March 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

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

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

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.

2 1

Adding user-defined SEI messages thru libavcodec/libx264
by SF 06 Jan '18

06 Jan '18

Hi, I am searching a way to insert private SEI messages before each picture when encoding with x264. As far as i see from the libx264.c code, it is not supported up to now. Am i correct ? If yes, i am willing to push patch to support this, but do you have a preferred way to do this (ie thru medata field of AVFrame ? thru sidedata of AVFrame ? ...) Thanks for your hints, Regards, SF

2 1

QSV Hevc to HLS not segmenting?
by Markku Vainio 31 Oct '17

31 Oct '17

Hello, Encoding with hevc_qsv results only one large 00000.ts. It seems to me that ffmpeg is unable to split QSV encoded HEVC stream. (Intel Core i7-6770HQ, Iris Pro 580, W10) ffmpeg -y -i alasin.mp4 -vf yadif=0 -c:v hevc_qsv -load_plugin hevc_hw -g 25 -q 20 -preset slow -pix_fmt nv12 -c:a aac -ac 2 -hls_time 2 -hls_wrap 10000 -hls_segment_filename "%%05d.ts" index.m3u8 With libx265 splitting works fine. I also successfully tested h264_qsv. ffmpeg -y -i masteri.mov -vf yadif=0 -c:v libx265 -g 25 -q 20 -preset slow -c:a aac -ac 2 -hls_time 2 -hls_wrap 10000 -hls_segment_filename "%%05d.ts" index.m3u8 Am I doing something wrong here or is the HEVC stream coming from QSV missing some information that is needed for splitting at GOP boundary? C:\!!>ffmpeg -y -i alasin.mp4 -vf yadif=0 -c:v hevc_qsv -load_plugin hevc_hw -g 25 -q 20 -preset slow -pix_fmt nv12 -c:a aac -ac 2 -hls_time 2 -hls_wrap 10000 -hls_segment_filename "%05d.ts" index.m3u8 -loglevel verbose ffmpeg version N-83410-gb1e2192 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 5.4.0 (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-decklink --enable-zlib libavutil 55. 46.100 / 55. 46.100 libavcodec 57. 75.100 / 57. 75.100 libavformat 57. 66.101 / 57. 66.101 libavdevice 57. 2.100 / 57. 2.100 libavfilter 6. 72.100 / 6. 72.100 libswscale 4. 3.101 / 4. 3.101 libswresample 2. 4.100 / 2. 4.100 libpostproc 54. 2.100 / 54. 2.100 [h264 @ 0000000000dd46e0] Reinit context to 3840x2176, pix_fmt: yuv420p Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'alasin.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf57.58.102 Duration: 00:02:15.06, start: 0.000000, bitrate: 89980 kb/s Stream #0:0(und): Video: h264 (High), 1 reference frame (avc1 / 0x31637661), yuv420p(left), 3840x2160 (3840x2176) [SAR 1:1 DAR 16:9], 89529 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default) Metadata: handler_name : VideoHandler timecode : 00:00:00:00 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 460 kb/s (default) Metadata: handler_name : SoundHandler Stream #0:2(eng): Data: none (tmcd / 0x64636D74) Metadata: handler_name : TimeCodeHandler timecode : 00:00:00:00 [graph 0 input from stream 0:0 @ 0000000000e0f080] w:3840 h:2160 pixfmt:yuv420p tb:1/12800 fr:25/1 sar:1/1 sws_param:flags=2 [auto_scaler_0 @ 0000000000e5c4e0] w:iw h:ih flags:'bicubic' interl:0 [format @ 0000000000e0f600] auto-inserting filter 'auto_scaler_0' between the filter 'Parsed_yadif_0' and the filter 'format' [auto_scaler_0 @ 0000000000e5c4e0] w:3840 h:2160 fmt:yuv420p sar:1/1 -> w:3840 h:2160 fmt:nv12 sar:1/1 flags:0x4 [hevc_qsv @ 0000000000ea7be0] Initialized an internal MFX session using hardware accelerated implementation [hevc_qsv @ 0000000000ea7be0] Using the constant quantization parameter (CQP) ratecontrol method [hevc_qsv @ 0000000000ea7be0] profile: simple; level: 306 [hevc_qsv @ 0000000000ea7be0] GopPicSize: 25; GopRefDist: 9; GopOptFlag: closed ; IdrInterval: 0 [hevc_qsv @ 0000000000ea7be0] TargetUsage: 4; RateControlMethod: CQP [hevc_qsv @ 0000000000ea7be0] QPI: 16; QPP: 20; QPB: 26 [hevc_qsv @ 0000000000ea7be0] NumSlice: 1; NumRefFrame: 5 [hevc_qsv @ 0000000000ea7be0] RateDistortionOpt: unknown [hevc_qsv @ 0000000000ea7be0] RecoveryPointSEI: unknown IntRefType: 0; IntRefCycleSize: 0; IntRefQPDelta: 0 [hevc_qsv @ 0000000000ea7be0] MaxFrameSize: 0; MaxSliceSize: 0; [hevc_qsv @ 0000000000ea7be0] BitrateLimit: unknown; MBBRC: OFF; ExtBRC: unknown [hevc_qsv @ 0000000000ea7be0] Trellis: auto [hevc_qsv @ 0000000000ea7be0] RepeatPPS: OFF; NumMbPerSlice: 0; LookAheadDS: unknown [hevc_qsv @ 0000000000ea7be0] AdaptiveI: unknown; AdaptiveB: unknown; BRefType: pyramid [hevc_qsv @ 0000000000ea7be0] MinQPI: 0; MaxQPI: 0; MinQPP: 0; MaxQPP: 0; MinQPB: 0; MaxQPB: 0 [graph_1_in_0_1 @ 0000000000e5c5a0] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3 [mpegts @ 0000000002a294e0] muxrate VBR, pcr every 2 pkts, sdt every 2147483647, pat/pmt every 2147483647 pkts Output #0, hls, to 'index.m3u8': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf57.66.101 Stream #0:0(und): Video: hevc (hevc_qsv), 1 reference frame, nv12(left), 3840x2160 [SAR 1:1 DAR 16:9], q=2-31, 1000 kb/s, 25 fps, 90k tbn, 25 tbc (default) Metadata: handler_name : VideoHandler timecode : 00:00:00:00 encoder : Lavc57.75.100 hevc_qsv Side data: cpb: bitrate max/min/avg: 0/0/1000000 buffer size: 0 vbv_delay: -1 Stream #0:1(und): Audio: aac (LC), 48000 Hz, stereo, fltp, delay 1024, 128 kb/s (default) Metadata: handler_name : SoundHandler encoder : Lavc57.75.100 aac Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> hevc (hevc_qsv)) Stream #0:1 -> #0:1 (aac (native) -> aac (native)) Press [q] to stop, [?] for help [h264 @ 0000000000eaa8e0] Reinit context to 3840x2176, pix_fmt: yuv420p frame= 846 fps= 35 q=-0.0 size=N/A time=00:00:34.15 bitrate=N/A speed=1.43x

3 17

Syncing clips based on audio waveform?
by Arthur Wait 05 Oct '17

05 Oct '17

Hi all, This is coming from out of left field I imagine, but I'm wondering if anyone knows of a filter (or chain of filters) which would allow me to align/sync two video clips based on matches in their audio waveforms. I'm shooting events with multiple cameras and would like to either trim the clips to a precise, matching starting frame or generate a metadata "offset" marker indicating how far off one clip is from the other in time. Then I can trim the clips to match. Is this possible in ffmpeg (or anywhere else for that matter!)? Thanks very much! Art

7 7

could not find codec parameters for stream 0 unspecified size
by Ashish Lal 09 Aug '17

09 Aug '17

I am trying to convert a live rtsp stream (H264 + MPEG2 TS ) into MP4 using ffmpeg. I am getting an error "could not find codec parameters for stream 0 unspecified size". Any help in fixing this is appreciated. ffmpeg -rtsp_transport tcp -i rtsp://AA.BB.CC.DD:5555/mpeg2TransportStreamFr omUDPSourceTest -analyzeduration 999999999 -probesize 99999999 -vcodec libx264 -s 176x144 -b:v 256k -f mp4 -movflags frag_keyframe+empty_moov op.mp4 ffmpeg version 2.2.2 Copyright (c) 2000-2014 the FFmpeg developers built on May 27 2014 09:12:51 with gcc 4.7 (Debian 4.7.2-5) configuration: --enable-libx264 --enable-encoder=libx264 --enable-libvpx --pre fix=/home/ashish/ffmpeg-2.2.2/mybuild --enable-gpl --enable-debug --disable-stri pping --disable-optimizations --enable-memalign-hack libavutil 52. 66.100 / 52. 66.100 libavcodec 55. 52.102 / 55. 52.102 libavformat 55. 33.100 / 55. 33.100 libavdevice 55. 10.100 / 55. 10.100 libavfilter 4. 2.100 / 4. 2.100 libswscale 2. 5.102 / 2. 5.102 libswresample 0. 18.100 / 0. 18.100 libpostproc 52. 3.100 / 52. 3.100 [NULL @ 0x235cd60] non-existing PPS 0 referenced [h264 @ 0x235cd60] non-existing PPS 0 referenced [h264 @ 0x235cd60] decode_slice_header error [h264 @ 0x235cd60] no frame! [h264 @ 0x235cd60] non-existing PPS 0 referenced ......... ......... Last message repeated 1 times [h264 @ 0x235cd60] decode_slice_header error [h264 @ 0x235cd60] no frame! [h264 @ 0x235cd60] non-existing PPS 0 referenced [rtsp @ 0x232fc40] decoding for stream 0 failed [rtsp @ 0x232fc40] Could not find codec parameters for stream 0 (Video: h264 ([2 7][0][0][0] / 0x001B)): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options rtsp://AA.BB.CC.DD:5555/mpeg2TransportStreamFromUDPSourceTest: could not find codec parameters Conversion failed! Regards Ashish

3 3

Converting a cassette audio tape (Lecture) to mp3
by jd1008 14 Jul '17

14 Jul '17

Hi all, I am running fedora 20 with ffmpeg version N-51556-ge278500 Copyright (c) 2000-2013 the FFmpeg developers I also installed in /opt ffmpeg version N-72985-g7c3f7e6 Copyright (c) 2000-2015 the FFmpeg developers I need a set of steps/commands to convert this tape. I have plenty of audio tools installed - but not certain what else I might need. If anyone has done this before, please share your script. Kind regards, JD

6 15

Compiling FFMPEG with ffplay support
by Stas Oskin 13 Jun '17

13 Jun '17

Hi. Did the compilation parameters changed in recent FFMPEG version? Configuring the FFMPEG like this: ./configure --prefix=/usr --enable-pthreads Does not produce ffplay anymore. I tried to add --enable-ffplay, without any result. Any idea what I'm missing? Thanks.

6 9

Create thumbnails from RTMP stream
by Grant Curtis 09 Jun '17

09 Jun '17

If I use Flash Media Live Encoder to send a stream to an RTMP server, and then, on that server, run the below command, I can create thumbnails from the stream. But when I try to do the same when I stream the file to the server using ffmpeg, I get errors, and no thumbnails are created Server command and output: $ /usr/bin/ffmpeg -report -i "rtmp://localhost:1935/osc-live-packager/cumin?offsetFromLive=10,userID=testUser1/mp4:rebroadcast_testUser1" -y -r 29.97 -threads 4 -c:v copy -c:a copy /mnt_ephemeral1/media/cumin/cumin_1000.mp4 -filter:v "scale=w=178:h=100" -f image2 -vf "fps=fps=1/5" "/mnt_ephemeral1/media/cumin/cumin/thumbs/thumb_%04d.png" ffmpeg started on 2016-04-25 at 18:28:39 Report written to "/var/log/ffmpeg/ffmpeg-20160425-182839.log" ffmpeg version 2.7.2-static http://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.3 (Debian 4.9.3-1) 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 --enable-libass --enable-gnutls --enable-libvidstab --enable-libsoxr --cc=gcc-4.9 libavutil 54. 27.100 / 54. 27.100 libavcodec 56. 41.100 / 56. 41.100 libavformat 56. 36.100 / 56. 36.100 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 16.101 / 5. 16.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.100 / 1. 2.100 libpostproc 53. 3.100 / 53. 3.100 [NULL @ 0x31b65e0] missing picture in access unit with size 110664 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2338 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2645 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2600 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 8421 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 2895 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 1759 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 776 [h264 @ 0x31b65e0] no frame! [h264 @ 0x31b65e0] missing picture in access unit with size 13391 [h264 @ 0x31b65e0] no frame! -truncated a bunch more similar messages- [flv @ 0x31b3c00] decoding for stream 0 failed [flv @ 0x31b3c00] Could not find codec parameters for stream 0 (Video: h264, none): unspecified size Consider increasing the value for the 'analyzeduration' and 'probesize' options Input #0, flv, from 'rtmp://localhost:1935/osc-live-packager/cumin?offsetFromLive=10,userID=testUser1/mp4:rebroadcast_testUser1': Metadata: moovPosition : 36 avcprofile : 100 avclevel : 31 videoframerate : 25 audiochannels : 2 timescale : 1000 length : 839 sampletype : .mp3 Duration: 00:00:00.84, start: 0.040000, bitrate: N/A Stream #0:0: Video: h264, none, 25 tbr, 1k tbn, 2k tbc Stream #0:1: Subtitle: text Stream #0:2: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s Output #1, image2, to '/mnt_ephemeral1/media/cumin/cumin/thumbs/thumb_%04d.png': Metadata: moovPosition : 36 avcprofile : 100 avclevel : 31 videoframerate : 25 audiochannels : 2 timescale : 1000 length : 839 sampletype : .mp3 Output file #1 does not contain any stream Received signal 2: terminating. --- Here's how I start the stream from my local machine: --- ffmpeg -re -i testVideo.mp4 -ar 44100 -c:v libx264 -f flv rtmp://xxxxxxxx:1935/xxxxx/xxxxx/xxxxx_1000.mp4 ffmpeg version 2.8.4 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-lzma --enable-decklink --enable-zlib libavutil 54. 31.100 / 54. 31.100 libavcodec 56. 60.100 / 56. 60.100 libavformat 56. 40.101 / 56. 40.101 libavdevice 56. 4.100 / 56. 4.100 libavfilter 5. 40.101 / 5. 40.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.101 / 1. 2.101 libpostproc 53. 3.100 / 53. 3.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testVideo.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf53.24.2 Duration: 00:01:02.32, start: 0.000000, bitrate: 1347 kb/s Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 959 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, 5.1, fltp, 383 kb/s (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : SoundHandler [libx264 @ 03052340] using SAR=1/1 [libx264 @ 03052340] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 AVX2 LZCNT BMI2 [libx264 @ 03052340] profile High, level 3.1 [libx264 @ 03052340] 264 - core 148 r2638 7599210 - H.264/MPEG-4 AVC codec - Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, flv, to 'rtmp://xxxxxx:1935/xxxx/xxxxx/xxxxx_1000.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf56.40.101 Stream #0:0(und): Video: h264 (libx264) ([7][0][0][0] / 0x0007), yuv420p, 1280x720 [SAR 1:1 DAR 16:9], q=-1--1, 25 fps, 1k tbn, 25 tbc (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler encoder : Lavc56.60.100 libx264 Stream #0:1(und): Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, stereo, fltp (default) Metadata: creation_time : 1970-01-01 00:00:00 handler_name : SoundHandler encoder : Lavc56.60.100 libmp3lame Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264)) Stream #0:1 -> #0:1 (aac (native) -> mp3 (libmp3lame)) Press [q] to stop, [?] for help frame= 1336 fps= 25 q=28.0 size= 7310kB time=00:00:53.63 bitrate=1116.6kbits/s

2 2

Re: [FFmpeg-user] FFMPEG & Panasonic OP-1B MXF decode issue
by Jeff Smith 06 Jun '17

06 Jun '17

> 2017-03-10 9:20 GMT+01:00 <jeff at jksmith.com <http://ffmpeg.org/mailman/listinfo/ffmpeg-user>>: > >/Happy to provide a sample. Is there an FTP site or something I /> >/can upload it to? /> > You can usehttp://streams.videolan.org/upload/ but you absolutely > have to provide the file name, the service is extremely impractical. > > Carl Eugen Hi Carl, I'm assuming from radio silence that you couldn't get the sample I uploaded. I've posted it on an anonymous FTP server here: ftp://5.56.49.144/085FV4BR.MXF . Grateful if you could acknowledge if this works for you. Thanks, Jeff --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

2 1

Float formatting in drawtext
by Wolfgang Hugemann 29 May '17

29 May '17

Hey, seems to be an easy question, but I have been searching for the answer quite a while without success: If I perform a float calculation via text=%{expr\:whatever} is there any way to format the output the printf way? I know, the result could be converted to integer and be formatted via text=%{eif\:whatever\:d} but this is restricted to integer results. Wolfgang Hugemann

2 4

How do I encode DVB Subtitles?
by Wagner Patriota 02 May '17

02 May '17

*QUESTION 1:* How do I add a STR file to a MPEG-TS file with the "DVB subtitle codec"? If so, could you give me a sample command line? Thank. I am trying: ffmpeg -i video.ts *-i subtitle.srt* -vcodec copy -acodec copy *-scodec dvbsub* output.ts The stream is created because I can see it with MediaInfo, but the subtitles don't appear! *QUESTION 2:* How do I use the command *slang*? I try to use it, but ffmpeg says the command doesn't exist. Thank you.

7 15

DVB H264 chunk transcoding => merging in MP4 => out of sync
by ffmpeg＠me.com 10 Apr '17

10 Apr '17

Viewing on VLC works great. Audio/Video perfectly in sync. Viewing in Quicktime (Version 10.4 MacOS sierra, Quicktime 7) the video get's out of sync after 3/4 minutes. Scrubbing in the timeline will fix the problem, then after 3/4 minutes I see the synchronization problem again. Using current FFmpeg version to transcode into H.264 and AAC TS chunks. ffmpeg -i "udp://224.5.2.5:10001?overrun_nonfatal=1&fifo_size=50000000" -map 0:3 -map 0:4 -s 1280x720 -acodec aac -b:a 192k -ac 2 -ar 48000 -vcodec libx264 -preset:v medium -r 50 -g 100 -vb 3200000 -y -fflags +genpts -f segment -segment_time 60 -segment_atclocktime 1 -strftime 1 "/videofiles/TV_HD_%Y-%m-%d_%H-%M-%S.ts Everything works fine... chunks are created. No errors in output. Audio/video TS pieces in sync. #### Details of a chunk file: ffprobe /videofiles/TV_HD_2017-03-13_14-25-00.ts ffprobe version 3.2.4-tessus Copyright (c) 2007-2017 the FFmpeg developers built with Apple LLVM version 8.0.0 (clang-800.0.42.1) configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzmq --enable-version3 --disable-ffplay --disable-indev=qtkit --disable-indev=x11grab_xcb libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 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 libpostproc 54. 1.100 / 54. 1.100 Input #0, mpegts, from '/Users/Andreas/Desktop/TEST/ARD_HD_2017-03-13_14-25-00.ts': Duration: 00:01:00.04, start: 115.402667, bitrate: 3720 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc Stream #0:1[0x101](deu): Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 204 kb/s #### Merging together: ffmpeg -f concat -safe 0 -i files.txt -y -metadata creation_time=now -c:v copy -c:a copy -fflags +genpts -movflags faststart -use_editlist 0 -f mp4 tvexport.mp4 #### and output: ffmpeg version 3.2.4-tessus Copyright (c) 2000-2017 the FFmpeg developers built with Apple LLVM version 8.0.0 (clang-800.0.42.1) configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl --enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopus --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzmq --enable-version3 --disable-ffplay --disable-indev=qtkit --disable-indev=x11grab_xcb libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 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 libpostproc 54. 1.100 / 54. 1.100 Input #0, concat, from 'files.txt': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc Stream #0:1(deu): Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 169 kb/s Output #0, mp4, to 'work_tmp0.mp4': Metadata: creation_time : now encoder : Lavf57.56.101 Stream #0:0: Video: h264 (High) ([33][0][0][0] / 0x0021), yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 50 fps, 50 tbr, 90k tbn, 90k tbc Stream #0:1(deu): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, stereo, 169 kb/s Stream mapping: Stream #0:0 -> #0:0 (copy) Stream #0:1 -> #0:1 (copy) Press [q] to stop, [?] for help [mp4 @ 0x7fff37800600] Starting second pass: moving the moov atom to the beginning of the filex frame=24000 fps=9825 q=-1.0 Lsize= 199633kB time=00:08:00.20 bitrate=3405.6kbits/s speed= 197x video:187817kB audio:11755kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.030912% What could be the problem?

2 7

Can't get a/v synchronisation right with mpeg transport stream that has gaps in it
by Erik Slagter 03 Apr '17

03 Apr '17

Hi people, I'm having this problem for years and years (so it's not a recent problem or regression). I make recordings on my enigma2 based settopbox. It saves the recordings as the original transport stream, but with filtering of only the requested service pids. There is a feature to edit the recorded services, in a basic way. Very helpful to edit out commercials. Problem is that it makes cuts in a very "rude" way, it justs cuts the file at a certain position, in the middle of all kinds of open packets and streams. On playback it isn't such a big deal, because the damaged packets are discarded and the DVB hardware resyncs on the timestamps pretty quickly. After the crude editing, I like to perform a real editing run, using kdenlive (or any other NLVE, for that case). I've found that all of them have problems when fed with a "damaged" transport stream, like this. So what I do, is transcode the stream to something with mjpeg and pcm codecs, intra frames only, very easy on the editor. This is the problem: after every timestamp discontinuity ("cut" or "gap"), ffmpeg resyncs audio and video BUT it doesn't do it 100% correctly. Sound remains a bit "in front" of the video, skewed a bit. Also, it gets worse at every cut, everytime about 200-300 ms is added to the skew. That's not what I'd expect? I've tried these options to force synchronisation: - -vsync 1 (not per sé required, but I want the audio and video streams in the container in such a way that they're not skewed, so if the editor decides to demux and throw away the timestamps, it still works) - -r 25 (material is 25 fps, tried with and without, doesn't matter) - filter:a " aresample=async=10000:min_comp=0.1:min_hard_comp=0.1:max_soft_comp=10000:first_pts=0" None of this helps. I can see ffmpeg dropping and dupping frames, but it doesn't help to fix the slow drift. I was under the impression that ffmpeg would automatically base the synchronisation of the (single) audio stream on the (single) video stream. Is this a correct assumption or do I need to specify an explicit "-map" statement? It looks like ffmpeg takes care that frames are dupped/dropped so the frame rate keeps "solid" 25 fps (or whatever the source frame rate is specified as or is specified with -r), but it seems it does not keep the audio stream synchronised. Thanks for help, Erik.

3 12

T shirt
by David Whitaker 03 Apr '17

03 Apr '17

Hi, I am a big fan of ffmpeg, I like the logo so I decided to have a single white t shirt with the logo printed for my self. The company I asked to print for me wants me to have permission to use the logo first. Here is the logo I want to use https://upload.wikimedia.org/wikipedia/commons/thumb/7/76/FFmpeg_icon.svg/2… ------------------------------- Here is there contact info if needed Casey Maxwell CustomInk Satisfaction Assurance Associate 800-497-2067 service(a)customink.com http://www.customink.com FAST, FRIENDLY HELP 7 DAYS A WEEK Mon-Fri: 8am - 10:30pm Sat: 10am - 6pm Sun: 10am - 9pm (all times eastern) Helpful Links... Frequently Asked Questions: http://customink.com/help_center ------------------------------- If someone could help me find out if it possible to get permission to use this logo for 1 shirt id definitely appreciate the effort. Thanks Dave

7 8

Building Error avcodec_send_packet
by יוסף אלון 03 Apr '17

03 Apr '17

I would like to compile to following program: With the flages: -lavcodec -lavutil -lavfilter -lavformat -lswscale -lz #include "libavcodec/avcodec.h" #include "libavformat/avformat.h" #include "libswscale/swscale.h" #include <stdio.h> // The flush packet is a non-NULL packet with size 0 and data NULL int decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame, AVPacket *pkt) { int ret; *got_frame = 0; if (pkt) { ret = avcodec_send_packet(avctx, pkt); // In particular, we don't expect AVERROR(EAGAIN), because we read all // decoded frames with avcodec_receive_frame() until done. if (ret < 0) return ret == AVERROR_EOF ? 0 : ret; } ret = avcodec_receive_frame(avctx, frame); if (ret < 0 && ret != AVERROR(EAGAIN) && ret != AVERROR_EOF) return ret; if (ret >= 0) *got_frame = 1; return 0; } int main(int argc, char *argv[]) { return 0; But I receive: undefined reference to `avcodec_send_packet' undefined reference to `avcodec_receive_frame' ffmpeg version is: ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609 configuration: libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 / 57. 56.101 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 Does some one know, Whats the problem? -- Yosef Alon

2 3

Maximum argument length for tee pseudo muxer
by luke harrison 02 Apr '17

02 Apr '17

I'm preparing a script to take a feed coming in over SDI and encode it into multiple video and audio streams, all for distribution to different stream paths using the tee muxer. I'm sending quite a few different mixes of the streams to an RTMP server which means that my tee muxer is taking quite a large argument. My command line is shown below. ffmpeg -threads 0 -f decklink -i 'UltraStudio Mini Recorder@11' -loglevel debug \ -filter_complex "[0:v]split=2[VID1][VID2];[0:a]channelsplit=channel_layout=stereo[AL][AR]" -map '[AR]' -map '[AL]' -map '[VID1]' -map '[VID2]' -flags +global_header -c:v libx264 -profile:v high -preset veryfast -pix_fmt yuv420p -r 25 -s:v:0 640x360 -s:v:1 426x240 -b:v:0 484k -b:v:1 215k -bufsize 1500k -coder 1 -g 125 -keyint_min 25 -c:a aac -b:a 64k -ar 44100 -async 1 -rtmp_flashver "FMLE/3.0(compatible;FMSc/1.0)" -rtmp_live live -f tee "[select=\'a:0,v:0\':f=flv]'${entrypoint}${streampath1}${datavid11}'| \ [select=\'a:1,v:0\':f=flv]'${entrypoint}${streampath2}${datavid11}'| \ [select=\'a:0,v:1\':f=flv]'${entrypoint}${streampath1}${datavid12}'| \ [select=\'a:1,v:1\':f=flv]'${entrypoint}${streampath2}${datavid12}'| \ [select=\'a:0\':f=flv]'${entrypoint}${streampath1}${dataaud11}'| \ [select=\'a:1\':f=flv]'${entrypoint}${streampath2}${dataaud11}'" All of the output streams up until the last one (with a:1) connect to the server and begin streaming. The issue comes when ffmpeg tries to connect to the final stream. The stream path is cut off for some reason. This results in ffmpeg trying to connect to a stream path that is incomplete and therefore failing. [tcp @ 0x7fa0dc414180] Failed to resolve hostname p.epXXXXXX.i.ak: nodename nor servname provided, or not known [rtmp @ 0x7fa0dc414100] Cannot open connection tcp://p.epXXXXX.i.ak:1935 [tee @ 0x7fa0de801800] Slave '[select='a:1':f=flv]rtmp://XXXXXXX:XXXXXXX@p.epXXXXXX.i.ak': error opening: Input/output error [tee @ 0x7fa0de801800] Slave muxer #5 failed, aborting. I have confirmed that removing characters within the inverted commas ("") after the tee statement allows more of the stream path to be processed by ffmpeg and inversely adding characters within the "" results in less of the stream path being processed by ffmpeg. The characters can be related to any of the commands within the "" and don't need to specifically be any part of the stream paths. To me this looks as if the tee muxer has an undocumented, maximum character input limit for the argument that it takes within the inverted commas. Would any devs or users be able to confirm this based on experience/a deeper knowledge of the code? -- Kind regards, Luke Harrison

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MLV-files
by Cecil Westerhof 31 Mar '17

31 Mar '17

Yesterday I discovered Magic Lantern. I read that ffmpeg supports the MLV format. Does anyone has experience with this? -- Cecil Westerhof Senior Software Engineer LinkedIn: http://www.linkedin.com/in/cecilwesterhof

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movement on ffmpeg .R3D v4 support?
by Rob Wentz 31 Mar '17

31 Mar '17

Hi, I work in video post production, our DPs all shoot on Red cameras. . . it's become the norm that if stuff is coming in shot on anything larger than 2K it's shot on a Red camera and the footage is in .R3D. I was hoping to be able to get somewhere with using ffmpeg to create proxies of .R3Ds before archiving them to LTO using Archiware P5 Archive. The feature works wonders for allowing us to generate scrubbable proxies of archived .mov, mp4, etc video fileswho's full res files live on LTO5 - but we currently can not use ffmpeg on the current (v4) R3D format (Red Camera). We shoot on Red Weapon and raster sizes are getting HUGE with the advent of uHD and 720/VR. It is not practical to cast a wide blind net of footage to unarchive as this material takes up so much space and takes considerable time to unarchive. Most recent tracks of any movement on this topic I found have been silent for a while: https://trac.ffmpeg.org/ticket/2690 I'm curious where things are at in terms of .R3D v4 support for ffmpeg. Thank you! Rob Wentz • Post-Production Supervisor • Pedersen Media Group 501 Second Street, Suite 212, San Francisco, CA 94107

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cross compile FFPLAY
by Lars Heinrichs 31 Mar '17

31 Mar '17

Hello! I can successfully build ffmpeg using a cross compile toolchain (linaro). What I need on my target system is mostly ffplay, however, FFPLAY is missing. Some google searches showed that SDL is required to build FFPLAY, so i installed libsdl2-dev. I am running ./configure using ./configure --enable-cross-compile --cross-prefix=<toolchain_path>/arm-linux-gnueabihf- --arch=arm --cpu=armv7-a --target-os=linux what generates a warning: WARNING: <toolchain_path>/arm-linux-gnueabihf-pkg-config not found, library detection may fail. what is correct, there is no -pkg-config in the toolchain. config.mak had a line !CONFIG_FFPLAY=yes that i removed the ! from. On 'make', everything is build just fine except for ffplay: CC ffplay.o ffplay.c:56:17: fatal error: SDL.h: No such file or directory compilation terminated. I don know if the error is related to the warning I got on running the configure script. Do I need SDL to be built for the target platform (= cross compile and then link the result (how would i link this?)?) or should it be working right away? If -pgk-config does not exist, is that a problem? If what I want is not possible for some reason, what might be an alternative to ffplay?

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warning cseq 10 expected 8 received
by Ankit Gupta 30 Mar '17

30 Mar '17

Hi All, using following command for RTSP to http streaming\ ffmpeg -rtsp_transport tcp -i {RTSP} -c:v libx264 -crf 35 -preset ultrafast -maxrate 3M -bufsize 300k -r 10 -g 20 -movflags +faststart -tune zerolatency -hls_time 1 -hls_list_size 4 -hls_wrap 4 -start_number 1 -hls_allow_cache 0 -threads 1 -loglevel warning -y {} but getting frequent warning "cseq 10 expected 8 received" Any help??????

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H.265/HEVC for Premiere Pro compatibility
by Chris Colton 29 Mar '17

29 Mar '17

Hi all, I tried to get an answer on this before I actually subscribed to the mailing list, so I'm told I didn't receive all the replies so I'm just starting over here. I'm using FFmpeg to transcode to H.265 because Premiere's native encoding is incompatible with Vimeo (Vimeo's transcoding causes a color shift in HEVC from Premiere). I also love the constant rate factor encoding option because it'll decide bitrate depending on the resolution needed. I used the command below and VLC and Vimeo both play it (vimeo with no color issues), but if you try to check the encode to the original in Premiere, it imports with sound playing correctly and just a blank green image instead of the video. Any ideas? -i "source.mov" -y -c:v libx265 -preset veryfast -crf 23 -vf format=yuv420p -c:a aac -b:a 320k -ar 48000 -ac 2 -max_muxing_queue_size 1000 "H265.mp4" ᐧ

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