April 2024 - FFmpeg-trac - lists.ffmpeg.org

#10988(undetermined:new): Acrossover crash
by FFmpeg 02 Aug '25

02 Aug '25

#10988: Acrossover crash -------------------------------------+------------------------------------- Reporter: JamesBond | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Setup: Windows 7 x64, FFMPEG g08781ebe1-2024-04-29 x64 Source: song.dsf https://fastupload.io/en/8hp1Dg1K3KNY3Zc/file Command: ffmpeg -i song.dsf -af "volume=6dB,aresample=96000:resampler=soxr:precision=28:osf=s32:dither_method=triangular_hp,acrossover=24000:order=8th[low][high];[low]anull;[high]anullsink" out.flac When the encoding is almost done, the red line appears. It says “Assertion best_input >= 0 failed at fftools/ffmpeg_filter.c:2075”. Then I see a GUI window that says “ffmpeg.exe has stopped working” with a button “Close the program”. An incomplete FLAC file remains on the disk, i.e. flac -t song.flac says “FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC” and “ FLAC__STREAM_DECODER_END_OF_STREAM”. If I remove that acrossover filter, then the encoding goes fine. -- Ticket URL: <https://trac.ffmpeg.org/ticket/10988> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#9195(avfilter:new): vibrance (video filter) incorrect defaults for luma
by FFmpeg 31 Jul '25

31 Jul '25

#9195: vibrance (video filter) incorrect defaults for luma -------------------------------------+------------------------------------- Reporter: Jim | Type: defect DeLaHunt | Status: new | Priority: normal Component: avfilter | Version: git- | master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: The "vibrance" video filter appears to have incorrect default values for the luma coefficient parameters "rlum" and "blum". They appear to be swapped. How to reproduce: {{{ % ffmpeg -h filter=vibrance ffmpeg version 4.4 Copyright (c) 2000-2021 the FFmpeg developers built with Apple LLVM version 10.0.0 (clang-1000.10.44.4) configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable- libopus --enable-librsvg --enable-libtheora --enable-libopenjpeg --enable- libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex --enable- libass --enable-libbluray --enable-lzma --enable-gnutls --enable- fontconfig --enable-libfreetype --enable-libfribidi --disable-libjack --disable-libopencore-amrnb --disable-libopencore-amrwb --disable- indev=jack --enable-opencl --disable-outdev=xv --enable-audiotoolbox --enable-videotoolbox --enable-sdl2 --disable-securetransport --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --enable-libdav1d --arch=x86_64 --enable-x86asm --enable-libx265 --enable-gpl --enable-postproc --enable-libx264 --enable- libxvid libavutil 56. 70.100 / 56. 70.100 libavcodec 58.134.100 / 58.134.100 libavformat 58. 76.100 / 58. 76.100 libavdevice 58. 13.100 / 58. 13.100 libavfilter 7.110.100 / 7.110.100 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 9.100 / 5. 9.100 libswresample 3. 9.100 / 3. 9.100 libpostproc 55. 9.100 / 55. 9.100 Filter vibrance Boost or alter saturation. slice threading supported Inputs: #0: default (video) Outputs: #0: default (video) vibrance AVOptions: intensity <float> ..FV.....T. set the intensity value (from -2 to 2) (default 0) rbal <float> ..FV.....T. set the red balance value (from -10 to 10) (default 1) gbal <float> ..FV.....T. set the green balance value (from -10 to 10) (default 1) bbal <float> ..FV.....T. set the blue balance value (from -10 to 10) (default 1) rlum <float> ..FV.....T. set the red luma coefficient (from 0 to 1) (default 0.072186) glum <float> ..FV.....T. set the green luma coefficient (from 0 to 1) (default 0.715158) blum <float> ..FV.....T. set the blue luma coefficient (from 0 to 1) (default 0.212656) alternate <boolean> ..FV.....T. use alternate colors (default false) This filter has support for timeline through the 'enable' option. }}} The vibrance filter is documented at https://ffmpeg.org/ffmpeg- filters.html#vibrance . It looks like coefficients "rlum", "glum", and "blum" are supposed to match the luma computation as defined in BT.709-6 (https://en.wikipedia.org/wiki/Rec._709) or something similar. Wikipedia gives this formula as: > R’G’B’ coefficients 0.2126, 0.7152, and 0.0722 (together they add to 1) So, `luma = 0.2126 * r + 0.7152 * g + 0.0722 * b`, a value in [0.0, 1.0] for r, g, b in [0.0, 1.0] But the default values I see in the FFmpeg vibrance filter are (per output above): > rlum (default 0.072186), glum (default 0.715158), blum (default 0.212656) Thus it looks like the default values for rlum and blum are swapped. Vibrance would still compute a luma a value in a range of [0.0, 1.0], but it wouldn't be the correct value per BT.709. Since the vibrance adjustments are relative to luma, they might be slightly off. I believe the fix would be the simple, obvious exchange of default values in ffmpeg/libavfilter/vf_vibrance.c:372-374. But I haven't tried this. Present code: {{{ { "rlum", "set the red luma coefficient", OFFSET(lcoeffs[2]), AV_OPT_TYPE_FLOAT, {.dbl=0.072186}, 0, 1, VF }, { "glum", "set the green luma coefficient", OFFSET(lcoeffs[0]), AV_OPT_TYPE_FLOAT, {.dbl=0.715158}, 0, 1, VF }, { "blum", "set the blue luma coefficient", OFFSET(lcoeffs[1]), AV_OPT_TYPE_FLOAT, {.dbl=0.212656}, 0, 1, VF }, }}} Code with possible fix: {{{ { "rlum", "set the red luma coefficient", OFFSET(lcoeffs[2]), AV_OPT_TYPE_FLOAT, {.dbl=0.212656}, 0, 1, VF }, { "glum", "set the green luma coefficient", OFFSET(lcoeffs[0]), AV_OPT_TYPE_FLOAT, {.dbl=0.715158}, 0, 1, VF }, { "blum", "set the blue luma coefficient", OFFSET(lcoeffs[1]), AV_OPT_TYPE_FLOAT, {.dbl=0.072186}, 0, 1, VF }, }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/9195> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#10209(undetermined:new): EBUR128 and LOUDNORM filters compute LRA differently
by FFmpeg 25 Jul '25

25 Jul '25

#10209: EBUR128 and LOUDNORM filters compute LRA differently --------------------------------------+---------------------------------- Reporter: hsn | Type: defect Status: new | Priority: normal Component: undetermined | Version: 5.1.2 Keywords: LRA, ebur128 | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | --------------------------------------+---------------------------------- Summary of the bug: How to reproduce: {{{ ffmpeg -i song192k.mp4 -af ebur128,loudnorm=print_format=summary -f null - }}} Outputs: {{{ [Parsed_ebur128_0 @ 000001f413d21a40] Summary: Integrated loudness: I: -14.0 LUFS Threshold: -24.1 LUFS Loudness range: LRA: 3.6 LU Threshold: -33.9 LUFS LRA low: -15.6 LUFS LRA high: -12.0 LUFS [Parsed_loudnorm_1 @ 000001f413fc4e40] Input Integrated: -14.2 LUFS Input True Peak: +0.1 dBTP Input LRA: 5.9 LU Input Threshold: -24.3 LUFS }}} One LRA is ''5.9'' and second is ''3.6''. Song analysed by https://youlean.co/file-loudness-meter/ returns: {{{ Momentary Max = -9.81 LUFS Short Term Max = -11.50 LUFS Integrated = -14.00 LUFS }}} LRA definition is here: https://tech.ebu.ch/docs/tech/tech3342v2_0.pdf -- Ticket URL: <https://trac.ffmpeg.org/ticket/10209> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#8349(avcodec:new): Dolby AC-4 Support
by FFmpeg 21 Jul '25

21 Jul '25

#8349: Dolby AC-4 Support ----------------------------------+--------------------------------------- Reporter: Nomis101 | Type: enhancement Status: new | Priority: normal Component: avcodec | Version: git-master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+--------------------------------------- It would be nice to support the detection, muxing and demuxing of Dolby AC-4 audio streams (in MP4 and MKV). Information: https://dolbyac4.com/ Specs: https://www.dolby.com/us/en/technologies/ac-4/Next-Generation- Entertainment-Services.pdf Sample files: http://testassets.dashif.org/#testcase/details/586fb3879ae9045678eab64f MediaInfo 19.09 added support for AC-4, so it would be nice to have support in FFmpeg as well. https://mediaarea.net/MediaInfo/ChangeLog -- Ticket URL: <https://trac.ffmpeg.org/ticket/8349> 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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#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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#10959(undetermined:new): ffmpeg 7.0: Assertion best_input >= 0 failed at fftools/ffmpeg_filter.c:1923
by FFmpeg 06 Jul '25

06 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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#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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