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• FFmpeg realtime volume changing with C API

  9 janvier 2019, par Tank2006

  My environment is FFmpeg 4.1(Prebuild package from an official site) on Windows 10/Visual Studio 2017.

  I want to change an audio volume as realtime according to a volume button operation from media player apps.

  Think simply, it will update if AVFilter parameters changes, but it seems doesn’t work.

  const char src[] = "C:\\sample.mp3";
  
  
  
  AVFilterGraph *graph = NULL;
  
  AVFilterContext *ctx_src, *ctx_sink;
  
  AVFilter *ctx_vol;
  
  
  
  int main()
  
  {
  
      int res;
  
      AVPacket *packet = av_packet_alloc();
  
      AVFrame *frame = av_frame_alloc();
  
      AVFrame *fframe = av_frame_alloc();
  
  
  
      AVCodec *codec = avcodec_find_decoder(AV_CODEC_ID_MP3);
  
      AVCodecParser *parser = av_parser_init(codec->id);
  
  
  
      AVCodecContext *cc = avcodec_alloc_context3(codec);
  
      avcodec_open2(cc, codec, NULL);
  
  
  
  
  
      FILE *fp; uint8_t buffer[1024]; int filterinit = 0;
  
      fopen_s(&fp, src, "rb");
  
      while (feof(fp) == 0) {
  
          int read = fread(buffer, 1, 1024, fp);
  
  
  
          res = av_parser_parse2(parser, cc, &packet->data, &packet->size,
  
              buffer, read, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
  
          if (packet->size) {
  
              res = avcodec_send_packet(cc, packet);
  
  
  
              while (res >= 0) {
  
                  res = avcodec_receive_frame(cc, frame);
  
  
  
                  if (res == AVERROR(EAGAIN) || res == AVERROR_EOF) {
  
                      break;
  
                  } else if (res < 0) {
  
                      fprintf(stderr, "Error during decoding\n");
  
                      exit(1);
  
                  }
  
  
  
                  if (filterinit == 0) {
  
                      // Create a volume filter, links and graph from AVCodecContext's parameter
  
                      if(init_filters(cc)) filterinit = 1;
  
                  }
  
  
  
                  if (filterinit == 1) {
  
                      res = av_buffersrc_add_frame(ctx_src, frame);
  
                      if(av_buffersink_get_frame(ctx_sink, frame) >= 0) {
  
  
  
                          // Change the volume realtime
  
                          av_opt_set(ctx_vol, "volume", AV_STRINGIFY(1.2), AV_OPT_SEARCH_CHILDREN);
  
  
  
  
  
                          int datasize = av_get_bytes_per_sample(cc->sample_fmt);
  
                          for (int i = 0; i < frame->nb_samples; i++) {
  
                              for (int ch = 0; ch < cc->channels; ch++) {
  
                                  //fwrite(frame->data[ch] + data_size * i, 1, data_size, outfile);
  
                              }
  
                          }
  
                      }
  
                  }
  
              }
  
          }
  
      }
  
  
  
      fclose(fp);
  
  
  
      avcodec_free_context(&cc);
  
      av_parser_close(parser);
  
      av_frame_free(&frame);
  
      av_frame_free(&fframe);
  
      av_packet_free(&packet);
  
  
  
      avfilter_graph_free(&graph);
  
  
  
      return 0;
  
  }

  Can I change the filter value in realtime with FFmpeg’s C API or I need to create a new filter link each time when it requires to update ?

• FFmpeg Has A Native VP8 Decoder

  24 juin 2010, par Multimedia Mike — VP8

  Thanks to David Conrad and Ronald Bultje who committed their native VP8 video decoder to the FFmpeg codebase yesterday. At this point, it can decode 14/17 of the VP8 test vectors that Google released during the initial open sourcing event. Work is ongoing on those 3 non-passing samples (missing bilinear filter). Meanwhile, FFmpeg’s optimization-obsessive personalities are hard at work optimizing the native decoder. The current decoder is already profiled to be faster than Google/On2’s official libvpx.

  Testing
  So it falls to FATE to test this on the ridiculous diversity of platforms that FFmpeg supports. I staged individual test specs for each of the 17 test vectors : vp8-test-vector-001 ... vp8-test-vector-017. After the samples have propagated through to the various FATE installations, I’ll activate the 14 test specs that are currently passing.

  Initial Testing Methodology
  Inspired by Ronald Bultje’s idea, I built the latest FFmpeg-SVN with libvpx enabled. Then I selected between the reference and native decoders as such :

  $ for i in 001 002 003 004 005 006 007 008 009 \
   010 011 012 013 014 015 016 017
  do
    echo vp80-00-comprehensive-$i.ivf
    ffmpeg -vcodec libvpx -i \
      /path/to/vp8-test-vectors-r1/vp80-00-comprehensive-$i.ivf \
      -f framemd5 - 2> /dev/null
  done > refs.txt
  $ for i in 001 002 003 004 005 006 007 008 009 \
  
   010 011 012 013 014 015 016 017
  
  do
  
    echo vp80-00-comprehensive-$i.ivf
  
    ffmpeg -vcodec vp8 -i \
  
      /path/to/vp8-test-vectors-r1/vp80-00-comprehensive-$i.ivf \
  
      -f framemd5 - 2> /dev/null
  
  done > native.txt
  $ diff -u refs.txt native.txt
  

  That reveals precisely which files differ.

• convert WAV to TETRA format

  14 juin 2020, par Ashish Arora

  I am trying to convert a wav file into TETRA encoded file (https://en.wikipedia.org/wiki/Terrestrial_Trunked_Radio). Tetra is used by fire-fighters, it provides a radio-like voice.

  



  I am trying to use the official tetra codec codes available at (https://www.etsi.org/deliver/etsi_en/300300_300399/30039502/01.03.01_60/) and we can easily compile it using the scripts available at https://github.com/sq5bpf/install-tetra-codec.

  



  However, I am not able to figure out how to convert a wav file into tetra codec files using these files. I tried going through the documentation of the compiled files (ccoder, cdecoder, scoder, sdecoder).

  



  I tried the following command -

  



  


  tetra/bin/scoder input.wav serial_file synth_file

  


  



  here serial_file and synth_file are the output files and have following documentation in the scoder.c file :

  



      INPUT   :   - Description : speech file to be analyzed
                - Format : binary file 16 bit-samples
                  240 samples per frame

serial_file :   - Description : serial stream output file 
            - Format : binary file 16 bit-samples
              each 16 bit-sample represents one encoded bit
              138 (= 1 + 137) bits per frame

synth_file  :   - Description : local synthesis output file 
            - Format : binary file 16 bit-samples


  



  For an input file of size 13M, I obtained serial_file and synth_file of size 8.0M and 16M. However, I thought since the wav file is getting converted into a walkie-talkie type signal the output file size will be alot smaller.

  



  I want to clarify if :

  



  

  • I used the correct code to convert an input wav file into a tetra format output file.

  


  • can you please help me understand, what is serial_file and synth_file.

  


  



  Thanks,
Ashish