December 2005 - ffmpeg-user - lists.ffmpeg.org

Bus Error (SDL Parachute Deployed)
by wk＠wojciechkosma.art.pl 07 Jan '06

07 Jan '06

hello list, please forgive if it was discussed before (although i couldn`t find it) i`m new to ffmpeg, i wanted to test some movies with ffplay, but with every clip i get: ./ffplay -x 320 -y 240 b b: I/O error occured Usually that means that input file is truncated and/or corrupted. Fatal signal: Bus Error (SDL Parachute Deployed) any ideas? thanks, wojciech

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ffmpeg seg faults when reading .ac3
by fperloff＠mind.net 03 Jan '06

03 Jan '06

I want to use ffmpeg to resample a ac3 file to .wav format. I used the following command line: $ ffmpeg -i ~/w2k/sample.ac3 -f wav ./sample.wav And here's the error message: ffmpeg version CVS, build 3277056, Copyright (c) 2000-2004 Fabrice Bellard configuration: --prefix=/usr --libdir=/usr/lib64 --mandir=/usr/share/man --extra-cflags=-fPIC --enable-mp3lame --enable-libogg --enable-vorbis --enable-theora --enable-faad --enable-faac --enable-libgsm --enable-xvid --enable-x264 --enable-a52 --enable-a52bin --enable-dts --enable-pp --enable-shared-pp --enable-shared --enable-pthreads --enable-gpl --disable-opts --disable-strip built on Dec 9 2005 06:02:52, gcc: 4.0.2 20051125 (Red Hat 4.0.2-8) Segmentation fault I checked to see if all the libraries were installed: $ ldd /usr/bin/ffmpeg libavformat.so => /usr/lib64/libavformat.so (0x00000034c2b00000) libavcodec.so => /usr/lib64/libavcodec.so (0x00000034c2600000) libavutil.so => /usr/lib64/libavutil.so (0x00000034c2000000) libm.so.6 => /lib64/libm.so.6 (0x0000003f90d00000) libmp3lame.so.0 => /usr/lib64/libmp3lame.so.0 (0x0000003f29800000) libogg.so.0 => /usr/lib64/libogg.so.0 (0x0000003f28b00000) libvorbis.so.0 => /usr/lib64/libvorbis.so.0 (0x0000003f28f00000) libvorbisenc.so.2 => /usr/lib64/libvorbisenc.so.2 (0x0000003f29500000) libtheora.so.0 => /usr/lib64/libtheora.so.0 (0x00000034c2400000) libfaad.so.0 => /usr/lib64/libfaad.so.0 (0x0000003f91900000) libfaac.so.0 => /usr/lib64/libfaac.so.0 (0x0000003f28700000) libgsm.so.1 => /usr/lib64/libgsm.so.1 (0x00000034c1e00000) libxvidcore.so.4 => /usr/lib64/libxvidcore.so.4 (0x0000003f92100000) libz.so.1 => /usr/lib64/libz.so.1 (0x0000003f91300000) libdl.so.2 => /lib64/libdl.so.2 (0x0000003f90b00000) libpthread.so.0 => /lib64/libpthread.so.0 (0x0000003f91500000) libc.so.6 => /lib64/libc.so.6 (0x0000003f90800000) liba52.so.0 => /usr/lib64/liba52.so.0 (0x0000003f91700000) libpostproc.so.0.0.1 => /usr/lib64/libpostproc.so.0.0.1 (0x00000034c2200000) /lib64/ld-linux-x86-64.so.2 (0x0000003f90600000) I noticed that if I used ffmpeg to resample a wav to an mp3, I was successful. ffmpeg -i ~/ipod/other-sample.wav -f mp3 ./other-sample.mp3 I played both the sample.ac3 and the other-sample.wav with mplayer and both played normally. ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ I used the ffmpeg from the rpm from freshrpms. (I noticed that his build was dated Dec 09, 2005) I had tried to recompile ffmpeg from CVS but received an error message. x264.c: In function ?X264_init?: x264.c:222: error: ?struct <anonymous>? has no member named ?b_fast_pskip? make[1]: *** [x264.o] Error 1 make[1]: Leaving directory `/home/baba/src/ffmpeg/libavcodec' make: *** [lib] Error 2 My configure command was: ./configure --enable-mp3lame --enable-libogg --enable-vorbis --enable-theora --enable-faad --enable-faadbin --enable-faac --enable-libgsm --enable-xvid --enable-x264 --enable-a52 --enable-a52bin --enable-dts --enable-pp --enable-shared-pp --enable-shared --enable-pthreads --enable-gpl --disable-strip Here's the gcc I used: $ gcc -v Using built-in specs. Target: x86_64-redhat-linux Configured with: ../configure --prefix=/usr --mandir=/usr/share/man --infodir=/usr/share/info --enable-shared --enable-threads=posix --enable-checking=release --with-system-zlib --enable-__cxa_atexit --disable-libunwind-exceptions --enable-libgcj-multifile --enable-languages=c,c++,objc,java,f95,ada --enable-java-awt=gtk --with-java-home=/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/jre --host=x86_64-redhat-linux Thread model: posix gcc version 4.0.2 20051125 (Red Hat 4.0.2-8) And here's the kernel: $ uname -a Linux localhost.localdomain 2.6.14-1.1653_FC4 #1 Tue Dec 13 21:34:16 EST 2005 x86_64 x86_64 x86_64 GNU/Linux I searched through the archives on either compiling ffmpeg, or on my resampling issue, plus I looked through the FAQ and the docs, but I didn't find anything to help. Thanks for any new ideas, Fred

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MPEG4 to FLV
by chris.songer＠getblaze.com 02 Jan '06

02 Jan '06

Has anyone successfully encoded a FLV that has both video and sound. This is a followup of a post I sent yesterday.

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stream screen in realtime
by peter.boot＠gmail.com 30 Dec '05

30 Dec '05

Is it possible to capture video from the desktop screen, with something like xvidcap, then stream it in real time with ffserver ? I asked the question on the xvidcap forum, they had tried but had not been able to "open an ffmpeg pipe properly" http://sourceforge.net/forum/forum.php?thread_id=1409933&forum_id=278401

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libfaad.so.0 wtf?
by chris.songer＠getblaze.com 30 Dec '05

30 Dec '05

i tried installing faad2 so that i could convert to ACC from mpeg4. after compiling i look for the libfaad.so.0 and it is no where to be found... has anyone else ran into this problem? any fixes?

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Segfault during conversion of 8 bit grayscale avi files
by cjmar227＠yahoo.com 29 Dec '05

29 Dec '05

Hi All, I am writing to see if anyone has a suggestion on how to solve a problem that I have converting 8 bit uncompressed gray-scale avi files to another format like divx or msmpeg4v2 using ffmpeg. I am doing a research project in which I capture video using a high speed camera (19,000 fps @ 480 x 288 pixels) and I save the movie as a tiff stack (usually 2000 frames). I open this stack in ImageJ for further processing and from there I can save the stack as an uncompressed 8 bit grayscale avi file. These files are large (>500 MB for 100 ms) and I want to compress then into a format that can be easily read on different platforms. The problem is that ffmpeg segfault every time I try to load these files for conversion. Here is the typical command that I execute and the corresponding output: > ffmpeg -i Drop_formation_Near_Tip_900-300-300.avi -vcodec msmpeg4v2 pp.avi > ffmpeg version CVS, build 3277056, Copyright (c) 2000-2004 Fabrice Bellard configuration: built on Nov 10 2005 14:50:42, gcc: 3.3.6 Segmentation fault Does anyone has a suggestion on how to solve this problem? One thing to note is that ffmpeg works when I save the tiff stack as an RGB color avi movie. I do not do this often because ImageJ usually runs out of memory during the grayscale -> RGB conversion. The frame rates that I use range from 5 fps to 60 fps. The image (video) resolutions are typically 480 x 288, 512 x 512 or 926 x 288. Also mplayer has no problem playing the uncompressed avi files. Here is some info about my system: Processor: Pentium 4 2.8GHz Distribution: Slackware 10.2 Linux Kernel: 2.4.31 Any info will be greatly appreciated. Thanks Carlos Martinez --------------------------------- Yahoo! for Good - Make a difference this year.

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MPEG4 to FLV... no sound?
by chris.songer＠getblaze.com 29 Dec '05

29 Dec '05

I am trying to encode FLV's on the fly from MPEG4 format. The video is fine however i am not able to get any sound. Has anyone ever had a similar problem?

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converting realmedia to mpeg
by vkwyau＠gmail.com 29 Dec '05

29 Dec '05

Hi: I am trying to convert realmedia files to mpeg. I have tried mplayer with (mplayer -vo mpegpes -ao mpegpes file.rmvb) and that did not work. I wonder if there is a way to do the conversion with ffmpeg. Any tips/pointers much appreciated. thanks --Vincent

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ffmpeg seeking in a FIFO pipe
by overdrigzed＠gmail.com 29 Dec '05

29 Dec '05

I am using the ffmpeg CVS from 20051216 (the gentoo ebuild). I want to convert a .ogm with XviD video and vorbis audio to iPod compatible video. ffmpeg fails to read this type of file. on some files it will segfault, others it sees the input like this: Input #0, ogg, from 'infile.ogm': Duration: 00:21:41.2, start: 0.000000, bitrate: 1122 kb/s Stream #0.0, 1000000.00 fps: Video: 0x0000 Stream #0.1: Audio: vorbis, 48000 Hz, stereo, 112 kb/s Stream #0.2: Audio: vorbis, 48000 Hz, stereo, 64 kb/s then dies with: Unsupported codec (id=0) for input stream #0.0 In order to convert this file, I am trying to use mplayer to read uncompressed audio into a file, then pipe uncompressed video into an ffmpeg process that will read the uncompressed video from the pipe and uncompressed audio from the file, and encode the two together. I use: mplayer "${infile}" -vc dummy -vo null -ao "pcm:file=${audio}" to get the audio. I then create a pipe with mkfifo ${video}, and use the following two commands concurrently: mplayer "${infile}" -vo "yuv4mpeg:file=${video}" -ac dummy -ao null ffmpeg -acodec aac -ab 96 -i "${audio}" -vcodec xvid -b 300 -qmin 3 -qmax 10 -bufsize 4096 -g 300 -i "${video}" -s 320x240 "${outfile} While this works correctly if I write the uncompressed video to a file and then read that with ffmpeg, when I write it to a FIFO, ffmpeg fails: drigz at ayro ~ $ ffmpeg -acodec aac -ab 96 -i /tmp/rawaudio5931 -vcodec xvid -b 300 -qmin 3 -qmax 10 -bufsize 4096 -g 300 -i /tmp/rawvideo5931 -s 320x240 XIRS.mov ffmpeg version CVS, build 3342336, Copyright (c) 2000-2004 Fabrice Bellard configuration: --prefix=/usr --host=i686-pc-linux-gnu --mandir=/usr/share/man --infodir=/usr/share/info --datadir=/usr/share --sysconfdir=/etc --localstatedir=/var/lib --enable-shared-pp --enable-shared --disable-static --enable-mmx --disable-altivec --disable-debug --enable-mp3lame --enable-a52 --disable-a52bin --enable-audio-oss --enable-v4l --disable-dv1394 --disable-dc1394 --disable-pthreads --enable-xvid --enable-libogg --enable-vorbis --enable-theora --enable-dts --enable-network --enable-zlib --enable-ffplay --enable-faad --enable-faac --disable-faadbin --enable-gpl --enable-pp --disable-opts --build=i686-pc-linux-gnu built on Dec 23 2005 17:22:16, gcc: 3.4.4 (Gentoo 3.4.4-r1, ssp-3.4.4-1.0, pie-8.7.8) Input #0, wav, from '/tmp/rawaudio5931': Duration: 00:01:55.2, start: 0.000000, bitrate: 1411 kb/s Stream #0.0: Audio: pcm_s16le, 44100 Hz, stereo, 1411 kb/s /tmp/rawvideo5931: Error while opening file I ran ffmpeg in strace (just the end): open("/tmp/rawvideo5931", O_RDONLY|O_LARGEFILE) = 4 read(4, "YUV4MPEG2 W320 H240 F23993471:10"..., 32768) = 46 read(4, "FRAME\n\20\20\20\20\20\20\20\20\17\17\17\17\17\17\17\17"..., 32768) = 32768 _llseek(4, 0, 0xbfaec4a0, SEEK_SET) = -1 ESPIPE (Illegal seek) read(4, "\17\17\17\17\17\17\17\17\17\17\17\17\17\17\17\17\17\17"..., 32768) = 32768 close(4) = 0 write(2, "/tmp/rawvideo5931: Error while o"..., 44) = 44 exit_group(1) = ? It seems that it is trying to seek in the FIFO, which is causing the error. How can I avoid this, or alternatively - what other ways could I perform this task?

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Encoding for the sony clie with -f psp broken.
by drbob＠gmx.co.uk 29 Dec '05

29 Dec '05

Hello, I encode videos for my sony clie PDA using ffmpeg. mp4 files for the clie need the same special sony headers as the psp mp4 files. However the clie PDAs are *not* limited to 76800 total pixels for their mp4 files. Until recently I was using an older version (cvs ~ march 05) and everything was fine, however upon switching to a recent cvs version I now get this error when trying to encode at 480x320: [psp @ 0x6d8984]PSP mode need width*height < 76800 Could not write header for output file #0 (incorrect codec parameters ?) This is all very helpful for psp owners but is there any way to override it? I need the sony mp4 headers, but don't need the psp resolution limit. Any help appreciated. regards, drbob

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