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

#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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#4078(ffmpeg:new): Specify audio bitrate per-channel
by FFmpeg 15 Apr '24

15 Apr '24

#4078: Specify audio bitrate per-channel ----------------------------------+--------------------------------------- Reporter: Haravikk | Type: enhancement Status: new | Priority: normal Component: ffmpeg | Version: unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+--------------------------------------- Fairly self-explanatory, but currently we can specify audio bit-rate for the entire stream, but this requires us to calculate the total bitrate from the number of input channels ourselves. This is fine for single files, but for batch-encoding a mixture of stereo and surround-sound content it's not as ideal, or even for encoding a file that contains multiple audio-streams. To accommodate these use-cases I'd like to request an alternative syntax for specifying audio channels with a per-channel bit-rate so that I can instead just specify say 64k per channel for AAC, with ffmpeg automatically using a total of 128k for stereo or 384k for 5.1 etc. I'm not sure what the best way to do this would be. I suppose one option would be to have a new parameter -bc for "bits per channel", so I could specify something like ffmpeg -i "$INFILE" -acodec aac -bc:a 64k "$OUTFILE" An alternative would be allowing some kind of suffix to the -b option such as: ffmpeg -i "$INFILE" -acodec aac -b:a 64k/c "$OUTFILE" -- Ticket URL: <https://trac.ffmpeg.org/ticket/4078> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4209(undetermined:new): GPS coordinates location and other iOS metadata in MOV are not copied to output MP4
by FFmpeg 10 Apr '24

10 Apr '24

#4209: GPS coordinates location and other iOS metadata in MOV are not copied to output MP4 -------------------------------------+------------------------------------- Reporter: elmimmo | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: Certain metadata in MOV files recorded with iOS devices, such as GPS location, is lost when converting to MP4. **ffmpeg** does state in its log that the output file will contain such metadata, but such is not the case as identified with **ffprobe** and **exiftool**. Some of the metadata present in the input MOV that is missing in the output MP4, according to **ffprobe**: - make ("Apple") - encoder - location (GPS latitude/longitude/altitude coordinates) - model (the iPhone/iPad/iPod model) **exiftool** identifies some more: - camera identifier ("Back" or "Front") - Software Version (iOS version) This happens irrespective of input video/audio being transcoded or just repackaged. How to reproduce: {{{ % ffmpeg -i INPUT.mov -map_metadata 0 -c copy OUTPUT.mp4 ffmpeg version 2.5.2 Copyright (c) 2000-2014 the FFmpeg developers built on Dec 26 2014 21:55:58 with Apple LLVM version 4.2 (clang-425.0.28) (based on LLVM 3.2svn) configuration: --prefix=/usr/local/Cellar/ffmpeg/2.5.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable- libx264 --enable-libmp3lame --enable-libxvid --enable-libvo-aacenc --enable-vda libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat 56. 15.102 / 56. 15.102 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 libavresample 2. 1. 0 / 2. 1. 0 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'INPUT.mov': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2014-12-26 20:54:22 model : iPhone 4 model-spa : iPhone 4 encoder : 7.1.1 encoder-spa : 7.1.1 date : 2014-12-26T21:54:22+0100 date-spa : 2014-12-26T21:54:22+0100 location : +41.4082+002.1852+027.119/ location-spa : +41.4082+002.1852+027.119/ make : Apple make-spa : Apple Duration: 00:00:01.07, start: 0.000000, bitrate: 11694 kb/s Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p(tv, bt709), 1280x720, 11890 kb/s, 24 fps, 24 tbr, 600 tbn, 1200 tbc (default) Metadata: rotate : 90 creation_time : 2014-12-26 20:54:22 handler_name : Core Media Data Handler encoder : H.264 Side data: displaymatrix: rotation of -90.00 degrees Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 62 kb/s (default) Metadata: creation_time : 2014-12-26 20:54:22 handler_name : Core Media Data Handler Output #0, mp4, to 'OUTPUT.mp4': Metadata: major_brand : qt minor_version : 0 compatible_brands: qt creation_time : 2014-12-26 20:54:22 model : iPhone 4 model-spa : iPhone 4 make-spa : Apple encoder : Lavf56.15.102 date : 2014-12-26T21:54:22+0100 date-spa : 2014-12-26T21:54:22+0100 location : +41.4082+002.1852+027.119/ location-spa : +41.4082+002.1852+027.119/ make : Apple Stream #0:0(und): Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1280x720, q=2-31, 11890 kb/s, 24 fps, 19200 tbn, 600 tbc (default) Metadata: rotate : 90 creation_time : 2014-12-26 20:54:22 handler_name : Core Media Data Handler encoder : H.264 Side data: displaymatrix: rotation of -90.00 degrees Stream #0:1(und): Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, mono, 62 kb/s (default) Metadata: creation_time : 2014-12-26 20:54:22 handler_name : Core Media Data Handler Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help frame= 25 fps=0.0 q=-1.0 Lsize= 1522kB time=00:00:01.06 bitrate=11673.4kbits/s video:1512kB audio:8kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.123020% }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/4209> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5771(undetermined:new): avi with defect header won't work
by FFmpeg 24 Mar '24

24 Mar '24

#5771: avi with defect header won't work -------------------------------------+------------------------------------- Reporter: the_gamer | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: I can't ffmpeg get to cooperate with any videos created with the catcam http://www.mr-lee-catcam.de/ Videos work in vlc, windows media player, mplayer and mencoder can convert them. But ffmpeg won't do anything. How to reproduce: $ ffmpeg -c:v avrn -loglevel trace -an -r:v 30 -i PICT0001.AVI -vf scale=640:480 -an katze.mpg ffmpeg version 3.1.1 Copyright (c) 2000-2016 the FFmpeg developers built with gcc 6.1.1 (GCC) 20160707 configuration: --prefix=/usr --disable-debug --disable-static --disable- stripping --enable-avisynth --enable-avresample --enable-fontconfig --enable-gmp --enable-gnutls --enable-gpl --enable-ladspa --enable-libass --enable-libbluray --enable-libfreetype --enable-libfribidi --enable- libgsm --enable-libiec61883 --enable-libmodplug --enable-libmp3lame --enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable- libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable- libv4l2 --enable-libvidstab --enable-libvorbis --enable-libvpx --enable- libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-netcdf --enable-shared --enable-version3 --enable-x11grab libavutil 55. 28.100 / 55. 28.100 libavcodec 57. 48.101 / 57. 48.101 libavformat 57. 41.100 / 57. 41.100 libavdevice 57. 0.101 / 57. 0.101 libavfilter 6. 47.100 / 6. 47.100 libavresample 3. 0. 0 / 3. 0. 0 libswscale 4. 1.100 / 4. 1.100 libswresample 2. 1.100 / 2. 1.100 libpostproc 54. 0.100 / 54. 0.100 Splitting the commandline. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'avrn'. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'trace'. Reading option '-an' ... matched as option 'an' (disable audio) with argument '1'. Reading option '-r:v' ... matched as option 'r' (set frame rate (Hz value, fraction or abbreviation)) with argument '30'. Reading option '-i' ... matched as input file with argument 'PICT0001.AVI'. Reading option '-vf' ... matched as option 'vf' (set video filters) with argument 'scale=640:480'. Reading option '-an' ... matched as option 'an' (disable audio) with argument '1'. Reading option 'katze.mpg' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument trace. Successfully parsed a group of options. Parsing a group of options: input file PICT0001.AVI. Applying option c:v (codec name) with argument avrn. Applying option an (disable audio) with argument 1. Applying option r:v (set frame rate (Hz value, fraction or abbreviation)) with argument 30. Successfully parsed a group of options. Opening an input file: PICT0001.AVI. [file @ 0x55e8ab72c060] Setting default whitelist 'file,crypto' Probing avi score:100 size:2048 [avi @ 0x55e8ab72b820] Format avi probed with size=2048 and score=100 [avi @ 0x55e8ab72bf80] use odml:1 pos:14 tag: tag=LIST size=0x154 pos:18 list: tag=hdrl size=0x0 pos:20 tag: tag=avih size=0x38 pos:60 tag: tag=LIST size=0xa4 pos:64 list: tag=strl size=0x0 pos:6C tag: tag=strh size=0x38 pos:74 strh: tag=vids size=0xffffffff [avi @ 0x55e8ab72b820] 1000000 33333 0 pos:AC tag: tag=strf size=0x28 pos:D4 video: tag=MJPG size=0x0 pos:DC tag: tag=strd size=0x28 pos:10C tag: tag=LIST size=0x5c pos:110 list: tag=strl size=0x0 pos:118 tag: tag=strh size=0x38 pos:120 strh: tag=auds size=0xffffffff [avi @ 0x55e8ab72b820] scale/rate is 1/0 which is invalid. (This file has been generated by broken software.) [avi @ 0x55e8ab72b820] 1000000 33333 1 pos:158 tag: tag=strf size=0x10 [avi @ 0x55e8ab72b820] Invalid sample rate: 0 [AVIOContext @ 0x55e8ab7343a0] Statistics: 32768 bytes read, 0 seeks PICT0001.AVI: Invalid data found when processing input -- Ticket URL: <https://trac.ffmpeg.org/ticket/5771> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4859(undetermined:new): TTML subtitles
by FFmpeg 14 Mar '24

14 Mar '24

#4859: TTML subtitles -------------------------------------+------------------------------------- Reporter: cehoyos | Owner: Type: | Status: new enhancement | Component: Priority: wish | undetermined Version: git- | Keywords: sub master | Blocking: Blocked By: | Analyzed by developer: 0 Reproduced by developer: 0 | -------------------------------------+------------------------------------- vlc ticket 14974: https://trac.videolan.org/vlc/ticket/14974 FFmpeg does not support EBU Timed-Text Markup Language (TTML) subtitles yet. -- Ticket URL: <https://trac.ffmpeg.org/ticket/4859> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5154(avformat:new): Demux h264 stream from Arecont camera
by FFmpeg 23 Feb '24

23 Feb '24

#5154: Demux h264 stream from Arecont camera ---------------------------------------+---------------------------------- Reporter: cehoyos | Owner: Type: enhancement | Status: new Priority: wish | Component: avformat Version: git-master | Keywords: h264 Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 ---------------------------------------+---------------------------------- http://ffmpeg.org/pipermail/ffmpeg-user/2016-January/030122.html A user provided a stream dump from his Arecont Vision AV20365CO. FFmpeg correctly autodetects the stream as h264 but every frame starts with a http content header that makes remuxing impossible: {{{ $ ffmpeg -r 25 -i h264stream.dump -vcodec copy out.mov ffmpeg version N-77810-g405abdb Copyright (c) 2000-2016 the FFmpeg developers built with gcc 4.7 (SUSE Linux) configuration: --enable-gpl --enable-libx264 libavutil 55. 13.100 / 55. 13.100 libavcodec 57. 22.100 / 57. 22.100 libavformat 57. 21.101 / 57. 21.101 libavdevice 57. 0.100 / 57. 0.100 libavfilter 6. 23.100 / 6. 23.100 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100 [h264 @ 0x29a9440] Stream #0: not enough frames to estimate rate; consider increasing probesize Input #0, h264, from 'h264stream.dump': Duration: N/A, bitrate: N/A Stream #0:0: Video: h264 (Baseline), yuv420p, 1280x960, 25 fps, 25 tbr, 1200k tbn, 50 tbc [mov @ 0x2a378e0] Codec for stream 0 does not use global headers but container format requires global headers Output #0, mov, to 'out.mov': Metadata: encoder : Lavf57.21.101 Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p, 1280x960, q=2-31, 25 fps, 25 tbr, 12800 tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help [mov @ 0x2a378e0] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly frame= 20 fps=0.0 q=-1.0 Lsize= 13897kB time=00:00:00.80 bitrate=142304.5kbits/s speed=34.4x video:13896kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.006613% }}} The output file is unreadable. The following works fine: {{{ $ cat h264stream.dump | sed -u -b -e '/^--fbdr\r$/ {N;N;N;d}' | ffmpeg -i - -vcodec copy out.mov ffmpeg version N-77810-g405abdb Copyright (c) 2000-2016 the FFmpeg developers built with gcc 4.7 (SUSE Linux) configuration: --enable-gpl --enable-libx264 libavutil 55. 13.100 / 55. 13.100 libavcodec 57. 22.100 / 57. 22.100 libavformat 57. 21.101 / 57. 21.101 libavdevice 57. 0.100 / 57. 0.100 libavfilter 6. 23.100 / 6. 23.100 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100 [h264 @ 0x27a33c0] Stream #0: not enough frames to estimate rate; consider increasing probesize Input #0, h264, from 'pipe:': Duration: N/A, bitrate: N/A Stream #0:0: Video: h264 (Baseline), yuv420p, 1280x960, 25 fps, 25 tbr, 1200k tbn, 50 tbc [mov @ 0x27aa460] Codec for stream 0 does not use global headers but container format requires global headers [mov @ 0x27aa460] WARNING codec timebase is very high. If duration is too long, file may not be playable by quicktime. Specify a shorter timebase or choose different container. Output #0, mov, to 'out.mov': Metadata: encoder : Lavf57.21.101 Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p, 1280x960, q=2-31, 25 fps, 25 tbr, 1200k tbn, 1200k tbc Stream mapping: Stream #0:0 -> #0:0 (copy) [mov @ 0x27aa460] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly frame= 20 fps=0.0 q=-1.0 Lsize= 13896kB time=00:00:00.76 bitrate=149782.4kbits/s speed=17.2x video:13895kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.006635% }}} http://www.stardot-tech.com/developer/api.html describes a different but similar format, it also distinguishes between "Content-type: video/H.264I" for I-frames and "Content-type: video/H.264P" for P-frames. -- Ticket URL: <https://trac.ffmpeg.org/ticket/5154> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5885(undetermined:new): 32bit ppc fate client
by FFmpeg 22 Feb '24

22 Feb '24

#5885: 32bit ppc fate client -------------------------------------+------------------------------------- Reporter: michael | Owner: Type: task | Status: new Priority: normal | Component: Version: | undetermined unspecified | Keywords: fateclient Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 -------------------------------------+------------------------------------- We test build for ppc32 but not the actual regression tests If someone has a 32bit ppc please setup a fate client -- Ticket URL: <https://trac.ffmpeg.org/ticket/5885> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5456(undetermined:new): Videos produced with -shortest are longer than the audio
by FFmpeg 20 Feb '24

20 Feb '24

#5456: Videos produced with -shortest are longer than the audio -------------------------------------+------------------------------------- Reporter: hmemcpy | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: I tried following the guide for converting an audio file to video for YouTube, and I noticed that when using the `-shortest` switch, the videos produced were about 10 seconds longer than the audio track, and it was impossible to skip. After some searching, I found an old thread that suggests this was a bug introduced sometime in 2012, and suggested using an older version. I was able to find that last good build, after which the bug happens! Last good build was: `ffmpeg-20120720-git-85761ef` found here: https://ffmpeg.zeranoe.com/builds/win64/static/ All subsequent versions after it, including the very latest, have this incorrect behavior. How to reproduce: Attached is an 8 second .flac file (and an image for still frame) Run the following: {{{ ffmpeg -loop 1 -framerate 1 -i bg.jpg -i recit24bit.flac -shortest output.mp4 }}} When done with the "good" build 85761ef, the resulting mp4 is 10 seconds long When done with any other newer version, the resulting mp4 is 1 minute 08 seconds on my machine. See attached logs for additional info. -- Ticket URL: <https://trac.ffmpeg.org/ticket/5456> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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