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• Restream m3u8 using ffmpeg to rtmp but just audio

  6 juin 2021, par raisedhand

  I'm trying to re-stream m3u8 link to rtmp but just need to get audio in output without video.

  


  below code worked fine for restream m3u8 to rtmp but can't just get audio in output.

  


  CMD ffmpeg -re -i http://127.0.0.1:8080/stream.m3u8 -c:v copy -c:a aac -ar 44100 -ab 128k -ac 2 -strict -2 -flags +global_header -bsf:a aac_adtstoasc -bufsize 3000k -f flv rtmp :...

  


  My question is how to re-stream m3u8 to rtmp but just audio like radio station.

  


• Decode opus audio using ffmpeg lib not giving proper audio

  31 octobre 2018, par Harshpal Gosavi

  This is my code in this code all data is printing correct and there is no error but its not playing correctly. I am getting audio like radio voice. Human voice is not audible as it is continuosly playing radio like sound. I am not sure whether all values are correct or not. But if I change it then it wont work. Please help me how can I get proper audio voice.

  Opus.cpp

   {   void Opus::Decode1_working(int8_t *inBuf, int inLen , int16_t *outBuf, int *decodedLen)
  
  {
  
  
  
  
  
   AVPacket avpkt;
  
   AVFrame *decoded_frame = NULL;
  
  
  
   av_init_packet(&avpkt);
  
   avpkt.data = (uint8_t*)inBuf;
  
   avpkt.size = inLen;
  
  
  
  
  
   decoded_frame = av_frame_alloc();
  
   av_frame_unref(decoded_frame);
  
  int len = avcodec_send_packet(c, &avpkt);
  
   if (len < 0)
  
   {
  
        //Error
  
        return;
  
   }
  
   else
  
   {
  
       int retval = avcodec_receive_frame(c, decoded_frame);
  
       if(retval >= 0)
  
       {
  
           const AVCodecDescriptor *codesc = avcodec_descriptor_get(c->codec_id);
  
  
  
           int planeSize;
  
           int data_size = av_samples_get_buffer_size
  
                           (&planeSize,
  
                            c->channels,
  
                            decoded_frame->nb_samples,
  
                            c->sample_fmt,
  
                            1);
  
  
  
           if(data_size < AUDIO_MAXBUF_SIZE)
  
           {
  
               memcpy(outBuf, decoded_frame->data[0], (data_size));// * sizeof(uint8_t)));
  
               *decodedLen = data_size;
  
           }
  
           else
  
           {
  
              //Not copied
  
          }
  
  
  
       }
  
  
  
   }
  
   av_frame_free(&decoded_frame);
  
  }
  
  
  
  int Opus::init(void *args) {
  
  
  
  i_sampleRate = 0;
  
  i_channels = 0;
  
  
  
  avcodec_register_all();
  
  codec = avcodec_find_decoder(AV_CODEC_ID_OPUS);
  
   if(codec ==NULL)
  
   {
  
      //Codec not open
  
   }
  
  
  
  c= avcodec_alloc_context3(codec);
  
  
  
   c->sample_rate = 48000;
  
   c->channels = 1;
  
   c->bit_rate = 48000;
  
  
  
  
  
      av_opt_set_int(c, "refcounted_frames", 1, 0);
  
      int codecOpen = 0;
  
       codecOpen = avcodec_open2(c, codec,NULL);//,&audioOptionsDict) ;
  
       if (codecOpen < 0)
  
       {
  
           //return error;
  
       }
  
  
  
  return 0;
  
  }
  
  
  
  int Opus::decode(Packet *packet) {
  
  
  
  int8_t *data = new int8_t[packet->size];
  
   memcpy(data, packet->data, (packet->size * sizeof(int8_t)));
  
  
  
  long length = packet->size;
  
  
  
  int16_t *pcmdata = new int16_t[AUDIO_MAXBUF_SIZE];//[outLen];
  
  
  
  i_sampleRate = 48000;
  
  i_channels = 1;
  
  
  
  int decodedLen = 0;
  
  Decode1_working(data,length,pcmdata, &decodedLen);
  
  
  
  if (p_callback)
  
  {
  
      ++i_framesDecoded;
  
      p_callback->AfterAudioDecode((u8*)pcmdata,
  
                                   //length * sizeof(u8),
  
                                    decodedLen,//* sizeof(u8),
  
                                   packet->pts);
  
  }
  
  return 1;
  
  }
  
  
  
  }

• FFmpeg save stream to mp3

  17 avril 2015, par Sergey92zp

  I have an iOS project that play online radio streams, it is use FFmpeg to play. Also I added ability to record streams, decode streams via avcodec_decode_audio4 function, and write output to .wav file. But this files are too big, because it is uncompressed format, so I want to decode files to .mp3.

  I have found couple ways to convert audio but only when audio it is ready file, but I want decode to some compressed format as soon as I get chunk of data from stream, not ready file.

  Is it possible ?
  Can you give me some advise how to achieve this ?