November 2017 - ffmpeg-user - lists.ffmpeg.org

Recording audio from mic and speakers from commandline
by Mohan G 28 Sep '21

28 Sep '21

Hi, I am using ffmpeg version 2.4.7 on Fedora-21 workstation. I want to record audio, simultaneously from mic and speakers as separate streams. I am using the following command for it. ffmpeg -f alsa -i pulse -f alsa -i default -acodec libmp3lame -map 0:0 -map 1:0 outfile.mkv But after executing this command in terminal, I have to goto "Volume Control" settings in GUI, then goto "Recording" tab, change the "capture from" of one channel to "Built-in Audio Analog Stereo" (i.e., Mic) and the "capture from" of another channel to "Monitor of Built-in Audio Analog Stereo" (i.e., Speakers). But how can this be done completely from command-line, for automating it to use in scripts. - Mohan

3 10

How to have a secure UDP BroadCast
by hassan shatnawi 19 Aug '20

19 Aug '20

Dears, How to have a secure connection with username and password while broadcasting "UPD" I've use this command ffmpeg -f dshow -video_size 1280x720 -rtbufsize 702000K -framerate 30 -i video="Video (00-0 Pro Capture Quad HDMI)" -r 30 -threads 4 -vcodec libx264 -crf 0 -preset ultrafast -f mpegts "udp://username:password@192.168.1.12:6666" but it didn't work can you advice me plz. Thanks in advance. *Hassan Shatnawi* Software Developer *Software Development Department* *Haupshy Establishment for Electronics* Al Madina Al Monawara St. bld 31 | P.O.Box 6875 Amman 11118, Jordan. *T: *+962 6 5820 927* | F: *+962 6 5820 926* | M: *+962 78 5792 096 *E*: hassan(a)haupshy.com *|* www.haupshy.com

7 7

problem transcoding raw g729 to wav format
by Dave Horton 20 Aug '18

20 Aug '18

I am trying to decode a raw g729 input stream to wav format like so: ffmpeg -loglevel debug -threads 1 -y -acodec g729 -f g729 -i pipe:0 -ar 44100 -acodec pcm_s32le -f wav pipe:1 As you can see, the raw g729 is piped to stdin and the wav output should be piped to stdout. I am seeing this output / errors in the log, and I get an empty wav file: ffmpeg version N-87327-g18821e3ba1-static http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2017 the FFmpeg developers built with gcc 6.4.0 (Debian 6.4.0-5) 20170906 configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gray --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-librtmp --enable-libsoxr --enable-libspeex --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzimg libavutil 55. 75.100 / 55. 75.100 libavcodec 57.106.101 / 57.106.101 libavformat 57. 82.100 / 57. 82.100 libavdevice 57. 8.101 / 57. 8.101 libavfilter 6.105.100 / 6.105.100 libswscale 4. 7.103 / 4. 7.103 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 Splitting the commandline. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug’. Reading option '-threads' ... matched as AVOption 'threads' with argument '1’. option '-y' ... matched as option 'y' (overwrite output files) with argument '1’. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'g729’. Reading option '-f' ... matched as option 'f' (force format) with argument 'g729’. Reading option '-i' ... matched as input url with argument 'pipe:0’. Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '44100’. Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'pcm_s32le’. Reading option '-f' ... matched as option 'f' (force format) with argument 'wav’. Reading option 'pipe:1' ... matched as output url. Finished splitting the commandline. Parsing a group of options: global . Applying option loglevel (set logging level) with argument debug. Applying option y (overwrite output files) with argument 1. Successfully parsed a group of options. Parsing a group of options: input url pipe:0. Applying option acodec (force audio codec ('copy' to copy stream)) with argument g729. Applying option f (force format) with argument g729. Successfully parsed a group of options. Opening an input file: pipe:0. [g729 @ 0x3e25da0] Opening 'pipe:0' for reading [pipe @ 0x3e265a0] Setting default whitelist 'crypto' [g729 @ 0x3e25da0] Before avformat_find_stream_info() pos: 0 bytes read:0 seeks:0 nb_streams:1 [g729 @ 0x3e25da0] After avformat_find_stream_info() pos: 0 bytes read:0 seeks:0 frames:0 Guessed Channel Layout for Input Stream #0.0 : mono Input #0, g729, from 'pipe:0': Duration: N/A, bitrate: 8 kb/s Stream #0:0, 0, 1/8000: Audio: g729, 8000 Hz, mono, s16 Successfully opened the file. Parsing a group of options: output url pipe:1. Applying option ar (set audio sampling rate (in Hz)) with argument 44100. Applying option acodec (force audio codec ('copy' to copy stream)) with argument pcm_s32le. Applying option f (force format) with argument wav. Successfully parsed a group of options. Opening an output file: pipe:1. [pipe @ 0x3e2aba0] Setting default whitelist ‘crypto' Successfully opened the file. Stream mapping: Stream #0:0 -> #0:0 (g729 (native) -> pcm_s32le (native)) cur_dts is invalid (this is harmless if it occurs once at the start per stream) No more output streams to write to, finishing. Finishing stream 0:0 without any data written to it. detected 8 logical cores [graph_0_in_0_0 @ 0x3e2daa0] Setting 'time_base' to value '1/8000’ [graph_0_in_0_0 @ 0x3e2daa0] Setting 'sample_rate' to value '8000’ graph_0_in_0_0 @ 0x3e2daa0] Setting 'sample_fmt' to value 's16’ [graph_0_in_0_0 @ 0x3e2daa0] Setting 'channel_layout' to value '0x4’ [graph_0_in_0_0 @ 0x3e2daa0] tb:1/8000 samplefmt:s16 samplerate:8000 chlayout:0x4 format_out_0_0 @ 0x3e25ca0] Setting 'sample_fmts' to value 's32’ [format_out_0_0 @ 0x3e25ca0] Setting 'sample_rates' to value '44100’ [format_out_0_0 @ 0x3e25ca0] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0' [AVFilterGraph @ 0x3e2c640] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed [auto_resampler_0 @ 0x3e3fd20] [SWR @ 0x3e400c0] Using fltp internally between filters [auto_resampler_0 @ 0x3e3fd20] ch:1 chl:mono fmt:s16 r:8000Hz -> ch:1 chl:mono fmt:s32 r:44100Hz Output #0, wav, to 'pipe:1’: Metadata: ISFT : Lavf57.82.100 Stream #0:0, 0, 1/44100: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s32, 1411 kb/s Metadata: encoder : Lavc57.106.101 pcm_s32le size= 0kB time=00:00:00.00 bitrate=N/A speed= 0x video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown Input file #0 (pipe:0): Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples); Total: 0 packets (0 bytes) demuxed Output file #0 (pipe:1): Output stream #0:0 (audio): 0 frames encoded (0 samples); 0 packets muxed (0 bytes); Total: 0 packets (0 bytes) muxed Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used) 0 frames successfully decoded, 0 decoding errors Is there something wrong with my command? Interestingly, if I transcode to flac instead of wav, while I get similar errors, an accurate output file _is_ created in that case. Dave

3 13

FFserver doesn't work with HLS
by SD76 11 Jul '18

11 Jul '18

Hi, I try since a few days to stream a live with ffserver and HLS. My ffserver.conf run with ffserver -f ffserver.conf -d --------------- HTTPPort 8081 MaxClients 10 MaxBandwidth 5000000 <Feed plop.ffm> File /tmp/plop.ffm FileMaxSize 100M </Feed> <Stream s> Format status </Stream> <Stream plop.m3u8> Feed plop.ffm Format HLS VideoCodec libx264 VideoFrameRate 30 VideoBitRate 128 VideoSize 704*480 #AVOptionVideo hls_time 10 #AVOptionVideo hls_list_size 0 AVOptionVideo flags +global_header NoAudio </Stream> --------------- If i enable AVOptionVideo hls_time 10 and AVOptionVideo hls_list_size 0 ffserver say : Option not found: 'hls_time' Option not found: 'hls_list_size' I launch for send stream this command : ffmpeg -r 25 -s 640x480 -f video4linux2 -i /dev/video0 -hls_time 10 -hls_list_size 6 -hls_wrap 10 -start_number 1 http://192.168.0.16:8081/plop.ffm I use a remote player on my galaxy tab: <video controls width="480" height="270" src=" http://192.168.0.16:8081/plop.m3u8"> The stream doesn't work and ffserver say: Sun Nov 29 18:15:15 2015 [hls @ 0x1a20810]failed to rename file .tmp to Do HLS works with ffserver ? How ? I use ffserver version 2.8.1-1~bpo8+1 and ffmpeg version 2.8.1-1~bpo8+ on debian 8. Regard, Ben

2 2

Low latency H.264 streaming with ffmpeg
by Greger Burman 27 Mar '18

27 Mar '18

I would like to hear from anyone who has experience with low latency video streaming and ffmpeg. In my application I take video from a PCIe capture device and stream live over network. The end to end latency must be below 200ms otherwise it is not usable. Let us assume that the network latency is near zero. This question is about encoder+decoder+buffering latency. Very low latency can be achieved when using the x264 encoder by using settings like tune zerolatency. From my own testing with x264 (without ffmpeg) I can confirm that this works. I have measured approx 90 ms (3 frames) of latency added by encoder+decoder combined. Awesome. Sadly I have been unable to get anywhere near those numbers when using ffmpeg and libx264. As player I use either ffplay or a dot-net player built on the ffmpeg libraries, with about equal results. The lowest end to end latency I have seen is 400-500ms. That includes capture latency also. Yes, there are many unknowns here. To help pin point the problem I used mjpeg streaming for comparison. Same everything except I change ffmpeg vcodec to mjpeg and format to mjpeg. I change nothing with ffplay. With mjpeg I now get approx 150ms end to end latency. Observation: x264 codec + mpegts format compared to mjpeg codec + mjpeg format adds about 300 ms (10 frames) latency. I have tried every ffplay-, ffmpeg- and libx264 setting that I believe relevant. Most have not made any noticable difference in latency. I will not list a lot of settings here. Here are the basic settings and commands that I have used. I can post more details on request. x264-preset: vcodec=libx264 thread_type=slice slices=1 # x264 profile=baseline level=32 preset=superfast tune=zerolatency intra-refresh=1 crf=15 x264-params=vbv-maxrate=5000:vbv-bufsize=1:slice-max-size=1500:keyint=60 $ ffmpeg -r 30 -f dshow -i video="devicename" -pix_fmt yuv420p -an -vpre x264-preset -f mpegts udp://127.0.0.1:8888 $ ffplay -fflags nobuffer -flags low_delay -framedrop -strict experimental udp://127.0.0.1:8888 Everything is run on Windows 7 64-bit, with 32-bit executables. ffmpeg version N-76539-g480bad7 Copyright (c) 2000-2015 the FFmpeg developers built with gcc 5.2.0 (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-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --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-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib libavutil 55. 5.100 / 55. 5.100 libavcodec 57. 15.100 / 57. 15.100 libavformat 57. 14.100 / 57. 14.100 libavdevice 57. 0.100 / 57. 0.100 libavfilter 6. 15.100 / 6. 15.100 libswscale 4. 0.100 / 4. 0.100 libswresample 2. 0.101 / 2. 0.101 libpostproc 54. 0.100 / 54. 0.100 cheers -- Greger Burman

4 6

hls_init_time problem
by ? ? 14 Mar '18

14 Mar '18

Hi, I’m record a stream from ip camera with H264 video encoding & PCMA audio encoding to m3u8 Ffmpeg version:3.2.4 OS:OSX Here is my command: ffmpeg -rtsp_transport tcp -i rtsp://192.168.0.40/1/h264major -vcodec copy -acodec aac -hls_init_time 2 -hls_time 60 -hls_list_size 0 -hls_flags +single_file+round_durations -hls_segment_filename test.ts -t 3600 test.m3u8 But in the m3u8 file,every segment entry’s duration is 2 seconds Which part did I do wrong? Thx in advance!

5 6

Failed to resolve hostname: Ubuntu static builds
by hdn8 03 Mar '18

03 Mar '18

Using the binaries from http://johnvansickle.com/ I get an error when using a http input, "failed to resolve hostname": (using ffprobe, but also with ffmpeg): ffprobe version N-41351-g676110e- http://johnvansickle.com/ffmpeg/ Copyright (c) 2007-2014 the FFmpeg developers built on Oct 15 2014 01:22:19 with gcc 4.8 (Debian 4.8.3-11) 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-4.8 libavutil 54. 10.100 / 54. 10.100 libavcodec 56. 5.100 / 56. 5.100 libavformat 56. 9.100 / 56. 9.100 libavdevice 56. 1.100 / 56. 1.100 libavfilter 5. 1.106 / 5. 1.106 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc 53. 3.100 / 53. 3.100 [tcp @ 0x3f82100] Failed to resolve hostname mirrorblender.top-ix.org: Name or service not known http://mirrorblender.top-ix.org/peach/bigbuckbunny_movies/big_buck_bunny_48…: Input/output error Note I have tried with various http sources/hosts, all same result. Using the builds from ffmpeg.gusari.org it works fine: ffprobe version N-63893-gc69defd Copyright (c) 2007-2014 the FFmpeg developers built on Jul 16 2014 05:38:01 with gcc 4.6 (Debian 4.6.3-1) configuration: --prefix=/root/ffmpeg-static/64bit --extra-cflags='-I/root/ffmpeg-static/64bit/include -static' --extra-ldflags='-L/root/ffmpeg-static/64bit/lib -static' --extra-libs='-lxml2 -lexpat -lfreetype' --enable-static --disable-shared --disable-ffserver --disable-doc --enable-bzlib --enable-zlib --enable-postproc --enable-runtime-cpudetect --enable-libx264 --enable-gpl --enable-libtheora --enable-libvorbis --enable-libmp3lame --enable-gray --enable-libass --enable-libfreetype --enable-libopenjpeg --enable-libspeex --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-version3 --enable-libvpx libavutil 52. 89.100 / 52. 89.100 libavcodec 55. 66.101 / 55. 66.101 libavformat 55. 43.100 / 55. 43.100 libavdevice 55. 13.101 / 55. 13.101 libavfilter 4. 8.100 / 4. 8.100 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 'http://mirrorblender.top-ix.org/peach/bigbuckbunny_movies/big_buck_bunny_48…': Metadata: major_brand : qt minor_version : 537199360 compatible_brands: qt creation_time : 2008-05-27 18:32:32 timecode : 00:00:00:00 Duration: 00:09:56.46, start: 0.000000, bitrate: 3342 kb/s Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, bt709), 854x480, 2899 kb/s, 24 fps, 24 tbr, 2400 tbn, 4800 tbc (default) Metadata: creation_time : 2008-05-27 18:32:32 handler_name : Apple Alias Data Handler encoder : H.264 Stream #0:1(eng): Data: none (tmcd / 0x64636D74) (default) Metadata: creation_time : 2008-05-27 18:32:32 handler_name : Apple Alias Data Handler timecode : 00:00:00:00 Stream #0:2(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 437 kb/s (default) Metadata: creation_time : 2008-05-27 18:32:32 handler_name : Apple Alias Data Handler The error from http://johnvansickle.com/ffmpeg/ is the same for the nightly and version 2.4. The gusari.org binaries havent been built since July 16, for now I am using ffprobe from gusari for http sources and for file sources johnvansickle. From where are you getting your (supposedly stable) binaries? Ubuntu x86_64 GNU/Linux -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Failed-to-resolve-hostname-Ubuntu-… Sent from the FFmpeg-users mailing list archive at Nabble.com.

3 3

Avid opAtom trouble
by Bouke (VideoToolShed) 29 Jan '18

29 Jan '18

Hi guys, Toying around with Avid opAtom mxf, generating test clips. When I look at the clip in Avid, the bin info shows a raster dimension of 1920x1088, pixel aspect ratio of 1.007 (expected 1920x1080 / 1.000) And of course, when editing the clip into a seqence, frameflex kicks in. same line but output to Mov and AMA the clip in does give the expected result. Am I missing something or is this a bug? Next, what kind of metadata besides the clip name / timecode is supported? thx, Bouke ffmpeg -f lavfi -i "testsrc=size=1920x1080:rate=25" -frames:v 10 -f mxf_opatom -vcodec dnxhd -pix_fmt yuv422p -vb 120M -timecode 01:02:03:04 -metadata material_package_name="rastertest" -flags +ildct -top 1 -y "M:\Avid Mediafiles\MXF\2\test.mxf" ffmpeg version N-85653-gb4330a0 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 6.3.0 (GCC) configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --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-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib libavutil 55. 61.100 / 55. 61.100 libavcodec 57. 93.100 / 57. 93.100 libavformat 57. 72.101 / 57. 72.101 libavdevice 57. 7.100 / 57. 7.100 libavfilter 6. 87.100 / 6. 87.100 libswscale 4. 7.101 / 4. 7.101 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 Input #0, lavfi, from 'testsrc=size=1920x1080:rate=25': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 1920x1080 [SAR 1:1 DAR 16:9], 25 tbr, 25 tbn, 25 tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo (native) -> dnxhd (native)) Press [q] to stop, [?] for help Output #0, mxf_opatom, to 'M:\Avid MediaFiles\MXF\2\tst1.mxf': Metadata: timecode : 01:02:03:04 material_package_name: raster test encoder : Lavf57.72.101 Stream #0:0: Video: dnxhd (DNXHD), yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-1024, 120000 kb/s, 25 fps, 25 tbn, 25 tbc Metadata: encoder : Lavc57.93.100 dnxhd frame= 10 fps=0.0 q=1.0 Lsize= 5927kB time=00:00:00.40 bitrate=121375.9kbits/s speed=2.38x video:5920kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.110738% --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

5 7

FFmpeg mosaic
by alexdjca 15 Jan '18

15 Jan '18

Hi All, following this help Create Mosaic with FFMPEG <https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20inp…> i've created a nice mosaic i run to keep a constant eye on a lot of different live streams I've published in my network, and the presence of audio in each of them. The final code i've built is the following: ffmpeg -probesize 1M \ -i udp://239.98.181.1:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.2:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.3:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.4:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.6:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.7:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.3:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.2:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.4:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.1:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.7:6000?fifo_size=100000?overrun_nonfatal=1 \ -i udp://239.98.181.2:6000?fifo_size=100000?overrun_nonfatal=1 \ -filter_complex "color=c=black:size=1280x720 [base];\ [0:v] setpts=PTS-STARTPTS, scale=224x192, drawtext=text='TESTO1'[r1p1];\ [0:a] showvolume=b=1:h=10:w=200:o=v [r1p1a];\ [1:v] setpts=PTS-STARTPTS, scale=224x192 [r1p2];\ [1:a] showvolume=b=1:h=10:w=200:o=v [r1p2a];\ [2:v] setpts=PTS-STARTPTS, scale=224x192 [r1p3];\ [2:a] showvolume=b=1:h=10:w=200:o=v [r1p3a];\ [3:v] setpts=PTS-STARTPTS, scale=224x192 [r1p4];\ [3:a] showvolume=b=1:h=10:w=200:o=v [r1p4a];\ [4:v] setpts=PTS-STARTPTS, scale=224x192 [r2p1];\ [4:a] showvolume=b=1:h=10:w=200:o=v [r2p1a];\ [5:v] setpts=PTS-STARTPTS, scale=224x192 [r2p2];\ [5:a] showvolume=b=1:h=10:w=200:o=v [r2p2a];\ [6:v] setpts=PTS-STARTPTS, scale=224x192 [r2p3];\ [6:a] showvolume=b=1:h=10:w=200:o=v [r2p3a];\ [7:v] setpts=PTS-STARTPTS, scale=224x192 [r2p4];\ [7:a] showvolume=b=1:h=10:w=200:o=v [r2p4a];\ [8:v] setpts=PTS-STARTPTS, scale=224x192 [r3p1];\ [8:a] showvolume=b=1:h=10:w=200:o=v [r3p1a];\ [9:v] setpts=PTS-STARTPTS, scale=224x192 [r3p2];\ [9:a] showvolume=b=1:h=10:w=200:o=v [r3p2a];\ [10:v] setpts=PTS-STARTPTS, scale=224x192 [r3p3];\ [10:a] showvolume=b=1:h=10:w=200:o=v [r3p3a];\ [11:v] setpts=PTS-STARTPTS, scale=224x192 [r3p4];\ [11:a] showvolume=b=1:h=10:w=200:o=v [r3p4a];\ [base][r1p1] overlay=shortest=0:x=50:y=10 [stop1]; [stop1][r1p1a] overlay=shortest=0:x=0:y=0 [tmp1];\ [tmp1][r1p2] overlay=shortest=0:x=370:y=10 [stop2]; [stop2][r1p2a] overlay=shortest=0:x=320:y=0 [tmp2];\ [tmp2][r1p3] overlay=shortest=0:x=690:y=10 [stop3]; [stop3][r1p3a] overlay=shortest=0:x=640:y=0 [tmp3];\ [tmp3][r1p4] overlay=shortest=0:x=1010:y=10 [stop4]; [stop4][r1p4a] overlay=shortest=0:x=960:y=0 [tmp4];\ [tmp4][r2p1] overlay=shortest=0:x=50:y=250 [stop5]; [stop5][r2p1a] overlay=shortest=0:x=0:y=240 [tmp5]; \ [tmp5][r2p2] overlay=shortest=0:x=370:y=250 [stop6]; [stop6][r2p2a] overlay=shortest=0:x=320:y=240 [tmp6]; \ [tmp6][r2p3] overlay=shortest=0:x=690:y=250 [stop7]; [stop7][r2p3a] overlay=shortest=0:x=640:y=240 [tmp7]; \ [tmp7][r2p4] overlay=shortest=0:x=1010:y=250 [stop8]; [stop8][r2p4a] overlay=shortest=0:x=960:y=240 [tmp8]; \ [tmp8][r3p1] overlay=shortest=0:x=50:y=490 [stop9]; [stop9][r3p1a] overlay=shortest=0:x=0:y=480 [tmp9]; \ [tmp9][r3p2] overlay=shortest=0:x=370:y=490 [stop10]; [stop10][r3p2a] overlay=shortest=0:x=320:y=480 [tmp10]; \ [tmp10][r3p3] overlay=shortest=0:x=690:y=490 [stop11]; [stop11][r3p3a] overlay=shortest=0:x=640:y=480 [tmp11]; \ [tmp11][r3p4] overlay=shortest=0:x=1010:y=490 [stop12]; [stop12][r3p4a] overlay=shortest=0:x=960:y=480" \ -c:v mpeg2video -b:v 4M -minrate 4M -maxrate 4M -map 0 -map 1 -map 2 -map 3 -map 4 -map 5 -map 6 -map 7 -map 8 -map 9 -map 10 -map 11 -f mpegts udp://239.98.181.20:6000 The result is quite good but there is a major issue i cannot figure out how to fix. Basically until all the streams are running fine the mosaic is perfect, but as soon as one of the input is missing all the mosaic stops and there is no way to let it start again. Another issue (minor actually) is that ffmpeg, before to compose the mosaic, needs to analyze all the streams so it takes a lot to start. Can anybody help me to find a solution please? Thanks in advance. Alex -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/FFmpeg-mosaic-tp4679120.html Sent from the FFmpeg-users mailing list archive at Nabble.com.

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FFMPEG sidechaincompress not working as it should?
by Frank ＠ Taapo 08 Jan '18

08 Jan '18

I have been trying to get sidechaining to work with FFMPEG, but my complex filter skills are lacking. I'm running two aif files into ffmpeg, hoping the voice would "duck" the background music when the voice-over is speaking. But I get an error with this command-line (on Windows, with latest FFMPEG binary): ffmpeg.exe -i temp_music.aif -i temp_voice.aif -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress=threshold=0.1:ratio=5[bg]; [bg][mix]amerge[fin‌al]" -map [final] final.mp3 Everything decodes fine: ffmpeg version N-87353-g183fd30 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 7.2.0 (GCC) configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --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-libxavs --enable-libxvid --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib libavutil 55. 76.100 / 55. 76.100 libavcodec 57.106.101 / 57.106.101 libavformat 57. 82.101 / 57. 82.101 libavdevice 57. 8.101 / 57. 8.101 libavfilter 6.105.100 / 6.105.100 libswscale 4. 7.103 / 4. 7.103 libswresample 2. 8.100 / 2. 8.100 libpostproc 54. 6.100 / 54. 6.100 Guessed Channel Layout for Input Stream #0.0 : stereo Input #0, aiff, from 'temp_music.aif': Duration: 00:01:26.68, start: 0.000000, bitrate: 1536 kb/s Stream #0:0: Audio: pcm_s16be, 48000 Hz, stereo, s16, 1536 kb/s Guessed Channel Layout for Input Stream #1.0 : stereo Input #1, aiff, from 'temp_voice.aif': Duration: 00:01:26.68, start: 0.000000, bitrate: 1536 kb/s Stream #1:0: Audio: pcm_s16be, 48000 Hz, stereo, s16, 1536 kb/s Output with label 'final' does not exist in any defined filter graph, or was already used elsewhere. But then I receive an error: Output with label 'final' does not exist in any defined filter graph, or was already used elsewhere. Anybody knows why? ᐧ

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