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• How can a desktop node.js app play audio that may be controlled by the user like a media player ? [closed]

  19 février, par eedefeed

  I'm building a playlist manager that plays music. How can Node.JS play audio files quickly, reliably and with all the basic level features you would expect from a media player, namely :

  


  

  • play

  


  • pause

  


  • seek

  


  • stop

  


  • adjust volume

  


  


  I'm targetting windows/linux, but a Windows-only solution but be okay (for now.)

  


  I have tried a number of libraries and methods to play audio but it seems none of them are good enough :

  


  

  • Audic : it's reasonably good, but buggy. The play and pause functions sometimes get switched around. I also recall that there are some issues with uncaught exceptions somewhere in the dependencies that crash the entire app.

  


  • OBS : since the app is designed with broadcasting in mind, I've tried to use OBS's API to get it to play media. Unfortunately, it sometimes stops playback during some tracks, which is surprising since its underlying library, FFmpeg, plays them without issue.

  


  • node-groove : seems like its underlying library, libgroove, only supports linux. I can't find any builds to download, regardless.

  


  


  Attempts to use Speaker (which seems pretty good) have also failed because all the decoders have big issues :

  


  

  • Anything using lame - I want support for all audio, not specific formats.

  


  • fluent-FFmpeg - this is a wrapper around FFmpeg's CLI interface. It has no play/pause function, but bonus library fluent-FFmpeg-util adds this feature. Unfortunately, its pause takes about 4 seconds to work, which I'm guessing is to do with a buffer being exhausted. This is just too latent. Seek would also work by stopping the CLI process and reloading the file, which seems massively inefficient.

  


  • Node Vlc - promising but ancient library that gives me reams of node-gyp errors on install. Poorly documented and no explanation of what the library to do

  


  • VLC Client - this library has uncaught exceptions that crash the app. Wrapping in try/catch doesn't help.

  


  • sound play - doesn't support play/pause

  


  


  NPM's search function is filled with audio players designed to work in browsers, but I'm not building a web app. I guess it's an option but it seems inelegant to the point of rediculous.

  


  So it seems the best option centres around FFmpeg. FFmpeg has libraries, and I'm aware that node has ways to hook into those libraries via some sort of C or C++ compatibility layer. Unfortunately, official documentation is rather dense. Different unofficial guides seem to be recommending conflicting approaches (and might be dated), and it's difficult to work out whether myriad technologies are working in tandem or are alternatives, renames or replacements : node-gyp, node-api, addons, windows-build-tools, nan, C vs C++, Visual Studio. It's difficult to make any decisions or know where to start.

  


  Perhaps, also, another option is to use a Python library to interact with FFmpeg, since initial searches have indicated this might be possible. I wouldn't know whether this is a good option.

  


  So my question is : what's my best option to play audio ? Is it another NPM module that I'm not aware of ? Is it a compatibility layer with FFmpeg libraries ?

  


• webM files shows green and purple effects on mobile

  11 octobre 2015, par Naveen Gamage

  I have converted several GIFs to webM files using ffmpeg on my Ubuntu 14.04 server.

  Heres the code I used for conversation.

  ffmpeg -i your_gif.gif -c:v libvpx -crf 12 -b:v 500K output.webm

  source https://gist.github.com/ndarville/10010916

  The problem is converted webM files shows perfectly fine on PCs but on my mobile it shows with green and purple shadows.

  PC

  

  Mobile

  

  I tried changing -crf and -b:v values to their max but nothing happens.

  webM file : http://d1pnsuxwa0it39.cloudfront.net/uploads/comments/webm/4673555.webm

  edit :

  also I can see webM files on some other sites fine. I think this has to do something with the way I convert files.

  edit :

  I have tried another code I found on stackoverflow but still the same.

  ffmpeg -f gif -i infile.gif outfile.mp4

  EDIT :

  If anyone think this has something to do with the way I installed FFMPEG, I followed the steps on FFMPEG official docs.

  https://trac.ffmpeg.org/wiki/CompilationGuide/Ubuntu

  EDIT :

  Input file :

  http://d1pnsuxwa0it39.cloudfront.net/test/1.gif

  Output file :

  http://d1pnsuxwa0it39.cloudfront.net/test/output.webm

  FFMPEG CLI output

   /home/naveencg/bin/ffmpeg -i 1.gif -c:v libvpx -crf 12 -b:v 500K output.webm
  
  ffmpeg version 2.5.git Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Dec 31 2014 14:37:15 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
  
    configuration: --prefix=/home/naveencg/ffmpeg_build --extra-cflags=-I/home/naveencg/ffmpeg_build/include --extra-ldflags=-L/home/naveencg/ffmpeg_build/lib --bindir=/home/naveencg/bin --enable-gpl --enable-libass --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree
  
    libavutil      54. 15.100 / 54. 15.100
  
    libavcodec     56. 19.100 / 56. 19.100
  
    libavformat    56. 16.102 / 56. 16.102
  
    libavdevice    56.  3.100 / 56.  3.100
  
    libavfilter     5.  6.100 /  5.  6.100
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, gif, from '1.gif':
  
    Duration: N/A, bitrate: N/A
  
      Stream #0:0: Video: gif, bgra, 350x169, 25 fps, 25 tbr, 100 tbn, 100 tbc
  
  [libvpx @ 0x1e2bf60] v1.3.0
  
  Output #0, webm, to 'output.webm':
  
    Metadata:
  
      encoder         : Lavf56.16.102
  
      Stream #0:0: Video: vp8 (libvpx), yuva420p, 350x169, q=-1--1, 500 kb/s, 25 fps, 1k tbn, 25 tbc
  
      Metadata:
  
        encoder         : Lavc56.19.100 libvpx
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (gif (native) -> vp8 (libvpx))
  
  Press [q] to stop, [?] for help
  
  frame=   21 fps=0.0 q=0.0 size=      58kB time=00:00:00.84 bitrate= 569.7kbits/sframe=   44 fps= 41 q=0.0 size=     110kB time=00:00:01.76 bitrate= 512.4kbits/sframe=   62 fps= 39 q=0.0 size=     153kB time=00:00:02.48 bitrate= 505.9kbits/sframe=   84 fps= 40 q=0.0 size=     210kB time=00:00:03.36 bitrate= 510.8kbits/sframe=   88 fps= 41 q=0.0 Lsize=     218kB time=00:00:03.52 bitrate= 508.3kbits/s
  
  video:216kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.971527%

• Is FFmpegAudioDecoder supposed to reinitialize upon append of new init segment

  14 novembre 2023, par martin

  I am attempting to switch audio tracks but when switching the FFmpegAudioDecoder never reinitializes like it does with video tracks of differing resolutions. I am not certain if this is the intended behavior of FFmpegAudioDecoder and would love to learn more about the expected behavior.

  


  When switching audio tracks I end up calling the following operations :

  


  if sourceBuffer.getIsUpdate() {sourceBuffer.abort()}
sourceBuffer.remove(0-videoDuration)
initSegmentDataStream = fetch init segment of new audio representation
sourceBuffer.appendBuffer(initSegmentDataStream)


  


  These are the Media tab messages from initial video load

  


  ChunkDemuxer
Selected FFmpegAudioDecoder for audio decoding, config: codec: aac, profile: unknown, bytes_per_channel: 2, channel_layout: STEREO, channels: 2, samples_per_second: 48000, sample_format: Signed 16-bit, bytes_per_frame: 4, seek_preroll: 0us, codec_delay: 0, has extra data: false, encryption scheme: Unencrypted, discard decoder delay: false, target_output_channel_layout: STEREO, target_output_sample_format: Unknown sample format, has aac extra data: true
Cannot select DecryptingVideoDecoder for video decoding
Cannot select VDAVideoDecoder for video decoding
Cannot select VpxVideoDecoder for video decoding
Selected Dav1dVideoDecoder for video decoding, config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1280,720], visible rect: [0,0,1280,720], natural size: [1280,720], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}
Dropping audio frame (DTS 0us PTS -105375us,-62709us) that is outside append window [0us,9223372036854775807us).
Dropping audio frame (DTS 42666us PTS -62708us,-20042us) that is outside append window [0us,9223372036854775807us).
Truncating audio buffer which overlaps append window start. PTS -20041us frame_end_timestamp 22625us append_window_start 0us
Effective playback rate changed from 0 to 1


  


  For comparison this is what I get when appending the init segment of a different video resolution / track

  


  video decoder config changed midstream, new config: codec: av1, profile: av1 profile main, level: not available, alpha_mode: is_opaque, coded size: [1920,1080], visible rect: [0,0,1920,1080], natural size: [1920,1080], has extra data: false, encryption scheme: Unencrypted, rotation: 0°, flipped: 0, color space: {primaries:BT709, transfer:BT709, matrix:BT709, range:LIMITED}



  


  Chrome version : Version 119.0.6045.123 (Official Build)

  


  When appending the new init segment of an audio track I was expecting the FFmpegAudioDecoder to be reinitialized like the Dav1dVideoDecoder does for video tracks