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• HLS encoded video (ffmpeg) breaks in Safari when segmented via byte-range

  15 juin 2019, par eschie

  Some of my HLS encoded videos via ffmpeg drop the audio when seeking past the buffer. The only way to sync the audio and video up again is to restart the video. What would be causing this ?

  Example Profile :

  bitrate: 4800, profile: 'high', level: '4.1', resolution: 1080, framerate: '24000/1001'

  ffmpeg
  
      '-y' 
  
      '-i' input_file.mov                                                 
  
      '-v' error 
  
      '-map' '0:0'                                                        
  
      '-c:v' libx264                                                  
  
      '-x264opts' f'
  
          keyint=23:                  
  
          min-keyint=23:          
  
          no-scenecut                                                     
  
      '
  
      '-vf' f'scale=-1:1080'
  
      '-preset' 'slow'
  
      '-profile:v' 'high'
  
      '-level' '4.1'
  
      '-b:v' '4800k'
  
      '-maxrate' '4800k'
  
      '-movflags' 'faststart'
  
      '-bufsize' '9600k' 
  
      '-write_tmcd', '0'
  
      '-r' '24000/1001'                                   
  
      output_dir                                                      

  Segmentation CMD :

  FFMPEG 
  
      '-i' output_dir 
  
      '-v' 'error' 
  
      '-acodec' 'copy'
  
      '-vcodec' 'copy'
  
      '-hls_time' '4' #seconds
  
      '-hls_list_size' '0'
  
      '-hls_flags' 'single_file' 
  
      os.path.join(output_dir, f'{run_id}_{bitrate}.m3u8'

  Added : apple’s mediastreamvalidator outputs a few different errors :

  Error: Playlist vs segment duration mismatch
  
  --> Detail:  Segment duration 98.0146, Playlist duration: 4.7968
  
  --> Source:  1559962503399_2200k.m3u8 - 1559962503399_2200k.ts:1746520@0

  Error: Measured peak bitrate compared to master playlist declared value exceeds error tolerance
  
  --> Detail:  Measured: 3182.61 kb/s, Master playlist: 2173.82 kb/s, Error: 46.41%, Combined rendition name: English
  
  --> Source:  ...playlist.m3u8
  
  --> Compare: 1559962503399_2200k.m3u8

  Error: Different target durations detected
  
  --> Detail:  Target duration: 5 vs Target duration: 4
  
  --> Source:  1559962503399_64k.m3u8
  
  --> Compare: 1559962503399_128k.m3u8

  UPDATE 1 :
  I modified the encoding command to utilize tee pseudo-muxer, but it seems the same issue still exists when segmenting HLS as byte-ranges rather than in separate .ts files :

  $ ffmpeg
  
  -hide_banner
  
  -report
  
  -benchmark
  
  -vstats
  
  -i "../Jane_shallowing_Top_v08.mp4"
  
  -dn
  
  -sn
  
  -filter_complex "[0:v]fps=fps=24.000,setpts=(PTS-STARTPTS),split=[vsplit1][vsplit2];[vsplit1]scale=-1:144[video_144];[vsplit2]scale=-1:1080[video_1080]"
  
  -map "[video_144]"
  
  -r:v:0 "24.000"
  
  -c:v:0 "libx264"
  
  -x264-params "keyint=144:min-keyint=144:scenecut=0:open_gop=0"
  
  -preset:v:0 "slow"
  
  -profile:v:0 "baseline"
  
  -refs:v:0 "2"
  
  -b-pyramid:v:0 "strict"
  
  -tune:v:0 "film"
  
  -b:v:0 "96000"
  
  -maxrate:v:0 "56000"
  
  -bufsize:v:0 "6*56000/8"
  
  -vsync:v:0 "cfr"
  
  -bsf:v:0 "h264_metadata=fixed_frame_rate_flag=1"
  
  -map "[video_1080]"
  
  -r:v:1 "24.000"
  
  -c:v:1 "libx264"
  
  -x264-params "keyint=144:min-keyint=144:scenecut=0:open_gop=0"
  
  -preset:v:1 "slow"
  
  -profile:v:1 "high"
  
  -refs:v:1 "2"
  
  -b-pyramid:v:1 "strict"
  
  -tune:v:1 "film"
  
  -b:v:1 "4800000"
  
  -maxrate:v:1 "4800000"
  
  -bufsize:v:1 "6*4800000/8"
  
  -vsync:v:1 "cfr"
  
  -bsf:v:1 "h264_metadata=fixed_frame_rate_flag=1"
  
  -map a:0 -map a:0
  
  -c:a "libfdk_aac"
  
  -ar "48000"
  
  -ab "128k"
  
  -af "aresample=async=1:min_hard_comp=0.100000:first_pts=0"
  
  -f "hls"
  
  -var_stream_map "v:1,a:0 v:0,a:1"
  
  -hls_time "6.000"
  
  -hls_segment_type "mpegts"
  
  -hls_flags "discont_start+temp_file+single_file"
  
  -hls_list_size "0"
  
  -master_pl_name "playlist.m3u8"
  
  -hls_segment_filename "out_%v.ts" "out_%v.m3u8" 

  Segment duration 98.0267, Playlist duration: 6.0000

• How to detect added scenes in Director's Cut Edition of movie ?

  17 juillet 2019, par B Lolo

  I’ve got huge collection of .mkv files. In every movie folder there are two files : one original movie and one extended edition of that movie.

  Two files can be different for example one is 480p and the second 1080p.

  One can be 1 hour long, second one 40 minutes.

  In those extended edition files scenes are added randomly, so it could be 2 minutes in beginning, 5 minutes after first 10 minutes of the film and so on.

  Is there a way to use Python and ffmpeg to detect scenes from extended edition files that are absent in original movie file ?

  I can also work with only audio if that is easier to do.

  For now I got idea to use ffmpeg and scene detection, I can manually search for differences between files, but I would like some hints where to look changes.

  This is python code with ffmpy library :

  from ffmpy import FFmpeg
  
  
  
  plik = "C:/special.mkv"
  
  png = re.sub("\.mkv","_changes.png",plik)
  
  
  
  ff = FFmpeg(executable='C:/ffmpeg.exe', global_options ='-v error', inputs={plik : ''}, outputs={png : "-vf select='gt(scene\,0.4)',scale=320:-1,tile=10x80 -frames:v 1 -y"})
  
  result = ff.run(stdout=PIPE, stderr=PIPE)

• Python : How to convert a mp3 chunk from webstream to .wav samples ?

  13 novembre 2019, par Bendzko

  I’m trying to catch chunks of an mp3 webstream and coverting them into wav samples for signal processing. I tried to catch the audio via requests and io.BytesIO to save the data as .wav file.
  But I think, that I have to convert the mp3 data to wav data, but I don’t know how. (My goal is not to record a .wav file, i am just doing this to test the algorithm.)

  I found the pymedia lib, but it is very old (last commit in 2006), using python 2.7 and for me not installable.

  Maybe it is possible with ffmpeg-python, but I have just seen examples using files as input and output.

  Here’s my code :

  import requests
  
  import io
  
  import soundfile as sf
  
  import struct
  
  import wave
  
  import numpy as np
  
  
  
  
  
  def main():
  
      stream_url = r'http://dg-wdr-http-dus-dtag-cdn.cast.addradio.de/wdr/1live/diggi/mp3/128/stream.mp3'
  
      r = requests.get(stream_url, stream=True)
  
      sample_array = []
  
      try:
  
          for block in r.iter_content(1024):
  
              data, samplerate = sf.read(io.BytesIO(block), format="RAW", channels=2, samplerate=44100, subtype='FLOAT',
  
                                         dtype='float32')
  
              sample_array = np.append(sample_array, data)
  
  
  
      except KeyboardInterrupt:
  
          print("...saving")
  
          obj = wave.open('sounds/stream1.wav', 'w')
  
          obj.setnchannels(1)  # mono
  
          obj.setsampwidth(2)  # bytes
  
          obj.setframerate(44100)
  
  
  
          data_max = np.nanmax(abs(sample_array))
  
  
  
          # fill WAV with samples from sample_array
  
          for sample in sample_array:
  
              if (np.isnan(sample) or np.isnan(32760 * sample / data_max)) is True:
  
                  continue
  
              try:
  
                  value = int(32760 * sample / data_max)  # normalization INT16
  
              except ValueError:
  
                  value = 1
  
              finally:
  
                  data = struct.pack('code>

  Do you have an idea how to handle this problem ?