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• convert pcm stream data to encoded aac data

  24 décembre 2019, par Kumar

  I tried to convert pulse-audio pcm stream data to aac encoded data using ffmpeg.
  But after encoding I get noise-full data, not the correct one. Here I post my code, anyone help me with some ideas.

  Initial configuration :

      av_register_all();
  
  
  
      int error;
  
      if ((error = avio_open(&output_io_context,"out.aac",AVIO_FLAG_WRITE))<0) {
  
              printf("could not open output file\n");
  
      }
  
  
  
      if (!(output_format_context = avformat_alloc_context())) {
  
              printf("output_format_context error\n");
  
      }
  
  
  
      output_format_context->pb = output_io_context;
  
  
  
      if(!(output_format_context->oformat = av_guess_format(NULL, "out.aac", NULL))) {
  
              printf("guess format error\n");
  
      }
  
  
  
      codec = avcodec_find_encoder(AV_CODEC_ID_AAC);
  
      if (codec == NULL) {
  
              printf("avcodec_find_encoder: ERROR\n");
  
      }
  
  
  
      if (!(stream = avformat_new_stream(output_format_context, NULL))) {
  
              printf("stream create error\n");
  
      }
  
      output_codec_context = avcodec_alloc_context3(codec);
  
      if(!output_codec_context) {
  
              printf("output_codec_context is null\n");
  
      }
  
  
  
      output_codec_context->channels       = CHANNELS;
  
      output_codec_context->channel_layout = av_get_default_channel_layout(CHANNELS);
  
      output_codec_context->sample_rate    = SAMPLE_RATE; //input_codec_context->sample_rate;
  
      output_codec_context->sample_fmt     = codec->sample_fmts[0];
  
      output_codec_context->bit_rate       = 48000; //OUTPUT_BIT_RATE;
  
  
  
      stream->time_base.den = SAMPLE_RATE;//input_codec_context->sample_rate;
  
      stream->time_base.num = 1;
  
  
  
      if(output_format_context->oformat->flags & AVFMT_GLOBALHEADER)
  
              output_codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
  
  
  
      if ((error = avcodec_open2(output_codec_context, codec, NULL)) < 0) {
  
          printf("error");
  
      }
  
  
  
      error = avcodec_parameters_from_context(stream->codecpar, output_codec_context);
  
  
  
      if (write_output_file_header(output_format_context)) {
  
              printf("write header failure...\n");
  
      }

  Data encoding :

  AVFrame *output_frame;
  
  int frame_pos = 0, ctx_frame_size = output_codec_context->frame_size;
  
  int size = av_samples_get_buffer_size(NULL, CHANNELS,
  
                output_codec_context->frame_size,output_codec_context->sample_fmt, 1);
  
  if((x =  avcodec_fill_audio_frame(output_frame, CHANNELS,
  
                 output_codec_context->sample_fmt, data, length, 1)) < 0) {
  
        printf("avcodec_fill_audio_frame error : %s\n", av_err2str(x));
  
  }
  
  
  
   int data_written;
  
   if (encode_audio_frame(output_frame, output_format_context,
  
              output_codec_context, &data_written)) {
  
          printf("encode_audio_frame error\n");
  
    }
  
  
  
    av_frame_free(&output_frame);

  helper_function :

   int encode_audio_frame(AVFrame *frame,AVFormatContext *output_format_context,
  
                AVCodecContext *output_codec_context, int *data_present)
  
   {
  
     AVPacket output_packet;
  
     int error;
  
     init_packet(&output_packet);
  
  
  
     if (frame) {
  
        frame->pts = pts;
  
        pts += frame->nb_samples;
  
     }
  
  
  
     error = avcodec_send_frame(output_codec_context, frame);
  
     if (error == AVERROR_EOF) {
  
       error = 0;
  
       goto cleanup;
  
     } else if (error < 0) {
  
        fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
  
              av_err2str(error));
  
        return error;
  
     }
  
  
  
     error = avcodec_receive_packet(output_codec_context, &output_packet);
  
     if (error == AVERROR(EAGAIN)) {
  
       error = 0;
  
       goto cleanup;
  
     } else if (error == AVERROR_EOF) {
  
        error = 0;
  
        goto cleanup;
  
     } else if (error < 0) {
  
        fprintf(stderr, "Could not encode frame (error '%s')\n",
  
              av_err2str(error));
  
        goto cleanup;
  
     } else {
  
        *data_present = 1;
  
    }
  
  
  
    if (*data_present &&
  
        (error = av_write_frame(output_format_context, &output_packet)) < 0) {
  
        fprintf(stderr, "Could not write frame (error '%s')\n",
  
               av_err2str(error));
  
        goto cleanup;
  
    }
  
  
  
    cleanup:
  
        av_packet_unref(&output_packet);
  
        return error;
  
   }

  • Do we need to fill AVFrame with sizeof(av_samples_get_buffer_size) or context->frame_size ?
    

  TYIA :) !!

• How to save synthesized audio during streaming distribution

  7 avril 2022, par its-ogawa

  I would like to stream a composite of microphone audio and digital sound sources by ffmpeg and save the delivery to an m3u8 file.

  


  Below are the commands I have actually tried.

  


  ffmpeg -rtbufsize 100M -f dshow -i video=<my webcam="webcam">:audio=<my microphone="microphone"> -re -stream_loop -1 -i <my sound="sound" source="source"> -filter_complex "[0]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,adelay=2100|2100,volume@voice=volume=10dB[voice],[1]aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo,azmq,volume@bgm=volume=0.2[bgm],[voice][bgm]amerge=inputs=2[out]" -map 0:v -map [out]:a -f mpegts -flush_packets 0 udp://XXX.XXX.XXX.XXX:XXX?pkt_size=1316 -f hls -hls_time 5 hls.m3u8&#xA;</my></my></my>

  


  I have played an m3u8 created this way, but I cannot hear the digital sound source that I am supposed to have synthesized.
Even though the synthesized audio was audible when streamed.

  


  Perhaps you need to set up something like -map 0:v -map [out]:a when saving to the m3u8 file, in which case the Output with label 'out' does not exist in any defined filter graph, or was already used elsewhere. message appears and does not work.

  


• Python ffmpeg does not save the mp4 clips to file

  30 janvier 2021, par oo92

  I'm using the ffmpeg Python library to save 60 second mp4 clips to file from Twitch livestreams using its API. This is my code :

  


  current_dir = os.getcwd()

client = TwitchClient(client_id='')
streams_now = client.streams.get_live_streams(limit=100, offset=900)
epoch = str(math.ceil(time.time()))

if not os.path.exists(current_dir + '/twitch_videos/'):
    os.mkdir(current_dir + '/twitch_videos/')

for i in range(0, 1):
    try:
        username = streams_now[i]['channel']['name']
        id = streams_now[i]['id']
        game = streams_now[i]['game']
        game = game.translate(str.maketrans({':': '-', ' ': '-', "'": '', '!': '', '&': '_', '.': '', '+': '_'}))
        streaming = streamlink.streams('http://twitch.tv/' + username)
        stream = streaming["best"].url

        twitch_stream = ffmpeg.input(stream)

        twitch_stream = ffmpeg.filter(twitch_stream,
                                      'fps',
                                      fps=1,
                                      round='up')

        twitch_stream = ffmpeg.output(twitch_stream,
                                      current_dir + '/twitch_videos/' + '%d_' + username + '_' + epoch + '.mp4',
                                      sc_threshold='0',
                                      g='60',
                                      f='segment',
                                      segment_time='600',
                                      segment_format_options='movflags=+faststart',
                                      reset_timestamps='1')frl3dqgn21bbpp6tajjvg5pdevczac
        ffmpeg.run(twitch_stream)


    except:
        pass


  


  I am concatenating the path it should go to, to the name of the file. But the folder is empty after I take a look.