March 2025 - 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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#10732(undetermined:new): avcodec_flush_buffers() not resetting E-AC-3 decoder
by FFmpeg 08 Jul '25

08 Jul '25

#10732: avcodec_flush_buffers() not resetting E-AC-3 decoder ----------------------------------------+---------------------------------- Reporter: Peter Krefting | Type: defect Status: new | Priority: minor Component: undetermined | Version: 5.0.3 Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------------+---------------------------------- Summary of the bug: I am reusing the same AVCodecContext to decode several parallel E-AC-3 audio streams. As per documentation I am calling [https://ffmpeg.org/doxygen/trunk/group__lavc__misc.html#gaf60b0e076f822abcb… avcodec_flush_buffers()] between each decoding run. This seems to work for most codecs, but when used with the E-AC-3 decoder the output is not identical between runs How to reproduce: {{{#!c #include <libavcodec/avcodec.h> #include <libavformat/avformat.h> #include <stdio.h> #include <assert.h> int main(int argc, char *argv[]) { const AVCodec *codec = avcodec_find_decoder(AV_CODEC_ID_EAC3); AVCodecContext *decoder = avcodec_alloc_context3(codec); avcodec_open2(decoder, codec, NULL); assert(decoder->sample_fmt == AV_SAMPLE_FMT_FLTP); for (int l = 0; l < 2; ++ l) { printf("[%d] Reading file\n", l); avcodec_flush_buffers(decoder); AVFormatContext *avf_ctx = NULL; avformat_open_input(&avf_ctx, "file:i28553p303t610156395819694.ac3", NULL, NULL); avformat_find_stream_info(avf_ctx, NULL); AVPacket avpkt = { 0 }; while (av_read_frame(avf_ctx, &avpkt) == 0) { avcodec_send_packet(decoder, &avpkt); AVFrame *frame = av_frame_alloc(); avcodec_receive_frame(decoder, frame); printf("Got %d samples in %d channels\n", frame->nb_samples, frame->channels); for (int i = 0; i < frame->nb_samples; ++ i) { printf("%4d:", i); for (int j = 0; j < frame->channels; ++ j) { printf(" [%d]%f", j, ((float *) frame->data[j])[i]); } printf("\n"); } av_frame_free(&frame); } avformat_close_input(&avf_ctx); } avcodec_free_context(&decoder); return 0; } }}} The test file is available here: https://e.pcloud.link/publink/show?code=XZS8y1ZlEryJIDYU0yIzDTUhw1sDfHDRtUV and contains a ten-second extract of an audio track from an MPEGTS container, captured off-air while looking at a problem where the encoder suddenly only outputs silence. I expect the output to be identical on the first and second run, but that is not what I am seeing: {{{ $ ./decode |grep -A5 'Reading file' [eac3 @ 0x55fc4b0a5ec0] Estimating duration from bitrate, this may be inaccurate [0] Reading file Got 1536 samples in 6 channels 0: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 1: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 2: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 3: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 [eac3 @ 0x55fc4b11a840] Estimating duration from bitrate, this may be inaccurate -- [1] Reading file Got 1536 samples in 6 channels 0: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 1: [0]-0.000000 [1]0.000000 [2]-0.000000 [3]-0.000000 [4]-0.000000 [5]-0.000000 2: [0]0.000000 [1]0.000000 [2]0.000000 [3]0.000000 [4]0.000000 [5]0.000000 3: [0]-0.000000 [1]-0.000000 [2]-0.000000 [3]-0.000000 [4]-0.000000 [5]-0.000000 }}} In real life we are also interleaving other E-AC-3 audio streams from the same MPEGTS between decodes, and see that the initial decoded samples differ even more than when just redecoding the same sample over and over. -- Ticket URL: <https://trac.ffmpeg.org/ticket/10732> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#11366(undetermined:new): Regression when prepending/concatenating still image to video
by FFmpeg 07 Jul '25

07 Jul '25

#11366: Regression when prepending/concatenating still image to video -------------------------------------+------------------------------------- Reporter: mahozad | Type: defect Status: new | Priority: normal Component: | Version: 7.1 undetermined | Keywords: concat | Blocked By: concatenate filter_complex | Blocking: | Reproduced by developer: 1 Analyzed by developer: 0 | -------------------------------------+------------------------------------- **Summary of the bug:** I'm using FFmpeg to create clips with ability to concat an image to be shown at the start of the clip for the specified duration: https://github.com/mahozad/cutcon It uses FFmpeg CLI. FFmpeg version 6.1.1 (and I think also version 5.1.2) worked as expected. When upgrading to FFmpeg 7.1, the result video just contains the intro image and then just ends. I'm not sure if this is a regression. Maybe my FFmpeg command is "wrong" and it should have failed in previous versions as well? **How to reproduce:** FFmpeg 7.1 and also FFmpeg-full 2024-12-19-git-494c961379 downloaded from https://www.gyan.dev/ffmpeg/builds/ both tested on Windows 11. Here is an example FFmpeg invocation (formatted, each option in a new line) to reproduce the problem. It tries to create a clip from an input MP4 file and show an image at the start for 8 seconds. The image size is coerced to be at maximum the size of video and is padded with black color if its aspect ratio is different than the video. There is some other filters too. {{{ path/to/ffmpeg.exe -y -nostdin -loglevel info -stats_period 0.1s -accurate_seek -ss 356500ms -to 378500ms -i path/to/a/video.mp4 -loop 1 -framerate 25 -t 8s -i path/to/a/picture.jpg -t 8s -f lavfi -i aevalsrc=0 -filter_complex [0] yadif=1 [media-de-interlaced]; [media-de-interlaced] scale=iw*sar:ih [media-anamorphic-fixed]; [media-anamorphic-fixed] pad=ceil(iw/2)*2:ceil(ih/2)*2 [media-odd- pixel-number-fixed]; [media-odd-pixel-number-fixed] setsar=1/1 [media-source-aspect- ratio-normalized]; [media-source-aspect-ratio-normalized] null [media]; [1][media] scale2ref='if(gt(main_a,a),min(main_w,iw),min(main_h,ih)*main_a)':'if(gt(main_a,a),min(main_w,iw)/main_a,min(main_h,ih))' [intro-image-max-size-coerced][media]; [1][media] scale2ref=iw:ih [size-template][media]; [size-template] drawbox=w=iw:h=ih:t=fill:color=#000000 [intro- background]; [intro-background][intro-image-max-size-coerced] overlay=x='(W-w)/2':y='(H-h)/2' [intro-with-background]; [intro-with-background] setsar=1/1 [intro-source-aspect-ratio- normalized]; [intro-source-aspect-ratio-normalized] null [intro]; [intro][2][media] concat=n=2:v=1:a=1 -c:v libx264 -c:a aac -crf 23 -r 25 -metadata encoding_tool=Cutcon v1.2.3 path/to/output.mp4 }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/11366> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#10959(undetermined:new): ffmpeg 7.0: Assertion best_input >= 0 failed at fftools/ffmpeg_filter.c:1923
by FFmpeg 07 Jul '25

07 Jul '25

#10959: ffmpeg 7.0: Assertion best_input >= 0 failed at fftools/ffmpeg_filter.c:1923 -------------------------------------+------------------------------------- Reporter: Andreas | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- I try to split a video into 3 parts: {{{ % C:/Java/ffmpeg-n7.0-latest-win64-gpl-7.0/bin/ffmpeg.exe -report -nostdin -y -i video.mp4 -filter_complex "[v:0]select=n='3':e='gte(t,0.00)*lt(t,10.45)*1 + gte(t,10.45)*lt(t,58.50)*2 + gte(t,58.50)*lt(t,3661.00)*3'[vout001][vout002][vout003]" -map [vout001] v00-001.mp4 -map [vout002] v00-002.mp4 -map [vout003] v00-003.mp4 ... [out_0_0 @ 0000021b889bd600] 100 buffers queued in out_0_0, something may be wrong. ... [out_1_0 @ 0000021b889be200] 100 buffers queued in out_1_0, something may be wrong. [out_1_0 @ 0000021b889be200] 1000 buffers queued in out_1_0, something may be wrong. ... Assertion best_input >= 0 failed at fftools/ffmpeg_filter.c:1923 }}} It is independent of the input video, but you can find the video here: https://a-voss.de/ffmpeg/video.mp4 -- Ticket URL: <https://trac.ffmpeg.org/ticket/10959> 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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#11421(undetermined:new): ffmpeg 7.1 fps bug
by FFmpeg 30 Jun '25

30 Jun '25

#11421: ffmpeg 7.1 fps bug --------------------------------------+---------------------------------- Reporter: 张文兵 | Type: defect Status: new | Priority: normal Component: undetermined | Version: 7.1 Keywords: fps | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | --------------------------------------+---------------------------------- ffmpeg version n7.1 Copyright (c) 2000-2024 the FFmpeg developers {{{ % ffmpeg -i xxx.mp4 -c:v libx264 -crf 23 -vf scale=1920:1080,fps=fps=25,format=yuv422p10le -map 0:v:0 -c:a pcm_s16le -b:a 1536000 -ar 48000 -map "0:a?" -y v4.mp4 }}} After adding the fps filter, the transcoded file will have one less frame of data. -- Ticket URL: <https://trac.ffmpeg.org/ticket/11421> 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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#11204(undetermined:new): ffplay cannot play raw .thd files or .mlp files
by FFmpeg 26 Jun '25

26 Jun '25

#11204: ffplay cannot play raw .thd files or .mlp files -------------------------------------+------------------------------------- Reporter: Balling | Type: defect Status: new | Priority: normal Component: | Version: git- undetermined | master Keywords: regression | Blocked By: thd mlp | Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: was working fine back in 4a11a6f4ccc7c56ccc82adf0c3ab4054d4c22d1e but then Paul broke it How to reproduce (there will be artefacts, thankfull does not happen when decoding to wav even with -c:a pcm_s24le), also happens with actual MLP https://trac.ffmpeg.org/raw-attachment/ticket/10540/7s-sinetone- sample.mlp: {{{ PS > ffplay https://trac.ffmpeg.org/raw- attachment/ticket/9511/truehd_regression.mlp ffplay version N-117043-g8707c8660d-20240915 Copyright (c) 2003-2024 the FFmpeg developers built with gcc 14.2.0 (crosstool-NG 1.26.0.106_ed12fa6) 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-zlib --enable-libfreetype --enable-libfribidi --enable-gmp --enable-libxml2 --enable-lzma --enable-fontconfig --enable- libharfbuzz --enable-libvorbis --enable-opencl --disable-libpulse --enable-libvmaf --disable-libxcb --disable-xlib --enable-amf --enable- libaom --enable-libaribb24 --enable-avisynth --enable-chromaprint --enable-libdav1d --enable-libdavs2 --enable-libdvdread --enable-libdvdnav --disable-libfdk-aac --enable-ffnvcodec --enable-cuda-llvm --enable-frei0r --enable-libgme --enable-libkvazaar --enable-libaribcaption --enable- libass --enable-libbluray --enable-libjxl --enable-libmp3lame --enable- libopus --enable-librist --enable-libssh --enable-libtheora --enable- libvpx --enable-libwebp --enable-libzmq --enable-lv2 --enable-libvpl --enable-openal --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopenmpt --enable- librav1e --enable-librubberband --enable-schannel --enable-sdl2 --enable- libsoxr --enable-libsrt --enable-libsvtav1 --enable-libtwolame --enable- libuavs3d --disable-libdrm --enable-vaapi --enable-libvidstab --enable- vulkan --enable-libshaderc --enable-libplacebo --enable-libvvenc --enable- libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable- libzimg --enable-libzvbi --extra-cflags=-DLIBTWOLAME_STATIC --extra- cxxflags= --extra-libs=-lgomp --extra-ldflags=-pthread --extra-ldexeflags= --cc=x86_64-w64-mingw32-gcc --cxx=x86_64-w64-mingw32-g++ --ar=x86_64-w64-mingw32-gcc-ar --ranlib=x86_64-w64-mingw32-gcc-ranlib --nm=x86_64-w64-mingw32-gcc-nm --extra-version=20240915 libavutil 59. 36.100 / 59. 36.100 libavcodec 61. 13.100 / 61. 13.100 libavformat 61. 5.101 / 61. 5.101 libavdevice 61. 2.101 / 61. 2.101 libavfilter 10. 2.102 / 10. 2.102 libswscale 8. 2.100 / 8. 2.100 libswresample 5. 2.100 / 5. 2.100 libpostproc 58. 2.100 / 58. 2.100 Input #0, truehd, from 'https://trac.ffmpeg.org/raw- attachment/ticket/9511/truehd_regression.mlp': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Audio: truehd, 48000 Hz, stereo, s32 (24 bit) 1.35 M-A: 0.000 fd= 0 aq= 67KB vq= 0KB sq= 0B }}} Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker. -- Ticket URL: <https://trac.ffmpeg.org/ticket/11204> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#11307(avcodec:new): [truehd_core] Application provided invalid, non monotonically increasing dts to muxer in stream 0
by FFmpeg 26 Jun '25

26 Jun '25

#11307: [truehd_core] Application provided invalid, non monotonically increasing dts to muxer in stream 0 -------------------------------------+------------------------------------- Reporter: microchip | Type: defect Status: new | Priority: normal Component: avcodec | Version: git- Keywords: truehd | master truehd_core Atmos | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: How to reproduce: Run it on any video containing TrueHD+Atmos audio {{{ % ffmpeg -i input -vn -c:a copy -bsf:a truehd_core output.thd [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241380 >= 241380 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241385 >= 241385 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241390 >= 241390 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241395 >= 241395 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241400 >= 241400 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241405 >= 241405 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241410 >= 241410 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241415 >= 241415 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241420 >= 241420 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241425 >= 241425 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241430 >= 241430 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241435 >= 241435 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241440 >= 241440 [truehd @ 0x1d768440] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 241445 >= 241445 ffmpeg version N-117812-gbbb0fdedb7 built on 17-novembre-2024 }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/11307> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#11250(ffmpeg:new): VVC MP4 Decoding Issue - Frames Are Missing
by FFmpeg 24 Jun '25

24 Jun '25

#11250: VVC MP4 Decoding Issue - Frames Are Missing -------------------------------------+------------------------------------- Reporter: | Type: defect Rupert_Williamson | Status: new | Priority: important Component: ffmpeg | Version: Keywords: vvc, vvenc, | unspecified livbbenc, mp4 | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: Attempting to extract frames from a vvenc-encoded .mp4 file does not extract all the frames. How to reproduce: {{{ Encode with: % ffmpeg -r 60 -ss 0 -i silver.mov -t 4 -y -c:v libvvenc vvc.mp4 Then decode with: ffmpeg -r 60 -i vvc.mp4 out/out_%04d.png ffmpeg version 2024-10-10-git-0f5592cfc7-full_build-www.gyan.dev Copyright (c) 2000-2024 the FFmpeg developers }}} Rather than getting 240 frames as I would expect (4 seconds of 60.000 fps input), I only get 209 frames out. In this case the input file is a lossless 8bpc RGB 60fps 4K uncompressed .mov file. However I have this issue with other files too, such as 16 bit .nut files or even 16bpc .tif sequences. Gyan thinks it "notionally appears to be a bug in the mp4 muxer" and sees an "editlist setting the start of the video track output from ~half- second." For more details, you can see a post I made here: https://www.reddit.com/r/ffmpeg/comments/1g684mo/vvc_decoding_issue_frames_… -- Ticket URL: <https://trac.ffmpeg.org/ticket/11250> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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