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• Why does mplayer not play my stream ?

  17 juillet 2016, par SirWindfield

  I wanted to use mplayer (a command line tool) to play a SoundCloud url that I fetched with there official API. The url looks like this

  https://cf-media.sndcdn.com/qop35iYKlHhs.128.mp3?Policy=eyJTdGF0ZW1lbnQiOlt7IlJlc291cmNlIjoiKjovL2NmLW1lZGlhLnNuZGNkbi5jb20vcW9wMzVpWUtsSGhzLjEyOC5tcDMiLCJDb25kaXRpb24iOnsiRGF0ZUxlc3NUaGFuIjp7IkFXUzpFcG9jaFRpbWUiOjE0Njg3NTM4NjB9fX1dfQ__&Signature=OZ4IawmjdU6hEN8KB8e~t3QTy4lBFVBbIDg-a6tuPkb65e0mPskkCFFASkTWbyy5lsaT9IrURan6y70sZhLXPbng1IfkTsGdX1dO938NwVYnKs-BS7IL4TiVFxTBXoKJgmCmUD0qmmGmYqm3YdGZQrP~Sj~mw9~fmtwdHQu0rhl3O-dKsgk497JAR6pMorQs7BSs0XIosV1Mmv2DMD6iifquCWV9Ezq4ekneQ1gfSjVmzjjnKvsxjPpgmU~5DTdewwzlNClVxdzHSQONWM7c0YMGlBcVz97NviaeZOQSCAW8QZS59WULXJBJ9OmEPctWtpe3O0mo-GYjwEkbVGYl9A__&Key-Pair-Id=APKAJAGZ7VMH2PFPW6UQ

  I use the command "mplayer ’url’" to play it but I always get the output :

  libavformat version 57.25.100 (internal)
  
  https protocol not found, recompile FFmpeg with openssl, gnutls,
  
  or securetransport enabled.

  Looking which protocols have been installed tells me that I have https input and output support (ffmpeg -protocols) :

  ffmpeg version 3.1.1 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with Apple LLVM version 7.3.0 (clang-703.0.31)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.1.1 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libxvid --enable-libtheora --enable-libvorbis --enable-libvpx --enable-librtmp --enable-libfaac --enable-ffplay --enable-libspeex --enable-openssl --enable-libopus --disable-lzma --enable-nonfree --enable-vda
  
    libavutil      55. 28.100 / 55. 28.100
  
    libavcodec     57. 48.101 / 57. 48.101
  
    libavformat    57. 41.100 / 57. 41.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 47.100 /  6. 47.100
  
    libavresample   3.  0.  0 /  3.  0.  0
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  1.100 /  2.  1.100
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Supported file protocols:
  
  Input:
  
    async
  
    cache
  
    concat
  
    crypto
  
    data
  
    file
  
    ftp
  
    gopher
  
    hls
  
    http
  
    httpproxy
  
    https
  
    mmsh
  
    mmst
  
    pipe
  
    rtp
  
    srtp
  
    subfile
  
    tcp
  
    tls
  
    udp
  
    udplite
  
    unix
  
    rtmp
  
    rtmpe
  
    rtmps
  
    rtmpt
  
    rtmpte
  
  Output:
  
    crypto
  
    file
  
    ftp
  
    gopher
  
    http
  
    httpproxy
  
    https
  
    icecast
  
    md5
  
    pipe
  
    rtp
  
    srtp
  
    tcp
  
    tls
  
    udp
  
    udplite
  
    unix
  
    rtmp
  
    rtmpe
  
    rtmps
  
    rtmpt
  
    rtmpte

• 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

  


• 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 ?