October 2018 - FFmpeg-trac - lists.ffmpeg.org

#7283(ffmpeg:new): Drops on the input when rtsp is relayed
by FFmpeg 04 Oct '18

04 Oct '18

#7283: Drops on the input when rtsp is relayed -------------------------------------+------------------------------------- Reporter: jidckii | Type: defect Status: new | Priority: important Component: ffmpeg | Version: git- Keywords: rtsp rtp | master streaming | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: How to reproduce: {{{ % ffmpeg -i input ... output ffmpeg version built on ... }}} Patches should be submitted to the ffmpeg-devel mailing list and not this bug tracker. Hello! I for a very long time I can not solve the problem of retransmission of live stream. According to my assumptions, this is the result of the input and output work in the same thread. Whatever buffers I set up in the end result, I'm getting early loss of packets when reordering RTP {{{RTP: misssed XX pacets}}} or errors like {{{jitter buffer is full}}} I know about the existence of the parameter async and cache https://ffmpeg.org/ffmpeg-all.html#async , but they can not be added to the rtsp input. I'm using the latest builds from git, assembled in the docker image from this repository: https://hub.docker.com/r/jrottenberg/ffmpeg/ version at the time of this post: {{{ ffmpeg -version ffmpeg version N-90807-g00099ef Copyright (c) 2000-2018 the FFmpeg developers built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.9) 20160609 configuration: --disable-debug --disable-doc --disable-ffplay --enable- shared --enable-avresample --enable-libopencore-amrnb --enable- libopencore-amrwb --enable-gpl --enable-libass --enable-libfreetype --enable-libvidstab --enable-libmp3lame --enable-libopenjpeg --enable- libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable- libx265 --enable-libxvid --enable-libx264 --enable-nonfree --enable- openssl --enable-libfdk_aac --enable-libkvazaar --enable-libaom --extra- libs=-lpthread --enable-postproc --enable-small --enable-version3 --extra- cflags=-I/opt/ffmpeg/include --extra-ldflags=-L/opt/ffmpeg/lib --extra- libs=-ldl --prefix=/opt/ffmpeg }}} Here are the startup parameters: {{{ ffmpeg -v 48 -nostats -analyzeduration 20000000 -fflags igndts -fflags genpts -fflags latm -max_delay 500000 -reorder_queue_size 10000 -rtsp_transport udp -r 15 -i rtsp: admin: @ 192.168.86.169: 554/0? .sdp -map 0 -r 15 -c: v copy -an -f mpegts -muxdelay 1 -max_interleave_delta 0 -fflags + genpts udp: 239.0.0.1: 1234? ttl = 1? pkt_size = 1316 }}} During the experiments, I changed the values of the parameters, responsible for buffering, but whatever I put, the packets are still lost and the quality of the video stream spoils. -- Ticket URL: <https://trac.ffmpeg.org/ticket/7283> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#7470(ffprobe:new): ffprobe: Program info output for a selected program
by FFmpeg 04 Oct '18

04 Oct '18

#7470: ffprobe: Program info output for a selected program ---------------------------------+--------------------------------------- Reporter: user243 | Type: defect Status: new | Priority: normal Component: ffprobe | Version: unspecified Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ---------------------------------+--------------------------------------- I am trying to output the info of a particular program from the input which is MPTS but {{{ffprobe}}} outputs the info of all the programs. Below is the code I am using to output {{{service_provider}}} info only for program 1 but the output displays {{{service_provider}}} info for all the programs. {{{ ffprobe -loglevel error -show_programs -select_streams p:1 -show_entries program_tags=service_provider input.ts }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/7470> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#7459(avformat:new): Source Specific Multicast not working on OSX
by FFmpeg 03 Oct '18

03 Oct '18

#7459: Source Specific Multicast not working on OSX -------------------------------------+------------------------------------- Reporter: jlsantiago0 | Type: defect Status: new | Priority: normal Component: avformat | Version: git- Keywords: ssm, source | master specific multicast | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | -------------------------------------+------------------------------------- Summary of the bug: Source specific multicast input works on Linux and Windows, but not on OSX. I have two hosts streaming an MPEG2 transport stream to the same multicast IP address and port. In my test setup the IP address of the sources are 10.66.133.33 and 10.66.133.34 they are both streaming to 239.36.34.100:4900. I can tune into the desired stream by selecting the sources as follows: {{{ ffplay 'udp://@239.36.34.100:4900?sources=10.66.133.34' }}} This works on Linux and Windows but does not work on OSX with the following error: {{{ setsockopt(MCAST_JOIN_SOURCE_GROUP): Can't assign requested address udp://@239.36.34.100:4900?sources=10.66.133.34: Input/output error }}} NOTE: SSM does work correctly on VLC on OSX. The VLC syntax is slightly different: {{{ /Applications/VLC.app/Contents/MacOS/VLC 'udp://10.66.133.34@239.36.34.100:4900' }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/7459> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#7460(undetermined:new): ffmpeg output filename
by FFmpeg 02 Oct '18

02 Oct '18

#7460: ffmpeg output filename -------------------------------------+------------------------------------- Reporter: user243 | Type: Status: new | enhancement Component: | Priority: normal undetermined | Version: Keywords: | unspecified Blocking: | Blocked By: Analyzed by developer: 0 | Reproduced by developer: 0 -------------------------------------+------------------------------------- Is there a way to append the metadata info from the input to the output filename? -- Ticket URL: <https://trac.ffmpeg.org/ticket/7460> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#7465(avformat:new): MP3 24-bit (fmt:s32p) is displayed in the log, but the transcoded file is not
by FFmpeg 01 Oct '18

01 Oct '18

#7465: MP3 24-bit (fmt:s32p) is displayed in the log, but the transcoded file is not -------------------------------------+------------------------------------- Reporter: Sami32 | Type: Status: new | enhancement Component: avformat | Priority: minor Keywords: MP3 24-bit | Version: git- Blocking: | master Analyzed by developer: 0 | Blocked By: | Reproduced by developer: 0 -------------------------------------+------------------------------------- Summary of the bug: Transcoding FLAC 24-bit stream to MP3 display a weird logging message for the output: "Stream #0:0, 0, 1/48000: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s32p (24 bit), 320 kb/s" But the output file is not, as expected by the standard AFAIK. How to reproduce: {{{ ffmpeg started on 2018-10-01 at 14:53:26 Report written to "ffmpeg-20181001-145326.log" Command line: ffmpeg.exe -report -y -i "F:\\_DEVELOPPEMENT_TOOLS\\!MEDIA_TEST_FILES\\AUDIO\\FLAC\\DOUBLE\\surround88 -24b-6ch.flac" -vn -dn -f mp3 -b:a 320000 NUL ffmpeg version N-91977-ge33b28cc79 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 8.2.0 (GCC) configuration: --enable-gpl --enable-version3 --enable-avisynth --enable-fontconfig --enable-gnutls --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libilbc --enable-libkvazaar --enable-libmfx --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable- librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable- libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libxml2 --enable-libzimg --enable-opengl --enable-pthreads --disable-w32threads --disable-dxva2 --disable-debug --pkg-config-flags=--static --extra- ldflags=-static --extra-cflags='-mtune=generic -mfpmath=sse -msse' --cpu=i686 --optflags='-O2 -finline-fun libavutil 56. 19.101 / 56. 19.101 libavcodec 58. 30.100 / 58. 30.100 libavformat 58. 18.101 / 58. 18.101 libavdevice 58. 4.103 / 58. 4.103 libavfilter 7. 32.100 / 7. 32.100 libswscale 5. 2.100 / 5. 2.100 libswresample 3. 2.100 / 3. 2.100 libpostproc 55. 2.100 / 55. 2.100 Splitting the commandline. Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'. Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'. Reading option '-i' ... matched as input url with argument 'F:\_DEVELOPPEMENT_TOOLS\!MEDIA_TEST_FILES\AUDIO\FLAC\DOUBLE\surround88 -24b-6ch.flac'. Reading option '-vn' ... matched as option 'vn' (disable video) with argument '1'. Reading option '-dn' ... matched as option 'dn' (disable data) with argument '1'. Reading option '-f' ... matched as option 'f' (force format) with argument 'mp3'. Reading option '-b:a' ... matched as option 'b' (video bitrate (please use -b:v)) with argument '320000'. Reading option 'NUL' ... matched as output url. Finished splitting the commandline. Parsing a group of options: global . Applying option report (generate a report) with argument 1. Applying option y (overwrite output files) with argument 1. Successfully parsed a group of options. Parsing a group of options: input url F:\_DEVELOPPEMENT_TOOLS\!MEDIA_TEST_FILES\AUDIO\FLAC\DOUBLE\surround88 -24b-6ch.flac. Successfully parsed a group of options. Opening an input file: F:\_DEVELOPPEMENT_TOOLS\!MEDIA_TEST_FILES\AUDIO\FLAC\DOUBLE\surround88 -24b-6ch.flac. [NULL @ 08021580] Opening 'F:\_DEVELOPPEMENT_TOOLS\!MEDIA_TEST_FILES\AUDIO\FLAC\DOUBLE\surround88 -24b-6ch.flac' for reading [file @ 080221c0] Setting default whitelist 'file,crypto' [flac @ 08021580] Format flac probed with size=2048 and score=100 [flac @ 08021580] Before avformat_find_stream_info() pos: 8438 bytes read:32768 seeks:0 nb_streams:1 [flac @ 08021580] All info found [flac @ 08021580] After avformat_find_stream_info() pos: 10486 bytes read:32768 seeks:0 frames:1 Input #0, flac, from 'F:\_DEVELOPPEMENT_TOOLS\!MEDIA_TEST_FILES\AUDIO\FLAC\DOUBLE\surround88 -24b-6ch.flac': Duration: 00:00:56.34, start: 0.000000, bitrate: 999 kb/s Stream #0:0, 1, 1/88200: Audio: flac, 88200 Hz, 5.1, s32 (24 bit) Successfully opened the file. Parsing a group of options: output url NUL. Applying option vn (disable video) with argument 1. Applying option dn (disable data) with argument 1. Applying option f (force format) with argument mp3. Applying option b:a (video bitrate (please use -b:v)) with argument 320000. Successfully parsed a group of options. Opening an output file: NUL. [file @ 08068a80] Setting default whitelist 'file,crypto' Successfully opened the file. detected 1 logical cores Stream mapping: Stream #0:0 -> #0:0 (flac (native) -> mp3 (libmp3lame)) Press [q] to stop, [?] for help cur_dts is invalid (this is harmless if it occurs once at the start per stream) [graph_0_in_0_0 @ 08067280] Setting 'time_base' to value '1/88200' [graph_0_in_0_0 @ 08067280] Setting 'sample_rate' to value '88200' [graph_0_in_0_0 @ 08067280] Setting 'sample_fmt' to value 's32' [graph_0_in_0_0 @ 08067280] Setting 'channel_layout' to value '0x3f' [graph_0_in_0_0 @ 08067280] tb:1/88200 samplefmt:s32 samplerate:88200 chlayout:0x3f [format_out_0_0 @ 08067f80] Setting 'sample_fmts' to value 's32p|fltp|s16p' [format_out_0_0 @ 08067f80] Setting 'sample_rates' to value '44100|48000|32000|22050|24000|16000|11025|12000|8000' [format_out_0_0 @ 08067f80] Setting 'channel_layouts' to value '0x4|0x3' [format_out_0_0 @ 08067f80] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0' [AVFilterGraph @ 08067080] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto_resampler_0 @ 080c4b80] picking s32p out of 3 ref:s32 [auto_resampler_0 @ 080c4b80] [SWR @ 080c5000] Using fltp internally between filters [auto_resampler_0 @ 080c4b80] [SWR @ 080c5000] Matrix coefficients: [auto_resampler_0 @ 080c4b80] [SWR @ 080c5000] FL: FL:0.414214 FR:0.000000 FC:0.292893 LFE:0.000000 BL:0.292893 BR:0.000000 [auto_resampler_0 @ 080c4b80] [SWR @ 080c5000] FR: FL:0.000000 FR:0.414214 FC:0.292893 LFE:0.000000 BL:0.000000 BR:0.292893 [auto_resampler_0 @ 080c4b80] ch:6 chl:5.1 fmt:s32 r:88200Hz -> ch:2 chl:stereo fmt:s32p r:48000Hz Output #0, mp3, to 'NUL': Metadata: TSSE : Lavf58.18.101 Stream #0:0, 0, 1/48000: Audio: mp3 (libmp3lame), 48000 Hz, stereo, s32p (24 bit), 320 kb/s Metadata: encoder : Lavc58.30.100 libmp3lame cur_dts is invalid (this is harmless if it occurs once at the start per stream) size= 91kB time=2.28 bitrate= 326.8kbits/s speed=4.53x size= 182kB time=4.61 bitrate= 323.3kbits/s speed=4.57x size= 256kB time=6.79 bitrate= 308.7kbits/s speed=4.49x size= 256kB time=8.98 bitrate= 233.6kbits/s speed=4.44x size= 256kB time=11.33 bitrate= 185.1kbits/s speed=4.49x size= 512kB time=13.61 bitrate= 308.2kbits/s speed= 4.5x size= 512kB time=15.98 bitrate= 262.4kbits/s speed=4.53x size= 512kB time=18.17 bitrate= 230.8kbits/s speed=4.51x size= 768kB time=20.59 bitrate= 305.5kbits/s speed=4.54x size= 768kB time=22.44 bitrate= 280.4kbits/s speed=4.45x size= 768kB time=23.78 bitrate= 264.5kbits/s speed=4.28x size= 768kB time=25.42 bitrate= 247.5kbits/s speed=4.19x size= 1024kB time=26.81 bitrate= 312.9kbits/s speed=4.08x size= 1024kB time=28.49 bitrate= 294.5kbits/s speed=4.01x size= 1024kB time=30.62 bitrate= 273.9kbits/s speed=4.03x size= 1024kB time=32.02 bitrate= 262.0kbits/s speed=3.95x size= 1280kB time=33.86 bitrate= 309.6kbits/s speed=3.93x size= 1280kB time=36.19 bitrate= 289.7kbits/s speed=3.97x size= 1280kB time=38.59 bitrate= 271.7kbits/s speed=4.01x size= 1536kB time=40.73 bitrate= 308.9kbits/s speed=4.02x [NULL @ 08033000] dropping low score -59 frame header from offset 35 to 6415 size= 1536kB time=42.91 bitrate= 293.2kbits/s speed=4.04x size= 1536kB time=44.18 bitrate= 284.8kbits/s speed=3.97x size= 1792kB time=46.13 bitrate= 318.2kbits/s speed=3.96x size= 1792kB time=48.48 bitrate= 302.8kbits/s speed=3.99x size= 1792kB time=50.50 bitrate= 290.7kbits/s speed=3.99x size= 2048kB time=52.44 bitrate= 319.9kbits/s speed=3.99x size= 2048kB time=54.53 bitrate= 307.7kbits/s speed=3.99x [out_0_0 @ 08067580] EOF on sink link out_0_0:default. No more output streams to write to, finishing. [libmp3lame @ 08024980] Trying to remove 436 more samples than there are in the queue size= 2203kB time=56.35 bitrate= 320.3kbits/s speed= 4x video:0kB audio:2202kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.044567% Input file #0 (F:\_DEVELOPPEMENT_TOOLS\!MEDIA_TEST_FILES\AUDIO\FLAC\DOUBLE\surround88 -24b-6ch.flac): Input stream #0:0 (audio): 1214 packets read (7032652 bytes); 1214 frames decoded (4969531 samples); Total: 1214 packets (7032652 bytes) demuxed Output file #0 (NUL): Output stream #0:0 (audio): 2348 frames encoded (2704507 samples); 2349 packets muxed (2255040 bytes); Total: 2349 packets (2255040 bytes) muxed 1214 frames successfully decoded, 0 decoding errors [AVIOContext @ 0804e740] Statistics: 1 seeks, 10 writeouts [AVIOContext @ 0802a500] Statistics: 7041090 bytes read, 0 seeks }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/7465> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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#7463(avformat:new): Cover art in wav file
by FFmpeg 01 Oct '18

01 Oct '18

#7463: Cover art in wav file ----------------------------------+--------------------------------------- Reporter: cgbug | Type: enhancement Status: new | Priority: normal Component: avformat | Version: git-master Keywords: | Blocked By: Blocking: | Reproduced by developer: 0 Analyzed by developer: 0 | ----------------------------------+--------------------------------------- VLC is able to display the embedded covert art. FFmpeg does not seem to find it. Sample: https://www.sendspace.com/file/5ojfix {{{ ffmpeg version N-92079-gb577153231 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 8.2.1 (GCC) 20180813 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable- fontconfig --enable-gnutls --enable-iconv --enable-libass --enable- libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore- amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable- libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable- ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth libavutil 56. 19.101 / 56. 19.101 libavcodec 58. 31.102 / 58. 31.102 libavformat 58. 18.103 / 58. 18.103 libavdevice 58. 4.104 / 58. 4.104 libavfilter 7. 33.100 / 7. 33.100 libswscale 5. 2.100 / 5. 2.100 libswresample 3. 2.100 / 3. 2.100 libpostproc 55. 2.100 / 55. 2.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, wav, from 'Dennis Ferrer - Hey Hey (David Pietras Remode).wav': Metadata: artist : Hey Hey (David Pietras Remode) album : Dennis Ferrer Duration: 00:04:54.19, bitrate: 1412 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s }}} -- Ticket URL: <https://trac.ffmpeg.org/ticket/7463> FFmpeg <https://ffmpeg.org> FFmpeg issue tracker

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