Recherche avancée

Sur d’autres sites (6670)

• Decode .wav file and write it into another file using ffmpeg

  16 février 2017, par deshu

  How to decode a .wav file and write it into another file using ffmpeg ?

  I got decoded data by this piece of code :

  -(void)audioDecode:(const char *)outfilename inFileName:(const char *)filename
  
  
  
  {
  
  
  
  const char* input_filename=filename;
  
  
  
      //avcodec_register_all();
  
      av_register_all();
  
      avcodec_register_all();
  
     // av_register_all();
  
      //av_ini
  
  
  
  //    AVFormatContext* container=avformat_alloc_context();
  
  //    if(avformat_open_input(&container,filename,NULL,NULL)<0){
  
  //        NSLog(@"Could not open file");
  
  //    }
  
  
  
      AVCodec *codec;
  
      AVCodecContext *c= NULL;
  
      int len;
  
      FILE *f, *outfile;
  
      uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
  
      AVPacket avpkt;
  
      AVFrame *decoded_frame = NULL;
  
  
  
      av_init_packet(&avpkt);
  
  
  
      printf("Decode audio file %s to %s\n", filename, outfilename);
  
      avcodec_register_all();
  
  
  
      AVFrame* frame = av_frame_alloc();
  
      if (!frame)
  
      {
  
          fprintf(stderr, "Could not allocate audio frame\n");
  
                          exit(1);
  
      }
  
  
  
      AVFormatContext* formatContext = NULL;
  
      /* Opening the file, and check if it has opened */
  
      if (avformat_open_input(&formatContext, filename, NULL, NULL) != 0)
  
      {
  
          av_frame_free(&frame);
  
          NSLog(@"Could not open file");
  
      }
  
  
  
      if (avformat_find_stream_info(formatContext, NULL) < 0)
  
      {
  
          av_frame_free(&frame);
  
          avformat_close_input(&formatContext);
  
  
  
          NSLog(@"Error finding the stream info");
  
      }
  
  
  
          outfile = fopen(outfilename, "wb");
  
          if (!outfile) {
  
              av_free(c);
  
              exit(1);
  
          }
  
  
  
      /* Find the audio Stream, if no audio stream are found, clean and exit */
  
      AVCodec* cdc = NULL;
  
      int streamIndex = av_find_best_stream(formatContext, AVMEDIA_TYPE_AUDIO, -1, -1, &cdc, 0);
  
      if (streamIndex < 0)
  
      {
  
          av_frame_free(&frame);
  
          avformat_close_input(&formatContext);
  
          NSLog(@"Could not find any audio stream in the file");
  
          exit(1);
  
      }
  
  
  
      /* Open the audio stream to read data  in audioStream */
  
      AVStream* audioStream = formatContext->streams[streamIndex];
  
  
  
      /* Initialize the codec context */
  
      AVCodecContext* codecContext = audioStream->codec;
  
      codecContext->codec = cdc;
  
      /* Open the codec, and verify if it has opened */
  
      if (avcodec_open2(codecContext, codecContext->codec, NULL) != 0)
  
      {
  
          av_frame_free(&frame);
  
          avformat_close_input(&formatContext);
  
          NSLog(@"Couldn't open the context with the decoder");
  
          exit(1);
  
      }
  
  
  
      /* Initialize buffer to store compressed packets */
  
      AVPacket readingPacket;
  
      av_init_packet(&readingPacket);
  
      int lenght = 1;
  
      long long int chunk = ((formatContext->bit_rate)*lenght/8);
  
      int j=0;
  
      int count = 0;
  
      //audioChunk output;
  
  
  
  
  
      while(av_read_frame(formatContext, &readingPacket)==0){
  
          //if((count+readingPacket.size)>start){
  
              if(readingPacket.stream_index == audioStream->index){
  
  
  
                  AVPacket decodingPacket = readingPacket;
  
  
  
                  // Audio packets can have multiple audio frames in a single packet
  
                  while (decodingPacket.size > 0){
  
                      // Try to decode the packet into a frame
  
                      // Some frames rely on multiple packets, so we have to make sure the frame is finished before
  
                      // we can use it
  
                      int gotFrame = 0;
  
                      int result = avcodec_decode_audio4(codecContext, frame, &gotFrame, &decodingPacket);
  
  
  
                      count += result;
  
  
  
                      if (result >= 0 && gotFrame)
  
                      {
  
                          decodingPacket.size -= result;
  
                          decodingPacket.data += result;
  
                          int a;
  
  
  
                          for(int i=0;idata[0], 1, decodingPacket.size, outfile);
  
  //                            *(output.data+j)=frame->data[0][i];
  
  //
  
                              j++;
  
                              if(j>=chunk) break;
  
                          }
  
  
  
                          // We now have a fully decoded audio frame
  
                      }
  
                      else
  
                      {
  
                          decodingPacket.size = 0;
  
                          decodingPacket.data = NULL;
  
                      }
  
                     // if(j>=chunk) break;
  
                  }
  
              }
  
  //        }else count+=readingPacket.size;
  
  //        
  
  //        // To prevent memory leak, must free packet.
  
  //        av_free_packet(&readingPacket);
  
  //        if(j>=chunk) break;
  
      }
  
      fclose(outfile);

  But the file is created with zero bytes. I don’t know what’s wrong with this code. And I got one more error : "Format adp detected only with low score of 25, misdetection possible !"

• get audio current freqency for each 0.1 second of audio file with ffmpeg sox or opus into file.txt

  15 mars 2019, par Vlad

  I am trying to get audio freqency statistic from call record in file.opus for each 0,1 second, have try ffmpeg (spectrum) ffprobe ,
  for example
  sox  '.$file.' −n rate 6k spectrogram −z 62 -w Hamming -o aaaaaa/test10.png stat -freq 2>aaaaaa/test10.txt
  return spectrum
  all of them return values but no freqency found any other library ? Or what I am doing wrong I now it retrun spectrum but I need concrete freqency of human voice it is possible with ffmpeg or ffprobe or sox ? I am working in php but running shel comands any help will be aprisiatied. What should be my next step ? I am lost I have no idea how could get freqency from this numbers or "it is possible " ??? What is the first step comand in sox etc.

• converting .mov file to .h264 file

  29 août 2011, par Robin Rye

  ok, this is the case, i actually want to parse frames from a mov file. get the encoded h264 frames. and i've managed to do so by using ffmpeg but when i try to make a movie again by using ffmpeg -i test* test.mov i get test00: Invalid data found when processing input so there is something not correct with the structure of the frames. as i understand it a frame should have the following appearance :

  00 00 00 01 XX data -------------

  where XX is say whether it is a I-,P- or B-frame. or more specifically type(XX) = 0x0F && XX says if it is I(type(XX) = 5 ?),P(type(XX) = 7 ?) or B(type(XX) = 8 ?) frame. I'm not sure about these number, i've been looking for it but not found good sources. so that's question number one, what number should the NALU be for the different frames ?

  anyway, when i use av_read_frame on the mov file, i get frame that look like this :

  4B = size, 1B = XX and then data. (at least this is what i think i get)

  the files where i store the frames are always size long when i look at them in a hexeditor(otherwise as well of course). and XX is always 65(ie. type(XX) = 5) in the first and then 61(ie. type(XX) = 1) for a couple of frames and then back to being 65 for one frame and so on.

  i guess that these are frames like : I P P P P P P I P P P P P P P I P P P P P P P .... however then my assumption about the type numbers for the different frame types are false, which is highly likely. (any suggestion on reading about this ? except the ISO/IEC 14496-10, i don't understand it really).

  I've tried to remove the size and append 00 00 00 01 before the XX byte and the data but without success. any tips on how i could modify the frames to be valid H264 encoded frames ?