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• Unhandled Rejection (Error) : ffmpeg.FS('readFile', [fileName]) error. Check if the path exists

  20 février 2021, par Rayhan Memon

  I'm learning how to use FFmpeg.wasm (the JS port of ffmpeg) following the instructions of this example, but adapting it for editing audio files.

  


  I've created a function called 'convertToWAV' (see below) that is called when the 'convert' button is pressed after an mp3 file is uploaded. The function writes the file to FFmpeg's memory as 'test.mp3', converts the file to a wav file named 'test.wav', reads the file, and then creates a URL for the 'test.wav' that can be read by the browser.

  


  I end up getting this error a few seconds after the convert button is pressed :


  


  I don't quite understand the error and the ffmpeg.wasm documentation doesn't indicate that I should be doing things any differently...

  


  Here is the function 'convertToWAV' :

  


     const convertToWAV = async () => {
    //Write the file to FFmpeg's memory as test.mp3
    //It will only stay in memory until the browser is refreshed
    ffmpeg.FS('writeFile', 'test.mp3', await fetchFile(audioFile));

    //Run ffmpeg command
    ffmpeg.run('-i', 'test.mp3', 'test.wav');

    //Read the resulting file
    const data = ffmpeg.FS('readFile', 'test.wav');

    //Create a URL that can be used by the browser
    const url = URL.createObjectURL(new Blob([data.buffer]), { type: 'audio/wav' });

    //set the outputFile state to the url
    setOutputFile(url);
    }


  


  Thank you in advance for the help !

  


• Streaming UDP packets to two different ports (for video and audio). Video works fine, but the audio does not show

  8 avril 2018, par Winston Chen

  I am taking a rtsp stream, split the video and audio out, and stream them to two different ports respectively using gstreamer so that my ffserver would be able to display the stream on my browser.

  My gstreamer pipeline :

  gst-launch-1.0 -v rtspsrc location=rtsp://wowzaec2demo.streamlock.net/vod/mp4:BigBuckBunny_115k.mov latency=300 timeout=0 drop-on-latency=true rtp-blocksize=4096 name=rtsp_source ! \
  
    queue ! capsfilter caps="application/x-rtp,media=video" ! rtph264depay ! h264parse ! rtph264pay config-interval=1 pt=96 ! udpsink host=127.0.0.1 port=9527 rtsp_source. ! \
  
    queue ! rtpmp4apay pt=97 ! udpsink host=127.0.0.1 port=9327

  Here comes the sdp and my ffmpeg commnad :

  m=video 9527 RTP/AVP 96
  
  a=rtpmap:96 H264/90000
  
  c=IN IP4 127.0.0.1
  
  
  
  m=audio 9327 RTP/AVP 97
  
  a=rtpmap:97 mpeg4-generic/48000/6
  
  c=IN IP4 127.0.0.1

  ffmpeg -protocol_whitelist "file,tcp,rtp,udp" -i ~/test.sdp -max_muxing_queue_size 1024 http://localhost:8090/feed1.ffm

  And finally, this is my ffserver config (the important part) :

  <feed>               # This is the input feed where FFmpeg will send
  
     File ./feed1.ffm            # video stream.
  
     FileMaxSize 1GB             # Maximum file size for buffering video
  
     ACL allow 127.0.0.1         # Allowed IPs
  
  </feed>
  
  
  
  <stream>              # Output stream URL definition
  
     Feed feed1.ffm              # Feed from which to receive video
  
     Format webm
  
  
  
     # NoDefaults
  
     # NoAudio
  
  
  
     # Audio settings
  
     AudioCodec vorbis
  
     AudioBitRate 64             # Audio bitrate
  
  
  
     # Video settings
  
     VideoCodec libvpx
  
     VideoSize 240x160           # Video resolution
  
     VideoFrameRate 10           # Video FPS
  
     AVOptionVideo flags +global_header  # Parameters passed to encoder
  
                                         # (same as ffmpeg command-line parameters)
  
  
  
     PreRoll 0
  
     StartSendOnKey
  
     VideoGopSize 12
  
     VideoBitRate 256
  
  </stream>

  The thing is that if I take away the audio part and apply NoAudio, the video streams fine. However, I could not get the audio to work. Am I doing anything wrong ?

• FFmpeg : Splitting video into segments of very small size (e.g. 5KB)

  3 avril 2019, par user528827

  I am trying to split a video file into segments that are less than 10KB (ideally 5KB) using FFmpeg.

  The video I am using is this one : https://download.blender.org/peach/bigbuckbunny_movies/BigBuckBunny_320x180.mp4. It already has a low resolution.

  I have tried specifying a very small -segment_time of 00:00:0.04 :

  ffmpeg -i BigBuckBunny_320x180.mp4 -c copy -map 0 -f segment -segment_time 00:00:0.04 BigBuckBunny_320x180_%03d.mp4

  I have tried specifying sequential -segment_frames e.g. 1,2,3,4,5,6,7... etc. :

  ffmpeg -i BigBuckBunny_320x180.mp4 -c copy -map 0 -f segment -segment_frames 1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30 -segment_list out.csv BigBuckBunny_320x180_%03d.mp4

  I have also tried to force key frames :

  ffmpeg -i BigBuckBunny_320x180.mp4 -force_key_frames "expr:gte(t,n_forced*0.04)" -c copy -map 0 -f segment -segment_frames 1,2,3,4,5,6,7,8,9,10,11,12,13,14,15,16,17,18,19,20,21,22,23,24,25,26,27,28,29,30 BigBuckBunny_320x180_%03d.mp4

  All of these result in segments 0.5s long with varying file sizes averaging about 50KB.

  username@My-Air:~/dev/video$ less out.csv 
  
  BigBuckBunny_320x180_000.mp4,0.000000,0.500000
  
  BigBuckBunny_320x180_001.mp4,0.500000,1.000000
  
  BigBuckBunny_320x180_002.mp4,1.000000,1.500000
  
  BigBuckBunny_320x180_003.mp4,1.500000,2.000000
  
  BigBuckBunny_320x180_004.mp4,2.000000,2.500000
  
  BigBuckBunny_320x180_005.mp4,2.500000,3.000000
  
  ...
  
  
  
  username@My-Air:~/dev/video/bunny_0.5s_splits$ ls -l
  
  total 134552
  
  -rw-r--r--  1 username  staff    16K 25 Mar 23:40 BigBuckBunny_320x180_000.mp4
  
  -rw-r--r--  1 username  staff    26K 25 Mar 23:40 BigBuckBunny_320x180_001.mp4
  
  -rw-r--r--  1 username  staff    36K 25 Mar 23:40 BigBuckBunny_320x180_002.mp4
  
  -rw-r--r--  1 username  staff    44K 25 Mar 23:40 BigBuckBunny_320x180_003.mp4
  
  -rw-r--r--  1 username  staff    40K 25 Mar 23:40 BigBuckBunny_320x180_004.mp4
  
  -rw-r--r--  1 username  staff    38K 25 Mar 23:40 BigBuckBunny_320x180_005.mp4
  
  -rw-r--r--  1 username  staff    43K 25 Mar 23:40 BigBuckBunny_320x180_006.mp4
  
  -rw-r--r--  1 username  staff    44K 25 Mar 23:40 BigBuckBunny_320x180_007.mp4
  
  -rw-r--r--  1 username  staff    50K 25 Mar 23:40 BigBuckBunny_320x180_008.mp4
  
  -rw-r--r--  1 username  staff    49K 25 Mar 23:40 BigBuckBunny_320x180_009.mp4
  
  -rw-r--r--  1 username  staff    50K 25 Mar 23:40 BigBuckBunny_320x180_010.mp4
  
  -rw-r--r--  1 username  staff    51K 25 Mar 23:40 BigBuckBunny_320x180_011.mp4
  
  -rw-r--r--  1 username  staff    51K 25 Mar 23:40 BigBuckBunny_320x180_012.mp4
  
  -rw-r--r--  1 username  staff    57K 25 Mar 23:40 BigBuckBunny_320x180_013.mp4
  
  -rw-r--r--  1 username  staff    53K 25 Mar 23:40 BigBuckBunny_320x180_014.mp4
  
  -rw-r--r--  1 username  staff    57K 25 Mar 23:40 BigBuckBunny_320x180_015.mp4
  
  -rw-r--r--  1 username  staff    57K 25 Mar 23:40 BigBuckBunny_320x180_016.mp4
  
  -rw-r--r--  1 username  staff    61K 25 Mar 23:40 BigBuckBunny_320x180_017.mp4
  
  ...

  It’s strange that the timestamps are at exactly 0s and 0.5s. Is this limitation of FFmpeg ? I couldn’t find any documentation to verify this.

  Is there any way to get these file sizes down to 5KB ? Is it possible with FFmpeg ? If not, is there another tool I could use ? I looked at GStreamer but couldn’t really see if it was possible and it seems very complicated.

  Any help appreciated. Thanks.