there's great chance that it comes from the 23976024/1000000 decimal which is not handled well). If it isn't the right place to report this bug, please tell me where I can do so ;-) Thanks. Noda

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Mac OS X compilaton
by techie＠malus.ca 04 Feb '08

04 Feb '08

Hello! I'm sorry, surely this has been asked and answered before... But I can't seem to find something that will work for me... I'd like to compile ffmpeg with faac, faad and lame on Mac OS X as a universal binary (so I assume I have to flag the faac, faad and lame libs too) and be able to move the resulting ffmpeg to any computer wihtout having to re-install all the libs - so basically a standalone ffmpeg... I've found a few pages with nnstructions - but either I can't compile everything, or, if I can, then the resulting package is not universal and won't work on PPC... I'm compiling form an Intel Mac)... Anyone can help me with this? Thanks! Jean

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ffplay won't compile on x86_64
by anze.vidmar＠interseek.si 04 Feb '08

04 Feb '08

Hello! I have troubles compiling ffplay from the latest svn. My configure/compile flags are: ./configure --enable-libvorbis --enable-libvorbis --enable-shared --enable-libmp3lame --extra-cflags=-I/include --prefix=/root/anze/ffmpeg/tmp/ That's all good since ffmpeg compiles with all the wanted support. The problem is because ffplay binary won't compile, nor do I receive any errors about it. This is happening *only* on x86_64 platform. On 32bit system ffplay compiles without a problem (from the same svn). So, what do I need to do to make ffplay compile on x86_64? Thanks and best regards, Anze

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Offset the screen capture time frame
by shawn.molloy＠gmail.com 04 Feb '08

04 Feb '08

Hello, I'm trying to capture a frame of a video that is say 10 seconds into the video. I see a switch called "itsoffset" in the docs. It looks like my syntax is correct (a negative number is actually forward in the video) but whatever I change that number to it still take the very first frame screen capture. Here is my command text: -vcodec png -vframes 1 -an -itsoffset -16 -f rawvideo -s 320x240 " + Path.ChangeExtension(targetPath_ss, ".png"); Thank you for taking a look at my problem. --shawn

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increase audio volume durinng the encoding
by airteknology＠gmail.com 02 Feb '08

02 Feb '08

Hi guys , I'm using ffmpeg for convert some avi to mpg with good result. The only problem is that the audio volume is very low. I use this oprion for encode the files: ffmpeg -i /root/video/sorgenti/xvid_virtualdub/1-10minuti.avi -vcodec mpeg2video -r 25.00 -aspect 4:3 -b 4000k -mbd rd -flags +trell -mv0 -cgop -cmp 2 -subcmp 2 -acodec mp2 -ab 192k -ar 48000 -ac 2 -s 720x576 -y /root/video/compressi/ffmpeg_agency/mp2/xvid_virtualdub/1-10minuti.mpg There is a command for increase audio volume? Thanks for your help Pierpaolo

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DVCPRO 50 input
by christophelorenz＠gmail.com 01 Feb '08

01 Feb '08

I'm trying to read an avi in DVCPRO 50 pal format fourcc DV50 in ffmpeg, but it doens't recognise it. I tried changing the fourcc to dv5p without success. Someone knows if dvcpro 50 is supported in avi ? (File comes from an EVS media exchange.) Thanks, Chris. ffmpeg -i "dvcpro50pal.avi" -bufsize 2000k -b 7600k -maxrate 8000k -vcodec mp eg2video output.m2v FFmpeg version SVN-r11629, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-memalign-hack --enable-gpl --enable-pp --enable-swscaler --cpu=i686 libavutil version: 49.6.0 libavcodec version: 51.49.0 libavformat version: 52.5.0 libavdevice version: 52.0.0 built on Jan 27 2008 11:40:40, gcc: 4.2.1 [Sherpya] Input #0, avi, from 'dvcpro50pal.avi': Duration: 00:00:05.1, start: 0.000000, bitrate: 60689 kb/s Stream #0.0: Video: dv50 / 0x30357664, 720x576 [PAR 0:1 DAR 0:1], 25.00 tb(r) Stream #0.1: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s Stream #0.2: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s swScaler: Unknown format is not supported as input format Cannot get resampling context

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How to calculate muxrate for mpeg2 ?
by christophelorenz＠gmail.com 01 Feb '08

01 Feb '08

I'd like to know of someone knowns how to calculate a realistic muxrate for mpeg2 ? I guess mux rate is something like = video bitrate + audio bitrate + codecs overhead + container overhead I know the two first ones, but the others.... So far, I've seen in the api the stream have a Stream->codec->extradata_size variable probably the codec overhead. But I found nothing regarding the mux overhead. Seems to be linked to packetsize somehow, but no idea how to estimate it. Chris.

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Audio problem
by ariel.amato.gm＠gmail.com 01 Feb '08

01 Feb '08

Hello, i'm having this problem when i try to convert a file with ffmpeg this is the command i execute: ffmpeg -i /home/files/input.flv -r 24 -b 1300k -s 704x576 -ac 1 -deinterlace -y /home/files/output.flv and i got this error: FFmpeg version SVN-r11735, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-liba52bin --enable-x11grab --enable-libx264 --enable-libfaad --enable-libfaac --enable-gpl --enable-pp --enable-swscaler --enable-libmp3lame libavutil version: 49.6.0 libavcodec version: 51.50.0 libavformat version: 52.7.0 libavdevice version: 52.0.0 built on Feb 1 2008 09:15:32, gcc: 3.4.6 20060404 (Red Hat 3.4.6-9) [flv @ 0x8360260]Unsupported audio codec (ffffffff) [flv @ 0x8360260]Unsupported video codec (ffffffff) [flv @ 0x8360260]Could not find codec parameters (Audio: 0xffffffff) Input #0, flv, from '/home/files/input.flv': Duration: 00:00:00.0, start: 0.000000, bitrate: N/A Stream #0.0: Video: flv, yuv420p, 352x211 [PAR 0:1 DAR 0:1], 1000.00tb(r) Stream #0.1: Audio: 0xffffffff Output #0, flv, to '/home/files/output.flv': Stream #0.0: Video: flv, yuv420p, 704x576 [PAR 0:1 DAR 0:1], q=2-31, 1300 kb/s, 24.00 tb(c) Stream #0.1: Audio: libmp3lame, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Error while opening codec for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height I continued investigating and found a site that stated that the error was due to the audio stream of the output and suggested that i added "-ar 44100" as a parameter for the output file, which id did, the result is the following FFmpeg version SVN-r11735, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-liba52bin --enable-x11grab --enable-libx264 --enable-libfaad --enable-libfaac --enable-gpl --enable-pp --enable-swscaler --enable-libmp3lame libavutil version: 49.6.0 libavcodec version: 51.50.0 libavformat version: 52.7.0 libavdevice version: 52.0.0 built on Feb 1 2008 09:15:32, gcc: 3.4.6 20060404 (Red Hat 3.4.6-9) [flv @ 0x8360260]Unsupported audio codec (ffffffff) [flv @ 0x8360260]Unsupported video codec (ffffffff) [flv @ 0x8360260]Could not find codec parameters (Audio: 0xffffffff) Input #0, flv, from '/home/files/input.flv' : Duration: 00:00:00.0, start: 0.000000, bitrate: N/A Stream #0.0: Video: flv, yuv420p, 352x211 [PAR 0:1 DAR 0:1], 1000.00tb(r) Stream #0.1: Audio: 0xffffffff Output #0, flv, to '/home/files/outputt.flv ': Stream #0.0: Video: flv, yuv420p, 704x576 [PAR 0:1 DAR 0:1], q=2-31, 1300 kb/s, 24.00 tb(c) Stream #0.1: Audio: libmp3lame, 44100 Hz, mono, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Unsupported codec (id=0) for input stream #0.1 ffmpeg is a built form svn sources. Thanks in advance for any ideas that may help Ariel

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FLV quality and EXIF preservation
by emailgrant＠gmail.com 01 Feb '08

01 Feb '08

I'm converting .mov files to .flv and I'm wondering if there is a way to increase the video quality? I've tried setting a higher video bitrate with -b but it doesn't seem to make a difference. Is there any way to get ffmpeg to copy the EXIF data found in the original .mov file to the new .flv file? - Grant

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