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• ffmpeg Output file #0 does not contain any stream when trying to access 1 of 2 audio streams

  7 juillet 2019, par nulltorpedo

  ffmpeg -i input.mkv  -map 0:2 -c copy -strict -2  audio.mkv

  Hi I have the above command. The output shows that there are 2 audio streams. I want to copy just the ac3 audio (actually I want to convert it but even this copy does not work). I have truncated the output print where there is metadata

  NEW updated sample with full log which results in same message

  ffmpeg -i input.mka -map 0:0 -c:a libfdk_aac   aac_out.m4a
  
  ffmpeg version 2.7.1 Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.9.3 (crosstool-NG 1.20.0) 20150311 (prerelease)
  
    configuration: --prefix=/usr --incdir='${prefix}/include/ffmpeg' --arch=i686 --target-os=linux --cross-prefix=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu- --enable-cross-compile --enable-optimizations --enable-pic --enable-gpl --enable-shared --disable-static --enable-version3 --enable-nonfree --enable-libfaac --enable-encoders --enable-pthreads --disable-bzlib --disable-protocol=rtp --disable-muxer=image2 --disable-muxer=image2pipe --disable-swscale-alpha --disable-ffserver --disable-ffplay --disable-devices --disable-bzlib --disable-altivec --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libmp3lame --disable-vaapi --disable-decoder=amrnb --disable-decoder=ac3 --disable-decoder=ac3_fixed --disable-encoder=zmbv --disable-encoder=dca --disable-encoder=ac3 --disable-encoder=ac3_fixed --disable-encoder=eac3 --disable-decoder=dca --disable-decoder=eac3 --disable-decoder=truehd --cc=/usr/local/x86_64-pc-linux-gnu/bin/x86_64-pc-linux-gnu-ccache-gcc --enable-yasm --enable-libx264 --enable-encoder=libx264
  
    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
  
  Input #0, matroska,webm, from '/volume1/..../input.mka':
  
    Metadata:
  
      encoder         : libebml v1.3.9 + libmatroska v1.5.2
  
      creation_time   : 2019-07-07 06:19:20
  
    Duration: 02:29:21.98, start: 0.000000, bitrate: 640 kb/s
  
      Stream #0:0(eng): Audio: ac3, 48000 Hz, 5.1(side), 640 kb/s
  
      Metadata:
  
        BPS-eng         : 640000
  
        DURATION-eng    : 02:29:21.984000480
  
        NUMBER_OF_FRAMES-eng: 280062
  
        NUMBER_OF_BYTES-eng: 716958720
  
        _STATISTICS_WRITING_APP-eng: mkvmerge v35.0.0 ('All The Love In The World') 64-bit
  
        _STATISTICS_WRITING_DATE_UTC-eng: 2019-07-07 06:19:20
  
        _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
  
  Output #0, ipod, to 'aac_out.m4a':
  
    Metadata:
  
      encoder         : libebml v1.3.9 + libmatroska v1.5.2
  
  Output file #0 does not contain any stream

• How to convert aac to ogg opus keeping bit rate and sample rate unchanged

  25 juin 2019, par Doovi

  I’m trying to convert a .aac file to .opus but after inspecting with ffprobe I get different bit and sample rates.

  While input file’s audio stream bit rate is 245995, the output file’s audio stream has no bit rate specified - "format" shows bit rate of 118788.

  While input file’s audio stream sample rate is 44100, the output’s is 48000.

  ffprobe  -v error -show_format -show_streams input.aac
  
  [STREAM]
  
  index=0
  
  codec_name=aac
  
  codec_long_name=AAC (Advanced Audio Coding)
  
  profile=LC
  
  codec_type=audio
  
  codec_time_base=1/44100
  
  codec_tag_string=[0][0][0][0]
  
  codec_tag=0x0000
  
  sample_fmt=fltp
  
  sample_rate=44100
  
  channels=2
  
  channel_layout=stereo
  
  bits_per_sample=0
  
  id=N/A
  
  r_frame_rate=0/0
  
  avg_frame_rate=0/0
  
  time_base=1/28224000
  
  start_pts=N/A
  
  start_time=N/A
  
  duration_ts=106533390807
  
  duration=3774.567418
  
  bit_rate=245995
  
  max_bit_rate=N/A
  
  bits_per_raw_sample=N/A
  
  nb_frames=N/A
  
  nb_read_frames=N/A
  
  nb_read_packets=N/A
  
  DISPOSITION:default=0
  
  DISPOSITION:dub=0
  
  DISPOSITION:original=0
  
  DISPOSITION:comment=0
  
  DISPOSITION:lyrics=0
  
  DISPOSITION:karaoke=0
  
  DISPOSITION:forced=0
  
  DISPOSITION:hearing_impaired=0
  
  DISPOSITION:visual_impaired=0
  
  DISPOSITION:clean_effects=0
  
  DISPOSITION:attached_pic=0
  
  DISPOSITION:timed_thumbnails=0
  
  [/STREAM]
  
  [FORMAT]
  
  filename=input.aac
  
  nb_streams=1
  
  nb_programs=0
  
  format_name=aac
  
  format_long_name=raw ADTS AAC (Advanced Audio Coding)
  
  start_time=N/A
  
  duration=3774.567418
  
  size=116065589
  
  bit_rate=245995
  
  probe_score=51
  
  [/FORMAT]

  ffmpeg -nostdin -i input.aac -c:a libopus output.opus
  
  ffmpeg version N-93449-g013f714 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04)
  
    configuration: --prefix=/home/vagrant/ffmpeg_build --pkg-config-flags=--static --extra-cflags=-I/home/vagrant/ffmpeg_build/include --extra-ldflags=-L/home/vagrant/ffmpeg_build/lib --extra-libs='-lpthread -lm' --bindir=/home/vagrant/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-nonfree
  
    libavutil      56. 26.100 / 56. 26.100
  
    libavcodec     58. 47.105 / 58. 47.105
  
    libavformat    58. 26.101 / 58. 26.101
  
    libavdevice    58.  7.100 / 58.  7.100
  
    libavfilter     7. 48.100 /  7. 48.100
  
    libswscale      5.  4.100 /  5.  4.100
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100
  
  [aac @ 0x55d4b7e21d80] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, aac, from 'input.aac':
  
    Duration: 01:02:54.57, bitrate: 245 kb/s
  
      Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 245 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (aac (native) -> opus (libopus))
  
  [libopus @ 0x55d4b7e3f8c0] No bit rate set. Defaulting to 96000 bps.
  
  Output #0, opus, to 'output.opus':
  
    Metadata:
  
      encoder         : Lavf58.26.101
  
      Stream #0:0: Audio: opus (libopus), 48000 Hz, stereo, flt, 96 kb/s
  
      Metadata:
  
        encoder         : Lavc58.47.105 libopus
  
  size=   52103kB time=00:59:53.21 bitrate= 118.8kbits/s speed=66.2x
  
  video:0kB audio:51733kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.715930%

  ffprobe -v error -show_format -show_streams output.opus
  
  [STREAM]
  
  index=0
  
  codec_name=opus
  
  codec_long_name=Opus (Opus Interactive Audio Codec)
  
  profile=unknown
  
  codec_type=audio
  
  codec_time_base=1/48000
  
  codec_tag_string=[0][0][0][0]
  
  codec_tag=0x0000
  
  sample_fmt=fltp
  
  sample_rate=48000
  
  channels=2
  
  channel_layout=stereo
  
  bits_per_sample=0
  
  id=N/A
  
  r_frame_rate=0/0
  
  avg_frame_rate=0/0
  
  time_base=1/48000
  
  start_pts=0
  
  start_time=0.000000
  
  duration_ts=172473677
  
  duration=3593.201604
  
  bit_rate=N/A
  
  max_bit_rate=N/A
  
  bits_per_raw_sample=N/A
  
  nb_frames=N/A
  
  nb_read_frames=N/A
  
  nb_read_packets=N/A
  
  DISPOSITION:default=0
  
  DISPOSITION:dub=0
  
  DISPOSITION:original=0
  
  DISPOSITION:comment=0
  
  DISPOSITION:lyrics=0
  
  DISPOSITION:karaoke=0
  
  DISPOSITION:forced=0
  
  DISPOSITION:hearing_impaired=0
  
  DISPOSITION:visual_impaired=0
  
  DISPOSITION:clean_effects=0
  
  DISPOSITION:attached_pic=0
  
  DISPOSITION:timed_thumbnails=0
  
  TAG:ENCODER=Lavc58.47.105 libopus
  
  [/STREAM]
  
  [FORMAT]
  
  filename=output.opus
  
  nb_streams=1
  
  nb_programs=0
  
  format_name=ogg
  
  format_long_name=Ogg
  
  start_time=0.000000
  
  duration=3593.201604
  
  size=53353867
  
  bit_rate=118788
  
  probe_score=100
  
  [/FORMAT]

  How can I preserve the quality of the input file ? Am I missing something in the ffmpeg cmd ?

• How to run the ffmpeg command using node.js ?

  12 juillet 2019, par Sachin Shah

  In node, When I got the request from /playMovie from App, I need to broadcast the video.

  When I execute this command in terminal it works fine.

  ffmpeg -re -i movie.mkv  -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f flv rtmp://192.168.1.13/live/myStream

  Now I’m going to setup this dynamic.

  app.use('/playMovie', function (req, res) {
  
    console.log("playMovie...");
  
    let filePaht = 'movie.mkv';
  
    let fileName = 'marvel-avengers';
  
  
  
    let ffmpeg = spawn(`ffmpeg -re -i ${filePaht}  -c:v libx264 -preset 
  
      superfast -tune zerolatency -c:a aac -ar 44100 -f flv 
  
  rtmp://192.168.1.13/live/${fileName}`);
  
       ffmpeg.on('exit', (statusCode) => {
  
     console.log("statusCode ::::::::::::::::::::::::::::::::: ",statusCode);
  
     if (statusCode === 0) {
  
        console.log('conversion successful')
  
     }
  
   })
  
  
  
  ffmpeg
  
    .stderr
  
    .on('data', (err) => {
  
      console.log('err:', new String(err))
  
    })
  
  }); 

  Refrence Link

  While run the app I got this error.

  playMovie...
  
  12/07/2019 15:27:17 31722 [ERROR] uncaughtException { Error: spawn ffmpeg -re -i movie.mkv  -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f flv rtmp://192.168.1.13/live/marvel-avengers ENOENT
  
  at _errnoException (util.js:1022:11)
  
  at Process.ChildProcess._handle.onexit (internal/child_process.js:190:19)
  
  at onErrorNT (internal/child_process.js:372:16)
  
  at _combinedTickCallback (internal/process/next_tick.js:138:11)
  
  at process._tickCallback (internal/process/next_tick.js:180:9)
  
  code: 'ENOENT',
  
  errno: 'ENOENT',
  
  syscall: 'spawn ffmpeg -re -i movie.mkv  -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f flv 
  
  rtmp://192.168.1.13/live/marvel-avengers',
  
  path: 'ffmpeg -re -i movie.mkv  -c:v libx264 -preset superfast -tune zerolatency -c:a aac -ar 44100 -f flv rtmp://192.168.1.13/live/marvel-avengers',
  
  spawnargs: [] }