October 2016 - FFmpeg-trac - lists.ffmpeg.org

#3626(undetermined:new): add 'extern "C"' to headers
by FFmpeg 12 Jul '24

12 Jul '24

#3626: add 'extern "C"' to headers -------------------------------------+------------------------------------- Reporter: spraff | Type: defect Status: new | Priority: wish Component: | Version: undetermined | unspecified Keywords: headers | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Your policy at the moment is to require applications to add 'extern "C"' to their C++ source files: http://trac.ffmpeg.org/wiki/Including%20FFmpeg%20headers%20in%20a%20C%2B%2B… This is silly and annoying. It is the header file's job to present its interface consistently to applications. decklink_enc.h does this, the other headers should too. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3626> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3674(avcodec:new): Just audio without video in windows decoder
by FFmpeg 11 Jul '24

11 Jul '24

#3674: Just audio without video in windows decoder -------------------------------------+------------------------------------- Reporter: mfz | Type: defect Status: new | Priority: normal Component: avcodec | Version: Keywords: h264, | unspecified height, audio, windows media | Blocked By: player, internet explorer | Reproduced by developer: 0 Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: I have a screen recorded video with frame rate 15, video codec H.264 (MP4) and AAC as audio. The target via FFmpeg should contain frame rate 25, height of 720 pixel and the codecs without change. After encoding, the file is playable in VLC player, Chrome browser (HTML5 video tag) and Firefox. In Windows Media player I just get the audio played (no video to see) and in Internet Explorer 11, I get an error ("Unknown source.") How to reproduce: {{{ % ffmpeg -i input_file.mp4 -vf "scale='(trunc((out_h*a)/2)*2):(min(720, in_h))'" output_file.mp4 VERSION: N-63439-g96470ca N-63439-g96470ca }}} Strange is: If I encode the video with a height of 718 or 722 (or any other height I tried), the video is playable anywhere - without problems. Test video file for input (short, length of 1 second): http://www.datafilehost.com/d/1ca920ac -- Ticket URL: <https://trac.ffmpeg.org/ticket/3674> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4995(undetermined:new): Error when inserting sidx atoms
by FFmpeg 08 Jul '24

08 Jul '24

#4995: Error when inserting sidx atoms -------------------------------------+------------------------------------- Reporter: alan1523 | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- bash-3.2$ /Volumes/storage/render/ffmpeg/ffmpeg -y -i "/Volumes/storage/render/media/in/input.mp4" -movflags global_sidx+frag_keyframe -frag_duration 1000 -codec:v libx264 -profile:v baseline -pix_fmt yuv420p -b:v 500k -codec:a aac -b:a 96k -strict -2 -s 426x240 -threads 0 "/Volumes/storage/render/media/out/output.mp4" ffmpeg version N-76633-g4746653-tessus Copyright (c) 2000-2015 the FFmpeg developers built with Apple LLVM version 6.0 (clang-600.0.57) (based on LLVM 3.5svn) configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --as=yasm --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable- libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable- libxavs --enable-libxvid --enable-libzmq --enable-version3 --disable- ffplay --disable-indev=qtkit --disable-indev=x11grab_xcb libavutil 55. 5.100 / 55. 5.100 libavcodec 57. 13.102 / 57. 13.102 libavformat 57. 13.100 / 57. 13.100 libavdevice 57. 0.100 / 57. 0.100 libavfilter 6. 14.101 / 6. 14.101 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.100 / 2. 0.100 libpostproc 54. 0.100 / 54. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/storage/render/media/in/input.mp4': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2015-10-27 16:59:17 Duration: 00:00:10.64, start: 0.000000, bitrate: 3098 kb/s Stream #0:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 83 kb/s (default) Metadata: creation_time : 2015-10-27 16:59:41 handler_name : Apple Alias Data Handler Stream #0:1(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720, 3008 kb/s, SAR 1:1 DAR 16:9, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc (default) Metadata: creation_time : 2015-10-27 16:59:41 handler_name : Apple Alias Data Handler encoder : H.264 [libx264 @ 0x7fc1d3012e00] using SAR=640/639 [libx264 @ 0x7fc1d3012e00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 0x7fc1d3012e00] profile Constrained Baseline, level 2.1 [libx264 @ 0x7fc1d3012e00] 264 - core 148 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=0 ref=3 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=7 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=500 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to '/Volumes/storage/render/media/out/output.mp4': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt encoder : Lavf57.13.100 Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 426x240 [SAR 640:639 DAR 16:9], q=-1--1, 500 kb/s, 29.97 fps, 30k tbn, 29.97 tbc (default) Metadata: creation_time : 2015-10-27 16:59:41 handler_name : Apple Alias Data Handler encoder : Lavc57.13.102 libx264 Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 96 kb/s (default) Metadata: creation_time : 2015-10-27 16:59:41 handler_name : Apple Alias Data Handler encoder : Lavc57.13.102 aac Stream mapping: Stream #0:1 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:0 -> #0:1 (aac (native) -> aac (native)) Press [q] to stop, [?] for help frame= 124 fps=0.0 q=25.0 size= 182kB time=00:00:04.88 bitrate= 305.0kbits/frame= 179 fps=179 q=27.0 size= 353kB time=00:00:06.61 bitrate= 437.2kbits/frame= 317 fps=211 q=25.0 size= 766kB time=00:00:10.64 bitrate= 589.2kbits/[mp4 @ 0x7fc1d3010400] Starting second pass: inserting sidx atoms Non-consecutive fragments, writing incorrect sidx Last message repeated 711 times frame= 319 fps=202 q=-1.0 Lsize= 913kB time=00:00:10.68 bitrate= 699.5kbits/s video:678kB audio:125kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 13.572538% [libx264 @ 0x7fc1d3012e00] frame I:5 Avg QP:20.79 size: 11995 [libx264 @ 0x7fc1d3012e00] frame P:314 Avg QP:23.62 size: 2019 [libx264 @ 0x7fc1d3012e00] mb I I16..4: 20.0% 0.0% 80.0% [libx264 @ 0x7fc1d3012e00] mb P I16..4: 0.9% 0.0% 2.6% P16..4: 36.6% 17.0% 6.8% 0.0% 0.0% skip:36.1% [libx264 @ 0x7fc1d3012e00] final ratefactor: 21.26 [libx264 @ 0x7fc1d3012e00] coded y,uvDC,uvAC intra: 66.7% 84.4% 57.9% inter: 16.3% 31.1% 3.8% [libx264 @ 0x7fc1d3012e00] i16 v,h,dc,p: 28% 26% 21% 25% [libx264 @ 0x7fc1d3012e00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 18% 25% 15% 6% 8% 7% 9% 6% 7% [libx264 @ 0x7fc1d3012e00] i8c dc,h,v,p: 50% 26% 15% 8% [libx264 @ 0x7fc1d3012e00] ref P L0: 75.7% 14.4% 9.9% [libx264 @ 0x7fc1d3012e00] kb/s:521.63 [aac @ 0x7fc1d3025a00] Qavg: 526.973 bash-3.2$ -- Ticket URL: <https://trac.ffmpeg.org/ticket/4995> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4478(avformat:new): PlayStation 2 movie files not supported (.pss)
by FFmpeg 08 Jul '24

08 Jul '24

#4478: PlayStation 2 movie files not supported (.pss) -------------------------------------+------------------------------------- Reporter: Melchior | Type: Status: new | enhancement Component: avformat | Priority: normal Keywords: PlayStation | Version: 2, PS2, .pss | unspecified Blocking: | Blocked By: Analyzed by developer: 0 | Reproduced by developer: 0 -------------------------------------+------------------------------------- I decided to dig out 2 of my Sony PlayStation 2 games: - Suikoden 3 - Suikoden 4 and extract the Movie (Opening/Closing) I found some str.bin files as well but all the movie files are .pss I get FFmpeg builds from and use them with http://ffmpeg.zeranoe.com/builds/ http://avanti.arrozcru.com/ OS used on: DesktopPC -- Windows XP Professional x64 LaptopPC -- Windows 7 Ultimate x64 these files I could barely extract the Video.. but the audio was not getting even seen never-mind getting extracted.. -- Ticket URL: <https://trac.ffmpeg.org/ticket/4478> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4683(undetermined:new): mjpeg: 12bpp fix artefacts
by FFmpeg 29 Jun '24

29 Jun '24

#4683: mjpeg: 12bpp fix artefacts -------------------------------------+------------------------------------- Reporter: ami_stuff | Owner: Type: defect | Status: new Priority: normal | Component: Version: | undetermined unspecified | Keywords: Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 -------------------------------------+------------------------------------- artefacts on the logo {{{ C:\>ffmpeg -i 12bpp.jpg out.bmp ffmpeg version N-73189-g7728d23 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable- libcaca -- enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-l ibilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab le-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable- librtmp --en able-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --ena ble-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo- amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enabl e-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --ena ble-decklink --enable-zlib libavutil 54. 27.100 / 54. 27.100 libavcodec 56. 45.100 / 56. 45.100 libavformat 56. 38.102 / 56. 38.102 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 18.100 / 5. 18.100 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.100 / 1. 2.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, image2, from '12bpp.jpg': Duration: 00:00:00.04, start: 0.000000, bitrate: 65308 kb/s Stream #0:0: Video: mjpeg, gray16le(12 bpc, bt470bg/unknown/unknown), 999x74 9 [SAR 300:300 DAR 999:749], 25 tbr, 25 tbn, 25 tbc Output #0, image2, to 'out.bmp': Metadata: encoder : Lavf56.38.102 Stream #0:0: Video: bmp, gray, 999x749 [SAR 1:1 DAR 999:749], q=2-31, 200 kb /s, 25 fps, 25 tbn, 25 tbc Metadata: encoder : Lavc56.45.100 bmp Stream mapping: Stream #0:0 -> #0:0 (mjpeg (native) -> bmp (native)) Press [q] to stop, [?] for help frame= 1 fps=0.0 q=0.0 Lsize=N/A time=00:00:00.04 bitrate=N/A video:732kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing o verhead: unknown }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/4683> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3386(undetermined:new): [H264] [Regression] illegal short term buffer state detected
by FFmpeg 19 Jun '24

19 Jun '24

#3386: [H264] [Regression] illegal short term buffer state detected -------------------------------------+------------------------------------- Reporter: | Type: defect Underground78 | Priority: normal Status: new | Version: git- Component: | master undetermined | Blocked By: Keywords: | Reproduced by developer: 0 Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- '''Summary of the bug:''' A regression in H264 decoding was apparently introduced about 3 months ago. It leads to artifacts appearing during playback with "illegal short term buffer state detected" errors being reported in the log. '''How to reproduce:''' Try [http://stfcc.org/misc/bbc2.sample.ts this sample] in ffplay for example. {{{ >ffplay.exe bbc2.sample.ts ffplay version N-60646-gd601106 Copyright (c) 2003-2014 the FFmpeg developers built on Feb 15 2014 03:11:41 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable- librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo- aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 63.101 / 52. 63.101 libavcodec 55. 52.101 / 55. 52.101 libavformat 55. 32.101 / 55. 32.101 libavdevice 55. 9.101 / 55. 9.101 libavfilter 4. 1.102 / 4. 1.102 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 [NULL @ 04011580] start time is not set in estimate_timings_from_pts [mpegts @ 0400d160] PES packet size mismatch Last message repeated 1 times Input #0, mpegts, from 'bbc2.sample.ts': Duration: 00:00:47.48, start: 0.200000, bitrate: 8424 kb/s Program 1 Stream #0:0[0x157c]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv , bt709), 1440x1080 [SAR 4:3 DAR 16:9], 23.75 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x157d](eng): Audio: ac3 ([6][0][0][0] / 0x0006), 48000 Hz, ster eo, fltp, 192 kb/s Stream #0:2[0x157f](eng): Subtitle: dvb_teletext ([6][0][0][0] / 0x0006) Stream #0:3[0x1580](eng): Subtitle: dvb_subtitle ([6][0][0][0] / 0x0006) No codec could be found with id 94215 [h264 @ 054c1980] illegal short term buffer state detected }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/3386> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5718(undetermined:new): ffmpeg not remapping channels for libopus automatically
by FFmpeg 16 Jun '24

16 Jun '24

#5718: ffmpeg not remapping channels for libopus automatically -------------------------------------+------------------------------------- Reporter: agressiv | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: libopus | Blocked By: channelmap | Reproduced by developer: 0 Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- {{{ [libopus @ 00000000026ad020] Invalid channel layout 5.1(side) for specified mapping family -1. }}} Summary of the bug: Encode a 5.1 audio file with 5.1(side) channel map How to reproduce: {{{ % ffmpeg -i movie-with-51side.mkv -c:a libopus -b:a 256k output.mkv ffmpeg version N-81025-g25ca74d built on July 15 2016 }}} ffmpeg version N-80101-gd970f7b automatically remapped these to the rear channels since libopus doesn't seem to allow side channels. I now have to explicitly add: {{{ -af "channelmap=channel_layout=5.1" }}} Not sure if this is expected behavior or not. Without this channelmap, it aborts. Command-line opusenc automatically remaps as well. -- Ticket URL: <https://trac.ffmpeg.org/ticket/5718> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5309(undetermined:new): Seek doesn't work in some .mp4 files
by FFmpeg 25 May '24

25 May '24

#5309: Seek doesn't work in some .mp4 files -------------------------------------+------------------------------------- Reporter: Arnaud | Type: defect Status: new | Priority: normal Component: | Version: git- undetermined | master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: How to reproduce: {{{ % ffplay -ss 5 ffmpeg version: ffplay version N-78958-g8c24523 built from latest git, on Ubuntu 14.04.3, on 2016-03-07 }}} File is seek_not_working.mp4 The file plays well but always from the beginning: the seek isn't working. -- Ticket URL: <https://trac.ffmpeg.org/ticket/5309> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5910(undetermined:new): AAC to PCM conversion inserts extra silence in the beginning
by FFmpeg 25 May '24

25 May '24

#5910: AAC to PCM conversion inserts extra silence in the beginning -------------------------------------+------------------------------------- Reporter: | Type: defect jwilhelmsson | Priority: normal Status: new | Version: Component: | unspecified undetermined | Blocked By: Keywords: aac pcm | Reproduced by developer: 0 Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: When converting AAC audio files/streams to PCM extra silence is inserted in the beginning of the output file. Note: This may very well be the same issue as ticket #2325, but since I believe I have more information I elected to create a new one. The long version: My company dub cartoons, so we receive many kinds of video formats from our various clients. Recently one of them complained that our final delivery was out of sync compared to the original material, and that's how this issue was discovered. The reference files from the client were mp4:s with aac audio, and when I converted said audio into wav files (for use in our recording software, Steinberg Nuendo) extra silence got inserted in the beginning, making us record everything out of sync. That it was ffmpeg that was in the wrong was concluded by comparing with files converted by ProTools, Nuendo, and QuickTime - which are all the same and different from the ffmpeg output. After lots of testing I concluded that it's the AAC to PCM conversion that's the culprit (ie. the video container format is mostly irrelevant), and also that the length of the inserted silence varies between different files. I haven't been able to pinpoint exactly what causes the difference. Attached are five aac files, plus wav files converted by ffmpeg (3.1.4) and QuickTime Pro (7.7.9) clearly showing the difference. Since the files come from commercial productions I've only included 7 to 10 seconds from each, but it's enough to see the error. Two of the files insert approximately 44 milliseconds (or about 2100 samples) of silence, two insert 108 milliseconds (about 5200 samples), and one oddly enough gets only 32 milliseconds of silence even though the audio is shifted 44 ms (this is easy to see since it starts with a test tone). How to reproduce: The aac files were converted by ffmpeg with the command (I'll attach outputs in separate messages below): {{{ ffmpeg -i input -c:a pcm_s24le -ar 48k output }}} They were also converted with QuickTime Pro with the same settings (24 bits, 48kHz). I then compared the waveforms in both Nuendo and Audacity. The offset values were measured by manually marking an area in Audacity, so they are very approximate. The files: The attached files come from one movie and two tv series (two episodes each). The movie files are called "g", and the series "tj" and "td". The aac files were extracted from the original mp4 files by stream copying: {{{ ffmpeg -i input -c:a copy -t 10 output }}} The movie file starts with a test tone, and is also the one which differs 32 ms in the beginning of the test tone, but 44 at the end of it. Random notes: The error is the same when converting directly from the mp4 file and when converting from an extracted aac. Extracting PCM wav from a mov container produces no errors. If I convert the aac stream to a new aac file there's still an error, but only half as long. I've only tested this on one file, but it produced a 22 ms gap instead of 44 ms. Compounded converting does not compound the error. Ie: Converting from aac to aac, and then converting that output file to aac again does not increase the error. Converting to a different bitrate/sample rate does not affect the result. Final words: I've done a lot of testing, but it's very possible that I've forgotten some vital information in this report, so please ask if you need more details. -- Ticket URL: <https://trac.ffmpeg.org/ticket/5910> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5733(undetermined:new): Dolby-E decoding
by FFmpeg 15 May '24

15 May '24

#5733: Dolby-E decoding -------------------------------------+------------------------------------- Reporter: | Type: smallishzulu | enhancement Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: dolby-e | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Hello, Does FFmpeg support Dolby-E decoding ? I have attached 2 files: a) Dolby-E data with SMPTE-337 b) Dolby-E data Kind Regards, -- Ticket URL: <https://trac.ffmpeg.org/ticket/5733> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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