Recherche avancée

Sur d’autres sites (6706)

• Anomalie #3921 : verifier url site

  23 mars 2017, par jluc -

  Oui, super ! ça marche bien avec demo.spip.net

  ou bien au choix :

  Oui, désolé ! excusez le dérangement...

• FFMPEG AAC encoding causes audio to be lower in pitch

  14 février 2017, par Paul Knopf

  I built a sample application that encodes AAC (from PortAudio) into a MP4 container (no video stream).

  The resulting audio is lower in pitch.

  #include "stdafx.h"
  
  #include "TestRecording.h"
  
  #include "libffmpeg.h"
  
  
  
  TestRecording::TestRecording()
  
  {
  
  }
  
  
  
  
  
  TestRecording::~TestRecording()
  
  {
  
  }
  
  
  
  struct RecordingContext
  
  {
  
      RecordingContext()
  
      {
  
          formatContext = NULL;
  
          audioStream = NULL;
  
          audioFrame = NULL;
  
          audioFrameframeNumber = 0;
  
      }
  
  
  
      libffmpeg::AVFormatContext* formatContext;
  
      libffmpeg::AVStream* audioStream;
  
      libffmpeg::AVFrame* audioFrame;
  
      int audioFrameframeNumber;
  
  };
  
  
  
  static int AudioRecordCallback(const void *inputBuffer, void *outputBuffer,
  
      unsigned long framesPerBuffer,
  
      const PaStreamCallbackTimeInfo* timeInfo,
  
      PaStreamCallbackFlags statusFlags,
  
      void *userData)
  
  {
  
      RecordingContext* recordingContext = (RecordingContext*)userData;
  
  
  
      libffmpeg::avcodec_fill_audio_frame(recordingContext->audioFrame,
  
          recordingContext->audioFrame->channels,
  
          recordingContext->audioStream->codec->sample_fmt,
  
          static_cast<const unsigned="unsigned">(inputBuffer),
  
          (framesPerBuffer * sizeof(float) * recordingContext->audioFrame->channels),
  
          0);
  
  
  
      libffmpeg::AVPacket pkt;
  
      libffmpeg::av_init_packet(&amp;pkt);
  
      pkt.data = NULL;
  
      pkt.size = 0;
  
  
  
      int gotpacket;
  
      int result = avcodec_encode_audio2(recordingContext->audioStream->codec, &amp;pkt, recordingContext->audioFrame, &amp;gotpacket);
  
  
  
      if (result &lt; 0)
  
      {
  
          LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't encode the audio frame to acc");
  
          return paContinue;
  
      }
  
  
  
      if (gotpacket)
  
      {
  
          pkt.stream_index = recordingContext->audioStream->index;
  
          recordingContext->audioFrameframeNumber++;
  
  
  
          // this codec requires no bitstream filter, just send it to the muxer!
  
          result = libffmpeg::av_write_frame(recordingContext->formatContext, &amp;pkt);
  
          if (result &lt; 0)
  
          {
  
              LOG(ERROR) &lt;&lt; "Couldn't write the encoded audio frame";
  
              libffmpeg::av_free_packet(&amp;pkt);
  
              return paContinue;
  
          }
  
  
  
          libffmpeg::av_free_packet(&amp;pkt);
  
      }
  
  
  
      return paContinue;
  
  }
  
  
  
  static bool InitializeRecordingContext(RecordingContext* recordingContext)
  
  {
  
      int result = libffmpeg::avformat_alloc_output_context2(&amp;recordingContext->formatContext, NULL, NULL, "C:\\Users\\Paul\\Desktop\\test.mp4");
  
      if (result &lt; 0)
  
      {
  
          LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't create output format context");
  
          return false;
  
      }
  
  
  
      libffmpeg::AVCodec *audioCodec;
  
      audioCodec = libffmpeg::avcodec_find_encoder(libffmpeg::AV_CODEC_ID_AAC);
  
      if (audioCodec == NULL)
  
      {
  
          LOG(ERROR) &lt;&lt; "Couldn't find the encoder for AAC";
  
      }
  
  
  
      recordingContext->audioStream = libffmpeg::avformat_new_stream(recordingContext->formatContext, audioCodec);
  
      if (!recordingContext->audioStream)
  
      {
  
          LOG(ERROR) &lt;&lt; "Couldn't create the audio stream";
  
          return false;
  
      }
  
  
  
      recordingContext->audioStream->codec->bit_rate = 64000;
  
      recordingContext->audioStream->codec->sample_fmt = libffmpeg::AV_SAMPLE_FMT_FLTP;
  
      recordingContext->audioStream->codec->sample_rate = 48000;
  
      recordingContext->audioStream->codec->channel_layout = AV_CH_LAYOUT_STEREO;
  
      recordingContext->audioStream->codec->channels = libffmpeg::av_get_channel_layout_nb_channels(recordingContext->audioStream->codec->channel_layout);
  
  
  
      recordingContext->audioStream->codecpar->bit_rate = recordingContext->audioStream->codec->bit_rate;
  
      recordingContext->audioStream->codecpar->format = recordingContext->audioStream->codec->sample_fmt;
  
      recordingContext->audioStream->codecpar->sample_rate = recordingContext->audioStream->codec->sample_rate;
  
      recordingContext->audioStream->codecpar->channel_layout = recordingContext->audioStream->codec->channel_layout;
  
      recordingContext->audioStream->codecpar->channels = recordingContext->audioStream->codec->channels;
  
  
  
      result = libffmpeg::avcodec_open2(recordingContext->audioStream->codec, audioCodec, NULL);
  
      if (result &lt; 0)
  
      {
  
          LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't open the audio codec");
  
          return false;
  
      }
  
  
  
      // create a new frame to store the audio samples
  
      recordingContext->audioFrame = libffmpeg::av_frame_alloc();
  
      if (!recordingContext->audioFrame)
  
      {
  
          LOG(ERROR) &lt;&lt; "Couldn't alloce the output audio frame";
  
          return false;
  
      }
  
  
  
      recordingContext->audioFrame->nb_samples = recordingContext->audioStream->codec->frame_size;
  
      recordingContext->audioFrame->channel_layout = recordingContext->audioStream->codec->channel_layout;
  
      recordingContext->audioFrame->channels = recordingContext->audioStream->codec->channels;
  
      recordingContext->audioFrame->format = recordingContext->audioStream->codec->sample_fmt;
  
      recordingContext->audioFrame->sample_rate = recordingContext->audioStream->codec->sample_rate;
  
  
  
      result = libffmpeg::av_frame_get_buffer(recordingContext->audioFrame, 0);
  
      if (result &lt; 0)
  
      {
  
          LOG(ERROR) &lt;&lt; "Coudln't initialize the output audio frame buffer";
  
          return false;
  
      }
  
  
  
      // some formats want video_stream headers to be separate  
  
      if (!strcmp(recordingContext->formatContext->oformat->name, "mp4") || !strcmp(recordingContext->formatContext->oformat->name, "mov") || !strcmp(recordingContext->formatContext->oformat->name, "3gp"))
  
      {
  
          recordingContext->audioStream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
  
      }
  
  
  
      // open the ouput file
  
      if (!(recordingContext->formatContext->oformat->flags &amp; AVFMT_NOFILE))
  
      {
  
          result = libffmpeg::avio_open(&amp;recordingContext->formatContext->pb, recordingContext->formatContext->filename, AVIO_FLAG_WRITE);
  
          if (result &lt; 0)
  
          {
  
              LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't open the output file");
  
              return false;
  
          }
  
      }
  
  
  
      // write the stream headers
  
      result = libffmpeg::avformat_write_header(recordingContext->formatContext, NULL);
  
      if (result &lt; 0)
  
      {
  
          LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't write the headers to the file");
  
          return false;
  
      }
  
  
  
      return true;
  
  }
  
  
  
  static bool FinalizeRecordingContext(RecordingContext* recordingContext)
  
  {
  
      int result = 0;
  
  
  
      // write the trailing information
  
      if (recordingContext->formatContext->pb)
  
      {
  
          result = libffmpeg::av_write_trailer(recordingContext->formatContext);
  
          if (result &lt; 0)
  
          {
  
              LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't write the trailer information");
  
              return false;
  
          }
  
      }
  
  
  
      // close all the codes
  
      for (int i = 0; i &lt; (int)recordingContext->formatContext->nb_streams; i++)
  
      {
  
          result = libffmpeg::avcodec_close(recordingContext->formatContext->streams[i]->codec);
  
          if (result &lt; 0)
  
          {
  
              LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't close the codec");
  
              return false;
  
          }
  
      }
  
  
  
      // close the output file
  
      if (recordingContext->formatContext->pb)
  
      {
  
          if (!(recordingContext->formatContext->oformat->flags &amp; AVFMT_NOFILE))
  
          {
  
              result = libffmpeg::avio_close(recordingContext->formatContext->pb);
  
              if (result &lt; 0)
  
              {
  
                  LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't close the output file");
  
                  return false;
  
              }
  
          }
  
      }
  
  
  
      // free the format context and all of its data
  
      libffmpeg::avformat_free_context(recordingContext->formatContext);
  
  
  
      recordingContext->formatContext = NULL;
  
      recordingContext->audioStream = NULL;
  
  
  
      if (recordingContext->audioFrame)
  
      {
  
          libffmpeg::av_frame_free(&amp;recordingContext->audioFrame);
  
          recordingContext->audioFrame = NULL;
  
      }
  
  
  
      return true;
  
  }
  
  
  
  int TestRecording::Test()
  
  {
  
      PaError result = paNoError;
  
  
  
      result = Pa_Initialize();
  
      if (result != paNoError) LOGINT_WITH_MESSAGE(ERROR, result, "Error initializing audio device framework");
  
  
  
      RecordingContext recordingContext;
  
      if (!InitializeRecordingContext(&amp;recordingContext))
  
      {
  
          LOG(ERROR) &lt;&lt; "Couldn't start recording file";
  
          return 0;
  
      }
  
  
  
      auto defaultDevice = Pa_GetDefaultInputDevice();
  
      auto deviceInfo = Pa_GetDeviceInfo(defaultDevice);
  
  
  
      PaStreamParameters  inputParameters;
  
      inputParameters.device = defaultDevice;
  
      inputParameters.channelCount = 2;
  
      inputParameters.sampleFormat = paFloat32;
  
      inputParameters.suggestedLatency = deviceInfo->defaultLowInputLatency;
  
      inputParameters.hostApiSpecificStreamInfo = NULL;
  
  
  
      PaStream* stream = NULL;
  
      result = Pa_OpenStream(
  
          &amp;stream,
  
          &amp;inputParameters,
  
          NULL,
  
          48000,
  
          1024,
  
          paClipOff,
  
          AudioRecordCallback,
  
          &amp;recordingContext);
  
      if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't open the audio stream");
  
  
  
      result = Pa_StartStream(stream);
  
      if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't start the audio stream");
  
  
  
      Sleep(1000 * 5);
  
  
  
      result = Pa_StopStream(stream);
  
      if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't stop the audio stream");
  
  
  
      if (!FinalizeRecordingContext(&amp;recordingContext)) LOG(ERROR) &lt;&lt; "Couldn't stop recording file";
  
  
  
      result = Pa_CloseStream(stream);
  
      if (result != paNoError)LOGINT_WITH_MESSAGE(ERROR, result, "Couldn't stop the audio stream");
  
  
  
      return 0;
  
  }
  
  </const>

  Here is the stdout, in case it helps.

  https://gist.github.com/pauldotknopf/9f24a604ce1f8a081aa68da1bf169e98

  Why is the audio lower in pitch ? I assume I am overlooking a parameter that needs to be configured between PortAudio and FFMPEG. Is there something super obvious that I am missing ?

• Documentation #3878 (Fermé) : generer_url_action() ne prend pas de tableau comme argument

  7 février 2017, par b b

  Super, merci, on ferme :)