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• hls.js starting a beginning with ANDROID mobile (chrome, webview also) and not live *** but works very nice in deskto, ios .. hls.js 1.0.0 2021-04-01

  27 avril 2021, par Jintor

  I'm streaming a .m3u8 with the latest hls.js 1.0.0 (not rc) but version of 2021-04-01...

  


  example : the stream began at 5pm, and now it's 5:15 pm...

  


  the stream start at live point in almost all browsers

  


  The pattern I see here : ALL browsers in android (tested in Android 10) won't start at live point, only at 0...

  


  I did all the tests

  


  • Safari desktop => stream live at 5:15

  


  • Safari mobile => stream live at 5:15

  


  • WebView (Android) => ••• ISSUE : the player starts the stream at 0 (5pm)

  


  • WKWebView (apple IOS iphone,ipad) => stream live at 5:15

  


  • Chrome Desktop (mac/win) => stream live at 5:15

  


  • Chrome MOBILE (Android) => ••• ISSUE : the player starts the stream at 0 (5pm)

  


  • Chrome MOBILE (iPhone) => stream live at 5:15

  


  • Microsoft EDGE Desktop => stream live at 5:15

  


  • Microsoft EDGE mobile (android) => ••• ISSUE : the player starts the stream at 0 (5pm)

  


  • Firefox Desktop (mac/win) => stream live at 5:15

  


  • Opera Desktop (mac/win) => stream live at 5:15

  


  • Opera Mini (iPhone) => stream live at 5:15

  


  • Opera Mini (android) => ••• ISSUE : the player starts the stream at 0 (5pm)

  


  • Brave Desktop (mac/win) => stream live at 5:15

  


  • Brave Mobile (iPhone) => stream live at 5:15

  


  • Brave Mobile (android) => ••• ISSUE : the player starts the stream at 0 (5pm)

  


  This code

  


  <code class="echappe-js">&lt;script src=&quot;https://cdn.jsdelivr.net/npm/hls.js@latest&quot;&gt;&lt;/script&gt;&#xA;    
  &#xA;    &lt;script&gt;&amp;#xA;      var video = document.getElementById(&quot;video&quot;);&amp;#xA;      var videoSrc = &quot;https://www.example1.com/streaming/index.m3u8&quot;;&amp;#xA;      if (video.canPlayType(&quot;application/vnd.apple.mpegurl&quot;)) {&amp;#xA;        video.src = videoSrc;&amp;#xA;      } else if (Hls.isSupported()) {&amp;#xA;         var config = {&amp;#xA;            autoStartLoad: true,&amp;#xA;            startPosition: -1,&amp;#xA;            debug: false,&amp;#xA;            capLevelOnFPSDrop: false,&amp;#xA;            capLevelToPlayerSize: false,&amp;#xA;            defaultAudioCodec: undefined,&amp;#xA;            initialLiveManifestSize: 1,&amp;#xA;            maxBufferLength: 30,&amp;#xA;            maxMaxBufferLength: 500,&amp;#xA;            backBufferLength: Infinity,&amp;#xA;            maxBufferSize: 60 * 1000 * 1000,&amp;#xA;            maxBufferHole: 0.5,&amp;#xA;            highBufferWatchdogPeriod: 2,&amp;#xA;            nudgeOffset: 0.1,&amp;#xA;            nudgeMaxRetry: 3,&amp;#xA;            maxFragLookUpTolerance: 0.25,&amp;#xA;            liveSyncDurationCount: 3,&amp;#xA;            liveMaxLatencyDurationCount: Infinity,&amp;#xA;            liveDurationInfinity: false,&amp;#xA;            enableWorker: true,&amp;#xA;            enableSoftwareAES: true,&amp;#xA;            manifestLoadingTimeOut: 10000,&amp;#xA;            manifestLoadingMaxRetry: 1,&amp;#xA;            manifestLoadingRetryDelay: 1000,&amp;#xA;            manifestLoadingMaxRetryTimeout: 64000,&amp;#xA;            startLevel: undefined,&amp;#xA;            levelLoadingTimeOut: 10000,&amp;#xA;            levelLoadingMaxRetry: 4,&amp;#xA;            levelLoadingRetryDelay: 1000,&amp;#xA;            levelLoadingMaxRetryTimeout: 64000,&amp;#xA;            fragLoadingTimeOut: 20000,&amp;#xA;            fragLoadingMaxRetry: 6,&amp;#xA;            fragLoadingRetryDelay: 1000,&amp;#xA;            fragLoadingMaxRetryTimeout: 64000,&amp;#xA;            startFragPrefetch: false,&amp;#xA;            testBandwidth: true,&amp;#xA;            progressive: false,&amp;#xA;            lowLatencyMode: true,&amp;#xA;            fpsDroppedMonitoringPeriod: 5000,&amp;#xA;            fpsDroppedMonitoringThreshold: 0.2,&amp;#xA;            appendErrorMaxRetry: 3,&amp;#xA;            enableWebVTT: true,&amp;#xA;            enableIMSC1: true,&amp;#xA;            enableCEA708Captions: true,&amp;#xA;            stretchShortVideoTrack: false,&amp;#xA;            maxAudioFramesDrift: 1,&amp;#xA;            forceKeyFrameOnDiscontinuity: true,&amp;#xA;            abrEwmaFastLive: 3.0,&amp;#xA;            abrEwmaSlowLive: 9.0,&amp;#xA;            abrEwmaFastVoD: 3.0,&amp;#xA;            abrEwmaSlowVoD: 9.0,&amp;#xA;            abrEwmaDefaultEstimate: 500000,&amp;#xA;            abrBandWidthFactor: 0.95,&amp;#xA;            abrBandWidthUpFactor: 0.7,&amp;#xA;            abrMaxWithRealBitrate: false,&amp;#xA;            maxStarvationDelay: 4,&amp;#xA;            maxLoadingDelay: 4,&amp;#xA;            minAutoBitrate: 0,&amp;#xA;            emeEnabled: false&amp;#xA;        };&amp;#xA;        var hls = new Hls(config);&amp;#xA;        hls.loadSource(videoSrc);&amp;#xA;        hls.attachMedia(video);&amp;#xA;      }   &amp;#xA;      video.addEventListener(&quot;loadedmetadata&quot;, function(){ video.muted = true; video.play(); }, false);&amp;#xA;    &lt;/script&gt;&#xA;
  

  


  // here I added video.muted = true ; video.play() ; to auto start, if I try to autoplay unmuted, many browsers refuse this command...

  


  // playsinline="true" is NEEDED for safari

  


  ••••••• THE FFMPEG COMMAND (working : it allows me to have 3 to 4 seconds delay ••••••

  


  ffmpeg -re -i input.x -c:a aac -c:v libx264 
-movflags +dash -preset ultrafast 
-crf 28 -refs 4 -qmin 4 -pix_fmt yuv420p 
-tune zerolatency -c:a aac -ac 2 -profile:v main 
-flags -global_header -bufsize 969k 
-hls_time 1 -hls_list_size 0 -g 30 
-start_number 0 -streaming 1 -hls_playlist 1 
-lhls 1 -hls_playlist_type event -f hls path_to_index.m3u8


  


  •••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••••

  


  How can this be fixed ?

  


  How can I make play at live point on load in android MOBILE ?

  


• FFMPEG command from Python 3.5 does not actually create audio file

  20 décembre 2017, par Nathan Blaine

  I have a Django web application that accepts user uploaded videos/audio and saves them into a folder ’../WebAppDirectory/media/recordings’.

  I am then using a speech to text API to get a rough transcription of the audio. This is working fine for .wav and .mp4 files, but the web app also accepts videos (.MOV) that I would like to first convert to .wav, then pass off to the API.

  Using ffmpeg from my command line like this

  ffmpeg -i C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.MOV -ab 160k -ac 2 -ar 44100 -vn upload_sample.wav

  Correctly creates the .wav file from the original .MOV.

  However, when I run this from python with

  subprocess.check_call(command, shell=True)

  ffmpeg responds with

  File ’upload_sample.wav’ already exists. Overwrite ? [y/N]

  While Python tells me

  FileNotFoundError : [Errno 2] No such file or directory : ’C :\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.wav’

  It is also worth noting that I do not see a ’upload_sample.wav’ file in the media/recordings/ directory.

  This leads me to believe that maybe Python and ffmpeg are looking in different folders, but I am not sure where I am going wrong. When I print the command from the subprocess.check_call and copy/paste it into cmd, the file is created as expected.

  Hoping someone with some experience with ffmpeg/Python subprocess can help shed some light ! Here are the files I am working with :

  Folder Structure

  DjangoWebApp
  
  |---media
  
  |---|---imgs
  
  |---|---recordings
  
  |---|---|---upload_sample.MOV
  
  |---uploaded_audio_to_text.py

  uploaded_audio_to_text.py

  import speech_recognition as sr
  
  from os import path
  
  import os
  
  import subprocess
  
  
  
  
  
  def speech_to_text(file_name):
  
      AUDIO_FILE = path.join(path.dirname(path.realpath(__file__)), 'media','recordings', file_name)
  
      print("Looking at path: ",AUDIO_FILE)
  
      # get extension
  
      AUDIO_FILE_EXT = os.path.splitext(AUDIO_FILE)[1]
  
  
  
      if(AUDIO_FILE_EXT == '.MOV'):
  
          print("File is not .wav: ", AUDIO_FILE_EXT, "found. Converting...")
  
          # We will use subprocess and ffmpeg to convert this .MOV file to .wav, so we can send to API
  
          temp_wav = os.path.splitext(file_name)[0] + '.wav'
  
          print("New audio file will be: ", temp_wav)
  
          # build CMD ffmpeg command
  
          command = "ffmpeg -i "
  
          command += AUDIO_FILE
  
          command += " -ab 160k -ac 2 -ar 44100 -vn "
  
          command += temp_wav
  
  
  
          print("Attempting to run this command: \n",command)
  
          print(subprocess.check_call(command, shell=True))
  
          print("Past Subprocess.call")
  
          AUDIO_FILE = path.join(path.dirname(path.realpath(__file__)), 'media','recordings', temp_wav)
  
          print("AUDIO_FILE now set to: ", AUDIO_FILE)
  
  
  
      else:
  
          # continue with what we are doing
  
          pass
  
  
  
  
  
      r = sr.Recognizer()
  
      with sr.AudioFile(AUDIO_FILE) as source:
  
          audio = r.record(source)  # read the entire audio file
  
          text_transcription = "Sentinel"
  
          # recognize speech using Microsoft Bing Voice Recognition
  
          BING_KEY = "MY_KEY_:)"
  
          try:
  
              text_transcription = r.recognize_bing(audio, key=BING_KEY)
  
          except sr.UnknownValueError:
  
              print("Microsoft Bing Voice Recognition could not understand audio")
  
          except sr.RequestError as e:
  
              print("Could not request results from Microsoft Bing Voice Recognition service; {0}".format(e))
  
  
  
      return text_transcription
  
  
  
  
  
  #my tests
  
  my_relative_file_path = "upload_sample.MOV"
  
  print(speech_to_text(my_relative_file_path))

  Console output (traceback and my print()’s)

  Looking at path:  C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.MOV 
  
  File is not .wav:  .MOV found. Converting... 
  
  New audio file will be:  upload_sample.wav Attempting to run this command:
  
   ffmpeg -i C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.MOV -ab 160k -ac 2 -ar 44100 -vn upload_sample.wav 
  
  ffmpeg version git-2017-12-18-74f408c Copyright (c) 2000-2017 the FFmpeg developers   built with gcc 7.2.0 (GCC)   
  
  ----REMOVED SOME FFMPEG OUTPUT FOR BREVITY----
  
  File 'upload_sample.wav' already exists. Overwrite ? [y/N] y 
  
  Stream mapping:   Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native)) Press [q] to stop, [?] for help Output #0, wav, to 'upload_sample.wav':   Metadata:
  
      major_brand     : qt  
  
      minor_version   : 0
  
      compatible_brands: qt  
  
      com.apple.quicktime.creationdate: 2017-12-19T16:06:10-0500
  
      com.apple.quicktime.make: Apple
  
      com.apple.quicktime.model: iPhone 6
  
      com.apple.quicktime.software: 10.3.3
  
      ISFT            : Lavf58.3.100
  
      Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s (default)
  
      Metadata:
  
        creation_time   : 2017-12-19T21:06:11.000000Z
  
        handler_name    : Core Media Data Handler
  
        encoder         : Lavc58.8.100 pcm_s16le size=    1036kB time=00:00:06.01 bitrate=1411.3kbits/s speed=N/A     video:0kB audio:1036kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.007352% 
  
  0 
  
  Traceback (most recent call last): Past Subprocess.call   
  
  File "C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\uploaded_audio_to_text.py", line 53, in <module> 
  
  AUDIO_FILE now set to:  C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\media\recordings\upload_sample.wav
  
      print(speech_to_text(my_relative_file_path))   
  
  File "C:\Users\Nathan\Desktop\MeetingRecorderWebAPP\uploaded_audio_to_text.py", line 36, in speech_to_text
  
      with sr.AudioFile(AUDIO_FILE) as source:   
  
  File "C:\Users\Nathan\AppData\Local\Programs\Python\Python36-32\lib\site-packages\speech_recognition\__init__.py", line 203, in __enter__
  
      self.audio_reader = wave.open(self.filename_or_fileobject, "rb")   
  
  File "C:\Users\Nathan\AppData\Local\Programs\Python\Python36-32\lib\wave.py", line 499, in open
  
      return Wave_read(f)   
  
  File "C:\Users\Nathan\AppData\Local\Programs\Python\Python36-32\lib\wave.py", line 159, in __init__
  
      f = builtins.open(f, 'rb') 
  
  FileNotFoundError: [Errno 2] No such file or directory: 'C:\\Users\\Nathan\\Desktop\\MeetingRecorderWebAPP\\media\\recordings\\upload_sample.wav'
  
  
  
  Process finished with exit code 1
  
  </module>

• Can I use avcodec_free_context() on an opened context ?

  23 mars 2017, par Ashe the human

  The latest documentation says here that opening a context that’s closed again is not supported any more. I see why. Some codecs don’t work properly when they’re reopened. So after finding this bug, I decided to not use avcodec_close() and call avcodec_free_context() on the contexts right away instead.

  But I’m not sure if it’s safe to do so with 2.8.4, the version that I linked to my program. The documentation from that time doesn’t clarify. Does anyone know ? At least empirically ?

  ffmpeg version 2.8.4 Copyright (c) 2000-2015 the FFmpeg developers
  
  built with Microsoft (R) C/C++ 최적화 컴파일러 버전 18.00.31101(x64)
  
  configuration: --toolchain=msvc --enable-gpl --enable-nonfree --enable-nvenc --enable-libvorbis --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-libx265 --enable-libxvid --enable-libopus --enable-libvpx --enable-static --disable-shared --disable-debug --extra-cflags=-MT --extra-cxxflags=-MT --extra-ldflags='/nodefaultlib:msvcrt.lib' --extra-libs='zlib.lib libogg_static.lib libvorbis_static.lib libmpghip-static.lib libmp3lame-static.lib libtheora_static.lib libx264.lib x265-static.lib libxvidcore.lib silk_fixed.lib silk_common.lib silk_float.lib celt.lib opus.lib vpxmt.lib'
  
  libavutil      54. 31.100 / 54. 31.100
  
  libavcodec     56. 60.100 / 56. 60.100
  
  libavformat    56. 40.101 / 56. 40.101
  
  libavdevice    56.  4.100 / 56.  4.100
  
  libavfilter     5. 40.101 /  5. 40.101
  
  libswscale      3.  1.101 /  3.  1.101
  
  libswresample   1.  2.101 /  1.  2.101
  
  libpostproc    53.  3.100 / 53.  3.100

  I know there’s a bunch of forums I could post on but I’d felt like to ask it here first.