November 2008 - ffmpeg-user - lists.ffmpeg.org

ffmpeg: symbol lookup error: /usr/lib/libavcodec.so.52: undefined symbol: av_lfg_init
by rickardo.net＠gmail.com 20 Nov '08

20 Nov '08

Hello, Tanim Islam. I use SuSE 10.3 and have ffmpeg installed on my system. Yesterday, I needed to use it and this same message was showed to me: "ffmpeg: symbol lookup error: /usr/lib/libavcodec.so.52: undefined symbol: av_lfg_init". The solution was to install the related library 'libavcodec' using YaST. Probably, you have just solved your problem. If not, maybe, you can do it in the same way with Ubuntu. Best regards, Rick.

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help with command line to convert video to be played on disney mix max
by hodder.jeremy＠gmail.com 20 Nov '08

20 Nov '08

i need help to convert a recorded cartoon into a playable format for my daughter's Disney Mix Max This file plays on the Disney Mix Max Complete name : C:\Users\spike\Documents\THE LAND BEFORE TIME.wmv Format : Windows Media File size : 65.5 MiB Overall bit rate mode : Constant Maximum Overall bit rate : 522 Kbps Encoded date : UTC 2008-11-13 11:55:29.950 Video Format : VC-1 Format profile : SP Codec ID : WMV3 Codec ID/Info : Windows Media Video 9 Codec ID/Hint : WMV3 Bit rate mode : Constant Bit rate : 384 Kbps Width : 220 pixels Height : 164 pixels Display aspect ratio : 4/3 Frame rate : 30.000 fps Resolution : 24 bits Scan type : Progressive Bits/(Pixel*Frame) : 0.355 Language : en-us Audio Format : WMA2 Format profile : L2 Codec ID : 161 Codec ID/Info : Windows Media Audio 2 Description of the codec : Windows Media Audio 9.2 - 128 kbps, 48 kHz, stereo (A/V) 1-pass CBR Bit rate mode : Constant Bit rate : 128 Kbps Channel(s) : 2 channels Sampling rate : 48.0 KHz Resolution : 16 bits Language : en-us C:\Users\spike\Desktop\ffmpeg-15815>ffmpeg.exe -i "C:\Users\hodder\Documents\TH E LAND BEFORE TIME.wmv" FFmpeg version SVN-r15815, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-memalign-hack --enable-postproc --enable-swscale --ena ble-gpl --enable-libfaac --enable-libfaad --enable-libgsm --enable-libmp3lame -- enable-libvorbis --enable-libtheora --enable-libx264 --enable-libxvid --disable- ffserver --disable-vhook --enable-avisynth --enable-pthreads libavutil 49.12. 0 / 49.12. 0 libavcodec 52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 6. 1 / 0. 6. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 13 2008 10:28:29, gcc: 4.2.4 (TDM-1 for MinGW) [wmv3 @ 0x146f4e0]Extra data: 8 bits left, value: 0 Seems stream 1 codec frame rate differs from container frame rate: 1000.00 (1000 /1) -> 30.00 (30/1) Input #0, asf, from 'C:\Users\spike\Documents\THE LAND BEFORE TIME.wmv': Duration: 00:17:53.75, start: 5.000000, bitrate: 512 kb/s Stream #0.0: Audio: wmav2, 48000 Hz, stereo, s16, 128 kb/s Stream #0.1: Video: wmv3, yuv420p, 220x164, 384 kb/s, 30.00 tb(r) Must supply at least one output file C:\Users\spike\Desktop\ffmpeg-15815> thank you for your help Jeremy Hodder

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Problem reading in JPEG files
by dennis.volodomanov＠conceiva.com 20 Nov '08

20 Nov '08

Hello, I'm getting an error trying to read in a jpeg file. Maybe my build is wrong (missing something), can you please advise me what the problem could be? ffmpeg -i photos/cimg3627.jpg movie.mp4 FFmpeg version UNKNOWN, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-gpl --enable-shared --disable-static --enable-memalign-hack --enable-libmp3lame --extra-cflags=-I/c/msys/local/include --extra-ldflags=-L/c/msys/local/lib --disable-mmx --enable-libamr-nb --enable-libamr-wb --enable-libfaac --enable-nonfree libavutil 49.12. 0 / 49.12. 0 libavcodec 52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 built on Nov 19 2008 14:46:00, gcc: 4.2.1-sjlj (mingw32-2) [image2 @ 0x2c9dc0]Could not find codec parameters (Video: mjpeg) photos/cimg3627.jpg: could not find codec parameters Thank you in advance, Dennis

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YUV file input?
by mark＠image-engine.com 20 Nov '08

20 Nov '08

Hi, I see that from the ffmpeg docs I can encode movies from either a "raw" planar yuv 4:2:0 file containing multiple frames (e.g. "input.yuv"), or from a sequence of files which have the Y/U/V planes split into individual files (e,g. "input.1.Y ", "input.1.U", "input.1.V", "input.2.Y", .....). Is it possible to specify an input sequence which has the Y/U/V planes together in the same file for each frame, but with a separate file for each frame (e.g. "input.1.yuv", "input.2.yuv" )? Currently I'm having to "cat" all my individual frames together, but I'd like to avoid the need for this huge intermediate file if possible. Thanks in advance, Mark

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Problem with help output
by dennis.volodomanov＠conceiva.com 20 Nov '08

20 Nov '08

> > Does anyone know what's wrong? > > It's a bug. Workaround: compile FFmpeg with --enable-swscale. Thank you, I'll try that. Dennis

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Problem with help output
by dennis.volodomanov＠conceiva.com 20 Nov '08

20 Nov '08

Hello, For some strange reason, I cannot get the full help output from ffmpeg (or it ends in a strange way in any case) - the last line I see is: imgresample AVOptions: And then nothing... This is my configuration: FFmpeg version UNKNOWN, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-gpl --enable-shared --disable-static --enable-memalign-hack --enable-libmp3lame --extra-cflags=-I/c/msys/local/include --extra-ldflags=-/c/msys/local/lib --disable-mmx libavutil 49.12. 0 / 49.12. 0 libavcodec 52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 built on Nov 20 2008 10:19:07, gcc: 4.2.1-sjlj (mingw32-2) Does anyone know what's wrong? Thanks in advance, Dennis

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Audio stutter when play FFmpeg generated Transport Streams using VLC
by anuradha＠rd.bbc.co.uk 19 Nov '08

19 Nov '08

Hi, I used FFmpeg to transcode a DV file to MPEG2 video + MP2 audio muxed in MPEG TS. FFplay plays this back well but when I try playing back the file using VLC the audio stutters. I get lots of "PTS out of range" messages for audio. When I analysed the transport stream I found that the first audio packet appears about after 10 video packets and this seems to cause the stutter. I had to double the file cache size in VLC to make the audio stutter go away. The stutter does not appear to be affected by the video size. It is present even when the video is 176x144 and I am using a dual-core Intel celeron processor. I dug into the FFmpeg MPEG TS muxing code and found that the default PES packet payload size is set to 2930. Since video payload is usually larger than this, several video PES packets get written out to the output file before audio does and this explains the audio packet lag in the output. Has anybody else encountered this problem? I wasn't sure where to report this so I chose the safest option of this mailing list. Should I report it as a bug? Or should I report it in the FFmpeg-devel mailing list? BTW, I am using the latest source from the SVN repository. Regards, Anuradha

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Creating streaming 3gpp files
by dennis.volodomanov＠conceiva.com 19 Nov '08

19 Nov '08

Hello, Is it possible to create a streaming 3gpp file using ffmpeg? >From what I understand, it needs to overwrite some fourcc data in the file like this (correct me, if I'm wrong): hint -> (not sure, but > 2?) ftyp -> mmp4 psiz -> set maximum packet size here pdur -> set maximum packet duration here Can I do that somehow using the ffmpeg command line? Thanks in advance, Dennis

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lgpl license breach?
by evert＠rooftopsolutions.nl 19 Nov '08

19 Nov '08

I noticed 'Wii Transfer' contains an ffmpeg binary http://www.riverfold.com/software/wiitransfer/ Would this be a breach of the LGPL license? Evert

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MPEG2 PS with PCM audio
by eric.buehl＠gmail.com 19 Nov '08

19 Nov '08

Hello, I have several mpeg2 videos authored with PCM audio tracks. However, ffmpeg (presumably libav) does not detect the audio tracks. Is this a bug? I am using a recent git checkout. I have uploaded a sample video to: ftp://upload.mplayerhq.hu/MPlayer/incoming/mpeg2-pcm-misdetect.mpg Any help/pointers would be greatly appreciated. Thanks Eric

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DTS to AC3 conversion problem
by slimerang＠gmail.com 19 Nov '08

19 Nov '08

Trying to convert DTS to AC3 and the resulting AC3 file is missing channels are something. Most notable difference is that all of the dialog seems to be only out of the left channel. Surround sound stuff seems to be where it should be, at least on headphones using media player classic. I don't have anything to really analyze multichannel audio. $ ffmpeg -i thfro.dts -acodec ac3 -ab 640k thfro.ac3 FFmpeg version SVN-r15821, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-gpl --enable-shared --enable-libx264 --enable-libfaac --enable-libmp3lame libavutil 49.12. 0 / 49.12. 0 libavcodec 52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 built on Nov 18 2008 16:43:19, gcc: 4.1.2 20070925 (Red Hat 4.1.2-33) Input #0, dts, from 'thfro.dts': Duration: 02:12:49.52, bitrate: 1535 kb/s Stream #0.0: Audio: dca, 48000 Hz, 5.1, s16, 1536 kb/s Output #0, ac3, to 'thfro.ac3': Stream #0.0: Audio: ac3, 48000 Hz, 5.1, s16, 640 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding size= 633445kB time=8108.10 bitrate= 640.0kbits/s video:0kB audio:633445kB global headers:0kB muxing overhead 0.000000% MediaInfo for DTS file General Complete name : U:\tmp\thfro.dts Format : DTS Format/Info : Digital Theater Systems File size : 1.43 GiB Duration : 2h 12mn Overall bit rate : 1536 Kbps Audio Format : DTS Format/Info : Digital Theater Systems Bit rate mode : Constant Bit rate : 1536 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Surround: L R, LFE Sampling rate : 48.0 KHz Resolution : 24 bits MediaInfo for AC3 File: General Complete name : U:\tmp\thfro.ac3 Format : AC-3 Format/Info : Audio Coding 3 File size : 619 MiB Duration : 2h 15mn Overall bit rate : 640 Kbps Audio Format : AC-3 Format/Info : Audio Coding 3 Bit rate mode : Constant Bit rate : 640 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Rear: L R, LFE Sampling rate : 48.0 KHz -- Charles Curley, slimerang at gmail.com

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windows media 8
by ceissen＠delasource.com 19 Nov '08

19 Nov '08

Hi all does someone know which arguments are needed to have the best quality with windows media 8 codec ? gop ? trellis ? bframe ?...... thank you

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ffmpeg > lame: output buffer too small
by stephen＠enscripter.com 19 Nov '08

19 Nov '08

Hi all - sorry to bother... I have an issue whilst performing a file conversion. Is there any way i can increase the lame output buffer size in order to get this to work? Here is what i did... any ideas? FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2004 Fabrice Bellard configuration: --enable-gpl --enable-pp --enable-pthreads --enable-mp3lame --enable-shared --enable-vorbis --enable-libogg --enable-a52 --enable-dts --enable-libgsm --enable-dc1394 --disable-debug --enable-shared --prefix=/usr libavutil version: 0d.49.0.0 libavcodec version: 0d.51.11.0 libavformat version: 0d.50.5.0 built on Nov 5 2008 11:36:33, gcc: 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) Input #0, mp3, from '/virtual/resonate/files/orig/1227088149__it is for freedom.mp3': Duration: 00:08:50.2, start: 0.000000, bitrate: 160 kb/s Stream #0.0: Audio: mp3, 48000 Hz, stereo, 160 kb/s Output #0, mp3, to '/virtual/resonate/files/mp3/1227088149__it is for freedom.mp3': Stream #0.0: Audio: mp3, 48000 Hz, mono, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding size=78kB time=10.0 bitrate= 64.0kbits/s ... size=3159kB time=404.4 bitrate= 64.0kbits/s [mp3 @ 0xb7e579c8]lame: output buffer too small (buffer index: 2112, free bytes: 192) size=3195kB time=408.9 bitrate= 64.0kbits/s video:0kB audio:3195kB global headers:0kB muxing overhead 0.000000% ,

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Miscompiled libavcodec
by novice.program＠gmail.com 19 Nov '08

19 Nov '08

Hello, during every conversion i do with ffmpeg, i get this error that libavcodec has been compiled, the compiler did not align variables. I can't update my gcc version. I dont know the consequences of this thing as the vidoes are getting encoded fine. Any ideas on how to solve this error? -- Thanks & Regards, Novice.

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libfaac
by mscarpa＠arsenale23.com 19 Nov '08

19 Nov '08

ffmpeg -i test.avi -vcodec libx264 -b 1500000 -refs 3 -bf 3 -r 25.000 -acodec libfaac -ac 2 -ab 112000 -ar 44100 -y "pippo.mp4" FFmpeg version SVN-r15872, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-gpl --enable-postproc --enable-libvorbis --enable-libtheora --disable-debug --enable-libmp3lame --enable-libfaad --enable-libfaac --enable-pthreads --enable-libx264 --enable-libdc1394 --enable-libgsm --disable-debug --enable-pthreads --disable-mmx --enable-shared --enable-libtheora --enable-libamr-nb --enable-libamr-wb --enable-nonfree libavutil 49.12. 0 / 49.12. 0 libavcodec 52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 17 2008 21:08:53, gcc: 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) [NULL @ 0x806b4a0]Invalid and inefficient vfw-avi packed B frames detected Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (30000/1) -> 30.00 (30/1) Input #0, avi, from 'test.avi': Duration: 00:01:44.99, start: 0.000000, bitrate: 881 kb/s Stream #0.0: Video: mpeg4, yuv420p, 320x240 [PAR 1:1 DAR 4:3], 30.00 tb(r) Stream #0.1: Audio: pcm_u8, 11024 Hz, mono, s16, 88 kb/s Output #0, mp4, to 'pippo.mp4': Stream #0.0: Video: libx264, yuv420p, 320x240 [PAR 1:1 DAR 4:3], q=2-31, 1500 kb/s, 25.00 tb(c) Stream #0.1: Audio: libfaac, 44100 Hz, stereo, s16, 112 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 [libx264 @ 0x80ba9f0]using SAR=1/1 [libx264 @ 0x80ba9f0]using cpu capabilities: none! Press [q] to stop encoding Audio resampler only works with 16 bits per sample, patch welcome.

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[Ffmpeg-user] How to make large FLV files by ffmpeg
by jimmyq＠hz.webex.com 19 Nov '08

19 Nov '08

Hi, friends The duration of FLV files converted by ffmpeg can't exceed 16777 seconds, I converted an asf file with 532 minutes duration to FLV file, but I only get a file with 252 minutes duration, but target file is very large, so I thought only its duartion is not stored correctly. And if I convert the asf file to mp3 with ffmpeg, all is ok. I referd a link: http://www.impossibilities.com/v4/2008/08/29/a-problem-and-solution-for-flv… , it seems that the duration is limited by 24-bit tag, how to break the limit? I hope I can use some special parameters or some other tools to fix them. ffmpeg information is below: libavutil version: 49.6.0 libavcodec version: 51.51.0 libavformat version: 52.9.0 libavdevice version: 52.0.0 built on Sep 25 2008 11:21:02, gcc: 3.4.6 20060404 (Red Hat 3.4.6-3) FFmpeg SVN-r12406 libavutil 3212800 libavcodec 3355392 libavformat 3410176 libavdevice 3407872 My command line looks something like: ffmpeg -i input.flv output.flv I look forward your helping. Thanks -- View this message in context: http://www.nabble.com/How-to-make-large-FLV-files-by-ffmpeg-tp20535365p2053… Sent from the FFmpeg-users mailing list archive at Nabble.com.

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Samsung HMX20C - Mpeg4v2 Warnings - AAC Audio Errors
by wwwshadow7＠yahoo.com 19 Nov '08

19 Nov '08

With the sample MP4 clips from here: http://www.engadgethd.com/2008/06/17/samsung-sc-hmx20c-hd-camcorder-review/ $ ffmpeg -r 30000/1001 -i Samsung_Garden_Clip.MP4 -an -sameq -s hd720 -f mpegts -vcodec mpeg2video -r 30000/1001 -y sg_video.mpeg I can convert the video with a lot of these errors/warnings/info: [h264 @ 0x806c100]AVC: Consumed only 320668 bytes instead of 320682 [h264 @ 0x806c100]AVC: Consumed only 35008 bytes instead of 35020 [h264 @ 0x806c100]AVC: Consumed only 37301 bytes instead of 37316 ... [h264 @ 0x806c100]AVC: Consumed only 44203 bytes instead of 44212 [h264 @ 0x806c100]AVC: Consumed only 39995 bytes instead of 40004 [h264 @ 0x806c100]AVC: Consumed only 45290 bytes instead of 45300 frame=? 590 fps=? 9 q=0.0 Lsize=?? 76122kB time=19.65 bitrate=31730.1kbits/s???? video:59697kB audio:0kB global headers:0kB muxing overhead 27.513211% $ ffmpeg -i Samsung_Garden_Clip.MP4 -vn -acodec libfaac -y sg_audio.aac But I can NOT convert the audio only.? I get these errors/warnings: Error while decoding stream #0.1 Error while decoding stream #0.1 Error while decoding stream #0.1 ... Error while decoding stream #0.1 Error while decoding stream #0.1 Error while decoding stream #0.1 size=?????? 0kB time=10000000000.00 bitrate=?? 0.0kbits/s???? video:0kB audio:0kB global headers:0kB muxing overhead nan% So....? Should I worry about the video warnings?? I get decent results otherwise.? As you can see above, I got no audio extracted.? It's there and plays fine with mplayer with audio and video.? Baring the cpu limitations with the higher resolution clips. I'm basically shopping around for an HD camcorder and I'm finding it difficult to process the content / results from any of them. $file Samsung_Garden_Clip.MP4 Samsung_Garden_Clip.MP4: ISO Media, MPEG v4 system, version 2 $mplayer -vo dummy -ao alsa Samsung_Garden_Clip.MP4 (plays the audio just fine) Progress - mplayer shows faad and assumes 128kbps, so this works: $ ffmpeg -ab 128k -i Samsung_Garden_Clip.MP4 -vn -acodec copy -y sg_audio.aac This works(now anyway, updated to svn version this afternoon).? Except for audacity or sox not recognizing aac as an input format. $ faad -o sg_audio.wav sg_audio.aac This works too.? I guess the only concern now is should I worry about the information messages from the video extraction?? And I still can't seem to convert video and audio at the same time.? I get the audio errors if I try to do so.? And outside of an -acodec copy I get the errors with audio. A functional workflow for the Samsung seems to be: $ ffmpeg -r 30000/1001 -i Samsung_Boardwalk_Clip.MP4 -an -sameq -vcodec mjpeg -r 30000/1001 -y sg_video.avi $ ffmpeg -i Samsung_Boardwalk_Clip.MP4 -vn -acodec copy -y sg_audio.aac $ faad -o sg_audio.wav sg_audio.aac $ ffmpeg -i sg_video.avi -sameq -s hd720 -i sg_audio.wav -qmin 4 -f avi -vcodec msmpeg4v2 -y sg_new.mp4 Which puts it at the front of the pack for usability.? Without upgrading to a quad quad core cpu based system.? The warnings / information messages from the video conversion does concern me.? Am I losing quality?? Is that what those messages imply? - James

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VFW audio capture?
by kensystem＠gmail.com 18 Nov '08

18 Nov '08

Hi, I know there is VFW video capture support on MinGW, but is there an VFW audio capture method that anyone is using -- or is video capture presently a standalone realtime capture on this platform? -ken

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Compile FFMPEG for ARMv6
by martinhering＠mac.com 18 Nov '08

18 Nov '08

Hi, has anyone successfully compiled FFMPEG for ARMv6 with all assembler options enabled? Does anybody know which toolchain or linux distribution is required? Maybe somebody already published a tutorial? The one I found provided by the videolan project is a little bit outdated. Thank you for your answers. Martin

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stream size is not the same after transcoding
by erwan＠mas.nom.fr 17 Nov '08

17 Nov '08

Hello , When i convert a mp3 file to wav file the length is not the same , and i can not force it . ffmpeg -i file.mp3 -acodec pcm_s16le -y file.wav mp3 file is 00:02:03.14 and the wav is 123.09 aka 00:02:03.09 but when i specify a length , i have the same behavior , i obtain a different value . ffmpeg -i file.mp3 -acodec pcm_s16le -t 123.08 -y file.wav so with specify a length of : 123.08 i obtain 123.09 123.04 i obtain 123.06 123.00 i obtain 123.01 122.96 i obtain 122.98 122.92 i obtain 122.93 I want a length that is a factor of 0.04 ( aka 1/25 ) . Some tips ? -- ____________________________________________________________ / Erwan MAS /\ | mailto:erwan at mas.nom.fr |_/ ___|________________________________________________________ | \___________________________________________________________\__/

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Compile error on head revision
by jan-jaap＠jan-jaap.net 17 Nov '08

17 Nov '08

Hi, I'm trying to compile the head revision of ffmpeg. I've been doing that almost daily for quite a long time now, but as of today I'm getting the following errors: gcc -L"/home/jfa/src/ffmpeg"/libavfilter -L"/home/jfa/src/ffmpeg"/libpostproc -L"/home/jfa/src/ffmpeg"/libswscale -L"/home/jfa/src/ffmpeg"/libavdevice -L"/home/jfa/src/ffmpeg"/libavformat -L"/home/jfa/src/ffmpeg"/libavcodec -L"/home/jfa/src/ffmpeg"/libavutil -rdynamic -export-dynamic -Wl,--warn-common -Wl,--as-needed -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libpostproc -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libswscale -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavfilter -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavdevice -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavformat -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavcodec -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavutil -Wl,-Bsymbolic -o ffmpeg_g ffmpeg.o cmdutils.o -lavfilter -lpostproc -lswscale -lavdevice -lavformat -lavcodec -lavutil -lz -lbz2 -pthread -lm -lmp3lame -lm -ltheora -logg -lvorbisenc -lvorbis -logg -ldl -ldl /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_3dn' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_sse' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_3dn2' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_interleave_3dn' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_interleave_sse' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_interleave_3dn2' collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1 I'm compiling using gcc (GCC) 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) My configure options: ./configure --prefix=/usr --enable-shared --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-avfilter-lavf --enable-pthreads --enable-libtheora --enable-libvorbis --enable-libmp3lame Is the head revision botched, or should I adjust something to fix this? Regards, Jan-Jaap

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AVM2 SWF
by tcs2201＠gmail.com 17 Nov '08

17 Nov '08

Apologies in advance for the newbie question.. I am trying to use ffmpeg to convert .flv to an AVM2 .swf ffmpeg - c:\foo.flv c:\foo.swf This does convert the flv to an SWF, but Flex reports that it is an AVM1 (which prevents it being coerced to a MovieClip instance). Are there any flags that need setting to make SWF output AVM2? Some configuration? I have compiled from 2008-11-14 sourcecode. thanks, tom

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2 Pass libx264 encode from DV file choking
by beekerstudios＠gmail.com 17 Nov '08

17 Nov '08

I can't seem to work out this problem I am having. (sorry for the long message) My source video is a self-enclosed DV .mov (out of FCP) Executing these commands: /opt/local/bin/ffmpeg -y -i ${input_filename} -an -v 1 -threads 8 -vcodec libx264 -deinterlace -b 1372k -r 29.97 -bt 500k -me full -bf 3 -g 300 -maxrate 1500k -bufsize 2M -cmp 1 -s 640X480 -f mp4 -pass 1 /dev/null /opt/local/bin/ffmpeg -y -i ${input_filename} -v 1 -threads 8 -vcodec libx264 -deinterlace -b 1372k -r 29.97 -bt 500k -me full -bf 3 -g 300 -maxrate 1500k -bufsize 2M -cmp 1 -s 640X480 -acodec libfaac -ab 128k -ar 48000 -ac 2 -f mp4 -pass 2 ${process_dir}${rename_filename}_lan.mp4 It chokes on the second pass, and just stalls, and the CPU is still crunching on something. I have now attempted this on 2 separate video files, and even on a file I previously encoded just fine, but now does not encode correctly. It's important to note that 2 other encodes that happen in this same process (a "DSL" encode, and a "podcast" encode, go through just fine). For some reason my bitrate or my other flags or something are wrong here. After doing an update something broke (or got fixed depending on how you look at it), and I had to modify my command line arguments, fine, seems to have simplified things nonetheless. My previous command for this encode looked like this: /opt/local/bin/ffmpeg -y -i ${input_filename} -an -v 1 -threads auto -vcodec h264 -deinterlace -b 1372k -bt 500k -refs 1 -loop 1 -deblockalpha 0 -deblockbeta 0 -parti4x4 1 -partp8x8 1 -me full -subq 1 -me_range 21 -chroma 1 -slice 2 -bf 0 -level 30 -g 300 -keyint_min 30 -sc_threshold 40 -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.7 -qmax 51 -qdiff 4 -i_qfactor 0.71428572 -maxrate 1500k -bufsize 2M -cmp 1 -s 640X480 -f mp4 -pass 1 /dev/null /opt/local/bin/ffmpeg -y -i ${input_filename} -v 1 -threads auto -vcodec h264 -deinterlace -b 1372k -bt 500k -refs 1 -loop 1 -deblockalpha 0 -deblockbeta 0 -parti4x4 1 -partp8x8 1 -me full -subq 6 -me_range 21 -chroma 1 -slice 2 -bf 0 -level 30 -g 300 -keyint_min 30 -sc_threshold 40 -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.7 -qmax 51 -qdiff 4 -i_qfactor 0.71428572 -maxrate 1500k -bufsize 2M -cmp 1 -s 640X480 -acodec aac -ab 128 -ar 48000 -ac 2 -f mp4 -pass 2 ${process_dir}${rename_filename}_lan.mp4 Now heres the part where I tell you I installed this on a "yucky" os x box via macports (previously darwin ports), and you tell me "that's your problem, it's on OS X". Well fine you got me there, but if someone on here is using FFMPEG via macports on os x 10.5, please let me know how I may be able to troubleshoot and resolve this issue. It's important to note, that I was previously running 10.4, and had installed ffmpeg via macports with the old version of xcode (2.?), but even with leopard, and the new xcode, it still has the same behavior.

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Compile error on head revision
by jan-jaap＠jan-jaap.net 17 Nov '08

17 Nov '08

Hi, I'm trying to compile the head revision of ffmpeg. I'm getting the following errors: gcc -L"/home/jfa/src/ffmpeg"/libavfilter -L"/home/jfa/src/ffmpeg"/libpostproc -L"/home/jfa/src/ffmpeg"/libswscale -L"/home/jfa/src/ffmpeg"/libavdevice -L"/home/jfa/src/ffmpeg"/libavformat -L"/home/jfa/src/ffmpeg"/libavcodec -L"/home/jfa/src/ffmpeg"/libavutil -rdynamic -export-dynamic -Wl,--warn-common -Wl,--as-needed -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libpostproc -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libswscale -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavfilter -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavdevice -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavformat -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavcodec -Wl,-rpath-link,"/home/jfa/src/ffmpeg"/libavutil -Wl,-Bsymbolic -o ffmpeg_g ffmpeg.o cmdutils.o -lavfilter -lpostproc -lswscale -lavdevice -lavformat -lavcodec -lavutil -lz -lbz2 -pthread -lm -lmp3lame -lm -ltheora -logg -lvorbisenc -lvorbis -logg -ldl -ldl /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_3dn' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_sse' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_3dn2' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_interleave_3dn' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_interleave_sse' /home/jfa/src/ffmpeg/libavcodec/libavcodec.so: undefined reference to `ff_fft_dispatch_interleave_3dn2' collect2: ld returned 1 exit status make: *** [ffmpeg_g] Error 1 My configure options: ./configure --prefix=/usr --enable-shared --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-avfilter-lavf --enable-pthreads --enable-libtheora --enable-libvorbis --enable-libmp3lame Is the head revision botched, or am I making a mistake somewhere? Regards, Jan-Jaap

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padding + mod16
by tobiasbrockamp＠gmx.de 17 Nov '08

17 Nov '08

hi everybody, i want to encode 16/9 input files to a 4/3 output file, so i need to add borders for correct aspect ratio. my question is how the encoding process look like when i use padding parameters. f.e. i have a 1024x576 input file and i want 640x480 output. my command line will have the following parameters: "-aspect 4:3 -s 640x360 -padtop 60 -padbottom 60 -padcolor 000000" will ffmpeg first encode the video 640x360 and then add the borders? so would it be better to specify the resolution according to mod16 (640x352)? are the borders fixed in the videostream or is it a "flag" in the container? or does ffmpeg encode the full 640x480 video, what for my understanding will be inefficient because of the black borders... kind regards tobias

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