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• C++ ffmpeg mp4 to mp3 transcoding

  2 juillet 2014, par Unknown

  I am working on an application that converts the audio of mp4 video to mp3 files. I managed to compile ffmpeg with libmp3lame.

  This is what I have so far

  // Step 1 - Register all formats and codecs
  
  avcodec_register_all();
  
  av_register_all();
  
  
  
  AVFormatContext* fmtCtx = avformat_alloc_context();
  
  
  
  // Step 2 - Open input file, and allocate format context
  
  if(avformat_open_input(&fmtCtx, filePath.toLocal8Bit().data(), NULL, NULL) < 0)
  
      qDebug() << "Error while opening " << filePath;
  
  
  
  // Step 3 - Retrieve stream information
  
  if(avformat_find_stream_info(fmtCtx, NULL) < 0)
  
      qDebug() << "Error while finding stream info";
  
  
  
  // Step 4 - Find the audiostream
  
  int audioStreamIdx = -1;
  
  for(uint i=0; inb_streams; i++) {
  
      if(fmtCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  
          audioStreamIdx = i;
  
          break;
  
      }
  
  }
  
  
  
  if(audioStreamIdx != -1) {
  
      AVCodecContext *audioDecCtx = fmtCtx->streams[audioStreamIdx]->codec;
  
  
  
      // Step 5
  
      AVCodec *aCodec = avcodec_find_decoder(audioDecCtx->codec_id);
  
      avcodec_open2(audioDecCtx, aCodec, NULL);
  
  
  
      // Step 6
  
      AVOutputFormat* outputFormat = av_guess_format(NULL, outPath.toLocal8Bit().data(), NULL);
  
  
  
      // Step 7
  
      AVFormatContext *outformatCtx = avformat_alloc_context();
  
  
  
      // Step 8
  
      AVCodec *encoder = avcodec_find_encoder(AV_CODEC_ID_MP3);
  
      //encoder->sample_fmts = AV_SAMPLE_FMT_S16;
  
      AVCodecContext *audioEncCtx = avcodec_alloc_context3(encoder);
  
      audioEncCtx->sample_fmt = AV_SAMPLE_FMT_S16P;
  
      audioEncCtx->sample_rate = 44100;
  
      audioEncCtx->channel_layout = av_get_default_channel_layout(audioDecCtx->channels);
  
  
  
      avcodec_open2(audioEncCtx, encoder, NULL);
  
  
  
      // Step 9
  
      AVFrame *frame = av_frame_alloc();
  
  
  
      AVPacket pkt;
  
      av_init_packet(&pkt);
  
  
  
      // Step 10
  
      while(av_read_frame(fmtCtx, &pkt) == 0) {
  
          int got_frame = 0;
  
  
  
          if(pkt.stream_index == audioStreamIdx) {
  
              int len = avcodec_decode_audio4(audioDecCtx, frame, &got_frame, &pkt);
  
  
  
              if(got_frame) {
  
                  // Step 11
  
                  AVPacket outPkt;
  
                  av_init_packet(&outPkt);
  
  
  
                  int got_packet = 0;
  
  
  
                  qDebug() << "This is printed";
  
  
  
                  int error = avcodec_encode_audio2(audioEncCtx, &outPkt, frame, &got_packet);
  
  
  
                  qDebug() << "This is not printed...";
  
             }
  
          }
  
      }
  
  
  
      av_free_packet(&pkt);
  
  }
  
  
  
  avformat_close_input(&fmtCtx);

  It goes wrong at the line avcodec_encode_audio2(). It seems like the function does not return. As you can see, the line after the encoding line is not printed. What am I doing wrong ?

  Thanks in advance !

• vc1dsp : Change remaining stride parameters to ptrdiff_t

  29 mars 2022, par Martin Storsjö

  vc1dsp : Change remaining stride parameters to ptrdiff_t
  The existing x86 assembly for loop filters uses the stride as a
  
  full register without clearing/sign extending the upper half
  
  of the registers on x86_64.
  This avoids crashes if the caller would have passed nonzero bits
  
  in the previously undefined upper 32 bits of the parameters.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libavcodec/mips/vc1dsp_mips.h
  • [DH] libavcodec/mips/vc1dsp_mmi.c
  • [DH] libavcodec/vc1dsp.c
  • [DH] libavcodec/vc1dsp.h
  • [DH] libavcodec/x86/vc1dsp_init.c
  • [DH] libavcodec/x86/vc1dsp_loopfilter.asm

• Decoding audio with FFMPEG

  8 juillet 2012, par Mariano

  I am using FFMPEG to decode an audio file, but I am not able to do it.

      AVFormatContext *pFormatCtx;
  
  AVCodec *pCodec_audio;
  
  AVCodecContext *aCodecCtx_audio = NULL;
  
  AVPacket packet;
  
  string outfilename = "salida.avi";
  
  string filename = "john.wav";
  
  int audioStream;
  
  int out_size, len, size;
  
  FILE *outfile;
  
  int16_t *outbuf;
  
  uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
  
  avcodec_init();
  
  /* register all the codecs */
  
  av_register_all();
  
  
  
  if(av_open_input_file(&pFormatCtx, filename.c_str(), NULL, 0, NULL) != 0) {
  
      cerr << "Archivo no encontrado" << endl;
  
      return -1; // Couldn't open file
  
  }
  
  
  
  if(av_find_stream_info(pFormatCtx) < 0) {
  
      cerr << " No encontro el stream de info" << endl;
  
      return -1;// Couldn't find stream information
  
  }
  
  
  
  dump_format(pFormatCtx, 0, filename.c_str(), 0);
  
  
  
  for(unsigned int i=0; i < pFormatCtx->nb_streams; i++)
  
    if(pFormatCtx->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO)
  
        audioStream = i;
  
  aCodecCtx_audio = pFormatCtx->streams[audioStream]->codec;
  
  pCodec_audio = avcodec_find_decoder(aCodecCtx_audio->codec_id);
  
  if(pCodec_audio == NULL) {
  
      cerr<< "Unsupported codec!" << endl;
  
      return -1; // Codec not found
  
  }
  
  
  
  if(avcodec_open(aCodecCtx_audio, pCodec_audio) < 0) {
  
      cerr << "No se pudo abrir el codec de audio" << endl;
  
      return -1;
  
  }
  
  
  
  outbuf = (int16_t*) av_malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
  
  
  
  outfile = fopen(outfilename.c_str(), "wb");
  
  if (!outfile) {
  
      exit(1);
  
  }
  
  
  
  av_init_packet(&packet);
  
  packet.data = inbuf;
  
  while(av_read_frame(pFormatCtx, &packet) == 0) {
  
      if(packet.stream_index == audioStream) {
  
          size = packet.size;
  
          if (size == 0) {
  
              cerr << "Size = 0 " << endl;
  
              break;
  
          }
  
          while(size > 0) {
  
              out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
  
              len = avcodec_decode_audio3(aCodecCtx_audio, outbuf, &out_size , &packet);
  
              //av_free_packet(&packet);
  
              cout << len << endl;
  
              if(len == -1) {
  
                  cerr << "Error while decoding" << endl;
  
                  return 1;
  
              }
  
              if(out_size > 0) {
  
                  fwrite(outbuf, 1, out_size, outfile);
  
              }
  
              size -= len;
  
              packet.data += len;
  
          }
  
      }
  
  }
  
  av_free_packet(&packet);
  
  
  
  
  
  fclose(outfile);
  
  free(outbuf);
  
  cout << "END CODE" << endl;
  
  avcodec_close(aCodecCtx_audio);
  
  av_free(aCodecCtx_audio);

  In the code I am trying to decode an .wav file to a .avi file. The output file ("salida.avi") is created , but does not play.