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• Révision 21750 : Un bug serieux sur la sauvegarde : lors de l’insertion on ne comptait pas sur la ...

  30 octobre 2014, par cedric -

  Pouvait provoquer une erreur avec abandon qui manquait de log et n’etait pas signale dans le status. On corrige donc les consequences d’une telle erreur.

• FFserver error : sample rate not set

  12 avril 2015, par Mese

  I’m trying to stream video with FFmpeg over FFserver to a webbrowser.
  FFmpeg is running on Windows and FFserver on Ubuntu. The streaming from FFmpeg to FFserver is working, but I get the following error when accessing the stream from a webbrowser :

  [webm @ 0x1da3c90]sample rate not set
  
  Error writing output header for stream 'test.webm': Invalid argument
  
  192.168.2.3 - - [GET] "/test.webm HTTP/1.1" 200 63
  
  [webm @ 0x1da4fd0]sample rate not set
  
  Error writing output header for stream 'test.webm': Invalid argument
  
  192.168.2.3 - - [GET] "/test.webm HTTP/1.1" 200 63

  I’m running FFmpeg with this command :

  ffmpeg -f gdigrab -video_size 800x600 -i desktop -f lavfi -i aevalsrc=0:1 http://192.168.2.5:8090/feed1.ffm

  My FFserver config file :

  Port 8090                      
  
  BindAddress 0.0.0.0
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 10000
  
  
  
  CustomLog -                  
  
  
  
  <feed>               
  
     File ./feed1.ffm            
  
     FileMaxSize 1G              
  
     #ACL allow 127.0.0.1         
  
  </feed>
  
  
  
  <stream>              
  
     Feed feed1.ffm             
  
     Format webm
  
  
  
     AudioCodec vorbis
  
     AudioBitRate 64
  
     AudioSampleRate 48000
  
     VideoCodec libvpx
  
     VideoSize 720x576          
  
     VideoFrameRate 25          
  
     AVOptionVideo flags +global_header  
  
     AVOptionVideo cpu-used 0
  
     AVOptionVideo qmin 10
  
     AVOptionVideo qmax 42
  
     AVOptionVideo quality good
  
     AVOptionAudio flags +global_header
  
     PreRoll 15
  
     StartSendOnKey
  
     VideoBitRate 400            
  
  </stream>
  
  
  
  <stream>            
  
     Format status
  
     # Only allow local people to get the status
  
     ACL allow localhost
  
     ACL allow 192.168.0.0 192.168.255.255
  
  </stream>
  
  
  
  <redirect>
  
     # Redirect index.html to the appropriate site
  
     URL http://www.ffmpeg.org/
  
  </redirect>

  Is there a solution for this problem ?

  Any help is appreciated

• 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