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• How to transcode MP3 files in django by using FFMPEG, Celery and RabbitMQ ?

  12 janvier 2017, par Srinivas 25

  I am trying to transcode user uploaded MP3 audio files into ogg, ac3 wav or other formats by using django, celery, rabbitMQ and FFMPEG. But i am getting the error with [WinError 10042] An unknown, invalid, or unsupported option or level was specified in a getsockopt or setsockopt call

  OS-Window 10-64bit
  Python 3.0
  Django - 1.10

  here is the code I followed :

  models.py
  
  import uuid
  
  from django.db import models
  
  
  
     # Create your models here.
  
  
  
      def unique_file_path(instance, filename):
  
      new_file_name = uuid.uuid4()
  
      return str(new_file_name)
  
  
  
      class AudioFile(models.Model):
  
      name = models.CharField(max_length=100, blank=True)
  
      mp3_file = models.FileField(upload_to=unique_file_path)
  
      ogg_file = models.FileField(blank=True, upload_to=unique_file_path)
  
      wav_file = models.FileField(blank=True, upload_to=unique_file_path)
  
      ac3_file = models.FileField(blank=True, upload_to=unique_file_path)
  
  
  
      def __str__(self):
  
          return self.name
  
  
  
      views.py
  
      from django.shortcuts import render
  
  
  
      # Create your views here.
  
  
  
      from django.views.generic.edit import FormView
  
      from django.http import HttpResponseRedirect
  
      from django.core.urlresolvers import reverse
  
  
  
      from django.views.generic import FormView
  
      from audio_transcoder.taskapp.tasks import transcode_mp3
  
  
  
      from .forms import AudioFileFrom
  
      from .models import AudioFile
  
  
  
      class UploadAudioFileView(FormView):
  
      template_name = 'upload/upload.html'
  
      form_class = AudioFileFrom
  
  
  
  
  
      def form_valid(self, form):
  
          audio_file = AudioFile(
  
              name=self.get_form_kwargs().get('data')['name'],
  
              mp3_file=self.get_form_kwargs().get('files')['mp3_file']
  
          )
  
          audio_file.save()
  
          transcode_mp3.delay(audio_file.id)
  
  
  
          return HttpResponseRedirect(self.get_success_url())
  
  
  
      def get_success_url(self):
  
          return reverse('/')
  
  
  
  
  
      tasks.py
  
      import os
  
      import os.path
  
      import subprocess
  
  
  
      from audio_transcoder.taskapp.celery import app
  
  
  
  
  
  
  
  
  
      from celery import Celery
  
  
  
      app = Celery('fftest',
  
               broker='amqp://guest@localhost//',
  
               include=['taskapp.tasks'])
  
  
  
      if __name__ == '__main__':
  
      app.start()
  
  
  
      from audio_transcoder.models import AudioFile
  
      import fftest.settings as settings
  
  
  
      @app.task
  
      def transcode_mp3(mp3_id):
  
      audio_file = AudioFile.objects.get(id=mp3_id)
  
      input_file_path = audio_file.mp3_file.path
  
      filename = os.path.basename(input_file_path)
  
  
  
      ogg_output_file_name = os.path.join('transcoded', '{}.ogg'.format(filename))
  
      ogg_output_file_path = os.path.join(settings.MEDIA_ROOT, ogg_`enter code     
  
      here`output_file_name)
  
      enter code here
  
      ac3_output_file_name = os.path.join('transcoded', '{}.ac3'.format(filename))
  
      ac3_output_file_path = os.path.join(settings.MEDIA_ROOT,  
  
      ac3_output_file_name)
  
  
  
      wav_output_file_name = os.path.join('transcoded', '{}.wav'.format(filename))
  
      wav_output_file_path = os.path.join(settings.MEDIA_ROOT, 
  
      wav_output_file_name)
  
  
  
      if not os.path.isdir(os.path.dirname(ogg_output_file_path)):
  
          os.makedirs(os.path.dirname(ogg_output_file_path))
  
  
  
      subprocess.call([
  
              settings.FFMPEG_PATH,
  
              '-i',
  
              input_file_path,
  
              ogg_output_file_path,
  
              ac3_output_file_path,
  
              wav_output_file_path
  
          ]
  
      )
  
  
  
      audio_file.ogg_file = ogg_output_file_name
  
      audio_file.ac3_file = ac3_output_file_name
  
      audio_file.wav_file = wav_output_file_name
  
      audio_file.save()

  Not sure where the mistake is happening. While uploading video it is showing below :

  OperationalError at /new/
  
  [WinError 10042] An unknown, invalid, or unsupported option or level was specified in a getsockopt or setsockopt call
  
  Request Method: POST
  
  Request URL:  http://127.0.0.1:8000/new/
  
  Django Version: 1.10.4
  
  Exception Type: OperationalError`enter code here`
  
  Exception Value:  
  
  [WinError 10042] An unknown, invalid, or unsupported option or level was specified in a getsockopt or setsockopt call
  
  Exception Location: C:\Users\RAMa2r3e4s5h6\fftest\lib\site-packages\amqp\transport.py in _set_socket_options, line 204

• FFMpeg C# wrapper "MediaToolkit 1.0.4.11". How can I do a conversion in memory ?

  8 décembre 2016, par Quantum_Kernel

  I am currently using the MediaToolkit 1.0.4.11 C# wrapper for FFMPEG to extract wav audio from a range of video files.

  Instead of creating a wav file on disk, I would like to create it in memory (as a MemoryStream). This will save me from creating temp files which I need to delete and should speed up the various analyses that I am doing on the audio data.

  Because of the way the call to the library works, I’m not sure how to ’trick’ it into doing this. Is there a way to do it or will I have to obtain and edit the wrapper source code ?

  Here is what I have for doing the conversion on disk :

  var inputFile = new MediaToolkit.Model.MediaFile { Filename = mediaFilePath };
  
              var outputFile = new MediaToolkit.Model.MediaFile { Filename = @"C:\Temp\audio.wav" };
  
  
  
              var conversionOptions = new MediaToolkit.Options.ConversionOptions
  
              {
  
                  MaxVideoDuration = TimeSpan.FromSeconds(30),
  
                  VideoAspectRatio = MediaToolkit.Options.VideoAspectRatio.R16_9,
  
                  VideoSize = MediaToolkit.Options.VideoSize.Hd1080,
  
                  AudioSampleRate = MediaToolkit.Options.AudioSampleRate.Hz48000
  
              };
  
  
  
              using (var engine = new Engine())
  
              {
  
                  engine.Convert(inputFile, outputFile, conversionOptions);
  
              }

• FFMPEG RTSP stream to MPEG4/H264 file using libx264

  4 décembre 2016, par Phi

  Heyo folks,

  I’m attempting to transcode/remux an RTSP stream in H264 format into a MPEG4 container, containing just the H264 video stream. Basically, webcam output into a MP4 container.

  I can get a poorly coded MP4 produced, using this code :

  // Variables here for demo
  
  AVFormatContext * video_file_output_format = nullptr;
  
  AVFormatContext * rtsp_format_context = nullptr;
  
  AVCodecContext * video_file_codec_context = nullptr;
  
  AVCodecContext * rtsp_vidstream_codec_context = nullptr;
  
  AVPacket packet = {0};
  
  AVStream * video_file_stream = nullptr;
  
  AVCodec * rtsp_decoder_codec = nullptr;
  
  int errorNum = 0, video_stream_index = 0;
  
  std::string outputMP4file = "D:\\somemp4file.mp4";
  
  
  
  // begin
  
  AVDictionary * opts = nullptr;
  
  av_dict_set(&opts, "rtsp_transport", "tcp", 0);
  
  
  
  if ((errorNum = avformat_open_input(&rtsp_format_context, uriANSI.c_str(), NULL, &opts)) < 0) {
  
      errOut << "Connection failed: avformat_open_input failed with error " << errorNum << ":\r\n" << ErrorRead(errorNum);
  
      TacticalAbort();
  
      return;
  
  }
  
  
  
  rtsp_format_context->max_analyze_duration = 50000;
  
  if ((errorNum = avformat_find_stream_info(rtsp_format_context, NULL)) < 0) {
  
      errOut << "Connection failed: avformat_find_stream_info failed with error " << errorNum << ":\r\n" << ErrorRead(errorNum);
  
      TacticalAbort();
  
      return;
  
  }
  
  
  
  video_stream_index = errorNum = av_find_best_stream(rtsp_format_context, AVMEDIA_TYPE_VIDEO, -1, -1, NULL, 0);
  
  
  
  if (video_stream_index < 0) {
  
      errOut << "Connection in unexpected state; made a connection, but there was no video stream.\r\n"
  
          "Attempts to find a video stream resulted in error " << errorNum << ": " << ErrorRead(errorNum);
  
      TacticalAbort();
  
      return;
  
  }
  
  
  
  rtsp_vidstream_codec_context = rtsp_format_context->streams[video_stream_index]->codec;
  
  
  
  av_init_packet(&packet);
  
  
  
  if (!(video_file_output_format = av_guess_format(NULL, outputMP4file.c_str(),  NULL))) {
  
      TacticalAbort();
  
      throw std::exception("av_guess_format");
  
  }
  
  
  
  if (!(rtsp_decoder_codec = avcodec_find_decoder(rtsp_vidstream_codec_context->codec_id))) {
  
      errOut << "Connection failed: connected, but avcodec_find_decoder returned null.\r\n"
  
          "Couldn't find codec with an AV_CODEC_ID value of " << rtsp_vidstream_codec_context->codec_id << ".";
  
      TacticalAbort();
  
      return;
  
  }
  
  
  
  video_file_format_context = avformat_alloc_context();
  
  video_file_format_context->oformat = video_file_output_format;
  
  
  
  if (strcpy_s(video_file_format_context->filename, sizeof(video_file_format_context->filename), outputMP4file.c_str())) {
  
      errOut << "Couldn't open video file: strcpy_s failed with error " << errno << ".";
  
      std::string log = errOut.str();
  
      TacticalAbort();
  
      throw std::exception("strcpy_s");
  
  }
  
  
  
  if (!(video_file_encoder_codec = avcodec_find_encoder(video_file_output_format->video_codec))) {
  
      TacticalAbort();
  
      throw std::exception("avcodec_find_encoder");
  
  }
  
  
  
  // MARKER ONE
  
  
  
  if (!outputMP4file.empty() &&
  
      !(video_file_output_format->flags & AVFMT_NOFILE) &&
  
      (errorNum = avio_open2(&video_file_format_context->pb, outputMP4file.c_str(), AVIO_FLAG_WRITE, nullptr, &opts)) < 0) {
  
      errOut << "Couldn't open video file \"" << outputMP4file << "\" for writing : avio_open2 failed with error " << errorNum << ": " << ErrorRead(errorNum);
  
      TacticalAbort();
  
      return;
  
  }
  
  
  
  // Create stream in MP4 file
  
  if (!(video_file_stream = avformat_new_stream(video_file_format_context, video_file_encoder_codec))) {
  
      TacticalAbort();
  
      return;
  
  }
  
  
  
  AVCodecContext * video_file_codec_context = video_file_stream->codec;
  
  
  
  // MARKER TWO
  
  
  
  // error -22/-21 in avio_open2 if this is skipped
  
  if ((errorNum = avcodec_copy_context(video_file_codec_context, rtsp_vidstream_codec_context)) != 0) {
  
      TacticalAbort();
  
      throw std::exception("avcodec_copy_context");
  
  }
  
  
  
  //video_file_codec_context->codec_tag = 0;
  
  
  
  /*
  
  // MARKER 3 - is this not needed? Examples suggest not.
  
  if ((errorNum = avcodec_open2(video_file_codec_context, video_file_encoder_codec, &opts)) < 0)
  
  {
  
      errOut << "Couldn't open video file codec context: avcodec_open2 failed with error " << errorNum << ": " << ErrorRead(errorNum);
  
      std::string log = errOut.str();
  
      TacticalAbort();
  
      throw std::exception("avcodec_open2, video file");
  
  }*/
  
  
  
  //video_file_format_context->flags |= AVFMT_FLAG_GENPTS;
  
  if (video_file_format_context->oformat->flags & AVFMT_GLOBALHEADER)
  
  {
  
      video_file_codec_context->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
  }
  
  
  
  if ((errorNum = avformat_write_header(video_file_format_context, &opts)) < 0) {
  
      errOut << "Couldn't open video file: avformat_write_header failed with error " << errorNum << ":\r\n" << ErrorRead(errorNum);
  
      std::string log = errOut.str();
  
      TacticalAbort();
  
      return;
  
  }

  However, there are several issues :

  1. I can’t pass any x264 options to the output file. The output H264 matches the input H264’s profile/level - switching cameras to a different model switches H264 level.
  2. The timing of the output file is off, noticeably.
  3. The duration of the output file is off, massively. A few seconds of footage becomes hours, although playtime doesn’t match. (FWIW, I’m using VLC to play them.)

  Passing x264 options

  If I manually increment PTS per packet, and set DTS equal to PTS, it plays too fast, 2-3 seconds’ worth of footage in one second playtime, and duration is hours long. The footage also blurs past several seconds, about 10 seconds’ footage in a second.

  If I let FFMPEG decide (with or without GENPTS flag), the file has a variable frame rate (probably as expected), but it plays the whole file in an instant and has a long duration too (over forty hours for a few seconds). The duration isn’t "real", as the file plays in an instant.

  At Marker One, I try to set the profile by passing options to avio_open2. The options are simply ignored by libx264. I’ve tried :

  av_dict_set(&opts, "vprofile", "main", 0);
  
  av_dict_set(&opts, "profile", "main", 0); // error, missing '('
  
  // FF_PROFILE_H264_MAIN equals 77, so I also tried
  
  av_dict_set(&opts, "vprofile", "77", 0); 
  
  av_dict_set(&opts, "profile", "77", 0);

  It does seem to read the profile setting, but it doesn’t use them. At Marker Two, I tried to set it after the avio_open2, before avformat_write_header .

  // I tried all 4 av_dict_set from earlier, passing it to avformat_write_header.
  
  // None had any effect, they weren't consumed.
  
  av_opt_set(video_file_codec_context, "profile", "77", 0);
  
  av_opt_set(video_file_codec_context, "profile", "main", 0);
  
  video_file_codec_context->profile = FF_PROFILE_H264_MAIN;
  
  av_opt_set(video_file_codec_context->priv_data, "profile", "77", 0);
  
  av_opt_set(video_file_codec_context->priv_data, "profile", "main", 0);

  Messing with privdata made the program unstable, but I was trying anything at that point.
  I’d like to solve issue 1 with passing settings, since I imagine it’d bottleneck any attempt to solve issues 2 or 3.

  I’ve been fiddling with this for the better part of a month now. I’ve been through dozens of documentation, Q&As, examples. It doesn’t help that quite a few are outdated.

  Any help would be appreciated.

  Cheers