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• Transcode to opus by Fluent-ffmpeg or ffmpeg from command line

  14 juillet 2017, par shamaleyte

  My purpose is to transcode a webm file into opus file.
  It works just fine as the following ;

  ffmpeg -i input.webm -vn -c:a copy output.opus

  But the generated opus file always starts from 4rd or 5th seconds when I play it. It seems like that the first seconds are lost. Any idea why it happens ?

  >ffmpeg -i x.webm -vn -c:a copy x1.opus
  
  ffmpeg version N-86175-g64ea4d1 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. 63.100 / 55. 63.100
  
    libavcodec     57. 96.101 / 57. 96.101
  
    libavformat    57. 72.101 / 57. 72.101
  
    libavdevice    57.  7.100 / 57.  7.100
  
    libavfilter     6. 90.100 /  6. 90.100
  
    libswscale      4.  7.101 /  4.  7.101
  
    libswresample   2.  8.100 /  2.  8.100
  
    libpostproc    54.  6.100 / 54.  6.100
  
  Input #0, matroska,webm, from 'x.webm':
  
    Metadata:
  
    encoder         : libwebm-0.2.1.0
  
    creation_time   : 2017-06-19T20:50:21.722000Z
  
  Duration: 00:00:32.33, start: 0.000000, bitrate: 134 kb/s
  
  Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)
  
  Stream #0:1(eng): Video: vp8, yuv420p(progressive), 640x480, SAR 1:1 DAR 
  
  4:3, 16.67 fps, 16.67 tbr, 1k tbn, 1k tbc (default)
  
  Output #0, opus, to 'x1.opus':
  
  Metadata:
  
  encoder         : Lavf57.72.101
  
  Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)
  
  Metadata:
  
    encoder         : Lavf57.72.101
  
  Stream mapping:
  
  Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  size=     114kB time=00:00:32.33 bitrate=  28.8kbits/s speed=3.22e+003x
  
  video:0kB audio:111kB subtitle:0kB other streams:0kB global headers:0kB 
  
  muxing overhead: 2.152229%

  It is jumping from 0 to 4th second .
  Please take a look at this screencast.
  https://www.screenmailer.com/v/52IXnpAarHavwJE

  This is the sample video file that I tried to transcode : https://drive.google.com/open?id=0B2sa3oV_Y3X_ZmVWX3MzTlRPSmc

  So I guess the transcoding starts right at the point that the voice comes in, why is that ?

• How can I play audio file(.mp3, .flac, .wav) and then loop over it (mix every few seconds) another audio file(wav) using ffmpeg

  11 mars 2019, par lukistar

  I got two different commands.

  ffmpeg -i input.mp3 -i second.mp3 -filter_complex "[0:a]atrim=end=10,asetpts=N/SR/TB[begin];[0:a]atrim=start=10,asetpts=N/SR/TB[end];[begin][1:a][end]concat=n=3:v=0:a=1[a]" -map "[a]" output

  This command inserts second.mp3 into input.mp3. It seems to always keep the parameters of input.mp3. It inserts it in exact 10 seconds of input.mp3.

  Here is the second command :

  ffmpeg -i input.mp3 -i second.mp3 -filter_complex "[1:a]adelay=10000|10000[1a];[0:a][1a]amix=duration:first" output

  This command is closer to my final goal. It plays input.mp3 and in exact 10 seconds it plays along second.mp3 without stopping input.mp3’s sound.(I think that’s called mixing ?)

  My final goal is to create final.mp3.

  Its duration must always equal input.mp3 duration. It must keep the samplerate, the count of channels, etc of input.mp3

  When playing final.mp3, it must play the whole input.mp3.
  But each 10-15 seconds, it must play second.mp3 without stopping input.mp3.(mix)
  It could be said that I must use "Second command" but in a loop.
  It would be great if there is one-line command for that in ffmpeg.
  I am working with flac, mp3 and wav and both of the commands were suitable for that.

  For example :

  input.mp3 could be 40 seconds long.

  second.mp3 could be 2 seconds long.

  When I play final.mp3 it will be 40 seconds long, but each 10-15 seconds(on random) it will play second.mp3 at the same time as input.mp3.

  Sadly I have no experience with ffmpeg, both of the commands I got are answers to questions here in stackoverflow. Hope somebody can help me. Thank you !

• Merge commit 'a1a143adb0fd11c474221431417cff25db7d920f'

  26 septembre 2017, par James Almer

  Merge commit 'a1a143adb0fd11c474221431417cff25db7d920f'
  * commit 'a1a143adb0fd11c474221431417cff25db7d920f' :
  
    rtmp : Rename packet types to closer match the spec
  Merged-by : James Almer <jamrial@gmail.com>
  

  • [DH] libavformat/rtmppkt.c
  • [DH] libavformat/rtmppkt.h
  • [DH] libavformat/rtmpproto.c