Android国标接入端如何播放GB28181平台端语音广播数据

Posted 2022-10-08

GB28181语音广播这块，我们依据GB/T28181-2016针对流程和实例代码，做过详细的描述，本次主要是探讨下，广播数据过来后，如何处理。

鉴于我们之前有非常成熟的RTMP|RTSP低延迟播放模块，语音广播数据过来后，调用startAudioPlay()，ntsOnInviteAudioBroadcastResponse()处理如下：

@Override
public void ntsOnInviteAudioBroadcastResponse(String sourceID, String targetID, int 
  handler_.postDelayed(new Runnable() 
    @Override
    public void run() 
      Log.i(TAG, "ntsOnInviteAudioBroadcastResponse, statusCode:" + status_code_ +" sourceID:" + source_id_ + ", targetID:" + target_id_);

      boolean is_need_destory_rtp = true;

      if (gb28181_agent_ != null ) 
        boolean is_need_bye = 200==status_code_;

        if (200 == status_code_ && session_description_ != null && rtp_receiver_handle_ != 0 ) 
          MediaSessionDescription audio_des = session_description_.getAudioDescription();

          SDPRtpMapAttribute audio_attr = null;
          if (audio_des != null && audio_des.getRtpMapAttributes() != null && !audio_des.getRtpMapAttributes().isEmpty() )
            audio_attr = audio_des.getRtpMapAttributes().get(0);

          if ( audio_des != null && audio_attr != null ) 
            lib_player_.SetRTPReceiverSSRC(rtp_receiver_handle_, audio_des.getSSRC());

            lib_player_.SetRTPReceiverPayloadType(rtp_receiver_handle_, audio_attr.getPayloadType(),
                audio_attr.getEncodingName(), 2, audio_attr.getClockRate());

            // 如果是PCMA, SDK会默认填 采样率8000, 通道1, 其他音频编码需要手动填入
            // lib_player_.SetRTPReceiverAudiosamplingRate(rtp_receiver_handle_, 8000);
            // lib_player_.SetRTPReceiverAudioChannels(rtp_receiver_handle_, 1);

            lib_player_.SetRTPReceiverRemoteAddress(rtp_receiver_handle_, audio_des.getAddress(), audio_des.getPort());
            lib_player_.InitRTPReceiver(rtp_receiver_handle_);

            if (startAudioPlay()) 
              is_need_bye = false;
              is_need_destory_rtp = false;

              gb_broadcast_source_id_ = source_id_;
              gb_broadcast_target_id_ = target_id_;
              btnGB28181AudioBroadcast.setText("终止GB28181语音广播");
              btnGB28181AudioBroadcast.setEnabled(true);
            
          

         else 
          btnGB28181AudioBroadcast.setText("GB28181语音广播");
        

        if (is_need_bye)
          gb28181_agent_.byeAudioBroadcast(source_id_, target_id_);
      

      if (is_need_destory_rtp)
        destoryRTPReceiver();
    

    private String source_id_;
    private String target_id_;
    private int status_code_;
    private PlaySessionDescription session_description_;

    public Runnable set(String source_id, String target_id, int 
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      this.status_code_ = status_code;
      this.session_description_ = session_description;
      return this;
    

  .set(sourceID, targetID, statusCode, sessionDescription),0);

startAudioPlay()初始化实例后，为了保证低延迟，拉流端设置0 buffer，处于调试方便，设置download speed回调2-5秒一次（可以看到是不是有音频数据过来），由于只需要播放音频，不需要视频，所以不要设置surface下去，然后设置拉流数据回调，需要注意的是，拉到的audio数据，不要转aac输出：

private boolean startAudioPlay() 
  if (player_handle_ != 0 )
    return false;

  player_handle_ = lib_player_.SmartPlayerOpen(context_);
  if (player_handle_ == 0)
    return false;

  lib_player_.SetSmartPlayerEventCallbackV2(player_handle_,new EventHandePlayerV2());

  // 缓存大小可以调整
  lib_player_.SmartPlayerSetBuffer(player_handle_, 0);

  // lib_player_.SmartPlayerSetFastStartup(player_handle_, 0);

  // set report download speed(默认2秒一次回调 用户可自行调整report间隔)
  lib_player_.SmartPlayerSetReportDownloadSpeed(player_handle_, 1, 20);

  lib_player_.SmartPlayerClearRtpReceivers(player_handle_);
  lib_player_.SmartPlayerAddRtpReceiver(player_handle_, rtp_receiver_handle_);

  lib_player_.SmartPlayerSetSurface(player_handle_, null);
  // lib_player_.SmartPlayerSetRenderScaleMode(player_handle_, 1);

  lib_player_.SmartPlayerSetAudioOutputType(player_handle_, 1);

  lib_player_.SmartPlayerSetMute(player_handle_, 0);

  lib_player_.SmartPlayerSetAudioVolume(player_handle_, 100);

  lib_player_.SmartPlayerSetExternalAudioOutput(player_handle_, new PlayerExternalPCMOutput());

  lib_player_.SmartPlayerSetUrl(player_handle_, "rtp://ntinternal/rtpreceiver/implemention0");

  if (0 != lib_player_.SmartPlayerStartPlay(player_handle_)) 
    lib_player_.SmartPlayerClose(player_handle_);
    player_handle_ = 0;

    Log.e(TAG,  "start audio paly failed");
    return false;
  

  lib_player_.SmartPlayerSetAudioDataCallback(player_handle_, new PlayerAudioDataOutput());
  lib_player_.SmartPlayerSetPullStreamAudioTranscodeAAC(player_handle_, 0);

  if (0 ==lib_player_.SmartPlayerStartPullStream(player_handle_) ) 
    // 启动定时器，长时间收不到音频数据，则停止播放，发送BYE
    last_received_audio_data_time_.set(SystemClock.elapsedRealtime());
    handler_.postDelayed(new AudioPlayerPCMTimer(player_handle_), AudioPlayerPCMTimer.INTERVAL_MS);
  

  return true;

调用StartPlay后，拿到的audio数据，塞到publisher端，做回音消除处理：

class PlayerExternalPCMOutput implements NTExternalAudioOutput 
  private int buffer_size_ = 0;
  private ByteBuffer pcm_buffer_ = null;

  @Override
  public ByteBuffer getPcmByteBuffer(int  
    //Log.i("getPcmByteBuffer", "size: " + size);

    if(size < 1)
      return null;

    if(buffer_size_ != size) 
      buffer_size_ = size;
      pcm_buffer_ = ByteBuffer.allocateDirect(buffer_size_);
    

    return pcm_buffer_;
  

  public void onGetPcmFrame(int ret, int sampleRate, int channel, int sampleSize, int 
    /*Log.i("onGetPcmFrame", "ret: " + ret + ", sampleRate: " + sampleRate + ", channel: " + channel + ", sampleSize: " + sampleSize +
        ",is_low_latency:" + is_low_latency + " buffer_size：" + buffer_size);*/

    if (null == pcm_buffer_)
      return;

    pcm_buffer_.rewind();

    if (ret == 0 && isGB28181StreamRunning && publisherHandle != 0 )
      libPublisher.SmartPublisherOnFarEndPCMData(publisherHandle, pcm_buffer_, sampleRate, channel, sampleSize, is_low_latency);
  


private static int align(int d, int  return (d + (a - 1)) & ~(a - 1); 

class PlayerAudioDataOutput implements NTAudioDataCallback 
  private int buffer_size_ = 0;
  private int param_info_size_ = 0;

  private ByteBuffer buffer_ = null;
  private ByteBuffer parameter_info_ = null;

  @Override
  public ByteBuffer getAudioByteBuffer(int 
    //Log.i("getAudioByteBuffer", "size: " + size);

    if( size < 1 ) return null;

    if (size <= buffer_size_ && buffer_ != null )
      return buffer_;

    buffer_size_ = align(size + 256, 16);
    buffer_ = ByteBuffer.allocateDirect(buffer_size_);
    // Log.i("getAudioByteBuffer", "size: " + size + " buffer_size:" + audio_buffer_size);

    return buffer_;
  

  @Override
  public ByteBuffer getAudioParameterInfo(int 
    //Log.i("getAudioParameterInfo", "size: " + size);

    if(size < 1) return null;

    if ( size <= param_info_size_ &&  parameter_info_ != null )
      return  parameter_info_;

    param_info_size_ = align(size + 32, 16);
    parameter_info_ = ByteBuffer.allocateDirect(param_info_size_);
    //Log.i("getAudioParameterInfo", "size: " + size + " buffer_size:" + param_info_size);

    return parameter_info_;
  

  public void onAudioDataCallback(int ret, int audio_codec_id, int sample_size, int is_key_frame, long timestamp, int sample_rate, int channel, int parameter_info_size, long  
    /*Log.i("onAudioDataCallback", "ret: " + ret + ", audio_codec_id: " + audio_codec_id + ", sample_size: " + sample_size + ", timestamp: " + timestamp +
        ",sample_rate:" + sample_rate);
     */

如果长时间收不到数据，主动断掉音频广播：

class AudioPlayerPCMTimer implements Runnable 
  public static final int THRESHOLD_MS = 60*1000; // 暂时设置到1分钟
  public static final int INTERVAL_MS = 10*1000; // 十秒一次, 太频繁影响主线程

  public AudioPlayerPCMTimer(long 
    handle_ = handle;
  

  @Override
  public void run() 
    if (0 == handle_)
      return;

    if (handle_ != player_handle_) 
      Log.i(TAG, "AudioPlayerPCMTimer handle changed, will stop this timer, handle:" + handle_ + " new handle:" + player_handle_);
      return;
    

    long last_update_time = last_received_audio_data_time_.get();
    long cur_time = SystemClock.elapsedRealtime();

    // Log.i(TAG, "AudioPlayerPCMTimer last_update_time:" + last_update_time + " cur_time:" + cur_time);

    if ( (last_update_time + this.THRESHOLD_MS) >  cur_time) 
      // 继续定时器
      handler_.postDelayed(new AudioPlayerPCMTimer(this.handle_), this.INTERVAL_MS);
      //  Log.i(TAG, "AudioPlayerPCMTimer running.");
    
    else 
      Log.i(TAG, "AudioPlayerPCMTimer,trigger threshold, bye audio, stop player.");

      if (gb_broadcast_source_id_ != null && gb_broadcast_target_id_ != null) 
        if (gb28181_agent_ != null)
          gb28181_agent_.byeAudioBroadcast(gb_broadcast_source_id_, gb_broadcast_target_id_);
      

      gb_broadcast_source_id_ = null;
      gb_broadcast_target_id_ = null;

      stopAudioPlayer();
      destoryRTPReceiver();

      btnGB28181AudioBroadcast.setText("GB28181语音广播");
      btnGB28181AudioBroadcast.setEnabled(false);
    
  

  private long

停止广播数据播放：

private void stopAudioPlayer() 
  if (player_handle_ != 0 ) 
    lib_player_.SmartPlayerStopPullStream(player_handle_);
    lib_player_.SmartPlayerStopPlay(player_handle_);
    lib_player_.SmartPlayerClose(player_handle_);
    player_handle_ = 0;
  

销毁RTPReceiver：

private void destoryRTPReceiver() 
  if (rtp_receiver_handle_ != 0) 
    lib_player_.UnInitRTPReceiver(rtp_receiver_handle_);
    lib_player_.DestoryRTPReceiverSession(rtp_receiver_handle_);
    lib_player_.DestoryRTPReceiver(rtp_receiver_handle_);
    rtp_receiver_handle_ = 0;
  

以上是针对GB28181平台端音频广播播放的一点说明，感兴趣的开发者，可以酌情参考，也可以和我探讨android平台GB28181接入模块的测试。