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• 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 ?

• FFMPEG & MistServer : RTMP Input/Output error

  17 octobre 2017, par brewcrazy

  I have an IP camera that outputs a RTSP stream that I’m trying to use to display a live feed on my website. This is a small site that only my wife and I will access so I’m trying to use a free streaming service. For that reason, I’ve decided to try MistServer’s open source option.

  I currently have downloaded MistServer and have it running without installation on my mac (sudo ./MistController). With MistServer running, I have a stream set up and default protocols configured. The stream is configured as follows :

  stream name: ipcam
  
  source: push://

  The configuration page gives me the following source to push to :

  RTMP full url: rtmp://127.0.0.1/live/ipcam
  
  RTMP url: rtmp://127.0.0.1/live/
  
  RTMP stream key: ipcam

  In the streams view, the stream’s status is unavailable, but I’m assuming this is because it isn’t receiving an input. I haven’t been able to confirm this via documentation.

  Here is the FFMPEG command that I am running and the error that I’m getting :

  ffmpeg -rtsp_transport tcp -i rtsp://<user>:@:554/live0.264 -acodec copy -vcodec copy -f flv rtmp://127.0.0.1/live/ipcam
  
  
  
  ffmpeg version 3.3.3 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with Apple LLVM version 8.1.0 (clang-802.0.42)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.3.3 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --enable-videotoolbox --disable-lzma --enable-vda
  
    libavutil      55. 58.100 / 55. 58.100
  
    libavcodec     57. 89.100 / 57. 89.100
  
    libavformat    57. 71.100 / 57. 71.100
  
    libavdevice    57.  6.100 / 57.  6.100
  
    libavfilter     6. 82.100 /  6. 82.100
  
    libavresample   3.  5.  0 /  3.  5.  0
  
    libswscale      4.  6.100 /  4.  6.100
  
    libswresample   2.  7.100 /  2.  7.100
  
    libpostproc    54.  5.100 / 54.  5.100
  
  Guessed Channel Layout for Input Stream #0.1 : mono
  
  Input #0, rtsp, from 'rtsp://<user>:@:554/live0.264':
  
    Metadata:
  
      title           : Session Streamed by LIBZRTSP
  
      comment         : live0.264
  
    Duration: N/A, start: 0.242000, bitrate: N/A
  
      Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1280x720, 25 fps, 24.83 tbr, 90k tbn, 50 tbc
  
      Stream #0:1: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
  
  rtmp://127.0.0.1/live/ipcam: Input/output error
  
  </user></user>

  I can’t determine from this error if the issue is the FFMPEG command or my MistServer configuration.

• Why is ffmpeg's hstack so much slower than overlay and pad ?

  27 janvier 2021, par cgenco

  I'm using ffmpeg to stitch together two videos of people chatting into a video with each of them side-by-side, like this :

  


  

  


  Here's the command I'm currently using to get this done, which runs at 2.5x on my 13" M1 MacBook Pro :

  


  ffmpeg -y -i left.mp4 -i right.mp4 -filter_complex "
  [0:v] crop=w=in_w/2 [croppedLeft];
  [1:v][1:v] overlay=x=overlay_w/4 [shiftedRight];
  [shiftedRight][croppedLeft] overlay [vout];
  [0:a][1:a] amix [aout]
" -map "[vout]" -map "[aout]" -ac 2 out.mp4


  


  This command crops the left video to half of its original width (cropping so the video is centered), then shifts the right video a quarter of its width to the right, then overlays the left video on the left half of the output merged with the shifted right video.

  


  One day on my weekly fun-time read-through the FFmpeg filters documentation I stumbled on a filter named hstack, which is described as being "faster than using overlay and pad filter to create same output."

  


  My ex wife can affirm that there are few higher priorities in my life than going faster, so I altered my ffmpeg script to use hstack instead of two overlays :

  


  ffmpeg -y -i left.mp4 -i right.mp4 -filter_complex "
  [0:v] crop=w=in_w/2 [croppedLeft];
  [1:v] crop=w=in_w/2 [croppedRight];
  [croppedLeft][croppedRight] vstack [vout];
  [0:a][1:a] amix [aout]
" -map "[vout]" -map "[aout]" -ac 2 out.mp4


  


  ...but that command runs painfully slowly, like 0.1x. It takes multiple minutes to render a single second.

  


  So uhhh what's going on here ? Why is hstack taking so long when it's supposed to be faster ?

  


  I've tried this on both the M1 native build from OSXExperts (version N-99816-g3da35b7) and the standard ffmpeg from brew and hstack is just as slow on each.