September 2011 - FFmpeg-trac - lists.ffmpeg.org

#420(undetermined:new): Sound fragments after seeking
by FFmpeg 20 Jan '14

20 Jan '14

#420: Sound fragments after seeking --------------------------+-------------------------- Reporter: kaptnole | Type: defect Status: new | Priority: important Component: undetermined | Version: unspecified Keywords: | Blocked By: Blocking: | Reproduced: 1 Analyzed: 1 | --------------------------+-------------------------- While trying to use FFmpeg libs as a developer I found the following: When decoding sound from some compressed audio formats and doing a seek, the first decoded packet after seeking is buggy. I had this with some mp4 files with aac audio codec and some wmv files with wma audio codec. Looks like some leftovers from internal buffers - but I am using 'avcodec_flush_buffers'! That of course leads to nasty sound noise/clicks. See attached sourcecode. Basically it shows the first few bytes of the first decoded sound packet then seeks back and shows the first packets decoding again. If you use it with some wmv or mp4 file with sound you should see see effect that both outputs at 0 are different but should be equal. Just make sure the sound is not only silence at the first 10 packets. Or am I missing some other flush...whatever function here? I am developing under windows using the actual ffmpeg git version. Confirmed for 32 and 64 bit version, self compiled and downloaded autobuilds. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/420> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#353(build system:new): Failed compiling FFmpeg in OS X 10.6.8
by FFmpeg 16 Jan '14

16 Jan '14

#353: Failed compiling FFmpeg in OS X 10.6.8 --------------------------+--------------------- Reporter: joao | Type: defect Status: new | Priority: normal Component: build system | Version: 0.8 Keywords: compile | Blocked By: Blocking: | Reproduced: 0 Analyzed: 0 | --------------------------+--------------------- Hi guys, Over at the Homebrew project (OS X package manager) we're running into issues compiling and installing FFmpeg on OS X. You can see our ticket here: https://github.com/mxcl/homebrew/issues/6278 The compilation error is seen here: https://gist.github.com/1059859 Someone managed to make it fail a bit later by doing this: https://gist.github.com/1072836 Can you provide us any help in getting this to compile? Thanks. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/353> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#43(undetermined:new): JACK input device doesn't link on OS X
by FFmpeg 10 Dec '13

10 Dec '13

#43: JACK input device doesn't link on OS X --------------------------+--------------------- Reporter: nieder | Type: defect Status: new | Priority: normal Component: undetermined | Version: git Keywords: | Blocked By: Blocking: | Reproduced: 0 Analyzed: 0 | --------------------------+--------------------- This is copied from the old roundup #976 https://roundup.libav.org/issue976 On OS X, using --enable-indev=jack, the build fails with {{{ CC libavdevice/jack_audio.o /src/ffmpeg/libavdevice/jack_audio.c: In function 'audio_read_packet': /src/ffmpeg/libavdevice/jack_audio.c:277: warning: implicit declaration of function 'sem_timedwait' AR libavdevice/libavdevice.a LD libavdevice/libavdevice.52.dylib Undefined symbols: "_sem_timedwait", referenced from: _audio_read_packet in jack_audio.o ld: symbol(s) not found collect2: ld returned 1 exit status make: *** [libavdevice/libavdevice.52.dylib] Error 1 }}} sem_* is semaphore_* on OS X. -- Ticket URL: <http://ffmpeg.org/trac/ffmpeg/ticket/43> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#314(avdevice:new): OpenAL device does not work on OS X
by FFmpeg 10 Dec '13

10 Dec '13

#314: OpenAL device does not work on OS X ----------------------+-------------------------- Reporter: nieder | Type: enhancement Status: new | Priority: normal Component: avdevice | Version: git-master Keywords: | Blocked By: Blocking: | Reproduced: 0 Analyzed: 0 | ----------------------+-------------------------- configure is not set to find openal on OS X. The fix for the linker check is easy: add "-Wl,-framework,OpenAL" to the list of flags to check {{{ -enabled openal && { { for al_libs in "${OPENAL_LIBS}" "-lopenal" "-lOpenAL32"; do +enabled openal && { { for al_libs in "${OPENAL_LIBS}" "-lopenal" "-lOpenAL32" "-Wl,-framework,OpenAL"; do }}} I don't know how to fix the header check to look for multiple possibilities, but on OS X the header in the OpenAL framework is found in OpenAL/al.h (not AL/al.h like in linux). If I change the header check in configure and the includes in libavdevice/openal-dec.c, the build and linkage works well. However, running 'ffmpeg -f openal' as suggested in the manual fails with an malloc error, so this might just not be doable on OS X. {{{ nieder $ /usr/local/ffmpeg-git/bin/ffmpeg -list_devices true -f openal -i dummy out.ogg ffmpeg version N-31036-g65eae2a, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 27 2011 10:29:19 with gcc 4.0.1 (Apple Inc. build 5493) configuration: --prefix=/usr/local/ffmpeg-git --mandir=/usr/local /ffmpeg-git/share/man --disable-static --enable-shared --enable-gpl --enable-version3 --enable-postproc --enable-pthreads --enable-x11grab --enable-bzlib --enable-libcelt --enable-libopencore-amrnb --enable- libopencore-amrwb --enable-libdirac --enable-libfreetype --enable- libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-openal --enable-zlib --disable- debug --extra-cflags='-I/sw/lib/libspeex1/include -I/sw/include -I/usr/X11R6/include' --extra-ldflags='-L/sw/lib/libspeex1/lib -L/sw/lib -L/usr/X11R6/lib' libavutil 51. 10. 0 / 51. 10. 0 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 24. 0 / 2. 24. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 [openal @ 0x200a800] List of OpenAL capture devices on this system: [openal @ 0x200a800] Built-in Microphone dummy: Immediate exit requested nieder $ /usr/local/ffmpeg-git/bin/ffmpeg -f openal -i 'Built-in Microphone' out.ogg ffmpeg version N-31036-g65eae2a, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 27 2011 10:29:19 with gcc 4.0.1 (Apple Inc. build 5493) configuration: --prefix=/usr/local/ffmpeg-git --mandir=/usr/local /ffmpeg-git/share/man --disable-static --enable-shared --enable-gpl --enable-version3 --enable-postproc --enable-pthreads --enable-x11grab --enable-bzlib --enable-libcelt --enable-libopencore-amrnb --enable- libopencore-amrwb --enable-libdirac --enable-libfreetype --enable- libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-openal --enable-zlib --disable- debug --extra-cflags='-I/sw/lib/libspeex1/include -I/sw/include -I/usr/X11R6/include' --extra-ldflags='-L/sw/lib/libspeex1/lib -L/sw/lib -L/usr/X11R6/lib' libavutil 51. 10. 0 / 51. 10. 0 libavcodec 53. 7. 0 / 53. 7. 0 libavformat 53. 4. 0 / 53. 4. 0 libavdevice 53. 2. 0 / 53. 2. 0 libavfilter 2. 24. 0 / 2. 24. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 mpeg(49840,0xa025b720) malloc: *** error for object 0x1e2c0e0: incorrect checksum for freed object - object was probably modified after being freed. *** set a breakpoint in malloc_error_break to debug mpeg(49840,0xa025b720) malloc: *** error for object 0x1e2c0e0: incorrect checksum for freed object - object was probably modified after being freed. *** set a breakpoint in malloc_error_break to debug [openal @ 0x200a800] Error: ALC Invalid Device Bus error }}} -- Ticket URL: <http://ffmpeg.org/trac/ffmpeg/ticket/314> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#228(FFserver:new): FFserver: reordering of "<Stream>"s in config file makes difference
by FFmpeg 10 Dec '13

10 Dec '13

#228: FFserver: reordering of "<Stream>"s in config file makes difference ------------------------+---------------------------------------- Reporter: burek | Owner: baptiste Type: defect | Status: new Priority: important | Component: FFserver Version: git | Keywords: ordering items config file Blocked By: | Blocking: Reproduced: 0 | Analyzed: 0 ------------------------+---------------------------------------- The first config file (ffserver1.conf) made ffserver drop rtsp connections, with error message "'''[NULL @ 0x1be7ba0]AAC with no global headers is currently not supported.'''" After just a simple reordering of <Stream> items (ffserver2.conf), everything started to work properly? Is this expected behavior? :) -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/228> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#512(avutil:new): libavutil\common.h conversion warnings
by FFmpeg 05 Dec '13

05 Dec '13

#512: libavutil\common.h conversion warnings ---------------------------------------+--------------------------------- Reporter: DonMoir | Owner: michael Type: enhancement | Status: new Priority: minor | Component: avutil Version: unspecified | Keywords: Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 ---------------------------------------+--------------------------------- Would you please make a couple minor changes to libavutil\common.h. static av_always_inline av_const int16_t av_clip_int16_c(int a) { if ((a+0x8000) & ~0xFFFF) return (a>>31) ^ 0x7FFF; else return a; } the returns cause me a conversion warning because they need to be cast to int16_t and so I have to change this each time I get a new include for ffmpeg. change to: static av_always_inline av_const int16_t av_clip_int16_c(int a) { if ((a+0x8000) & ~0xFFFF) return (int16_t)((a>>31) ^ 0x7FFF); else return (int16_t)a; } cast return values to (int32_t) for the following: static av_always_inline av_const int32_t av_clipl_int32_c(int64_t a) { if ((a+0x80000000u) & ~UINT64_C(0xFFFFFFFF)) return (a>>63) ^ 0x7FFFFFFF; else return a; } also for this one in libavutil\rational.h cast return values to (int): static inline int av_cmp_q(AVRational a, AVRational b){ const int64_t tmp= a.num * (int64_t)b.den - b.num * (int64_t)a.den; if(tmp) return (int)(((tmp ^ a.den ^ b.den)>>63)|1); else if(b.den && a.den) return 0; else if(a.num && b.num) return (int)((a.num>>31) - (b.num>>31)); else return INT_MIN; } There is no other case I have come across the produces any warnings but these. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/512> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#282(FFmpeg:new): undefined reference to `_avdevice_register_all'
by FFmpeg 20 Nov '13

20 Nov '13

#282: undefined reference to `_avdevice_register_all' -------------------------+------------------------------------------------- Reporter: KSHawkEye | Owner: michael Type: defect | Status: new Priority: critical | Component: FFmpeg Version: git | Keywords: undefined reference Blocked By: | avdevice_register_all Reproduced: 0 | Blocking: | Analyzed: 0 -------------------------+------------------------------------------------- I'm getting: {{{ ffmpeg.c:4514: undefined reference to `_avdevice_register_all' }}} when --disable-static --enable-shared is used, it seems to only happen when --disable-static --enable-shared is used, static builds fine. I believe it is related to the leading underscore _ before avdevice_register_all but I can't confirm that. I'm building for Windows and the toolchain is not the issue. I gave this a critical priority because FFmpeg will not build a shared version for Windows. I'm unsure of what patch broke this, but the last proven working build I know of is git-39dbe9b. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/282> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#227(FFmpeg:new): Extend -ss and -t to support seeking to EOF time position (and backwards)?
by FFmpeg 20 Oct '13

20 Oct '13

#227: Extend -ss and -t to support seeking to EOF time position (and backwards)? --------------------------+--------------------------------------- Reporter: burek | Owner: michael Type: enhancement | Status: new Priority: wish | Component: FFmpeg Version: git | Keywords: seeking eof time position Blocked By: | Blocking: Reproduced: 0 | Analyzed: 0 --------------------------+--------------------------------------- Maybe, it would be useful if options -ss and -t could support something like this: ffmpeg -ss EOF-00:00:10 -i input ... (meaning the start time is 10 seconds before the end of the file). This would be useful in automated use of ffmpeg, where users don't need to set time positions absolutely (manually), but rather relatively to the end of file. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/227> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#113(avcodec:new): AAC - Audio object type 23 is not supported
by FFmpeg 19 Sep '13

19 Sep '13

#113: AAC - Audio object type 23 is not supported -------------------------+--------------------- Reporter: k_s_johnson | Type: defect Status: new | Priority: normal Component: avcodec | Version: git Keywords: AAC-LD | Blocked By: Blocking: | Reproduced: 0 Analyzed: 0 | -------------------------+--------------------- ffmpeg AAC codec (decoder) doesn't seem to support AAC-LD and libfaad or libfaad2 (which is documented to support AAC-LD) is no longer an audio codec option. So there is no way to decode AAC-LD format audio track anymore. However, vlc on windows seems to play the file ffmpeg build from git snapshot of a few days ago( ffmpeg-HEAD-e16665b) on Fedora 2.6.30.9-96.fc11.x86_6411. command line: ffmpeg -y -i po_arrive...mov -acodec pcm_s24le extracted.wav Error message is: [aac @ 0x1e9bfa0] Audio object type 23 is not supported. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/113> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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#232(undetermined:new): bogus error message with amr files: [amrnb @ 0x626e90] dtx mode not implemented.
by FFmpeg 11 Sep '13

11 Sep '13

#232: bogus error message with amr files: [amrnb @ 0x626e90] dtx mode not implemented. ------------------------------+-------------------------- Reporter: matteosistisette | Type: defect Status: new | Priority: critical Component: undetermined | Version: unspecified Keywords: | Blocked By: Blocking: | Reproduced: 0 Analyzed: 0 | ------------------------------+-------------------------- I've just compiled the latest version of ffmpeg from the SVN on a CentOS machine and when I try to decode amr audio files I get this error (repeated a lot of times): {{{ [amrnb @ 0x626e90] dtx mode not implemented. Update your FFmpeg version to the newest one from SVN. If the problem still occurs, it means that your file has a feature which has not been implemented.If you want to help, upload a sample of this file to ftp://upload.ffmpeg.org/MPlayer/incoming/ and contact the ffmpeg-devel mailing list. }}} On another box, an Ubuntu one, I have installed the ffmpeg package from the repositories and it works fine with the same files. Also, on yet another CentOS box I am using a several-year-old version of ffmpeg that was compiled ages ago which can decode the same amr files just fine. So either the error message is completely wrong and the real problem with the files is different, or there has been some regression in that a feature that had already been implemented ages ago has been dropped. -- Ticket URL: <https://ffmpeg.org/trac/ffmpeg/ticket/232> FFmpeg <http://ffmpeg.org> FFmpeg issue tracker

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