March 2015 - FFmpeg-trac - lists.ffmpeg.org

#4178(undetermined:new): Opus audio in MKV container
by FFmpeg 13 Mar '20

13 Mar '20

#4178: Opus audio in MKV container -------------------------------------+------------------------------------- Reporter: agressiv | Type: defect Status: new | Priority: normal Component: | Version: 2.5 undetermined | Blocked By: Keywords: libopus, | Reproduced by developer: 0 mkv, matroska, opus, crackling | Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: Opus Audio in MKV/WEBM/MKA containers is has crackling and lots of clicks/artifacts when played in ffplay, mplayer or using shared libraries (e.g. Kodi) How to reproduce: {{{ Mux Opus Audio into MKV. Play it with ffplay, mplayer or anything leveraging ffmpeg shared libraries. }}} MPC-HC has zero problems playing these files at all. I've taken the first minute of a bad MKA (for sizes' sake) and have attached it. If you extract the Opus, ffplay will play it fine. -- Ticket URL: <https://trac.ffmpeg.org/ticket/4178> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4184(ffmpeg:new): No warning is shown if -vf is specified too often.
by FFmpeg 08 Mar '20

08 Mar '20

#4184: No warning is shown if -vf is specified too often. --------------------------------------+-------------------------------- Reporter: cehoyos | Owner: Type: defect | Status: new Priority: minor | Component: ffmpeg Version: git-master | Keywords: Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 --------------------------------------+-------------------------------- The following works as expected: {{{ $ ffmpeg -i fate-suite/lena.pnm -vf hflip out1.png -vf vflip out2.png }}} The following should show a warning, some users expect that one filter chains' output is used as input for the next filter: {{{ $ ffmpeg -i fate-suite/lena.pnm -vf hflip -vf vflip out.png ffmpeg version N-68401-g792a500 Copyright (c) 2000-2014 the FFmpeg developers built on Dec 13 2014 12:28:49 with gcc 4.7 (SUSE Linux) configuration: --enable-gpl libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 14.100 / 56. 14.100 libavformat 56. 15.104 / 56. 15.104 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 3.101 / 5. 3.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 Input #0, image2, from 'fate-suite/lena.pnm': Duration: 00:00:00.04, start: 0.000000, bitrate: 39333 kb/s Stream #0:0: Video: ppm, rgb24, 256x256, 25 tbr, 25 tbn, 25 tbc Output #0, image2, to 'out.png': Metadata: encoder : Lavf56.15.104 Stream #0:0: Video: png, rgb24, 256x256, q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc Metadata: encoder : Lavc56.14.100 png Stream mapping: Stream #0:0 -> #0:0 (ppm (native) -> png (native)) Press [q] to stop, [?] for help frame= 1 fps=0.0 q=0.0 Lsize=N/A time=00:00:00.04 bitrate=N/A video:177kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/4184> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4392(avformat:new): Reconnect on lost connection for TCP based protocols
by FFmpeg 20 Feb '20

20 Feb '20

#4392: Reconnect on lost connection for TCP based protocols ----------------------------------+--------------------------------------- Reporter: liquider | Type: enhancement Status: new | Priority: normal Component: avformat | Version: unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+--------------------------------------- ffmpeg doesn't reconnect when streaming to a remote server (e.g. rtmp://, tcp://) if the connection is lost. When running e.g. {{{ $ ffmpeg -i ... -f flv rtmp://... }}} if the connection drops for some reason (e.g. `ifdown eth0`, network cable unplugged), ffmpeg doesn't reestablish the session when the connection becomes available again. In fact nothing happens, the process just hangs. `-loglevel debug` doesn't report anything, the `RTMP_SendPacket` just stops. This seems to have been discussed before: http://ffmpeg.org/pipermail/ffmpeg-user/2012-October/010137.html A pair of options like `-reconnect` and `-disconnect_exit` would make sense for TCP-based protocols? -- Ticket URL: <https://trac.ffmpeg.org/ticket/4392> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4335(avformat:new): mpeg fifo assert while remuxing
by FFmpeg 25 Jan '20

25 Jan '20

#4335: mpeg fifo assert while remuxing -------------------------------------+------------------------------------- Reporter: ubitux | Owner: Type: defect | Status: new Priority: important | Component: avformat Version: git- | Keywords: mpeg dvd master | dvdsub assert Blocked By: | Blocking: Reproduced by developer: 0 | Analyzed by developer: 0 -------------------------------------+------------------------------------- Get a long DVD input (might be a bit slow): {{{ % ./ffmpeg -f lavfi -i testsrc=s=pal -t 00:30:00 -target pal-dvd -y test.mpg ffmpeg version N-70272-g3e0ae19 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.2 (GCC) 20150204 (prerelease) configuration: --prefix=/home/ux/local --enable-gpl --enable-nonfree --enable-libx264 --enable-libfaac --enable-libvorbis --enable- libfontconfig --enable-libfreetype --enable-libass --enable-libmp3lame --samples=/home/ux/fate-samples --assert-level=2 libavutil 54. 19.100 / 54. 19.100 libavcodec 56. 26.100 / 56. 26.100 libavformat 56. 23.105 / 56. 23.105 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, lavfi, from 'testsrc=s=pal': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 720x576 [SAR 1:1 DAR 5:4], 25 tbr, 25 tbn, 25 tbc Output #0, dvd, to 'test.mpg': Metadata: encoder : Lavf56.23.105 Stream #0:0: Video: mpeg2video, yuv420p, 720x576 [SAR 1:1 DAR 5:4], q=2-31, 6000 kb/s, 25 fps, 90k tbn, 25 tbc Metadata: encoder : Lavc56.26.100 mpeg2video Stream mapping: Stream #0:0 -> #0:0 (rawvideo (native) -> mpeg2video (native)) Press [q] to stop, [?] for help frame=45000 fps=247 q=2.0 Lsize= 208886kB time=00:29:59.96 bitrate= 950.7kbits/s video:197278kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 5.883934% }}} Remux it with DVD subs: {{{ % ./ffmpeg -i test.mpg -i gt-en.sub -map 0:1 -map 1:0 -c copy -f mpeg -y /dev/null ffmpeg version N-70272-g3e0ae19 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.2 (GCC) 20150204 (prerelease) configuration: --prefix=/home/ux/local --enable-gpl --enable-nonfree --enable-libx264 --enable-libfaac --enable-libvorbis --enable- libfontconfig --enable-libfreetype --enable-libass --enable-libmp3lame --samples=/home/ux/fate-samples --assert-level=2 libavutil 54. 19.100 / 54. 19.100 libavcodec 56. 26.100 / 56. 26.100 libavformat 56. 23.105 / 56. 23.105 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 [NULL @ 0x3202880] start time for stream 0 is not set in estimate_timings_from_pts Input #0, mpeg, from 'test.mpg': Duration: 00:30:00.00, start: 0.540000, bitrate: 950 kb/s Stream #0:0[0x1bf]: Data: dvd_nav_packet Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p(tv), 720x576 [SAR 1:1 DAR 5:4], max. 9000 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Input #1, mpeg, from 'gt-en.sub': Duration: 01:45:07.80, start: 69.600000, bitrate: 5 kb/s Stream #1:0[0x20]: Subtitle: dvd_subtitle Output #0, mpeg, to '/dev/null': Metadata: encoder : Lavf56.23.105 Stream #0:0: Video: mpeg2video, yuv420p, 720x576 [SAR 1:1 DAR 5:4], q=2-31, max. 9000 kb/s, 25 fps, 25 tbr, 90k tbn, 25 tbc Stream #0:1: Subtitle: dvd_subtitle Stream mapping: Stream #0:1 -> #0:0 (copy) Stream #1:0 -> #0:1 (copy) Press [q] to stop, [?] for help Assertion av_fifo_size(stream->fifo) == 0 failed at libavformat/mpegenc.c:1185s/s zsh: abort (core dumped) ./ffmpeg -i test.mpg -i gt-en.sub -map 0:1 -map 1:0 -c copy -f mpeg -y }}} Sub can be found at http://lucy.pkh.me/samples/gt-en.sub -- Ticket URL: <https://trac.ffmpeg.org/ticket/4335> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4270(undetermined:new): tee muxing problem when -vcodec copy
by FFmpeg 21 Jan '20

21 Jan '20

#4270: tee muxing problem when -vcodec copy -------------------------------------+------------------------------------- Reporter: arsen | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: muxing, tee | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: Tacking input stream, then sending without video transcoding to 1 or 2 servers. As result ffmpeg cannot start. In case if using trans-coding it works well How to reproduce: {{{ ffmpeg/ffmpeg -i http://InputIP:8042/InputStream -loglevel debug -preset fast -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec copy -f tee -map 0:v -map 0:a -flags +global_header "[f=flv]rtmp://OutputIP/live/ars|[f=flv]rtmp://204.174.104.72/live/ars" ffmpeg version 2.5.3- http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2015 the FFmpeg developers built on Jan 10 2015 09:24:27 with gcc 4.9.2 (Debian 4.9.2-10) configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable- libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable- libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable- gray --enable-libopenjpeg --enable-libopus --disable-ffserver --enable- libass --enable-gnutls --cc=gcc libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat 56. 15.102 / 56. 15.102 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 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 '-i' ... matched as input file with argument 'http://InputIP:8042/InputStream'. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'. Reading option '-preset' ... matched as AVOption 'preset' with argument 'fast'. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'. Reading option '-strict' ...Routing option strict to both codec and muxer layer matched as AVOption 'strict' with argument 'experimental'. Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100'. Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'. Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '96k'. Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'copy'. Reading option '-f' ... matched as option 'f' (force format) with argument 'tee'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:v'. Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0:a'. Reading option '-flags' ... matched as AVOption 'flags' with argument '+global_header'. Reading option '[f=flv]rtmp://OutputIP/live/ars|[f=flv]rtmp://204.174.104.72/live/ars' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument debug. Successfully parsed a group of options. Parsing a group of options: input file http://InputIP:8042/InputStream. Successfully parsed a group of options. Opening an input file: http://InputIP:8042/InputStream. [http @ 0x40f2460] request: GET /InputStream HTTP/1.1 User-Agent: Lavf/56.15.102 Accept: */* Range: bytes=0- Connection: close Host: InputIP:8042 Icy-MetaData: 1 [http @ 0x40f2460] header='HTTP/1.0 200 OK' [http @ 0x40f2460] http_code=200 [http @ 0x40f2460] header='Content-Type: application/octet-stream' [http @ 0x40f2460] header='Server: Icecast 2.3.2-kh29' [http @ 0x40f2460] header='Cache-Control: no-cache' [http @ 0x40f2460] header='Expires: Mon, 26 Jul 1997 05:00:00 GMT' [http @ 0x40f2460] header='Pragma: no-cache' [http @ 0x40f2460] header='' [mpegts @ 0x40f2f80] Format mpegts probed with size=2048 and score=100 [mpegts @ 0x40f2f80] stream=0 stream_type=f pid=4a prog_reg_desc= [mpegts @ 0x40f2f80] stream=1 stream_type=1b pid=4b prog_reg_desc= [mpegts @ 0x40f2f80] Before avformat_find_stream_info() pos: 0 bytes read:61996 seeks:0 [h264 @ 0x40f9260] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x40f9260] decode_slice_header error [h264 @ 0x40f9260] no frame! [h264 @ 0x40f9260] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x40f9260] decode_slice_header error [h264 @ 0x40f9260] no frame! [h264 @ 0x40f9260] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x40f9260] decode_slice_header error [h264 @ 0x40f9260] no frame! [h264 @ 0x40f9260] no picture [mpegts @ 0x40f2f80] max_analyze_duration 5000000 reached at 5013333 microseconds [mpegts @ 0x40f2f80] After avformat_find_stream_info() pos: 1301336 bytes read:1301524 seeks:0 frames:362 Input #0, mpegts, from 'http://InputIP:8042/InputStream': Duration: N/A, start: 54280.012100, bitrate: 114 kb/s Program 1 Stream #0:0[0x4a](rus), 237, 1/90000: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 114 kb/s Stream #0:1[0x4b], 125, 1/90000: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(left), 704x576 [SAR 16:11 DAR 16:9], 1/50, 25 fps, 25 tbr, 90k tbn, 50 tbc Successfully opened the file. Parsing a group of options: output file [f=flv]rtmp://OutputIP/live/ars|[f=flv]rtmp://204.174.104.72/live/ars. Applying option acodec (force audio codec ('copy' to copy stream)) with argument aac. Applying option ar (set audio sampling rate (in Hz)) with argument 44100. Applying option ac (set number of audio channels) with argument 2. Applying option b:a (video bitrate (please use -b:v)) with argument 96k. Applying option vcodec (force video codec ('copy' to copy stream)) with argument copy. Applying option f (force format) with argument tee. Applying option map (set input stream mapping) with argument 0:v. Applying option map (set input stream mapping) with argument 0:a. Successfully parsed a group of options. Opening an output file: [f=flv]rtmp://OutputIP/live/ars|[f=flv]rtmp://204.174.104.72/live/ars. Successfully opened the file. detected 4 logical cores [graph 0 input from stream 0:0 @ 0x40f91a0] Setting 'time_base' to value '1/48000' [graph 0 input from stream 0:0 @ 0x40f91a0] Setting 'sample_rate' to value '48000' [graph 0 input from stream 0:0 @ 0x40f91a0] Setting 'sample_fmt' to value 'fltp' [graph 0 input from stream 0:0 @ 0x40f91a0] Setting 'channel_layout' to value '0x3' [graph 0 input from stream 0:0 @ 0x40f91a0] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3 [audio format for output stream 0:1 @ 0x40de660] Setting 'sample_fmts' to value 'fltp' [audio format for output stream 0:1 @ 0x40de660] Setting 'sample_rates' to value '44100' [audio format for output stream 0:1 @ 0x40de660] Setting 'channel_layouts' to value '0x3' [audio format for output stream 0:1 @ 0x40de660] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1' [AVFilterGraph @ 0x4102640] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto-inserted resampler 0 @ 0x4118000] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:fltp r:44100Hz [rtmp @ 0x4101ee0] Handshaking... [rtmp @ 0x4101ee0] Type answer 3 [rtmp @ 0x4101ee0] Server version 3.0.1.1 [rtmp @ 0x4101ee0] Proto = rtmp, path = /live/ars, app = live, fname = ars [rtmp @ 0x4101ee0] Server bandwidth = 2500000 [rtmp @ 0x4101ee0] Client bandwidth = 2500000 [rtmp @ 0x4101ee0] New incoming chunk size = 4096 [rtmp @ 0x4101ee0] Releasing stream... [rtmp @ 0x4101ee0] FCPublish stream... [rtmp @ 0x4101ee0] Creating stream... [rtmp @ 0x4101ee0] Sending publish command for 'ars' [flv @ 0x4101380] Tag [27][0][0][0]/0x0000001b incompatible with output codec id '28' ([7][0][0][0]) [tee @ 0x433b180] Slave '[f=flv]rtmp://OutputIP/live/ars': error writing header: Invalid data found when processing input Output #0, tee, to '[f=flv]rtmp://OutputIP/live/ars|[f=flv]rtmp://204.174.104.72/live/ars': Metadata: encoder : Lavf56.15.102 Stream #0:0, 0, 1/25: Video: h264 ([27][0][0][0] / 0x001B), yuv420p(left), 704x576 (0x0) [SAR 16:11 DAR 16:9], 1/25, q=2-31, 25 fps, 25 tbn, 25 tbc Stream #0:1(rus), 0, 1/44100: Audio: aac, 44100 Hz, stereo, fltp, 96 kb/s Metadata: encoder : Lavc56.13.100 aac Stream mapping: Stream #0:1 -> #0:0 (copy) Stream #0:0 -> #0:1 (aac (native) -> aac (native)) Could not write header for output file #0 (incorrect codec parameters ?): Invalid data found when processing input [AVIOContext @ 0x40f27a0] Statistics: 1301524 bytes read, 0 seeks }}} Same stream Sending to 1 server without muxing (works well): {{{ ffmpeg/ffmpeg -i http://InputIP:8042/InputStream -loglevel debug -acodec aac -strict experimental -ar 44100 -ac 2 -b:a 96k -vcodec copy -f flv rtmp://OutputIP:10000/live/ars ffmpeg version 2.5.3- http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2015 the FFmpeg developers built on Jan 10 2015 09:24:27 with gcc 4.9.2 (Debian 4.9.2-10) configuration: --enable-gpl --enable-version3 --disable-shared --disable-debug --enable-runtime-cpudetect --enable-libmp3lame --enable- libx264 --enable-libx265 --enable-libwebp --enable-libspeex --enable- libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable- gray --enable-libopenjpeg --enable-libopus --disable-ffserver --enable- libass --enable-gnutls --cc=gcc libavutil 54. 15.100 / 54. 15.100 libavcodec 56. 13.100 / 56. 13.100 libavformat 56. 15.102 / 56. 15.102 libavdevice 56. 3.100 / 56. 3.100 libavfilter 5. 2.103 / 5. 2.103 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 '-i' ... matched as input file with argument 'http://InputIP:8042/InputStream'. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'. Reading option '-strict' ...Routing option strict to both codec and muxer layer matched as AVOption 'strict' with argument 'experimental'. Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100'. Reading option '-ac' ... matched as option 'ac' (set number of audio channels) with argument '2'. Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '96k'. Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'copy'. Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'. Reading option 'rtmp://OutputIP:10000/live/ars' ... matched as output file. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument debug. Successfully parsed a group of options. Parsing a group of options: input file http://InputIP:8042/InputStream. Successfully parsed a group of options. Opening an input file: http://InputIP:8042/InputStream. [http @ 0x31a0460] request: GET /InputStream HTTP/1.1 User-Agent: Lavf/56.15.102 Accept: */* Range: bytes=0- Connection: close Host: InputIP:8042 Icy-MetaData: 1 [http @ 0x31a0460] header='HTTP/1.0 200 OK' [http @ 0x31a0460] http_code=200 [http @ 0x31a0460] header='Content-Type: application/octet-stream' [http @ 0x31a0460] header='Server: Icecast 2.3.2-kh29' [http @ 0x31a0460] header='Cache-Control: no-cache' [http @ 0x31a0460] header='Expires: Mon, 26 Jul 1997 05:00:00 GMT' [http @ 0x31a0460] header='Pragma: no-cache' [http @ 0x31a0460] header='' [mpegts @ 0x31a0e00] Format mpegts probed with size=2048 and score=100 [mpegts @ 0x31a0e00] stream=0 stream_type=f pid=4a prog_reg_desc= [mpegts @ 0x31a0e00] stream=1 stream_type=1b pid=4b prog_reg_desc= [mpegts @ 0x31a0e00] Before avformat_find_stream_info() pos: 0 bytes read:9872 seeks:0 [h264 @ 0x31a71a0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x31a71a0] decode_slice_header error [h264 @ 0x31a71a0] no frame! [h264 @ 0x31a71a0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x31a71a0] decode_slice_header error [h264 @ 0x31a71a0] no frame! [h264 @ 0x31a71a0] non-existing PPS 0 referenced Last message repeated 1 times [h264 @ 0x31a71a0] decode_slice_header error [h264 @ 0x31a71a0] no frame! [h264 @ 0x31a71a0] no picture [mpegts @ 0x31a0e00] max_analyze_duration 5000000 reached at 5013333 microseconds [mpegts @ 0x31a0e00] After avformat_find_stream_info() pos: 1182896 bytes read:1184272 seeks:0 frames:362 Input #0, mpegts, from 'http://InputIP:8042/InputStream': Duration: N/A, start: 54069.750767, bitrate: 149 kb/s Program 1 Stream #0:0[0x4a](rus), 237, 1/90000: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 149 kb/s Stream #0:1[0x4b], 125, 1/90000: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(left), 704x576 [SAR 16:11 DAR 16:9], 1/50, 25 fps, 25 tbr, 90k tbn, 50 tbc Successfully opened the file. Parsing a group of options: output file rtmp://OutputIP:10000/live/ars. Applying option acodec (force audio codec ('copy' to copy stream)) with argument aac. Applying option ar (set audio sampling rate (in Hz)) with argument 44100. Applying option ac (set number of audio channels) with argument 2. Applying option b:a (video bitrate (please use -b:v)) with argument 96k. Applying option vcodec (force video codec ('copy' to copy stream)) with argument copy. Applying option f (force format) with argument flv. Successfully parsed a group of options. Opening an output file: rtmp://OutputIP:10000/live/ars. [rtmp @ 0x31a7620] Handshaking... [rtmp @ 0x31a7620] Type answer 3 [rtmp @ 0x31a7620] Server version 3.0.1.1 [rtmp @ 0x31a7620] Proto = rtmp, path = /live/ars, app = live, fname = ars [rtmp @ 0x31a7620] Server bandwidth = 2500000 [rtmp @ 0x31a7620] Client bandwidth = 2500000 [rtmp @ 0x31a7620] New incoming chunk size = 4096 [rtmp @ 0x31a7620] Releasing stream... [rtmp @ 0x31a7620] FCPublish stream... [rtmp @ 0x31a7620] Creating stream... [rtmp @ 0x31a7620] Sending publish command for 'ars' Successfully opened the file. detected 4 logical cores [graph 0 input from stream 0:0 @ 0x31a0940] Setting 'time_base' to value '1/48000' [graph 0 input from stream 0:0 @ 0x31a0940] Setting 'sample_rate' to value '48000' [graph 0 input from stream 0:0 @ 0x31a0940] Setting 'sample_fmt' to value 'fltp' [graph 0 input from stream 0:0 @ 0x31a0940] Setting 'channel_layout' to value '0x3' [graph 0 input from stream 0:0 @ 0x31a0940] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3 [audio format for output stream 0:1 @ 0x318ca00] Setting 'sample_fmts' to value 'fltp' [audio format for output stream 0:1 @ 0x318ca00] Setting 'sample_rates' to value '44100' [audio format for output stream 0:1 @ 0x318ca00] Setting 'channel_layouts' to value '0x3' [audio format for output stream 0:1 @ 0x318ca00] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1' [AVFilterGraph @ 0x31a0500] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto-inserted resampler 0 @ 0x3288a60] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:2 chl:stereo fmt:fltp r:44100Hz Output #0, flv, to 'rtmp://OutputIP:10000/live/ars': Metadata: encoder : Lavf56.15.102 Stream #0:0, 0, 1/1000: Video: h264 ([7][0][0][0] / 0x0007), yuv420p(left), 704x576 (0x0) [SAR 16:11 DAR 16:9], 1/90000, q=2-31, 25 fps, 1k tbn, 90k tbc Stream #0:1(rus), 0, 1/1000: Audio: aac ([10][0][0][0] / 0x000A), 44100 Hz, stereo, fltp, 96 kb/s Metadata: encoder : Lavc56.13.100 aac Stream mapping: Stream #0:1 -> #0:0 (copy) Stream #0:0 -> #0:1 (aac (native) -> aac (native)) Press [q] to stop, [?] for help [flv @ 0x31c9a20] Failed to update header with correct duration.ate=1726.5kbits/s [flv @ 0x31c9a20] Failed to update header with correct filesize. frame= 758 fps= 30 q=-1.0 Lsize= 6413kB time=00:00:30.48 bitrate=1723.0kbits/s video:6017kB audio:358kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.581407% Input file #0 (http://InputIP:8042/InputStream) Input stream #0:0 (audio): 1430 packets read (498746 bytes); 1430 frames decoded (1464320 samples); Input stream #0:1 (video): 761 packets read (6174587 bytes); Total: 2191 packets (6673333 bytes) demuxed Output file #0 (rtmp://OutputIP:10000/live/ars): Output stream #0:0 (video): 758 packets muxed (6161396 bytes); Output stream #0:1 (audio): 1313 frames encoded (1344512 samples); 1314 packets muxed (367077 bytes); Total: 2072 packets (6528473 bytes) muxed 1430 frames successfully decoded, 0 decoding errors [AVIOContext @ 0x328b7e0] Statistics: 0 seeks, 2097 writeouts [rtmp @ 0x31a7620] UnPublishing stream... [rtmp @ 0x31a7620] Deleting stream... [AVIOContext @ 0x31a06e0] Statistics: 7114428 bytes read, 0 seeks }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/4270> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3615(ffmpeg:new): -y option affects audio/video skew during screen capture
by FFmpeg 20 Jan '20

20 Jan '20

#3615: -y option affects audio/video skew during screen capture ---------------------------------+------------------------------------- Reporter: grepfor | Type: defect Status: new | Priority: important Component: ffmpeg | Version: 2.2.1 Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ---------------------------------+------------------------------------- Example: On my system, the following script works fine and consistently renders normally synchronized audio and video: {{{ ffmpeg -f alsa -ac 2 -i hw:0 \ -f x11grab -r 30 -s 1600x900 -i :0.0 \ -acodec mp3 \ -vcodec libx264 -preset ultrafast -threads 0 \ -f matroska -y ofile.mkv }}} However, removing the **-y** option -- and assuming that a file named `ofile.mkv` already exists, so that ffmpeg asks whether to overwrite it -- results in skew between audio and video. The observed skew seems to be roughly equal to the time it takes for the user to respond to the //File exists. Overwrite?// question. This suggests that the source of the problem may be that the audio timestamp is being initialized prior to the //Overwrite?// question and the video timestamp initialized after the user response. Just guessing. In any case, omitting **-y** produces gross skew effects, and is 100% reproducible on my system. Logfiles obtained using the **-report** option are attached, one for each case (with and without **-y** option). **NOTE:** This ticket was filed with priority "important" because during my own investigation of the issue, I came across a large number of reports by others also stumped by A/V skew peculiarities when attempting aud+vid screencaps using `ffmpeg`. Many were frustrated at being unable to get any of the (numerous) cargo-cult 'recipe' commandline approaches to work. A few of the recipies do explicitly specify using **-y**, but I didn't see any explanations as to why it was necessary, and it seems (at least to me) extremely counterintuitive that that option should have any such effect. In short, aud+vid screencap seems like a not-uncommon application of `ffmpeg` and fixing this sooner rather than later may help quite a few people avoid tripping over the same issue. Just my 2c, and perhaps exaggerating the need for priority. Obviously re- prioritize as you see fit. NOTES: 0. Demo system was Arch linux, synched to repos about 2 days ago. 1. Effect has been observed using **mp3** and **pcm_s16le** audio codecs. 2. Have not tried it with video codecs other than **libx264**. -- Ticket URL: <https://trac.ffmpeg.org/ticket/3615> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#4213(undetermined:new): sps 32 refrence erro
by FFmpeg 20 Jan '20

20 Jan '20

#4213: sps 32 refrence erro -------------------------------------+------------------------------------- Reporter: bajaj | Type: defect Status: new | Priority: important Component: | Version: 2.4.4 undetermined | Blocked By: Keywords: | Reproduced by developer: 0 Blocking: | Analyzed by developer: 0 | -------------------------------------+------------------------------------- I am getting this msg constantly. How to eliminiate it. ? Last message repeated 12 times [NULL @ 0x277af20] non-existing SPS 32 referenced in buffering period2301.0kbits/s dup=39 drop=0 Last message repeated 11 times [NULL @ 0x277af20] non-existing SPS 32 referenced in buffering period2301.0kbits/s dup=39 drop=0 Last message repeated 11 times -- Ticket URL: <https://trac.ffmpeg.org/ticket/4213> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3493(ffmpeg:new): ffmpeg unable to download when using AES-128 with https protocol
by FFmpeg 20 Jan '20

20 Jan '20

#3493: ffmpeg unable to download when using AES-128 with https protocol ----------------------------------+-------------------------------------- Reporter: adi10289 | Type: defect Status: new | Priority: critical Component: ffmpeg | Version: git-master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+-------------------------------------- Hi Recently I was able to download AES-128 encrypted videos using ffmpeg with normal http protocol but since protocol is changed to secure https I am unable to download any video. Pls find below logs and help. ffmpeg -i "http://hls.spuul.com/va/movies/12610001-c9e0f902/0011/s_renditions.m3u8" -c copy -absf aac_adtstoasc "Ankhon Dekhi 1080p.mkv" -loglevel verbose ffmpeg version N-61722-gfc567ac Copyright (c) 2000-2014 the FFmpeg developers built on Mar 22 2014 22:01:33 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable- librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo- aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 67.100 / 52. 67.100 libavcodec 55. 52.103 / 55. 52.103 libavformat 55. 34.101 / 55. 34.101 libavdevice 55. 11.100 / 55. 11.100 libavfilter 4. 3.100 / 4. 3.100 libswscale 2. 5.102 / 2. 5.102 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 [https @ 02a0e160] HTTP error 404 Not Found Unable to open key file https://secure.spuul.com/keys/va/movies/12610001-c9e0f902/0011/renditions- encryption-00001.key Format mp3 detected only with low score of 1, misdetection possible! [mp3 @ 02a07180] Header missing Last message repeated 1057 times [mp3 @ 046e9a60] decoding for stream 0 failed [mp3 @ 046e9a60] Could not find codec parameters for stream 0 (Audio: mp3, 0 channels, s16p): unspecified frame size Consider increasing the value for the 'analyzeduration' and 'probesize' options http://hls.spuul.com/va/movies/12610001-c9e0f902/0011/s_renditions.m3u8: Operation not permitted -- Ticket URL: <https://trac.ffmpeg.org/ticket/3493> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3707(avcodec:new): Support decoding transparency information in dnxhd streams
by FFmpeg 13 Dec '19

13 Dec '19

#3707: Support decoding transparency information in dnxhd streams -------------------------------------+------------------------------------- Reporter: cehoyos | Owner: Type: | Status: new enhancement | Component: avcodec Priority: wish | Keywords: dnxhd alpha Version: git- | Blocking: master | Analyzed by developer: 0 Blocked By: | Reproduced by developer: 0 | -------------------------------------+------------------------------------- http://thread.gmane.org/gmane.comp.video.ffmpeg.libav.user/13205 A user uploaded a dnxhd sample that contains transparency information (the black background is supposed to be transparent), FFmpeg reports yuv422p10le. {{{ $ ffmpeg -i Untitled_dnhd.mov ffmpeg version N-63852-gf341340 Copyright (c) 2000-2014 the FFmpeg developers built on Jun 9 2014 16:59:00 with gcc 4.7 (SUSE Linux) configuration: --enable-gpl libavutil 52. 89.100 / 52. 89.100 libavcodec 55. 66.100 / 55. 66.100 libavformat 55. 42.101 / 55. 42.101 libavdevice 55. 13.101 / 55. 13.101 libavfilter 4. 7.100 / 4. 7.100 libswscale 2. 6.100 / 2. 6.100 libswresample 0. 19.100 / 0. 19.100 libpostproc 52. 3.100 / 52. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Untitled_dnhd.mov': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2014-05-29 21:36:09 Duration: 00:00:01.03, start: 0.000000, bitrate: 842317 kb/s Stream #0:0(eng): Video: dnxhd (AVdn / 0x6E645641), yuv422p10le, 1920x1080, 841432 kb/s, 29.97 fps, 29.97 tbr, 1000k tbn, 1000k tbc (default) Metadata: creation_time : 2014-05-29 21:36:09 handler_name : Apple Alias Data Handler encoder : DNxHD 709 At least one output file must be specified }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/3707> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#3664(undetermined:new): ffprobe can't read packet DTS/PTS
by FFmpeg 07 Nov '19

07 Nov '19

#3664: ffprobe can't read packet DTS/PTS -------------------------------------+------------------------------------- Reporter: hxuanyu | Type: defect Status: new | Priority: normal Component: | Version: undetermined | unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- ffprobe built from latest git head can't read out all DTS/PTS. attached txt are output by calling ffprobe -show_packets -select_streams v INPUT hxuanyu@ubuntu:~/Downloads/ffmpeg$ ./ffprobe -show_packets -select_streams v v_h264_30_yuv420p_dar30x17_0.mpeg >latest.txt ffprobe version N-63381-g64bf845 Copyright (c) 2007-2014 the FFmpeg developers built on May 23 2014 16:39:38 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --enable-static --disable-shared --enable-small libavutil 52. 86.100 / 52. 86.100 libavcodec 55. 63.100 / 55. 63.100 libavformat 55. 40.100 / 55. 40.100 libavdevice 55. 13.101 / 55. 13.101 libavfilter 4. 5.100 / 4. 5.100 libswscale 2. 6.100 / 2. 6.100 libswresample 0. 19.100 / 0. 19.100 Input #0, mpeg, from 'v_h264_30_yuv420p_dar30x17_0.mpeg': Duration: 00:00:06.01, start: 1.000000, bitrate: 703 kb/s Stream #0:0[0x1e0]: Video: h264, yuv420p, 480x272 [SAR 1:1 DAR 30:17], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc but ffprobe built from ffmpeg 2.0.1 could read out all DTS/PTS -- Ticket URL: <https://trac.ffmpeg.org/ticket/3664> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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