December 2017 - FFmpeg-trac - lists.ffmpeg.org

#5406(undetermined:new): adpcm_vima: problem with stereo files
by FFmpeg 20 Aug '25

20 Aug '25

#5406: adpcm_vima: problem with stereo files -------------------------------------+------------------------------------- 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 -------------------------------------+------------------------------------- audio in the output file = noise http://www.datafilehost.com/d/afef00c6 {{{ C:\>ffmpeg -i D:\crushed_1.snm out.avi ffmpeg version N-78758-g5156578 Copyright (c) 2000-2016 the FFmpeg developers built with gcc 5.3.0 (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-amrwbenc --enable- libvorbis -- enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-lib x265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-d ecklink --enable-zlib libavutil 55. 19.100 / 55. 19.100 libavcodec 57. 27.100 / 57. 27.100 libavformat 57. 26.100 / 57. 26.100 libavdevice 57. 0.101 / 57. 0.101 libavfilter 6. 37.100 / 6. 37.100 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100 Guessed Channel Layout for Input Stream #0.1 : stereo Input #0, smush, from 'D:\crushed_1.snm': Duration: 00:01:40.40, start: 0.000000, bitrate: 2239 kb/s Stream #0:0: Video: sanm, rgb565le, 640x480, 15 fps, 15 tbr, 15 tbn, 15 tbc Stream #0:1: Audio: adpcm_vima, 22050 Hz, 2 channels, s16 File 'out.avi' already exists. Overwrite ? [y/N] y Output #0, avi, to 'out.avi': Metadata: ISFT : Lavf57.26.100 Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 640x480, q=2-31, 200 kb/s, 15 fps, 15 tbn, 15 tbc Metadata: encoder : Lavc57.27.100 mpeg4 Side data: cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1 Stream #0:1: Audio: mp3 (libmp3lame) (U[0][0][0] / 0x0055), 22050 Hz, stereo , s16p Metadata: encoder : Lavc57.27.100 libmp3lame Stream mapping: Stream #0:0 -> #0:0 (sanm (native) -> mpeg4 (native)) Stream #0:1 -> #0:1 (adpcm_vima (native) -> mp3 (libmp3lame)) Press [q] to stop, [?] for help [output stream 0:1 @ 04fb0180] 100 buffers queued in output stream 0:1, somethin g may be wrong. frame= 80 fps=0.0 q=31.0 size= 346kB time=00:00:08.46 bitrate= 335.3kbits/ frame= 194 fps=0.0 q=31.0 size= 673kB time=00:00:16.11 bitrate= 341.9kbits/ frame= 308 fps=205 q=31.0 size= 984kB time=00:00:23.71 bitrate= 339.9kbits/ [sanm @ 039ad9a0] Ignoring invalid motion vector (657, 472)->(480, 472), block s ize = 8 frame= 421 fps=210 q=31.0 size= 1337kB time=00:00:31.19 bitrate= 351.1kbits/ frame= 531 fps=212 q=31.0 size= 1754kB time=00:00:38.58 bitrate= 372.4kbits/ frame= 649 fps=216 q=31.0 size= 2002kB time=00:00:46.39 bitrate= 353.6kbits/ frame= 769 fps=220 q=31.0 size= 2239kB time=00:00:54.46 bitrate= 336.8kbits/ frame= 891 fps=223 q=31.0 size= 2420kB time=00:01:02.58 bitrate= 316.7kbits/ frame= 1014 fps=225 q=31.0 size= 2650kB time=00:01:10.79 bitrate= 306.6kbits/ frame= 1142 fps=228 q=31.0 size= 2831kB time=00:01:19.33 bitrate= 292.3kbits/ frame= 1261 fps=229 q=31.0 size= 3054kB time=00:01:27.24 bitrate= 286.7kbits/ frame= 1380 fps=230 q=31.0 size= 3245kB time=00:01:35.19 bitrate= 279.2kbits/ frame= 1499 fps=231 q=19.1 size= 3499kB time=00:01:43.13 bitrate= 277.9kbits/ frame= 1506 fps=231 q=13.8 Lsize= 3634kB time=00:01:43.68 bitrate= 287.2kbits /s speed=15.9x video:2681kB audio:810kB subtitle:0kB other streams:0kB global headers:0kB muxin g overhead: 4.083068% }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/5406> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5419(avformat:new): HLS EXT-X-DISCONTINUITY tag is not supported
by FFmpeg 05 Aug '25

05 Aug '25

#5419: HLS EXT-X-DISCONTINUITY tag is not supported -----------------------------------+--------------------------------------- Reporter: xyq204101 | Type: enhancement Status: new | Priority: normal Component: avformat | Version: git-master Keywords: hls | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -----------------------------------+--------------------------------------- Summary of the bug: I'm trying to read input from HLS and output to RTMP, this works fine if the HLS segments are all in one time sequence. However if the time changes e.g. segments generated from two video files, ffmpeg got error 'Non- monotonous DTS in output stream 0:1;'. According to the HLS documentation, EXT-X-DISCONTINUITY should be used when time sequence changes. (https://tools.ietf.org/html/draft-pantos- http-live-streaming-13#section-3.4.11) But I still got the same error after I added that tag to the m3u8 file. Look at hls.c file in libavformat, it doesn't handle EXT-X-DISCONTINUITY tag. How to reproduce: {{{ ffmpeg -re -i http://hls-server/playlist.m3u8 -acodec copy -vcodec copy -bsf:a aac_adtstoasc -f flv rtmp://rtmp-server/app }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/5419> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5413(ffmpeg:new): Image size, pixel format cannot be detected ( 0x0, -1 ) for short h264 video
by FFmpeg 03 Aug '25

03 Aug '25

#5413: Image size, pixel format cannot be detected ( 0x0,-1 ) for short h264 video -------------------------------------+------------------------------------- Reporter: sadensmol | Type: defect Status: new | Priority: normal Component: ffmpeg | Version: Keywords: image size | unspecified 0x0 | Blocked By: Blocking: | Reproduced by developer: 1 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Cannot convert video which plays fine with Windows Media Player. ffmpeg parameters ( splitted by comma) -probesize,100M,-analyzeduration,100M,-i,00443.MTS,-an,-y,-c:v,mjpeg,-q:v,8,frame%d.jpg ffmpeg output: [mpegts @ 0x7ffcd9808800] Could not find codec parameters for stream 0 (Video: h264 (HDMV / 0x564D4448), none): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, mpegts, from '/Users/saden/Documents/asimon/simplevideoed/test/folder8/00443.MTS': Duration: 00:00:00.51, start: 1.033367, bitrate: 9216 kb/s Program 1 Stream #0:0[0x1011]: Video: h264 (HDMV / 0x564D4448), none, 29.97 fps, 29.97 tbr, 90k tbn, 180k tbc Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz, 5.1(side), fltp, 448 kb/s Stream #0:2[0x1200]: Subtitle: hdmv_pgs_subtitle ([144][0][0][0] / 0x0090), 1920x1080 [buffer @ 0x7ffcd96013c0] Unable to parse option value "0x0" as image size [buffer @ 0x7ffcd96013c0] Unable to parse option value "-1" as pixel format [buffer @ 0x7ffcd96013c0] Unable to parse option value "0x0" as image size [buffer @ 0x7ffcd96013c0] Error setting option video_size to value 0x0. [graph 0 input from stream 0:0 @ 0x7ffcd9601500] Error applying options to the filter. This problem is easily reproducible with the latest version of ffmpeg on both OSX and windows systems. At the same time exiftool returns correct values for size. -- Ticket URL: <https://trac.ffmpeg.org/ticket/5413> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#5970(undetermined:new): No valid multicast port obtained from RTSP
by FFmpeg 11 Jul '25

11 Jul '25

#5970: No valid multicast port obtained from RTSP -------------------------------------+------------------------------------- Reporter: ioeir2 | 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: Cannot get multicast RTSP stream from camera via ffmpeg/ffplay. ffmpeg/ffplay tries to connect to UDP port 0 instead of 40000. It sends in SETUP: Transport: RTP/AVP/UDP;multicast But should something like Transport: RTP/AVP;multicast;destination=224.1.2.4;port=40000-40001 How to reproduce: IP camera: Amcrest IPM-HX1, sw. version 2.420.AC00.15.R, build : 2016-09-08. Very popular at the moment IP camera. {{{ ./ffplay-3.2 -rtsp_transport udp_multicast 'rtsp://admin:password@10.0.1.20/cam/realmonitor?channel=1&subtype=0&proto=Dahua3&unicast=false' ffplay version 3.2 Copyright (c) 2003-2016 the FFmpeg developers built with Apple LLVM version 8.0.0 (clang-800.0.42.1) configuration: libavutil 55. 34.100 / 55. 34.100 libavcodec 57. 64.100 / 57. 64.100 libavformat 57. 56.100 / 57. 56.100 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 [udp @ 0x7fb20bc9d500] bind failed: Permission denied 0B f=0/0 [udp @ 0x7fb20bc9b1a0] bind failed: Permission denied [udp @ 0x7fb20bc9b220] bind failed: Permission denied Segmentation fault: 11 }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/5970> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4566(ffmpeg:new): ffmpeg won't copy metadata ("StreamTitle") from audio live stream (transcoding)
by FFmpeg 29 Jun '25

29 Jun '25

#4566: ffmpeg won't copy metadata ("StreamTitle") from audio live stream (transcoding) ----------------------------------+------------------------------------- Reporter: easydoor | Type: defect Status: new | Priority: important Component: ffmpeg | Version: 2.6.2 Keywords: metadata | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+------------------------------------- It won't copy "StreamTitle" from input to output while transcoding input from mp3 to aac. I'm using version v2.6.1 compiled by me on ubuntu 14. I try simpliest transcoding with -c:a copy option and sending it to icecast v2.4.1 and as a client I'm using latest winamp and vlc. In booth player as title is showing only the mount point (in this case mystream) from icecast: {{{ icecast://source:mypass@x.x.x.x:8000/mystream }}} and not the StreamTitle from input file. Why and how can I just pass it from input to output? I don't think that problem is Icecast, because I try to stream with SAM Broadcaster and in Winamp I see updated StreamTitle in real time. Here is the command line: {{{ /root/ffmpeg_sources/ffmpeg*/./ffmpeg -stats -report -i http://streaming.streamonomy.com/ElectricFM -c:a copy -content_type audio/mpeg -f mp3 icecast://source:mypass@x.x.x.x:8000/mystream }}} and here is the output: {{{ root@vs4063:~# /root/ffmpeg_sources/ffmpeg*/./ffmpeg -stats -report -i http://streaming.streamonomy.com/ElectricFM -c:a copy -content_type audio/mpeg -f mp3 icecast://source:mypass@x.x.x.x:8000/mystream ffmpeg started on 2015-03-23 at 03:01:01 Report written to "ffmpeg-20150323-030101.log" ffmpeg version 2.6.1 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --prefix=/root/ffmpeg_build --extra- cflags=-I/root/ffmpeg_build/include --extra- ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --enable-gpl --enable- libass --enable-libfdk-aac --enable-libfaac --enable-libfreetype --enable- libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libaacplus --enable-nonfree libavutil 54. 20.100 / 54. 20.100 libavcodec 56. 26.100 / 56. 26.100 libavformat 56. 25.101 / 56. 25.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 11.102 / 5. 11.102 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, mp3, from 'http://streaming.streamonomy.com/ElectricFM': Metadata: icy-br : 128 icy-description : ElectricFM plays today's dance hits! icy-genre : Electronic Dance Pop icy-name : ElectricFM.com - America's Real Dance! icy-pub : 1 icy-url : http://www.electricfm.com StreamTitle : KYGO f/ CONRAD - FIRESTONE Duration: N/A, start: 0.000000, bitrate: 128 kb/s Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s Output #0, mp3, to 'icecast://source:mypass@x.x.x.x:8000/mystream': Metadata: icy-br : 128 icy-description : ElectricFM plays today's dance hits! icy-genre : Electronic Dance Pop icy-name : ElectricFM.com - America's Real Dance! icy-pub : 1 icy-url : http://www.electricfm.com StreamTitle : KYGO f/ CONRAD - FIRESTONE TSSE : Lavf56.25.101 Stream #0:0: Audio: mp3, 44100 Hz, stereo, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help size= 2093kB time=00:02:13.95 bitrate= 128.0kbits/s video:0kB audio:2093kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.016237% }}} [[Image(http://ffmpeg.gusari.org/download/file.php?id=103)]] In winamp first track is original and second transcoded.. Here is with debug: {{{ /root/ffmpeg_sources/ffmpeg*/./ffmpeg -stats -v 512 -report -i http://streaming.streamonomy.com/ElectricFM -c:a copy -content_type audio/mpeg -f mp3 icecast://source:mypass@x.x.x.x:8000/mystream/mystream ffmpeg started on 2015-03-23 at 03:51:36 Report written to "ffmpeg-20150323-035136.log" ffmpeg version 2.6.1 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1) configuration: --prefix=/root/ffmpeg_build --extra- cflags=-I/root/ffmpeg_build/include --extra- ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --enable-gpl --enable- libass --enable-libfdk-aac --enable-libfaac --enable-libfreetype --enable- libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libaacplus --enable-nonfree libavutil 54. 20.100 / 54. 20.100 libavcodec 56. 26.100 / 56. 26.100 libavformat 56. 25.101 / 56. 25.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 11.102 / 5. 11.102 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Splitting the commandline. Reading option '-stats' ... matched as option 'stats' (print progress report during encoding) with argument '1'. Reading option '-v' ... matched as option 'v' (set logging level) with argument '512'. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option '-i' ... matched as input file with argument 'http://streaming.streamonomy.com/ElectricFM'. Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'copy'. Reading option '-content_type' ... matched as AVOption 'content_type' with argument 'audio/mpeg'. Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'. Reading option 'icecast://source:mypass@x.x.x.x:8000/mystream' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option stats (print progress report during encoding) with argument 1. Applying option v (set logging level) with argument 512. Applying option report (generate a report) with argument 1. Successfully parsed a group of options. Parsing a group of options: input file http://streaming.streamonomy.com/ElectricFM. Successfully parsed a group of options. Opening an input file: http://streaming.streamonomy.com/ElectricFM. [http @ 0x217cf80] request: GET /ElectricFM HTTP/1.1 User-Agent: Lavf/56.25.101 Accept: */* Range: bytes=0- Connection: close Host: streaming.streamonomy.com Icy-MetaData: 1 [http @ 0x217cf80] header='HTTP/1.0 200 OK' [http @ 0x217cf80] http_code=200 [http @ 0x217cf80] header='Accept-Ranges: none' [http @ 0x217cf80] header='Content-Type: audio/mpeg' [http @ 0x217cf80] header='icy-br:128' [http @ 0x217cf80] header='ice-audio-info: ice-samplerate=44100;ice- bitrate=128;ice-channels=2' [http @ 0x217cf80] header='icy-br:128' [http @ 0x217cf80] header='icy-description:ElectricFM plays today's dance hits!' [http @ 0x217cf80] header='icy-genre:Electronic Dance Pop' [http @ 0x217cf80] header='icy-name:ElectricFM.com - America's Real Dance!' [http @ 0x217cf80] header='icy-pub:1' [http @ 0x217cf80] header='icy-url:http://www.electricfm.com' [http @ 0x217cf80] header='Server: Icecast 2.3.3-kh8' [http @ 0x217cf80] header='Cache-Control: no-cache, no-store' [http @ 0x217cf80] header='Pragma: no-cache' [http @ 0x217cf80] header='Access-Control-Allow-Origin: *' [http @ 0x217cf80] header='Access-Control-Allow-Headers: Origin, Accept, X-Requested-With, Content-Type' [http @ 0x217cf80] header='Access-Control-Allow-Methods: GET, OPTIONS, HEAD' [http @ 0x217cf80] header='Connection: close' [http @ 0x217cf80] header='Expires: Mon, 26 Jul 1997 05:00:00 GMT' [http @ 0x217cf80] header='icy-metaint:16000' [http @ 0x217cf80] header='' [mp3 @ 0x217daa0] Format mp3 probed with size=2048 and score=51 [mp3 @ 0x217daa0] Before avformat_find_stream_info() pos: 0 bytes read:2508 seeks:0 [mp3 @ 0x217daa0] All info found [mp3 @ 0x217daa0] After avformat_find_stream_info() pos: 21504 bytes read:23824 seeks:0 frames:50 Input #0, mp3, from 'http://streaming.streamonomy.com/ElectricFM': Metadata: icy-br : 128 icy-description : ElectricFM plays today's dance hits! icy-genre : Electronic Dance Pop icy-name : ElectricFM.com - America's Real Dance! icy-pub : 1 icy-url : http://www.electricfm.com StreamTitle : PAUL OAKENFOLD and DISFUNKTION f/ SPITFIRE - BEAUTIFUL WORLD Duration: N/A, start: 0.000000, bitrate: 128 kb/s Stream #0:0, 50, 1/14112000: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s Successfully opened the file. Parsing a group of options: output file icecast://source:mypass@x.x.x.x:8000/mystream. Applying option c:a (codec name) with argument copy. Applying option f (force format) with argument mp3. Successfully parsed a group of options. Opening an output file: icecast://source:mypass@x.x.x.x:8000/mystream. [http @ 0x2191520] request: PUT /mystream HTTP/1.1 User-Agent: Lavf/56.25.101 Accept: */* Expect: 100-continue Connection: close Host: x.x.x.x:8000 Content-Type: audio/mpeg Icy-MetaData: 1 Ice-Public: 0 Authorization: Basic c291cmNlOnNpbnVzZm04MA== [http @ 0x2191520] header='HTTP/1.1 100 Continue' [http @ 0x2191520] http_code=100 [http @ 0x2191520] header='' Successfully opened the file. Output #0, mp3, to 'icecast://source:mypass@x.x.x.x:8000/mystream5.3': Metadata: icy-br : 128 icy-description : ElectricFM plays today's dance hits! icy-genre : Electronic Dance Pop icy-name : ElectricFM.com - America's Real Dance! icy-pub : 1 icy-url : http://www.electricfm.com StreamTitle : PAUL OAKENFOLD and DISFUNKTION f/ SPITFIRE - BEAUTIFUL WORLD TSSE : Lavf56.25.101 Stream #0:0, 0, 1/14112000: Audio: mp3, 44100 Hz, stereo, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Press [q] to stop, [?] for help size= 101kB time=00:00:06.42 bitrate= 128.5kbits/s video:0kB audio:100kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.371530% Input file #0 (http://streaming.streamonomy.com/ElectricFM) Input stream #0:0 (audio): 246 packets read (102818 bytes); Total: 246 packets (102818 bytes) demuxed Output file #0 (icecast://source:mypass@x.x.x.x:8000/mystream): Output stream #0:0 (audio): 246 packets muxed (102818 bytes); Total: 246 packets (102818 bytes) muxed 0 frames successfully decoded, 0 decoding errors [AVIOContext @ 0x2192e80] Statistics: 0 seeks, 247 writeouts [AVIOContext @ 0x21884c0] Statistics: 105326 bytes read, 0 seeks }}} The main usage of transcoding would be: {{{ /root/ffmpeg_sources/ffmpeg/./ffmpeg -stats -v 1024 -report -i http://stream2.electricfm.com -acodec libaacplus -ar 44.1k -ab 40k -ac 2 -content_type audio/aacp -f adts icecast://source:mypass@x.x.x.x:8000/electricfm@40k -acodec libaacplus -ar 44.1k -ab 64k -ac 2 -content_type audio/aacp -f adts icecast://source:mypass@x.x.x.x:8000/electricfm@64k < /dev/null >/dev/null 2>/var/log/ffmpeg.log & }}} but it gave me same result..not showing "StreamTitle" from input to output! I try with options -map_metada 0:s:0 and others option, but without success. -- Ticket URL: <https://trac.ffmpeg.org/ticket/4566> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#6468(undetermined:new): ffmpeg doesn't maintain chapters when concatenating two MP4 files using the concat demuxer
by FFmpeg 21 Jun '25

21 Jun '25

#6468: ffmpeg doesn't maintain chapters when concatenating two MP4 files using the concat demuxer -------------------------------------+------------------------------------- Reporter: whereswardy | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- I'm trying to concatenate two MP4 files using the concat demuxer and want to retain the chapter markers that exist in both files so that the output file contains all the chapters from both of my input files. The source files contain chapter markers according to `mediainfo` which I want to be maintained in the output, but when using the `-map_chapters` option in ffmpeg they don't appear in the output file. I'm concatenating the following two files from `streams.txt`: {{{ file 'episode2.mp4' file 'episode3.mp4' }}} which contain the following chapter/menu data (output cut down for brevity): {{{ $ mediainfo episode2.mp4 ... Menu ID : 3 Codec ID : text Duration : 22mn 25s Language : English Encoded date : UTC 2017-06-18 15:27:40 Tagged date : UTC 2017-06-18 15:27:40 Bit rate mode : CBR 00:00:00.000 : Chapter 1 00:00:59.720 : Chapter 2 ... }}} and I'm using the following ffmpeg command: {{{ ffmpeg -f concat -safe 0 -i streams.txt -map_metadata 0 -map_chapters 0 -c copy output.mp4 }}} however the output.mp4 file contains no chapter markers according to `mediainfo` and there's nothing viewable in any media player. I'm using ffmpeg 3.3.2 on Ubuntu 14.04.5: {{{ $ ffmpeg -version ffmpeg version 3.3.1 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3) configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man --enable-avresample --disable-debug --enable- nonfree --enable-gpl --enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse --enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 --enable-libfdk-aac --enable-libvorbis --enable- libmp3lame --enable-libopus --enable-libvpx --enable-libspeex --enable- libass --enable-avisynth --enable-libsoxr --enable-libxvid --enable- libvidstab --enable-libwavpack --enable-nvenc libavutil 55. 58.100 / 55. 58.100 libavcodec 57. 89.100 / 57. 89.100 libavformat 57. 71.100 / 57. 71.100 libavdevice 57. 6.100 / 57. 6.100 libavfilter 6. 82.100 / 6. 82.100 libavresample 3. 5. 0 / 3. 5. 0 libswscale 4. 6.100 / 4. 6.100 libswresample 2. 7.100 / 2. 7.100 libpostproc 54. 5.100 / 54. 5.100 }}} {{{ $ uname -a Linux server 3.13.0-119-generic #166-Ubuntu SMP Wed May 3 12:18:55 UTC 2017 x86_64 x86_64 x86_64 GNU/Linux }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/6468> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4908(avformat:new): Memory leak in ff_img_read_header() with custom io
by FFmpeg 06 Jun '25

06 Jun '25

#4908: Memory leak in ff_img_read_header() with custom io -------------------------------------+------------------------------------- Reporter: | Type: defect personalmountains | Priority: normal Status: new | Version: Component: avformat | unspecified Keywords: memory leak | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- [https://github.com/FFmpeg/FFmpeg/commit/726e253b004de9511f63625ce95feee4d9a… This patch] changed `probe_buffer` from an array to a pointer allocated with `av_realloc()`. It was correctly freed at the end of the block. However, [https://github.com/FFmpeg/FFmpeg/commit/6691eee48c7f08caf479ce3562a1c834475… that one] buried the `av_free()` call into `ffio_rewind_with_probe_data()`. Unsurprisingly, [https://github.com/FFmpeg/FFmpeg/commit/e6e8cc8ce9c2a398fbb51254a5067f4bd3c… a subsequent patch] made it so this call is skipped in the case of custom io, so the buffer is never freed. A simple `av_free()` after the call to `avio_seek()` on [https://github.com/FFmpeg/FFmpeg/blob/master/libavformat/img2dec.c#L346 img2dec.c:346] might do the job. {{{ int probe_buffer_size = 2048; uint8_t *probe_buffer = av_realloc(NULL, probe_buffer_size + AVPROBE_PADDING_SIZE); // ... if (s1->flags & AVFMT_FLAG_CUSTOM_IO) { // this leaks probe_buffer because av_free() isn't called avio_seek(s1->pb, 0, SEEK_SET); } else ffio_rewind_with_probe_data(s1->pb, &probe_buffer, probe_buffer_size); }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/4908> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3643(undetermined:new): XYZ to RGB color space
by FFmpeg 27 May '25

27 May '25

#3643: XYZ to RGB color space -------------------------------------+------------------------------------- Reporter: tteam | Type: Status: new | enhancement Component: | Priority: normal undetermined | Version: Keywords: more XYZ to | unspecified RGB matrix | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: XYZ to RGB colorspace conversion use only sRGB matrix and sometimes the conversion is deffected in red channel and the picture is dark. Hi all! Is it possible to add more color space matrix (in option) to XYZ to RGB conversion? Because the FFMPEG use only one matrix (sRGB) but lots of available, and best if the user can choose one. Because the problem is: sometimes the conversion is deffected the red is much more and the picture is dark. If possible please add some from this list. The default is the sRGB but optional the others. XYZ to RGB matrices: http://www.brucelindbloom.com/index.html?Eqn_RGB_XYZ_Matrix.html Thanks -- Ticket URL: <https://trac.ffmpeg.org/ticket/3643> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4448(avformat:new): Properly support cover art in ogg
by FFmpeg 22 May '25

22 May '25

#4448: Properly support cover art in ogg --------------------------------------+---------------------------------- Reporter: llogan | Owner: Type: defect | Status: new Priority: normal | Component: avformat Version: git-master | Keywords: ogg Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 --------------------------------------+---------------------------------- {{{ $ ./ffmpeg -i ../fate-suite/cover_art/cover_art.flac ../out.ogg ffmpeg version N-71319-g43b4342 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.2 (GCC) 20150304 (prerelease) configuration: --enable-gpl --disable-doc libavutil 54. 22.100 / 54. 22.100 libavcodec 56. 34.100 / 56. 34.100 libavformat 56. 29.100 / 56. 29.100 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 13.101 / 5. 13.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, flac, from '../fate-suite/cover_art/cover_art.flac': Metadata: RATING : 0 ALBUM : Ангелофрения ARTIST : Мельница TITLE : Дороги TRACKTOTAL : 11 TOTALTRACKS : 11 GENRE : Folk DATE : 2012 track : 2 Duration: 00:03:09.05, start: 0.000000, bitrate: 1 kb/s Stream #0:0: Audio: flac, 44100 Hz, stereo, s16 Stream #0:1: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 350x350 [SAR 1:1 DAR 1:1], 90k tbr, 90k tbn, 90k tbc Metadata: comment : Other Output #0, ogg, to '../out.ogg': Metadata: RATING : 0 ALBUM : Ангелофрения ARTIST : Мельница TITLE : Дороги TRACKTOTAL : 11 TOTALTRACKS : 11 GENRE : Folk DATE : 2012 track : 2 Stream #0:0: Video: theora, none, q=2-31, 128 kb/s Metadata: comment : Other Stream #0:1: Audio: flac, 0 channels, 128 kb/s Stream mapping: Stream #0:1 -> #0:0 (mjpeg (native) -> ? (?)) Stream #0:0 -> #0:1 (flac (native) -> flac (native)) Encoder (codec theora) not found for output stream #0:0 }}} Of course if Theora is supported by the build then a Theora video stream will be outputted (90k frame rate in this case). Possibly related to #4442 since I assume the cover art mechanics would be similar: https://wiki.xiph.org/VorbisComment#Cover_art -- Ticket URL: <https://trac.ffmpeg.org/ticket/4448> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#6091(avformat:new): support ds2 audio (dss pro audio) file format
by FFmpeg 03 May '25

03 May '25

#6091: support ds2 audio (dss pro audio) file format -----------------------------------+--------------------------------------- Reporter: vineet156 | Type: enhancement Status: new | Priority: normal Component: avformat | Version: git-master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -----------------------------------+--------------------------------------- Summary of the bug: ffmpeg already supports dss audio file decoding. Please add support for ds2 as well, which is an improved version of dss and called "dss pro" How to reproduce: {{{ ffmpeg started on 2017-01-19 at 04:20:46 Report written to "ffmpeg-20170119-042046.log" Command line: ../bin/ffmpeg -report -i VoiceRec11.ds2 VoiceRec11.mp3 ffmpeg version N-83132-g9561de4 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609 configuration: --prefix=/home/ubuntu/ffmpeg_build --pkg-config- flags=--static --extra-cflags='-I/home/ubuntu/ffmpeg_build/include -static' --extra-ldflags='-L/home/ubuntu/ffmpeg_build/lib -static' --bindir=/home/ubuntu/bin --enable-libmp3lame libavutil 55. 43.100 / 55. 43.100 libavcodec 57. 73.100 / 57. 73.100 libavformat 57. 62.100 / 57. 62.100 libavdevice 57. 2.100 / 57. 2.100 libavfilter 6. 69.100 / 6. 69.100 libswscale 4. 3.101 / 4. 3.101 libswresample 2. 4.100 / 2. 4.100 Splitting the commandline. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option '-i' ... matched as input url with argument 'VoiceRec11.ds2'. Reading option 'VoiceRec11.mp3' ... matched as output url. Finished splitting the commandline. Parsing a group of options: global . Applying option report (generate a report) with argument 1. Successfully parsed a group of options. Parsing a group of options: input url VoiceRec11.ds2. Successfully parsed a group of options. Opening an input file: VoiceRec11.ds2. [file @ 0x3e5db40] Setting default whitelist 'file,crypto' [AVIOContext @ 0x3e66f60] Statistics: 1048576 bytes read, 0 seeks VoiceRec11.ds2: Invalid data found when processing input }}} A ds2 sample file is attached for testing. Also making a small donation to the project thank you, -- Ticket URL: <https://trac.ffmpeg.org/ticket/6091> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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