Recherche avancée

Sur d’autres sites (5279)

• ValueError When Reading Video Frame

  27 juin 2016, par Bassie

  I am following this article, from where I got this code :

  FFMPEG_BIN ="Z:\ffmpeg\bin\ffmpeg.exe"
  
  
  
  import subprocess as sp
  
  command = [ FFMPEG_BIN,
  
              '-i', 'video.mp4',
  
              '-f', 'image2pipe',
  
              '-pix_fmt', 'rgb24',
  
              '-vcodec', 'rawvideo', '-']
  
  pipe = sp.Popen(command, stdout = sp.PIPE, bufsize=10**8, shell=True)
  
  
  
  import numpy
  
  # read 420*360*3 bytes (= 1 frame)
  
  raw_image = pipe.stdout.read(420*360*3)
  
  # transform the byte read into a numpy array
  
  image =  numpy.fromstring(raw_image, dtype='uint8')
  
  image = image.reshape((360,420,3))
  
  # throw away the data in the pipe's buffer.
  
  pipe.stdout.flush()

  When I run it I see this error :

  Traceback (most recent call last):
  
    File "Z:\py\ffmtest\test.py", line 16, in <module>
  
      image = image.reshape((360,420,3))
  
  ValueError: total size of new array must be unchanged
  
  </module>

  Where line 16 is image = image.reshape((360,420,3)). I think this error is produced by numpy, but probably because I am calculating the values for my video incorrectly.

  Output :

  raw_image : b’ ’

  len(raw_image) : 0

  image : [ ]

  len(image) : 0

  I am not sure whether I am passing in the correct values for read or reshape functions - any help at all is much appreciated !

• Trying to open file with PHP-FFMpeg after it was encoded once

  19 mai 2016, par Shillo Ben David

  in the same PHP process I’m trying to open a file that was manipulated and saved, and then I’m trying to open it with as a new FFMpeg\Video. For example, in the same process :

  Open -> original.MOV
  
    Manipulate & save to -> new.mp4
  
      Open -> new.mp4

  However when I’m trying to open the manipulated file I get this InvalidArgumentException exception :

  InvalidArgumentException: Unable to detect file format, only audio and video supported

  It’s thrown by the FFMpeg::open() after it could not detect that it’s a either Video or Audio stream.

  FFMpeg::open()

  public function open($pathfile)
  
  {
  
      if (null === $streams = $this->ffprobe->streams($pathfile)) {
  
          throw new RuntimeException(sprintf('Unable to probe "%s".', $pathfile));
  
      }
  
  
  
      if (0 < count($streams->videos())) {
  
          return new Video($pathfile, $this->driver, $this->ffprobe);
  
      } elseif (0 < count($streams->audios())) {
  
          return new Audio($pathfile, $this->driver, $this->ffprobe);
  
      }
  
  
  
      throw new InvalidArgumentException('Unable to detect file format, only audio and video supported');
  
  }

  The filters I applied to the video are audio mute and speedup (setpts).

  So I wonder, why FFMpeg doesn’t recognise it as video ?

• C++ FFmpeg pitch issue

  24 janvier 2016, par David Andrei Norgren

  I’m using swr_convert to lower/raise the pitch of incoming audio and store it in a .mp3. To change the pitch, I’m dividing the out sample rate by a factor. However, the resulting audio is slightly distorted when this factor is anything other than 1. Here’s my conversion code :

  ...
  
  
  
  // Set up resample context
  
  swrContext = swr_alloc();
  
  if (!swrContext)
  
      throw -15;
  
  
  
  av_opt_set_int(swrContext, "in_channel_count", codecContext->channels, 0);
  
  av_opt_set_int(swrContext, "in_channel_layout", codecContext->channel_layout, 0);
  
  av_opt_set_int(swrContext, "in_sample_rate", codecContext->sample_rate, 0);
  
  av_opt_set_sample_fmt(swrContext, "in_sample_fmt", codecContext->sample_fmt, 0);
  
  
  
  av_opt_set_int(swrContext, "out_channel_count", STREAM_AUDIO_CHANNELS, 0);
  
  av_opt_set_int(swrContext, "out_channel_layout", STREAM_AUDIO_CHANNEL_LAYOUT, 0);
  
  av_opt_set_int(swrContext, "out_sample_rate", STREAM_AUDIO_SAMPLE_RATE / pitch, 0);
  
  av_opt_set_sample_fmt(swrContext, "out_sample_fmt", STREAM_AUDIO_SAMPLE_FORMAT_GM, 0);
  
  
  
  if (swr_init(swrContext))
  
      throw -16;
  
  
  
  // Allocate re-usable frame
  
  frameDecoded = av_frame_alloc();
  
  if (!frameDecoded)
  
      throw -17;
  
  
  
  frameDecoded->format = codecContext->sample_fmt;
  
  frameDecoded->channel_layout = codecContext->channel_layout;
  
  frameDecoded->channels = codecContext->channels;
  
  frameDecoded->sample_rate = codecContext->sample_rate;
  
  
  
  // Load frames
  
  inPacket.data = NULL;
  
  inPacket.size = 0;
  
  
  
  int gotFrame, samples = 0;
  
  
  
  while (av_read_frame(formatContext, &inPacket) >= 0) {
  
  
  
      if (inPacket.stream_index != streamId) 
  
          continue;
  
  
  
      if (avcodec_decode_audio4(codecContext, frameDecoded, &gotFrame, &inPacket) < 0)
  
          throw -18;
  
  
  
      if (!gotFrame)
  
          continue;
  
  
  
      // Begin conversion
  
      if (swr_convert(swrContext, NULL, 0, (const uint8_t **)frameDecoded->data, frameDecoded->nb_samples) < 0)
  
          throw -19;
  
  
  
      while (swr_get_out_samples(swrContext, 0) >= RAW_AUDIO_FRAME_SIZE) {
  
  
  
          // Allocate data
  
          uint8_t **convertedData = NULL;
  
          if (av_samples_alloc_array_and_samples(&convertedData, NULL, STREAM_AUDIO_CHANNELS, RAW_AUDIO_FRAME_SIZE, STREAM_AUDIO_SAMPLE_FORMAT_GM, 0) < 0)
  
              throw -20;
  
  
  
          // Convert
  
          if (swr_convert(swrContext, convertedData, RAW_AUDIO_FRAME_SIZE, NULL, 0) < 0)
  
              throw -21;
  
  
  
          // Calculate buffer size
  
          size_t bufferSize = av_samples_get_buffer_size(NULL, STREAM_AUDIO_CHANNELS, RAW_AUDIO_FRAME_SIZE, STREAM_AUDIO_SAMPLE_FORMAT_GM, 0);
  
          if (bufferSize < 0)
  
              throw -22;
  
  
  
          fwrite(convertedData[0], 1, bufferSize, outStream);
  
          av_free(convertedData);
  
      }
  
  }
  
  
  
  ...

  STREAM_AUDIO_SAMPLE_RATE is defined as 44100.
  Here’s the entire program if it helps : http://pastebin.com/5akEwNg4

  The program generates a .mp3 with 25 notes that decrease in pitch.
  Here’s an example of the distortion : http://www.stuffbydavid.com/dl/30256478.mp3

  Can you spot anything incorrect about my conversion, or is my method of changing the pitch incorrect ? Is there another way ?