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• FFMPEG multi livestream - recorded stream send to different services like YT and Twitch at different time (on different button clicks )

  4 octobre 2022, par Ganesh

  Trying for the last 10 days and still no success, I am creating a python application that will accept the URL and visit that URL using chromium, capture that screen and send that real-time screen recording to different live stream acceptors as youtube live, twitch Twitter, Facebook live or some other sources and many of these could be multiple.

  


  There are two challenges (both challenges depend on a user action like different button clicks) -

  


  

  • The time of starting the Livestream we know only one Livestream acceptor and other acceptors could be sent via another API at any time or may not be sent on the whole live stream.

  


  • Any of the streams could be stopped at any moment including the first one which started the original live streaming service

  


  


  To Solve these challenges I am trying the following process (i took mp4 as a source for simplifying)

  


  

  • create a stream and store it into PIPE.stdout

  


  


  ffmpeg_Command_get_stream = 'ffmpeg -re -i test.mp4 -f flv pipe:1'
ffmpeg_Command_get_stream=ffmpeg_Command_get_stream.split()
pipe = sp.Popen(ffmpeg_Command_get_stream,
            stdout=sp.PIPE,
            stderr=sp.PIPE,
            bufsize=8000000,
            shell=True,
            universal_newlines=True
            )
out,err = pipe.communicate()


  


  


  • and send that stream with the help of FFMPEG to the Livestream acceptor with the click of the youtube Livestream button

    


    ffmpeg_Command_send_stream = ['ffmpeg','-i',pipe.stdout,'-f','flv',RTMPURL_YOUTUBE]

    


  


  


  Update Trying to Explain it a little more :

  


  step 1 - I need a real-time stream from the first command, so I used -re in FFMPEG

  


  step 2 - Use above stream as an input for other command and send that as an output as a Livestream to youtube (or twitch/Facebook), But the second step would happen only when the user click on the button "YT LiveStream", Here the tricky thing is there are multiple buttons (YT LiveStream, Twitch LiveStream, Facebook LiveStream) and user can click any time on any of button, also can click on all button one by one.

  


  

  


  sorry for bad explaination

  


  what I am doing wrong ? , Is this Possible ? or need to go with another process,

  


  any help would be greatly appreciated

  


• libavfilter/dnn : determine dnn output during execute_model instead of set_input_output

  25 avril 2019, par Guo, Yejun

  libavfilter/dnn : determine dnn output during execute_model instead of set_input_output
  Currently, within interface set_input_output, the dims/memory of the tensorflow
  
  dnn model output is determined by executing the model with zero input,
  
  actually, the output dims might vary with different input data for networks
  
  such as object detection models faster-rcnn, ssd and yolo.
  This patch moves the logic from set_input_output to execute_model which
  
  is suitable for all the cases. Since interface changed, and so dnn_backend_native
  
  also changes.
  In vf_sr.c, it knows it's srcnn or espcn by executing the model with zero input,
  
  so execute_model has to be called in function config_props
  Signed-off-by : Guo, Yejun <yejun.guo@intel.com>
  
  Signed-off-by : Pedro Arthur <bygrandao@gmail.com>
  

  • [DH] libavfilter/dnn_backend_native.c
  • [DH] libavfilter/dnn_backend_native.h
  • [DH] libavfilter/dnn_backend_tf.c
  • [DH] libavfilter/dnn_backend_tf.h
  • [DH] libavfilter/dnn_interface.h
  • [DH] libavfilter/vf_sr.c

• A ffmpeg comman canwork in cmd but not in Python using subprocess.call() or os.system()

  6 juin 2018, par Starrysky

  I wanna transfer a .mp3 to .wav. This is my command :
  ffmpeg -i a.mp3 -ar 16000 -ac 1 -acodec pcm_s16le a.wav

  It worked well in cmd

  C:\Users\starrysky\Documents\GitHub\bing_pic\html>ffmpeg -i a.mp3 -ar 16000 -ac 1 -acodec pcm_s16le a.wav
  
  ffmpeg version N-86482-gbc40674 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 7.1.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
  
    libavutil      55. 66.100 / 55. 66.100
  
    libavcodec     57. 99.100 / 57. 99.100
  
    libavformat    57. 73.100 / 57. 73.100
  
    libavdevice    57.  7.100 / 57.  7.100
  
    libavfilter     6. 92.100 /  6. 92.100
  
    libswscale      4.  7.101 /  4.  7.101
  
    libswresample   2.  8.100 /  2.  8.100
  
    libpostproc    54.  6.100 / 54.  6.100
  
  Input #0, mp3, from 'a.mp3':
  
    Metadata:
  
      encoder         : Lavf54.6.100
  
    Duration: 00:00:01.87, start: 0.000000, bitrate: 8 kb/s
  
      Stream #0:0: Audio: mp3, 8000 Hz, mono, s16p, 8 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (native) -> pcm_s16le (native))
  
  Press [q] to stop, [?] for help
  
  Output #0, wav, to 'a.wav':
  
    Metadata:
  
      ISFT            : Lavf57.73.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s
  
      Metadata:
  
        encoder         : Lavc57.99.100 pcm_s16le
  
  size=      59kB time=00:00:01.87 bitrate= 256.3kbits/s speed= 187x
  
  video:0kB audio:58kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.130208%

  but when I moved it into my python program, something strange happened.

  >>> C:\Users\starrysky\Documents\GitHub\bing_pic\html\
  
  'ffmpeg' �����ڲ����ⲿ���Ҳ���ǿ����еij���
  
  ���������ļ���
  
  1 Command 'ffmpeg -i a.mp3 -ar 16000 -ac 1 -acodec pcm_s16le a.wav' returned non-zero exit status 1.
  
  文件错误啊，亲
  
  [WinError 2] 系统找不到指定的文件。: 'a.wav'

  This is part of my python code :

  @bot.register(wife, RECORDING)
  
  def translate_sound(msg):
  
      msg.get_file(save_path='a.mp3')
  
      path = os.path.abspath('.')+'\\'
  
      print(path)
  
      try:
  
          subprocess.check_call('ffmpeg -i a.mp3 -ar 16000 -ac 1 -acodec pcm_s16le a.wav', shell=True)
  
          # ''
  
      except Exception as e:
  
          print(1, e)
  
      wav_to_text('a.wav')
  
      try:
  
          os.remove('a.wav')
  
      except Exception as e:
  
          print(e)

  ---

  # 调用百度语音识别API
  
  def get_token():
  
      URL = 'http://openapi.baidu.com/oauth/2.0/token'
  
      _params = urllib.parse.urlencode({'grant_type': b'client_credentials',
  
                                        'client_id': b''
  
                                        'client_secret': b''})
  
      _res = urllib.request.Request(URL, _params.encode())
  
      _response = urllib.request.urlopen(_res)
  
      _data = _response.read()
  
      _data = json.loads(_data)
  
      return _data['access_token']
  
  
  
  
  
  def wav_to_text(wav_file):
  
      try:
  
          wav_file = open(wav_file, 'rb')
  
      except IOError:
  
          print('文件错误啊，亲')
  
          return
  
      wav_file = wave.open(wav_file)
  
      n_frames = wav_file.getnframes()
  
      print('n_frames ', n_frames)
  
      frame_rate = wav_file.getframerate()
  
      print("frame_rate ", frame_rate)
  
      if n_frames == 1 or frame_rate not in (8000, 16000):
  
          print('不符合格式')
  
          return
  
      audio = wav_file.readframes(n_frames)
  
      seconds = n_frames/frame_rate+1
  
      minute = int(seconds/60 + 1)
  
      for i in range(0, minute):
  
          sub_audio = audio[i*60*frame_rate:(i+1)*60*frame_rate]
  
          base_data = base64.b64encode(sub_audio)
  
          data = {"format": "wav",
  
                  "token": get_token(),
  
                  "len": len(sub_audio),
  
                  "rate": frame_rate,
  
                  "speech": base_data.decode(),
  
                  "cuid": "B8-AC-6F-2D-7A-94",
  
                  "channel": 1}
  
          data = json.dumps(data)
  
          res = urllib.request.Request('http://vop.baidu.com/server_api',
  
                                data.encode(),
  
                                {'content-type': 'application/json'})
  
          response = urllib.request.urlopen(res)
  
          res_data = json.loads(response.read())
  
          try:
  
              print(res_data['result'][0])
  
          except Exception as e:
  
              print(e)

  What happened ?