February 2014 - FFmpeg-trac - lists.ffmpeg.org

#3421(avformat:new): non-interleaved avi regression
by FFmpeg 11 Mar '14

11 Mar '14

#3421: non-interleaved avi regression -------------------------------------+------------------------------------- Reporter: cehoyos | Owner: Type: defect | Status: new Priority: important | Component: avformat Version: git- | Keywords: avi master | regression Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 -------------------------------------+------------------------------------- ffplay plays audio badly for one avi sample here because it fails detecting non-interleaved avi. This is a regression since ef8cc06d I will upload the sample, it is unfortunately very large. Every audio underrun is clearly audible, no problems with {{{ffmpeg}}}. {{{ $ ffplay TERA\ 2013-08-26\ 17-41-52-39.avi -autoexit ffplay version N-60928-g8f946ac Copyright (c) 2003-2014 the FFmpeg developers built on Feb 26 2014 15:07:15 with gcc 4.7 (SUSE Linux) configuration: --enable-gpl libavutil 52. 66.100 / 52. 66.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 33.100 / 55. 33.100 libavdevice 55. 10.100 / 55. 10.100 libavfilter 4. 2.100 / 4. 2.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 [avi @ 0x7f8998000920] Stream #0: not enough frames to estimate rate; consider increasing probesize Input #0, avi, from 'TERA 2013-08-26 17-41-52-39.avi': Duration: 00:00:30.33, start: 0.000000, bitrate: 589592 kb/s Stream #0:0: Video: fraps (FPS1 / 0x31535046), bgr24, 1920x1080, 60 tbr, 60 tbn, 60 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 192000 Hz, 2 channels, s16, 6144 kb/s ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 ALSA lib pcm.c:7339:(snd_pcm_recover) underrun occurred 0B f=0/0 30.33 A-V: -0.005 fd= 0 aq= 0KB vq= 0KB sq= 0B f=0/0 }}} Increasing the probesize makes the warning disappear but does not help with the audio problem. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3421> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3429(avcodec:new): fate-acodec-ra144 fails with cpuflags mmxext/sse
by FFmpeg 07 Mar '14

07 Mar '14

#3429: fate-acodec-ra144 fails with cpuflags mmxext/sse --------------------------------------+--------------------------------- Reporter: jamal | Owner: Type: defect | Status: new Priority: normal | Component: avcodec Version: git-master | Keywords: Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 --------------------------------------+--------------------------------- How to reproduce: {{{ $ make fate-acodec-ra144 V=1 CPUFLAGS="mmx" TEST acodec-ra144 /ffmpeg/src/tests/fate-run.sh fate-acodec-ra144 "" "" "/ffmpeg/build" 'enc_dec_pcm rm wav s16le tests/data/asynth-8000-1.wav -c:a real_144' 'stddev' 'tests/data/asynth-8000-1.wav' '' '' '' 'mmx' '-320' '4777' '' '2' '' /ffmpeg/build/ffmpeg -nostats -cpuflags mmx -threads 1 -thread_type frame+slice -i /ffmpeg/build/tests/data/asynth-8000-1.wav -c:a real_144 -f rm -y /ffmpeg/build/tests/data/fate/acodec-ra144.rm /ffmpeg/build/ffmpeg -nostats -cpuflags mmx -flags +bitexact -threads 1 -thread_type frame+slice -i /ffmpeg/build/tests/data/fate/acodec-ra144.rm -c:a pcm_s16le -f wav - $ make fate-acodec-ra144 V=1 CPUFLAGS="mmx+mmxext" TEST acodec-ra144 /ffmpeg/src/tests/fate-run.sh fate-acodec-ra144 "" "" "/ffmpeg/build" 'enc_dec_pcm rm wav s16le tests/data/asynth-8000-1.wav -c:a real_144' 'stddev' 'tests/data/asynth-8000-1.wav' '' '' '' 'mmx+mmxext' '-320' '4777' '' '2' '' /ffmpeg/build/ffmpeg -nostats -cpuflags mmx+mmxext -threads 1 -thread_type frame+slice -i /ffmpeg/build/tests/data/asynth-8000-1.wav -c:a real_144 -f rm -y /ffmpeg/build/tests/data/fate/acodec-ra144.rm /ffmpeg/build/ffmpeg -nostats -cpuflags mmx+mmxext -flags +bitexact -threads 1 -thread_type frame+slice -i /ffmpeg/build/tests/data/fate /acodec-ra144.rm -c:a pcm_s16le -f wav - stddev: 6011.07 PSNR: 20.75 MAXDIFF:45550 bytes: 96000/ 96000 Test acodec-ra144 failed. Look at tests/data/fate/acodec-ra144.err for details. make: *** [fate-acodec-ra144] Error 1 $ make fate-acodec-ra144 V=1 CPUFLAGS="mmx+mmxext+sse+sse2" TEST acodec-ra144 /ffmpeg/src/tests/fate-run.sh fate-acodec-ra144 "" "" "/ffmpeg/build" 'enc_dec_pcm rm wav s16le tests/data/asynth-8000-1.wav -c:a real_144' 'stddev' 'tests/data/asynth-8000-1.wav' '' '' '' 'mmx+mmxext+sse+sse2' '-320' '4777' '' '2' '' /ffmpeg/build/ffmpeg -nostats -cpuflags mmx+mmxext+sse+sse2 -threads 1 -thread_type frame+slice -i /ffmpeg/build/tests/data/asynth-8000-1.wav -c:a real_144 -frm -y /ffmpeg/build/tests/data/fate/acodec-ra144.rm /ffmpeg/build/ffmpeg -nostats -cpuflags mmx+mmxext+sse+sse2 -flags +bitexact -threads 1 -thread_type frame+slice -i /ffmpeg/build/tests/data/fate/acodec-ra144.rm -c:a pcm_s16le -f wav - }}} The problem seems to be in ff_scalarproduct_int16_mmxext(), which is not used in the first example above, and is replaced by ff_scalarproduct_int16_sse2() in the third. I however couldn't reproduce similar failures with other codecs using scalarproduct_int16, like for example fate-lossless-tak. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3429> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3391(undetermined:new): Gaps in input timestamps cause invalid Ogg Opus output
by FFmpeg 02 Mar '14

02 Mar '14

#3391: Gaps in input timestamps cause invalid Ogg Opus output -------------------------------------+------------------------------------- Reporter: MarkZV | Type: defect Status: new | Priority: normal Component: | Version: git- undetermined | master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- When transcoding to Ogg Opus from a format with per-packet timestamps, such as mp4 audio, if the input contains a gap in packet timestamps (e.g. due to a lost/corrupted packet), FFmpeg writes invalid Ogg Opus output. Specifically, the Ogg granule position written by FFmpeg increments by more than the number of samples contained in packets that complete on the page. This violates [https://tools.ietf.org/html/draft-ietf-codec- oggopus-03#section-4 section 4 of the specification] and may cause synchronization problems. Because the Ogg container does not have explicit per-frame or per-packet timestamps, Ogg Opus does not allow gaps in the audio. Instead, an equal duration of silence or predicted samples may be substituted for the missing samples when encoding. If pre-encoded Opus is being copied to Ogg without encoding, any missing Opus packets may be replaced with Opus PLC (packet loss concealment) packets as described in [https://tools.ietf.org/html/draft-ietf-codec-oggopus-03#section-4.1 section 4.1 of the specification]. {{{ $ ffmpeg -v 9 -loglevel 99 -i lostpacket.m4a out.opus ffmpeg version N-60696-g38a08e0 Copyright (c) 2000-2014 the FFmpeg developers built on Feb 17 2014 01:24:12 with gcc 4.8.2 (MacPots gcc48 4.8.2_0) configuration: --enable-debug --assert-level=2 --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable- libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-fontconfig --enable-libfreetype --mandir=/opt/local/share/man --enable-pthreads --cc=/opt/local/bin/gcc-mp-4.8 --arch=x86_64 --enable- yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-nonfree --enable-libfdk-aac --enable-libfaac libavutil 52. 64.100 / 52. 64.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 33.100 / 55. 33.100 libavdevice 55. 10.100 / 55. 10.100 libavfilter 4. 1.102 / 4. 1.102 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Splitting the commandline. Reading option '-v' ... matched as option 'v' (set logging level) with argument '9'. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument '99'. Reading option '-i' ... matched as input file with argument 'lostpacket.m4a'. Reading option 'out.opus' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option v (set logging level) with argument 9. Successfully parsed a group of options. Parsing a group of options: input file lostpacket.m4a. Successfully parsed a group of options. Opening an input file: lostpacket.m4a. [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] ISO: File Type Major Brand: M4A [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] Before avformat_find_stream_info() pos: 28861 bytes read:28861 seeks:0 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 0x103009800] After avformat_find_stream_info() pos: 48 bytes read:28861 seeks:0 frames:1 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'lostpacket.m4a': Metadata: major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf55.32.101 Duration: 00:00:03.54, start: 0.042667, bitrate: 65 kb/s Stream #0:0(und), 1, 1/48000: Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 61 kb/s (default) Metadata: handler_name : SoundHandler Successfully opened the file. Parsing a group of options: output file out.opus. Successfully parsed a group of options. Opening an output file: out.opus. Successfully opened the file. detected 4 logical cores [graph 0 input from stream 0:0 @ 0x102a12fc0] Setting 'time_base' to value '1/48000' [graph 0 input from stream 0:0 @ 0x102a12fc0] Setting 'sample_rate' to value '48000' [graph 0 input from stream 0:0 @ 0x102a12fc0] Setting 'sample_fmt' to value 'fltp' [graph 0 input from stream 0:0 @ 0x102a12fc0] Setting 'channel_layout' to value '0x3' [graph 0 input from stream 0:0 @ 0x102a12fc0] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3 [audio format for output stream 0:0 @ 0x102a138a0] Setting 'sample_fmts' to value 's16|flt' [audio format for output stream 0:0 @ 0x102a138a0] Setting 'sample_rates' to value '48000|24000|16000|12000|8000' [audio format for output stream 0:0 @ 0x102a138a0] Setting 'channel_layouts' to value '0x4|0x3|0x7|0x33|0x37|0x3f|0x70f|0x63f' [audio format for output stream 0:0 @ 0x102a138a0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0' [AVFilterGraph @ 0x102a11be0] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto-inserted resampler 0 @ 0x102a14080] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:flt r:48000Hz [libopus @ 0x10300f800] No bit rate set. Defaulting to 96000 bps. Output #0, opus, to 'out.opus': Metadata: major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf55.33.100 Stream #0:0(und), 0, 1/48000: Audio: opus (libopus), 48000 Hz, stereo, flt, 96 kb/s (default) Metadata: handler_name : SoundHandler major_brand : M4A minor_version : 512 compatible_brands: isomiso2 encoder : Lavf55.33.100 Stream mapping: Stream #0:0 -> #0:0 (aac -> libopus) Press [q] to stop, [?] for help [libopus @ 0x10300f800] Trying to remove 584 more samples than there are in the queue [output stream 0:0 @ 0x102a13300] EOF on sink link output stream 0:0:default. No more output streams to write to, finishing. size= 49kB time=00:00:03.57 bitrate= 112.8kbits/s video:0kB audio:49kB subtitle:0 data:0 global headers:0kB muxing overhead 1.290400% 166 frames successfully decoded, 0 decoding errors [AVIOContext @ 0x102a12c80] Statistics: 0 seeks, 6 writeouts [AVIOContext @ 0x102a11dc0] Statistics: 28861 bytes read, 0 seeks $ }}} The incorrect Ogg Opus file produces warnings from the opusinfo tool (part of [http://www.opus-codec.org/downloads/ opus-tools]); "Sample count behind granule": {{{ $ opusinfo out.opus Processing file "out.opus"... New logical stream (#1, serial: f039efde): type opus Encoded with Lavf55.33.100 User comments section follows... language=und handler_name=SoundHandler major_brand=M4A minor_version=512 compatible_brands=isomiso2 encoder=Lavf55.33.100 WARNING: Sample count behind granule (95040>96064) in stream 1 WARNING: Sample count behind granule (143040>144064) in stream 1 WARNING: Sample count behind granule (170880>171320) in stream 1 Opus stream 1: Pre-skip: 312 Playback gain: 0 dB Channels: 2 Original sample rate: 48000Hz Packet duration: 20.0ms (max), 20.0ms (avg), 20.0ms (min) Page duration: 1000.0ms (max), 890.0ms (avg), 580.0ms (min) Total data length: 50394 bytes (overhead: 1.31%) Playback length: 0m:03.562s Average bitrate: 113.2 kb/s, w/o overhead: 111.7 kb/s Logical stream 1 ended $ }}} It is expected that FFmpeg would produce a valid Ogg Opus file, in this case by inserting silence or predicted samples in place of the missing samples before encoding. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3391> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3376(ffmpeg:new): -hls_list_size 0 does not write playlist entries
by FFmpeg 01 Mar '14

01 Mar '14

#3376: -hls_list_size 0 does not write playlist entries ------------------------------------+-------------------------------------- Reporter: blacktrash | Type: defect Status: new | Priority: normal Component: ffmpeg | Version: git-master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ------------------------------------+-------------------------------------- Summary of the bug: How to reproduce: {{{ $ ffmpeg -i testhls.mp4 -bsf:v h264_mp4toannexb -c copy -map 0 -f hls -hls_list_size 0 testhls0.m3u8 ffmpeg version N-60230-gcfe282e Copyright (c) 2000-2014 the FFmpeg developers built on Jan 29 2014 14:17:03 with Apple LLVM version 5.0 (clang-500.2.79) (based on LLVM 3.3svn) configuration: --enable-gpl --enable-version3 --enable-nonfree --disable-network --disable-ffserver --enable-shared --enable-libfdk-aac --enable-libx264 --enable-libxvid --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libvpx --enable-librtmp --extra-cflags=-I/sw/include --extra-libs=-L/sw/lib libavutil 52. 63.100 / 52. 63.100 libavcodec 55. 49.101 / 55. 49.101 libavformat 55. 28.101 / 55. 28.101 libavdevice 55. 7.100 / 55. 7.100 libavfilter 4. 1.101 / 4. 1.101 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testhls.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf55.28.101 Duration: 00:00:35.23, start: 0.229388, bitrate: 314 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 282 kb/s, 15 fps, 15 tbr, 15360 tbn, 30 tbc (default) Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 22050 Hz, stereo, fltp, 32 kb/s (default) Metadata: handler_name : SoundHandler Output #0, hls, to 'testhls0.m3u8': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf55.28.101 Stream #0:0(und): Video: h264 (avc1 / 0x31637661), yuv420p, 320x240 [SAR 1:1 DAR 4:3], q=2-31, 282 kb/s, 15 fps, 90k tbn, 15 tbc (default) Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 22050 Hz, stereo, 32 kb/s (default) Metadata: handler_name : SoundHandler Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help [adts @ 0x7fb72883c000] Encoder did not produce proper pts, making some up. frame= 524 fps=0.0 q=-1.0 Lsize=N/A time=00:00:35.29 bitrate=N/A video:1203kB audio:138kB subtitle:0 global headers:0kB muxing overhead -100.001603% $ cat testhls0.m3u8 #EXTM3U #EXT-X-VERSION:3 #EXT-X-TARGETDURATION:0 #EXT-X-MEDIA-SEQUENCE:3 #EXT-X-ENDLIST }}} The segments are created, but the list is empty. However man ffmpeg- formats says: hls_list_size size Set the maximum number of playlist entries. If set to 0 the list file will contain all the segments. Default value is 5. Same behaviour with ffmpeg version N-60430-gc72f587. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3376> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #2457(avcodec:closed): MAX_SLICES value too low for a number of RTP endpoints
by FFmpeg 28 Feb '14

28 Feb '14

#2457: MAX_SLICES value too low for a number of RTP endpoints -------------------------------------+------------------------------------- Reporter: tgreenwood | Owner: Type: enhancement | Status: closed Priority: wish | Component: avcodec Version: git-master | Resolution: Keywords: h264 | needs_more_info Blocking: | Blocked By: Analyzed by developer: 0 | Reproduced by developer: 0 -------------------------------------+------------------------------------- Changes (by cehoyos): * status: new => closed * resolution: => needs_more_info Comment: If this is still reproducible, please provide a sample. -- Ticket URL: <https://trac.ffmpeg.org/ticket/2457#comment:6> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #2686(avcodec:open): Native AAC encoder collapses at high bitrates on some samples
by FFmpeg 28 Feb '14

28 Feb '14

#2686: Native AAC encoder collapses at high bitrates on some samples -------------------------------------+------------------------------------- Reporter: Kamedo2 | Owner: Type: defect | Status: open Priority: normal | Component: avcodec Version: git-master | Resolution: Keywords: aac | Blocked By: regression | Reproduced by developer: 1 Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- Comment (by klaussfreire): It's probably the same bug, it happens sometimes with short windows I noticed. -- Ticket URL: <https://trac.ffmpeg.org/ticket/2686#comment:259> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #2686(avcodec:open): Native AAC encoder collapses at high bitrates on some samples
by FFmpeg 28 Feb '14

28 Feb '14

#2686: Native AAC encoder collapses at high bitrates on some samples -------------------------------------+------------------------------------- Reporter: Kamedo2 | Owner: Type: defect | Status: open Priority: normal | Component: avcodec Version: git-master | Resolution: Keywords: aac | Blocked By: regression | Reproduced by developer: 1 Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- Comment (by Kamedo2): The stereo image of Mama.wv, itCouldBeSweet.wv is strange in v8 ABR 192kbps. http://www.rarewares.org/test_samples/ -- Ticket URL: <https://trac.ffmpeg.org/ticket/2686#comment:258> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3212(undetermined:closed): initial seek in MSVIDEO1 causes failure to display on some avi files
by FFmpeg 28 Feb '14

28 Feb '14

#3212: initial seek in MSVIDEO1 causes failure to display on some avi files -------------------------------------+------------------------------------- Reporter: DonMoir | Owner: Type: defect | Status: closed Priority: normal | Component: Version: unspecified | undetermined Keywords: | Resolution: fixed Blocking: | Blocked By: Analyzed by developer: 0 | Reproduced by developer: 1 -------------------------------------+------------------------------------- Changes (by michael): * status: reopened => closed * resolution: => fixed * reproduced: 0 => 1 Comment: Fixed in f3eef02746ef9585400f58de7803c2db7efd7b4a -- Ticket URL: <https://trac.ffmpeg.org/ticket/3212#comment:12> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3212(undetermined:reopened): initial seek in MSVIDEO1 causes failure to display on some avi files
by FFmpeg 28 Feb '14

28 Feb '14

#3212: initial seek in MSVIDEO1 causes failure to display on some avi files -------------------------------------+------------------------------------- Reporter: DonMoir | Owner: Type: defect | Status: reopened Priority: normal | Component: Version: unspecified | undetermined Keywords: | Resolution: Blocking: | Blocked By: Analyzed by developer: 0 | Reproduced by developer: 0 -------------------------------------+------------------------------------- Changes (by richardpl): * status: closed => reopened * resolution: duplicate => Old description: > Same as ticket #1273 but different codec ID. > > If you do a seek prior to calling avcodec_decode_video2 at least once, > then you will no longer get any video stream packets with the attached > files. The seek works fine after you call avcodec_decode_video2 but not > if you tried to seek prior to this. Reading a first video packet is not > enough. > > Use this sample to test. > > http://sms.pangolin.com/temp/EVBPIC.AVI New description: Same as ticket #1273 but different codec ID. If you do a seek prior to calling avcodec_decode_video2 at least once, then you will no longer get any video stream packets with the attached files. The seek works fine after you call avcodec_decode_video2 but not if you tried to seek prior to this. Reading a first video packet is not enough. Use this sample to test. *http://sms.pangolin.com/temp/EVBPIC.AVI -- -- Ticket URL: <https://trac.ffmpeg.org/ticket/3212#comment:11> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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Re: [FFmpeg-trac] #3192(undetermined:new): Dropped Frames With Overlay Filter
by FFmpeg 28 Feb '14

28 Feb '14

#3192: Dropped Frames With Overlay Filter -------------------------------------+------------------------------------- Reporter: ian_m | Owner: Type: defect | Status: new Priority: normal | Component: Version: git-master | undetermined Keywords: x11grab | Resolution: Blocking: | Blocked By: Analyzed by developer: 0 | Reproduced by developer: 0 -------------------------------------+------------------------------------- Comment (by jnvsor): I can confirm it occurs when both streams are passed through filter_complex, even if they are null filters that shouldn't effect performance at all. Low fps: {{{ ffmpeg -f x11grab -framerate 30 -s hd1080 -i :0.0 -f v4l2 -framerate 30 -i /dev/video0 -filter_complex '[0]null[cap];[1]null[webcam]' -map '[cap]' -map '[webcam]' -c:v libx264 -pix_fmt yuv444p -preset ultrafast -qp 0 -threads 2 -y out.avi }}} Remove either of the null filters and it will work at full framerate: {{{ - '[0]null[cap];[1]null[webcam]' - -map '[cap]' + '[1]null[webcam]' + -map 0 OR + '[0]null[cap]' + -map 1 }}} Add the v4l2 input first and it will work at full framerate: {{{ - -f x11grab -framerate 30 -s hd1080 -i :0.0 -f v4l2 -framerate 30 -i /dev/video0 + -f v4l2 -framerate 30 -i /dev/video0 -f x11grab -framerate 30 -s hd1080 -i :0.0 }}} It only occurs if that filter output is actually used, piping one to a nullsink also makes it work at full framerate: {{{ - '[0]null[cap];[1]null[webcam]' - -map '[cap]' + '[0]null[cap];[cap]nullsink;[1]null[webcam]' + -map 0 }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/3192#comment:11> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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