Recherche avancée

Sur d’autres sites (7844)

• How to append fMP4 chunks to SourceBuffer ?

  24 octobre 2020, par Stefan Falk

  I have finally managed to create an fMP4 but now I am not able to seek or play the file depending on what I do in the file.

  


  On my backend I am taking the file and convert it to MP4 or fragmented MP4.

  


  The file gets send to the clients chunk-wise but this approach does not seem to work as it used to work on Chrome (bot not on Firefox) when using MP3.

  


  How I played MP3

  


  Say we have a 10 seconds track that is 1 MB in size which I want to start playing from second five. I want to load chunks of 1 second.

  


  Thus, I have offset = 5 / 10 * file_size and chunkSize = 1 / 10 * file_size`.

  


  With this I just started loading the MP3-file at an offset of 0.5 MB and loaded the chunks as needed where each chunk was 0.1 MB in size.

  


  This worked because before actually playing the file, I loaded the first bytes of the file and appended it to the SourceBuffer as well s.t. it was able to load the meta-information of the file. However, this approach is just not working for fMP4.

  


  What I tried with fMP4

  


  So, I have been converting MP3 to fMP4 with the MP3-approach ..

  


  .. using +dash (can play but not seek)

  


  ffmpeg -i input.mp3 -acodec aac -b:a 256k -f mp4 -movflags +dash output.mp4


  


  .. using frag_keyframe+empty_moov (cannot play on Chrome)

  


  ffmpeg -i input.mp3 -acodec aac -b:a 256k -f mp4 -movflags frag_keyframe+empty_moov output.mp4


  


  On the client the chunks get appended to a SourceBuffer (as explained above) after creating it with the Mime-Type audio/mp4; codecs="mp4a.40.2" :

  


  this.sourceBuffer = this.mediaSource
                        .addSourceBuffer('audio/mp4; codecs="mp4a.40.2"');


  


  and

  


  private appendSegment = (chunk) => {
  try {
    this.sourceBuffer.appendBuffer(chunk);
  } catch {
    return;
  }
}


  


  The problem is that I can only play the +dash converted file if I start reading it from the start and continue adding chunks.

  


  However, if I start reading the file from further down, the audio gets never played.

  


  playTrack(track, 0.0);  // Start at second 0 works
playTrack(track, 10.0); // Start at second 10 does not work


  


• Why doesn't FFmpeg work when using yt-dlp in python script ?

  11 mai 2022, par spelle

  I'm trying to download a video using yt-dlp in python.

  


  ydl_opts = {'format': 'bv+ba/b'}
with YoutubeDL(ydl_opts) as ydl:
     ydl.download('https://www.reddit.com/r/cats/comments/re37dn/weve_been_feeding_this_stray_for_several_years/')


  


  But I'm reaching an FFmpeg error in the log

  


  [generic] 1o8t9ollwx481: Requesting header
[redirect] Following redirect to https://www.reddit.com/r/cats/comments/re37dn/weve_been_feeding_this_stray_for_several_years/
[Reddit] re37dn: Downloading JSON metadata
[Reddit] re37dn: Downloading m3u8 information
[Reddit] re37dn: Downloading MPD manifest
[info] 1o8t9ollwx481: Downloading 1 format(s): dash-video_4419291+dash-audio_0_133951
WARNING: You have requested merging of multiple formats but ffmpeg is not installed. The formats won't be merged.
[download] Destination: We’ve been feeding this stray for several years, but she’s lost a lot of weight and I don’t think she would last outside for another winter, so I brought her in. [1o8t9ollwx481].fdash-video_4419291.mp4
[download] 100% of 5.18MiB in 00:00               
[download] Destination: We’ve been feeding this stray for several years, but she’s lost a lot of weight and I don’t think she would last outside for another winter, so I brought her in. [1o8t9ollwx481].fdash-audio_0_133951.m4a
[download] 100% of 161.32KiB in 00:00


  


  FFmpeg is installed through pip and added in PATH.

  


• How to stop ffmpeg when there's no incoming rtmp stream

  5 juillet 2016, par M. Irich

  I use ffmpeg together with nginx-rtmp.
  The thing is ffmpeg doesn’t finish the process when the stream’s finished

  I use the following command :

  ffmpeg  -i 'rtmp://localhost:443/live/test' -loglevel debug  -c:a libfdk_aac -b:a 192k -c:v libx264 -profile baseline -preset superfast -tune zerolatency -b:v 2500k -maxrate 4500k -minrate 1500k -bufsize 9000k -keyint_min 15 -g 15 -f dash -use_timeline 1 -use_template 1 -min_seg_duration 5000 -y /tmp/dash/test/test.mpd

  but even the stream’s not running ffmpeg still can’t finish the process and is waiting for the rtmp stream

  Successfully parsed a group of options.
  
  Opening an input file: rtmp://localhost:443/live/test.
  
  [rtmp @ 0x2ba2160] No default whitelist set
  
  [tcp @ 0x2ba2720] No default whitelist set
  
  [rtmp @ 0x2ba2160] Handshaking...
  
  [rtmp @ 0x2ba2160] Type answer 3
  
  [rtmp @ 0x2ba2160] Server version 13.14.10.13
  
  [rtmp @ 0x2ba2160] Proto = rtmp, path = /live/test, app = live, fname = test
  
  [rtmp @ 0x2ba2160] Server bandwidth = 5000000
  
  [rtmp @ 0x2ba2160] Client bandwidth = 5000000
  
  [rtmp @ 0x2ba2160] New incoming chunk size = 4096
  
  [rtmp @ 0x2ba2160] Creating stream...
  
  [rtmp @ 0x2ba2160] Sending play command for 'test'

  Is it possible to limit the latency time to several seconds ?

  Sorry for any possible mistakes - English’s not my native language.