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• Bad file descriptor decoding mp3 to pipe with ffmpeg

  1er février 2017, par Pete Bleackley

  Can any FFMpeg gurus help me with the following ?

  I’m trying to convert a podcast to PCM and downsample it to 16KHz mono before feeding it to a speech recognition system for transcription. The command line

  ffmpeg -i http://media.blubrry.com/conlangery/content.blubrry.com/conlangery/Conlangery01.mp3 -f s16le -ac 1 -ar 16000 pipe:0

  fails with

  av_interleaved_write_frame() : Bad file descriptor

  What is the problem here and how do I fix it ?

  EDIT

  Full error message is

  ffmpeg version N-83189-gd5d474aea5-static http://johnvansickle.com/ffmpeg/  Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 5.4.1 (Debian 5.4.1-4) 20161202
  
    configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-5 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gray --enable-libass --enable-libfreetype --enable-libfribidi --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzimg
  
    libavutil      55. 44.100 / 55. 44.100
  
    libavcodec     57. 75.100 / 57. 75.100
  
    libavformat    57. 62.100 / 57. 62.100
  
    libavdevice    57.  2.100 / 57.  2.100
  
    libavfilter     6. 69.100 /  6. 69.100
  
    libswscale      4.  3.101 /  4.  3.101
  
    libswresample   2.  4.100 /  2.  4.100
  
    libpostproc    54.  2.100 / 54.  2.100
  
  Input #0, mp3, from 'http://media.blubrry.com/conlangery/content.blubrry.com/conlangery/Conlangery01.mp3':
  
    Metadata:
  
      track           : 1
  
      album           : Conlangery Podcast
  
      title           : Conlangery 01
  
      artist          : George Corley
  
      date            : 2011
  
    Duration: 00:44:58.08, start: 0.025057, bitrate: 128 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
  
      Metadata:
  
        encoder         : LAME3.98r
  
  Output #0, s16le, to 'pipe:0':
  
    Metadata:
  
      track           : 1
  
      album           : Conlangery Podcast
  
      title           : Conlangery 01
  
      artist          : George Corley
  
      date            : 2011
  
      encoder         : Lavf57.62.100
  
      Stream #0:0: Audio: pcm_s16le, 16000 Hz, mono, s16, 256 kb/s
  
      Metadata:
  
        encoder         : Lavc57.75.100 pcm_s16le
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (native) -> pcm_s16le (native))
  
  Press [q] to stop, [?] for help
  
  av_interleaved_write_frame(): Bad file descriptor
  
  Error writing trailer of pipe:0: Bad file descriptorsize=       1kB time=00:00:00.02 bitrate= 256.0kbits/s speed=95.2x    
  
  video:0kB audio:1kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000000%
  
  Conversion failed!
  
  Traceback (most recent call last):
  
    File "./PodcastTranscriber.py", line 206, in <module>
  
      PodcastTranscriber('http://conlangery.com/feed/',upload)()        
  
    File "./PodcastTranscriber.py", line 101, in __call__
  
      self.process(item)
  
    File "./PodcastTranscriber.py", line 136, in process
  
      (audio,errors)=mp3.run(stdout=subprocess.PIPE)
  
    File "/usr/local/lib/python2.7/site-packages/ffmpy.py", line 105, in run
  
      raise FFRuntimeError(self.cmd, self.process.returncode, out[0], out[1])
  
  ffmpy.FFRuntimeError: `ffmpeg -i http://media.blubrry.com/conlangery/content.blubrry.com/conlangery/Conlangery01.mp3 -f s16le -ac 1 -ar 16000 pipe:0` exited with status 1
  
  </module>

  The invoking code is

  def process(self,item):
  
      """Downloads the audio and transcribes it"""
  
      audio_url=None
  
      print item['title']
  
      for link in item.links:
  
          if link['rel']=='enclosure':
  
              audio_url=link['href']
  
      if audio_url is not None:
  
          pubDate=to_date(item.published_parsed)
  
          if self.lastUploadDate is None or pubDate>self.lastUploadDate:
  
              self.lastUploadDate=pubDate
  
          mp3=ffmpy.FFmpeg(inputs={audio_url:None},
  
                           outputs={'pipe:0':['-f','s16le','-ac','1','-ar','16000']})
  
          (audio,errors)=mp3.run(stdout=subprocess.PIPE)
  
          sphinx=subprocess.Popen(['java','-jar','transcriber.jar'],
  
                                  stdin=audio,
  
                                  stdout=subprocess.PIPE)
  
          wiki=threading.Thread(target=self.callback,args=(item,sphinx.stdout))
  
          wiki.start()
  
          #mp3.start()
  
          #mp3.join()
  
          sphinx.stdin.close()
  
          wiki.join()
  
          sphinx.wait()

• FFMpeg - Combine multiple filter_complex and overlay functions

  7 juin 2016, par Mike Johnson

  I am having trouble combining these 3 passes in ffmpeg into a single process.

  Is this even possible ?

  Pass 1

  ffmpeg -y -i C:\Users\MJ\Downloads\20151211_pmoney_pmpod.mp3 -loop 1 -i C:\Users\MJ\Documents\pm1080.png  -filter_complex "[0:a]showwaves=s=1920x1080:mode=line,colorkey=0x000000:0.01:0.1,format=yuva420p[v];[1:v][v]overlay=0:270[outv]" -map "[outv]" -pix_fmt yuv420p -map 0:a -c:v libx264 -c:a copy -shortest C:\Users\MJ\Documents\20151211_pmoney_pmpod4.mp4

  Pass 2

  ffmpeg -i "C:\Users\MJ\Documents\20151211_pmoney_pmpod4.mp4" -vf drawtext="fontsize=50:fontcolor=white:fontfile=/Windows/Fonts/impact.ttf:text=Planet Money Podcast on NPR - A/B Split Testing:x=(w-text_w)/2:y=200" -acodec copy "C:\Users\MJ\Documents\20151211_pmoney_pmpod-overlay-text.mp4"

  Pass 3

  ffmpeg -i "C:\Users\MJ\Documents\20151211_pmoney_pmpod-overlay-text.mp4" -i C:\Users\MJ\Downloads\6.png -filter_complex "overlay=10:10" C:\Users\MJ\Documents\20151211_pmoney_pmpod-overlay-text1.mp4"

  Thanks !

• How to Synchronize Audio with Video Frames [Python]

  19 septembre 2023, par Ростислав

  I want to stream video from URL to a server via socket, which then restreams it to all clients in the room.

  


  This code streams video frame by frame :

  


  async def stream_video(room, url):
    cap = cv2.VideoCapture(url)
    fps = round(cap.get(cv2.CAP_PROP_FPS))

    while True:
        ret, frame = cap.read()
        if not ret: break
        _, img_bytes = cv2.imencode(".jpg", frame)
        img_base64 = base64.b64encode(img_bytes).decode('utf-8')
        img_data_url = f"data:image/jpeg;base64,{img_base64}"

        await socket.emit('segment', { 'room': room, 'type': 'video', 'stream': img_data_url})
        await asyncio.sleep(1/fps)
    
    cap.release()


  


  And this is code for stream audio :

  


  async def stream_audio(room, url):
    sample_size = 14000
    cmd_audio = [
        "ffmpeg",
        "-i", url,
        '-vn',
        '-f', 's16le',
        '-c:a', 'pcm_s16le',
        "-ac", "2",
        "-sample_rate","48000",
        '-ar','48000',
        "-acodec","libmp3lame",
        "pipe:1"
    ]
    proc_audio = await asyncio.create_subprocess_exec(
        *cmd_audio, stdout=subprocess.PIPE, stderr=False
    )

    while True:
        audio_data = await proc_audio.stdout.read(sample_size)
        if audio_data:
            await socket.emit('segment', { 'room': room, 'type': 'audio', 'stream': audio_data})
        await asyncio.sleep(1)



  


  But the problem is : how to synchronize them ? How many bytes need to be read every second from ffmpeg so that the audio matches the frames.

  


  I tried to do this, but the problem with the number of chunks still remained :

  


  while True:
    audio_data = await proc_audio.stdout.read(sample_size)
    if audio_data:
        await socket.emit('segment', { 'room': room, 'type': 'audio', 'stream': audio_data})

        for i in range(fps):
            ret, frame = cap.read()
            if not ret: break
            _, img_bytes = cv2.imencode(".jpg", frame)
            img_base64 = base64.b64encode(img_bytes).decode('utf-8')
            img_data_url = f"data:image/jpeg;base64,{img_base64}"

            await socket.emit('segment', { 'room': room, 'type': 'video', 'stream': img_data_url})
            await asyncio.sleep(1/fps)


  


  I also tried loading a chunk of audio into pydub, but it shows that the duration of my 14000 chunk is 0.07s, which is very small. And if you increase the number of chunks for reading to 192k (as the gpt chat says), then the audio will simply play very, very quickly.
The ideal number of chunks that I was able to achieve is approximately 14000, but the audio is still not synchronous.