September 2022 - FFmpeg-trac - lists.ffmpeg.org

#9846(ffmpeg:new): DNxHR doesn't work with LB in Shutter Encoder
by FFmpeg 10 Jul '25

10 Jul '25

#9846: DNxHR doesn't work with LB in Shutter Encoder -------------------------------------+------------------------------------- Reporter: Soto Soto | Type: defect Status: new | Priority: normal Component: ffmpeg | Version: Keywords: ffmpeg | unspecified dnxhr | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- '''Summary of the bug:''' I've just been forwarded by the creator of Shutter Encoder who uses FFmpeg as the backend of their software. I wanted to create a small 480x270 file at low bitrate and the render stops after a few minutes. The last line the console says is "Assertion s->buf_ptr < s->buf_end failed at src/libavcodec/put_bits.h:150". How to reproduce: 1. Used a 1920x1080p, 60fps, h.264 10gb file. 2. Dropped it in Shutter Encoder, set to DNxHR > LB ("Low Bitrate" doesn't state the number of the bitrate itself). Change scale to 480x270p 3. Render 4. Error shown in the last line of the console (right click in progress bar and click "Console"). -- Ticket URL: <https://trac.ffmpeg.org/ticket/9846> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9281(ffmpeg:new): wrong $Number$ value while reading mpd
by FFmpeg 04 Jul '25

04 Jul '25

#9281: wrong $Number$ value while reading mpd -------------------------------------+------------------------------------- Reporter: Kostik | Type: defect Status: new | Priority: normal Component: ffmpeg | Version: Keywords: mpd | unspecified manifest | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Hello, using the latest static build ffmpeg version N-57631-gebedd26eef-static I want to read mpd and convert to mpegts - this part works ok, except one manifest looks like ffmpeg make wrong calculation for $Number$ In my case I have availabilityStartTime="2021-05-27T11:59:16Z" media="$RepresentationID$-$Number$.m4s" duration="180180" startNumber="487925" presentationTimeOffset="89360526151" I expect $Number$ to be converted to 488430 (and other players do so), but ffmpeg try retrieve 397033568 here is full manifest <?xml version="1.0" encoding="UTF-8"?> <MPD xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xmlns="urn:mpeg:dash:schema:mpd:2011" xmlns:cenc="urn:mpeg:cenc:2013" xmlns:mspr="urn:microsoft:playready" xsi:schemaLocation="urn:mpeg:dash:schema:mpd:2011 DASH-MPD.xsd" type="dynamic" minimumUpdatePeriod="PT6S" minBufferTime="PT1.000S" maxSegmentDuration="PT3.003S" availabilityStartTime="2021-05-27T11:59:16Z" timeShiftBufferDepth="PT36.036S" publishTime="2021-06-07T19:25:48Z" profiles="urn:mpeg:dash:profile:isoff-live:2011"> <Period id="487925" start="PT976813.47S"> <AdaptationSet mimeType="video/mp4" segmentAlignment="true" startWithSAP="1" maxWidth="1280" maxHeight="720" maxFrameRate="60000/1001" par="16:9"> <InbandEventStream schemeIdUri="urn:mpeg:dash:event:2012" value="1"/> <Accessibility schemeIdUri="urn:scte:dash:cc:cea-608:2015" value="CC1:und;CC2:und"/> <SegmentTemplate timescale="90000" media="$RepresentationID$-$Number$.m4s" startNumber="487925"/> <Representation id="video1" width="1280" height="720" frameRate="60000/1001" sar="1:1" scanType="progressive" bandwidth="6562944" codecs="avc1.640029"/> </AdaptationSet> <AdaptationSet mimeType="audio/mp4" lang="en" segmentAlignment="true" startWithSAP="1"> <Role schemeIdUri="urn:mpeg:dash:role:2011" value="main"/> <SegmentTemplate timescale="90000" media="$RepresentationID$-$Number$.m4s" startNumber="487925"/> <Representation id="audio1" audioSamplingRate="44100" bandwidth="96000" codecs="mp4a.40.2"> <AudioChannelConfiguration schemeIdUri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"/> </Representation> </AdaptationSet> <AdaptationSet mimeType="audio/mp4" lang="es" segmentAlignment="true" startWithSAP="1"> <Role schemeIdUri="urn:mpeg:dash:role:2011" value="dub"/> <SegmentTemplate timescale="90000" media="$RepresentationID$-$Number$.m4s" startNumber="487925"/> <Representation id="audio2" audioSamplingRate="44100" bandwidth="96000" codecs="mp4a.40.2"> <AudioChannelConfiguration schemeIdUri="urn:mpeg:dash:23003:3:audio_channel_configuration:2011" value="2"/> </Representation> </AdaptationSet> </Period> <UTCTiming schemeIdUri="urn:mpeg:dash:utc:direct:2014" value="2021-06-07T19:25:48Z"/> </MPD> -- Ticket URL: <https://trac.ffmpeg.org/ticket/9281> 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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#9522(undetermined:new): Certain H.264 videos cannot decode with videotoolbox
by FFmpeg 19 Jun '25

19 Jun '25

#9522: Certain H.264 videos cannot decode with videotoolbox -------------------------------------+------------------------------------- Reporter: forthrin | Type: defect Status: new | Priority: normal Component: | Version: git- undetermined | master Keywords: | Blocked By: videotoolbox gpu | Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- For reasons unknown, a percentage of H.264 videos fail to hardware decode with videotoolbox. Enclosed is a dozen different single frame specimens that all cause the same problem and should thus help quickly identify and resolve the issue. {{{ $ ffmpeg -hwaccel videotoolbox -i 01.mp4 -xerror -f mp4 -y /dev/null ffmpeg version git-2021-11-21-18f22bf Copyright (c) 2000-2021 the FFmpeg developers built with Apple clang version 12.0.5 (clang-1205.0.22.9) configuration: --enable-nonfree --enable-gpl --enable-libass --enable- libx264 --enable-libx265 --enable-libssh --enable-libzimg --enable-opencl --enable-libmp3lame libavutil 57. 9.100 / 57. 9.100 libavcodec 59. 13.100 / 59. 13.100 libavformat 59. 9.101 / 59. 9.101 libavdevice 59. 0.101 / 59. 0.101 libavfilter 8. 17.100 / 8. 17.100 libswscale 6. 1.100 / 6. 1.100 libswresample 4. 0.100 / 4. 0.100 libpostproc 56. 0.100 / 56. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '01.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 Duration: 00:00:00.04, start: 0.000000, bitrate: 402 kb/s Stream #0:0[0x1](und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 244 kb/s, 23.99 fps, 23.99 tbr, 16k tbn (default) Metadata: handler_name : VideoHandler vendor_id : [0][0][0][0] Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Press [q] to stop, [?] for help Error creating Videotoolbox decoder. videotoolbox hwaccel requested for input stream #0:0, but cannot be initialized. [h264 @ 0x7fce00d13580] decode_slice_header error [h264 @ 0x7fce00d13580] no frame! Conversion failed! $ }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/9522> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9473(ffmpeg:new): gdigrab screen captures are interrupted by secure desktop pop-ups
by FFmpeg 15 Jun '25

15 Jun '25

#9473: gdigrab screen captures are interrupted by secure desktop pop-ups -----------------------------------+-------------------------------------- Reporter: leopold84 | Type: defect Status: new | Priority: important Component: ffmpeg | Version: git-master Keywords: gdigrab | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -----------------------------------+-------------------------------------- '''Summary of the bug:''' If you start screen capture with gdigrab and any secure desktop pop-up will appear (UAC or anything else), capture immediately stops. '''How to reproduce:''' 1. Start screen capture using '''gdigrab''' device as input {{{ ffmpeg -f gdigrab -framerate 30 -i desktop output.mkv }}} 2. Launch any program as administrator or make a secure desktop to appear any other way 3. Once returned to the normal desktop, you will see that capture has stopped with the error similar to this one {{{ Failed to capture image (error 5)trate= 2.2kbits/s speed=0.54x desktop: I/O error }}} '''Expect result:''' If it's not possible to capture secure desktop with gdigrab, at least make it so recording doesn't stop and capture a black screen instead of secure desktop -- Ticket URL: <https://trac.ffmpeg.org/ticket/9473> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9751(swscale:new): Incorrect YUV->RGB24 conversion results on IBM Power9
by FFmpeg 09 Jun '25

09 Jun '25

#9751: Incorrect YUV->RGB24 conversion results on IBM Power9 -------------------------------------+------------------------------------- Reporter: gciaparrone | Type: defect Status: new | Priority: normal Component: swscale | Version: git- | master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Converting a YUV frame into an RGB24 image seems to produce incorrect results on an IBM Power9 processor. Compiling ffmpeg with --disable- altivec solves the issue. The problem might be related to this old issue pointed out by Hong Bo Peng here: https://trac.ffmpeg.org/ticket/7124#comment:7 This can be more easily seen by converting a completely black image from RGB to YUV and then back to RGB on an IBM Power9 (or possibly on any Little Endian processor with ALTIVEC support?). The output image has all pixels set to [16, 16, 16] instead of [0, 0, 0]. The issue spans multiple versions of ffmpeg, including the current git master version. Using --disable-altivec or running the command on a non-ALTIVEC processor outputs the correct image with all [0, 0, 0] pixels. Input image: Output image (IBM Power9): Output image (AMD Ryzen 5 1600X processor): Output image (IBM Power9, --disable-altivec): How to reproduce: {{{ # convert RGB image to YUV420p $ ffmpeg -i black.png -pix_fmt yuv420p black.yuv # convert it back to RGB $ ffmpeg -s:v 64x64 -pix_fmt yuv420p -i black.yuv -frames:v 1 -pix_fmt rgb24 reblack.png }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/9751> 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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#9471(ffmpeg:new): EAC3 native encoder is only gapless in the beginning, not in the end
by FFmpeg 27 May '25

27 May '25

#9471: EAC3 native encoder is only gapless in the beginning, not in the end -------------------------------------+------------------------------------- Reporter: Balling | Type: defect Status: new | Priority: normal Component: ffmpeg | Version: git- Keywords: mp4 eac3 | master editlist gapless | Blocked By: Blocking: | Reproduced by developer: 1 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: edts atmos (editlist) has media time and media duration, yet even though media time is correctly written for EAC3 and AAC (native EAC3 encoder has 256 sample of silence (a.k.a. encoder delay) that are then removed from the beginning with native encoder and native AAC has 1024 samples that are also working great) the media duration is not correctly written. Even if it were to be correctly written media duration is not applied on decoding even for AAC, see for example https://bugs.chromium.org/p/chromium/issues/detail?id=668999 that is still present in git-master!!! How to reproduce: {{{ % ffmpeg -f lavfi -i "sine=frequency=1000:duration=5" -c:a eac3 outeac3.mp4 ffmpeg version N-104341-g933765aa0e-20211013 Copyright (c) 2000-2021 the FFmpeg developers built with gcc 10-win32 (GCC) 20210408 configuration: --prefix=/ffbuild/prefix --pkg-config-flags=--static --pkg-config=pkg-config --cross-prefix=x86_64-w64-mingw32- --arch=x86_64 --target-os=mingw32 --enable-gpl --enable-version3 --disable-debug --enable-shared --disable-static --disable-w32threads --enable-pthreads --enable-iconv --enable-libxml2 --enable-zlib --enable-libfreetype --enable-libfribidi --enable-gmp --enable-lzma --enable-fontconfig --enable-libvorbis --enable-opencl --enable-libvmaf --enable-vulkan --disable-libxcb --disable-xlib --enable-amf --enable-libaom --enable- avisynth --enable-libdav1d --enable-libdavs2 --disable-libfdk-aac --enable-ffnvcodec --enable-cuda-llvm --enable-frei0r --enable-libglslang --enable-libgme --enable-libass --enable-libbluray --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvpx --enable-libwebp --enable-lv2 --enable-libmfx --enable-libopencore-amrnb --enable- libopencore-amrwb --enable-libopenjpeg --enable-librav1e --enable- librubberband --enable-schannel --enable-sdl2 --enable-libsoxr --enable- libsrt --enable-libsvtav1 --enable-libtwolame --enable-libuavs3d --disable-libdrm --disable-vaapi --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libzimg --enable-libzvbi --extra-cflags=-DLIBTWOLAME_STATIC --extra-cxxflags= --extra-ldflags=-pthread --extra-ldexeflags= --extra-libs=-lgomp --extra- version=20211013 libavutil 57. 7.100 / 57. 7.100 libavcodec 59. 12.100 / 59. 12.100 libavformat 59. 6.100 / 59. 6.100 libavdevice 59. 0.101 / 59. 0.101 libavfilter 8. 14.100 / 8. 14.100 libswscale 6. 1.100 / 6. 1.100 libswresample 4. 0.100 / 4. 0.100 libpostproc 56. 0.100 / 56. 0.100 Input #0, lavfi, from 'sine=frequency=1000:duration=5': Duration: N/A, start: 0.000000, bitrate: 705 kb/s Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s Stream mapping: Stream #0:0 -> #0:0 (pcm_s16le (native) -> eac3 (native)) Press [q] to stop, [?] for help Output #0, mp4, to 'outeac3.mp4': Metadata: encoder : Lavf59.6.100 Stream #0:0: Audio: eac3 (ec-3 / 0x332D6365), 44100 Hz, mono, fltp, 96 kb/s Metadata: encoder : Lavc59.12.100 eac3 size= 60kB time=00:00:05.00 bitrate= 98.2kbits/s speed= 487x video:0kB audio:59kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.166617% }}} {{{ Track duration: 5010 (0x00001392) - 5010 (0x1392) ms Media time: 256 (0x00000100) - 256 (0x100) ms Media rate: 65536 (0x00010000) - 1.000 }}} Now compare it to aac: ffmpeg -f lavfi -i "sine=frequency=1000:duration=5" -c:a aac fileaac.mp4: {{{ Track duration: 5000 (0x00001388) - 5000 (0x1388) ms Media time: 1024 (0x00000400) - 1024 (0x400) Media rate: 65536 (0x00010000) - 1.000 }}} Unfortunately a) Mediainfo tracer is buggy in that part: https://github.com/MediaArea/MediaInfoLib/issues/1441 b) I am not sure that media duration is really buggy since it is not applied anyway! c) I checked it all decoding to wav and checking in Audacity. d) I dunno whether sbgp and sgpd are needed (whether EAC3 depends on previous frames) Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker. -- Ticket URL: <https://trac.ffmpeg.org/ticket/9471> 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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