April 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 26 Mar '18

26 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 02 Mar '18

02 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 05 Jan '18

05 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

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