August 2007 - ffmpeg-user - lists.ffmpeg.org

Ffmpeg aac audio encoding limited to 152 kbps
by michael＠gideoncorp.com 28 Aug '07

28 Aug '07

Before I get flamed for asking an obvious question, please understand that I am not asking for step-by-step help in resolving anything. I am *not* running the absolute latest SVN of ffmpeg, because this is for a production environment where we cannot take the risk of breaking something each day. I would just like a quick answer, if anyone is familiar with this issue. When running the command to convert audio formats... ffmpeg -i "AudioInput.aif" -ab 160 "AudioOutput.m4a" ...the maximum bitrate I can get is 152 kbps. No matter whether I change the -ab value to "256", "160k", or "160000" (that last one doesn't work at all; incorrect parameters error). With any bitrate value less than 152, ffmpeg nails it. Is this some sort of limitation within ffmpeg? Libfaac? Something else? I have spent a great deal of time searching around for the answer to this before posting, but came up empty handed. So your consideration would be greatly appreciated, lest I unnecessarily begin taking this system apart piece by piece to locate the problem.... Thanks!

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Converting m4v to flv: Internal Buffer Inconsistency
by joem＠tempomg.com 28 Aug '07

28 Aug '07

Hi all, I've been dealing with this problem occurring with only some videos. Any advice on where I might be going wrong? Is this a generic error, or is there something specific going wrong here that I might be able to fix? FFmpeg version SVN-r9465, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-libfaad --enable-libmp3lame --enable-gpl libavutil version: 49.4.1 libavcodec version: 51.40.4 libavformat version: 51.12.1 built on Jul 3 2007 12:28:46, gcc: 4.0.2 20051125 (Red Hat 4.0.2-8) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Flannelgraph_Part_12_0.m4v': Duration: 00:22:39.5, start: 0.000000, bitrate: 443 kb/s Stream #0.0(eng): Video: h264, yuv420p, 320x240, 29.97 fps(r) Stream #0.1(eng): Audio: mpeg4aac, 44100 Hz, stereo Output #0, flv, to 'Flannelgraph_Part_12_0.flv': Stream #0.0: Video: flv, yuv420p, 300x224, q=2-31, 320 kb/s, 25.00 fps(c) Stream #0.1: Audio: libmp3lame, 22050 Hz, stereo, 96 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding Internal buffer inconsistency. flushbits <> ResvSizebit reservoir error: /s l3_side->main_data_begin: 1536 Resvoir size: -98304 resv drain (post) -4 resv drain (pre) 0 header and sideinfo: 168 data bits: 2389 total bits: 2553 (remainder: 1) bitsperframe: 2512 This is a fatal error. It has several possible causes:90 0XB.7B85008000009DP-16385ME compiled with buggy version of gcc using advanced optimizations 9% Your system is overclocked 1% bug in LAME encoding libraryInternal buffer inconsistency. flushbits <> ResvSizeInternal buffer inconsistency. flushbits <> ResvSize./convert: line 3: 8111 Segmentation fault ffmpeg -i $1.m4v -s 300x224 -r 25 -b 320k -ab 96k -ar 22050 $1.flv Thanks for the help.

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how to encoded sevc audio codec (mobile phone - samsung) ?
by romeo74＠gmail.com 28 Aug '07

28 Aug '07

Hi all I have a one problem. How can I encoded sevc audio codec to flv? Source movie have a mpeg4 video and sevc audio codec. Help me~ please~ This is error messages. FFmpeg version SVN-r10253, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-libmp3lame --enable-libogg --enable-libvorbis -- disable-mmx --enable-shared --enable-liba52 --enable-liba52bin --enable- libamr-nb --enable-libamr-wb --enable-libfaac --enable-libfaad --enable- libfaadbin --enable-libxvid --enable-gpl --enable-libx264 libavutil version: 49.5.0 libavcodec version: 51.42.0 libavformat version: 51.12.2 built on Aug 28 2007 11:17:53, gcc: 4.1.1 20070105 (Red Hat 4.1.1-52) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.skm': Duration: 00:03:48.2, start: 0.000000, bitrate: 140 kb/s Stream #0.0(eng): Video: mpeg4, yuv420p, 176x144, 15.00 fps(r) Stream #0.1(eng): Audio: sevc / 0x63766573, 8000 Hz, stereo File 'a.flv' already exists. Overwrite ? [y/N] y Output #0, flv, to 'a.flv': Stream #0.0(eng): Video: flv, yuv420p, 176x144, q=2-31, 1000 kb/s, 30.00 fps(c) Stream #0.1(eng): Audio: libmp3lame, 8000 Hz, stereo, 0 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Unsupported codec (id=0) for input stream #0.1

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64bit install problem
by parker23＠cc.wwu.edu 28 Aug '07

28 Aug '07

Good afternoon, I'm having some difficultly in trying to get the latest (today's) svn build compiling on a new 64bit server of mine. I think i have narrowed the problem to the --enable-shared option which I have heard has troubles with 64 bit architecture. I have configured and installed ffmpeg a handful of times both success stories and complete failures. In short I am trying to get mp3 encoding working on the 64 bit machine; while it works in the 32 bit platform the only difference is the --enable-shared option along with a few others that I don't think are needed (enable-pthreads, enable-pp, etc). ffmpeg -formats | grep mp3 FFmpeg version SVN-r10214, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-gpl --enable-pp --enable-pthreads --enable-libogg --enable-libmp3lame --enable-libxvid --enable-libx264 libavutil version: 49.5.0 libavcodec version: 51.42.0 libavformat version: 51.12.2 built on Aug 24 2007 17:35:23, gcc: 4.1.2 20070626 (Red Hat 4.1.2-13) DE mp3 MPEG audio layer 3 EA libmp3lame D A mp3 D A mp3adu D A mp3on4 The error I am getting from making the build is in mathematics.o and it is: /usr/bin/ld: mathematics.o: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC mathematics.o: could not read symbols: Bad value collect2: ld returned 1 exit status make[1]: *** [libavutil.so.49] Error 1 Any advice or input would be greatly appreciated. First time posting to the mailing list but I've been reading it for quite some time...hopefully I'm not continuing a dead topic. Thanks in advance, Justin

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very newbie question about two-pass ffmeg and resizing videos
by johnwren＠anet.net.th 28 Aug '07

28 Aug '07

I posted the following on the mencoder list, but have been correctly referred here. I must apologize for my ignorance of ffmpeg and video encoding generally. I really am new in this area. I have two questions, one about doing two-pass conversions, and the other about the size flag. Note that is the copy of the post below I do realize ffmpeg has highlighted the error,: [mpeg4 <at> 0x2b29c483c6d0]requested bitrate is to low but I simply don't understand. In the one-pass conversion I did not specify the bit rate (nor actually do I understand the implications of the 'bit rate'), and don't understand why it failed on the two-pass mode. I am learning, but really could use some gentle guidance. Thanks ------ copy of mencoder post ---- My immediate question was posted on a ubuntu linux (Feisty Fawn, 7.04 AMD64) forum, and was about ffmpeg rather than mencoder. Both are working on my system, and an answer to the question using either would be fine. I am an intermediate linux user, but a newbie with video conversion. Here is what I posted on the ubuntu forum: I am working on an instructional webpage and want to show some videos. We have chosen quicktime containers (either .mp4 or .mov) to support as many systems as possible. One objective of the project is to use only open-source software. The videos were captured using Kino and exported as mpeg (DVD) files. I'm learning to use ffmpeg to encode them to .mov containers. My system is feisty AMD64 and has working copies of ffmpeg (from medibuntu) mencoder and mplayer. The following works fine: ffmpeg -i test.mpg -sameq test4.mov But I am having trouble doing two pass encoding. I think I just don't understand the syntax for two-pass work. I tried the following: ffmpeg -i test.mpg -sameq -pass 1 temp.mov ffmpeg -i test.mpg -sameq -pass 2 test.mov The ffmpeg2pass-0.log is created on the first pass, but the second pass dies with johnwren <at> johnwren-AMD64:~/test$ ffmpeg -i test.mpg -sameq -pass 2 test.mov FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2004 Fabrice Bellard configuration: --enable-gpl --enable-pp --enable-pthreads --enable-vorbis --enable-libogg --enable-a52 --enable-dts --enable-libgsm --enable-dc1394 --disable-debug --enable-mp3lame --enable-faadbin --enable-faad --enable-faac --enable-xvid --enable-x264 --enable-amr_nb --enable-amr_wb --enable-shared --prefix=/usr libavutil version: 0d.49.0.0 libavcodec version: 0d.51.11.0 libavformat version: 0d.50.5.0 built on Mar 25 2007 22:39:13, gcc: 4.1.2 (Ubuntu 4.1.2-0ubuntu4) Input #0, mpeg, from 'test.mpg': Duration: 00:01:21.6, start: 0.184656, bitrate: 2656 kb/s Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 720x576, 7500 kb/s, 25.00 fps(r) Stream #0.1[0x1c0]: Audio: mp2, 48000 Hz, stereo, 192 kb/s Output #0, mov, to 'test.mov': Stream #0.0: Video: mpeg4, yuv420p, 720x576, q=2-31, pass 2, 200 kb/s, 25.00 fps(c) Stream #0.1: Audio: aac, 48000 Hz, stereo, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 [mpeg4 <at> 0x2b29c483c6d0]removing common factors from framerate [mpeg4 <at> 0x2b29c483c6d0]requested bitrate is to low Error while opening codec for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height

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need some hints to encode h264 level 1b video formy cell
by stefan_ell＠hotmail.com 27 Aug '07

27 Aug '07

Hello Brian, thanks for this input! Well, using mpeg4 instead of h.264 as vcodec made the video playable on the cell, but since I still believe the SE K610i should be able to play at least h.264 in level 1b standart I gave your suggestion a try as well. So I installed gpac and tried to alter the 3gp video with MP4Box. I tried the following command line(s): ffmpeg -i inputfile.avi -acodec aac -vcodec h264 -b 128k -s qcif -r 15 outputfile.3gp ... to create the 3gp container with h2.64 and narrow it down to "level 1b" as good as I can and then: MP4Box -hint -isma -3gp outputfile.3gp ... It procesed very well (without errors), the altered file was a little bigger than the original one. But it refused to play on the cell again (linux still played it though). well yeah... it was worth a try... maybe the guys on doom9 forum are wrong and the cell is not able to process any video encoded in the h.264 standart. It would be great to really encode the video with "level 1b" corresponding macroblock settings, but I still don't know that much of ffmpeg to do so. but thanks for your input anyway! _________________________________________________________________ JETZT die neueste Version des Windows Live Messenger downloaden! http://get.live.com/de-ch/messenger/overview

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live encoding - limiting factors
by pitman.pete＠gmail.com 27 Aug '07

27 Aug '07

Hello, I am encoding from an mpeg ts video device to dv format: ffmpeg -i /dev/mydevice -vcodec dvvideo out.dv however, I get glitches in the output video, and errors on the command line like: [mpeg2video @ 0x84d1cc8]00 motion_type at 7 33 [mpeg2video @ 0x84d1cc8]invalid cbp at 1 34 [mpeg2video @ 0x84d1cc8]ac-tex damaged at 0 35 [mpeg2video @ 0x84d1cc8]Warning MVs not available [mpeg2video @ 0x84d1cc8]concealing 675 DC, 675 AC, 675 MV errors If I do: cat /dev/mydevice > myfile.ts then: ffmpeg -i myfile.ts -vcodec dvvideo out2.dv all is well. CPU and memory usage is low (ish ~ 30%), what other limiting factors are there? Many thanks for your help, Peter PS. Full cmd line: ffmpeg -i /dev/mydevice -vcodec dvvideo ~/out.dv FFmpeg version SVN-r10129, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --enable-liba52bin --enable-pp --enable-pthreads -- enable-gpl --disable-strip --extra-cflags=-O2 --extra-cflags=-g -- extra-cflags=-pipe --extra-cflags=-Wall --extra-cflags=-Wp,- D_FORTIFY_SOURCE=2 --extra-cflags=-fexceptions --extra-cflags=-fstack- protector --extra-cflags=--param=ssp-buffer-size=4 --extra-cflags=- m32 --extra-cflags=-march=i386 --extra-cflags=-mtune=generic --extra- cflags=-fasynchronous-unwind-tables libavutil version: 49.5.0 libavcodec version: 51.41.0 libavformat version: 51.12.2 built on Aug 20 2007 17:53:31, gcc: 4.1.2 20070502 (Red Hat 4.1.2-12) Input #0, mpegts, from '/dev/mydevice': Duration: N/A, bitrate: 8172 kb/s Stream #0.0[0x44](eng): Audio: mp2, 48000 Hz, stereo, 192 kb/s Stream #0.1[0x45]: Video: mpeg2video, yuv420p, 720x576, 7980 kb/s, 25.00 fps(r) Output #0, dv, to '/video/out.dv': Stream #0.0: Video: dvvideo, yuv420p, 720x576, q=2-31, 200 kb/s, 25.00 fps(c) Stream #0.1(eng): Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1

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64 bit compile fails
by ampenner＠gmail.com 27 Aug '07

27 Aug '07

Can anyone offer advice on compiling libavcodec as 64 bit? I configured (svn revision 10252) with --arch=x86_64, and the assembler fails. gcc -m64 -I"/export/home/apenner/junk/ffmpeg64"/libswscale -I"/export/home/apenner/junk/ffmpeg64"/libavcodec -DHAVE_AV_CONFIG_H -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_ISOC9X_SOURCE -I"ffmpeg64" -I"ffmpeg64" -I"ffmpeg64"/libavutil -m64 -fomit-frame-pointer -g -Wdeclaration-after-statement -Wall -Wno-switch -Wdisabled-optimization -Wpointer-arith -Wredundant-decls -O3 -c -o h264.o h264.c (lots of warnings omitted) /var/tmp//ccaqDPAL.s: Assembler messages: /var/tmp//ccaqDPAL.s:7271: Error: `ff_h264_lps_range(%eax,%esi,2)' is not a valid 64 bit base/index expression /var/tmp//ccaqDPAL.s:7284: Error: `ff_h264_norm_shift(%esi)' is not a valid 64 bit base/index expression /var/tmp//ccaqDPAL.s:7286: Error: `ff_h264_mlps_state+128(%eax)' is not a valid 64 bit base/index expression /var/tmp//ccaqDPAL.s:7298: Error: `-1(%ebx)' is not a valid 64 bit base/index expression (same few errors above repeated over and over...)

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[3gp @ 0xa216aea0]fatal error, input is not a single packet
by webmaster＠smsjava.de 27 Aug '07

27 Aug '07

Hello i'm working on muxing H.263 and GSM-AMR together to 3GP since two weeks now and have no clue, why it doesn't work ?-( I have a MP4 File: ~ # file 1187966953.mp4 1187966953.mp4: Apple QuickTime movie (unoptimized) ~ # mp4creator -list 1187966953.mp4 Track Type Info 2 audio AAC from .mov, 58.752 secs, 14 kbps, 8000 Hz 1 video MPEG-4 Unknown Profile(255), 58.533 secs, 872 kbps, 320x240 @ 5.637845 fps When i try to convert this to 3gp i'm getting follwing output: ~ # ffmpeg -i 1187966953.263 -i 1187966953.wav -vcodec h263 -r 12 -s qcif -b 64 -acodec amr_nb -ac 1 -ar 8000 -ab 12200 -map 0.0 -map 1.0 1187966953.3gp FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2007 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --enable-static --enable-shared --cc=i686-pc-linux -gnu-gcc --disable-mmx --disable-altivec --disable-debug --disable-audio-oss --disable-v4l --disable-v4l2 --disable-dv1394 --disable-opts --ena ble-libmp3lame --enable-libvorbis --enable-libogg --enable-liba52 --enable-xvid --enable-libogg --enable-libdts --enable-x264 --enable-libfaad --enable-libfaac --enable-amr-nb --enable-amr-wb --enable-amr-if2 --enable-gpl --enable-pp --disable-strip libavutil version: 49.4.0 libavcodec version: 51.40.2 libavformat version: 51.11.0 built on Aug 24 2007 18:44:31, gcc: 3.4.6 (Gentoo Hardened 3.4.6-r2, ssp-3.4.6-1.0, pie-8.7.10) Input #0, avi, from '1187966953.263': Duration: 00:00:58.5, start: 0.000000, bitrate: 62 kb/s Stream #0.0: Video: h263, yuv420p, 176x144, 15.00 fps(r) Input #1, wav, from '1187966953.wav': Duration: 00:00:58.6, start: 0.000000, bitrate: 256 kb/s Stream #1.0: Audio: pcm_s16le, 8000 Hz, stereo, 256 kb/s File '1187966953.3gp' already exists. Overwrite ? [y/N] Output #0, 3gp, to '1187966953.3gp': Stream #0.0: Video: h263, yuv420p, 176x144, q=2-31, 0 kb/s, 12.00 fps(c) Stream #0.1: Audio: amr_nb, 8000 Hz, mono, 12 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #1.0 -> #0.1 Press [q] to stop encoding [3gp @ 0xb6576ea0]fatal error, input is not a single packet, inplement a AVParser for it [last meaage repeats many times] frame= 701 fps=434 q=31.0 Lsize= 132kB time=58.4 bitrate= 18.4kbits/s video:41kB audio:89kB global headers:0kB muxing overhead 1.668541% The resulting file does not play in VLC or Quicktime Player. My System is a Gentoo With Kernel 2.6.20-hardened-r6. Does anyone have a suggestion why this does not work? With greetings from Germany David

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"rc buffer underflow" message
by rachid＠mbdsys.com 27 Aug '07

27 Aug '07

Hi, I built ffmpeg, and when i make a video call: I have a good video quality, but every time: i have "rc buffer underflow" message in my console. you have any idea what this message means, and how i can disable it?? Thanks, Rachid.

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