2016-08-25 53 views
0

我在android中使用libjingle(version-9127)实现webrtc,问题是在一个应用程序生命周期中,当我尝试拨打电话时,第一个电话成功建立并结束。但是当第二个电话被接通时,当电话被接听时,视频被显示,但是音频流不被传送,然后在3-7秒之后应用程序崩溃给出这个错误。 下面的完整logcat的错误是:java.lang.AssertionError:预期的条件是真正的webrtc android

08-25 13:47:12.099 20887-23918/com.justtotaltech.tagove.app D/AudioRecordJni: [email protected][tid=23918] 
08-25 13:47:12.099 20887-23918/com.justtotaltech.tagove.app D/AudioManager: SetCommunicationMode(1)@[tid=23918] 
08-25 13:47:12.099 20887-23918/com.justtotaltech.tagove.app D/WebRtcAudioManager: setCommunicationMode(true)@[name=Thread-1692 - 23918, id=1699] 
08-25 13:47:12.099 20887-23918/com.justtotaltech.tagove.app D/AudioRecordJni: [email protected][tid=23918] 
08-25 13:47:12.099 20887-23918/com.justtotaltech.tagove.app D/WebRtcAudioRecord: InitRecording(sampleRate=48000, channels=1) 
08-25 13:47:12.099 20887-23918/com.justtotaltech.tagove.app D/WebRtcAudioRecord: byteBuffer.capacity: 960 
08-25 13:47:12.099 20887-23918/com.justtotaltech.tagove.app D/AudioRecordJni: OnCacheDirectBufferAddress 
08-25 13:47:12.099 20887-23918/com.justtotaltech.tagove.app D/AudioRecordJni: direct buffer capacity: 960 
08-25 13:47:12.109 20887-23918/com.justtotaltech.tagove.app D/WebRtcAudioRecord: AudioRecord.getMinBufferSize: 4096 
08-25 13:47:12.109 20887-23918/com.justtotaltech.tagove.app D/WebRtcAudioRecord: bufferSizeInBytes: 4096 
08-25 13:47:12.109 20887-23918/com.justtotaltech.tagove.app D/WebRtcAudioRecord: AudioRecord session ID: 299, audio format: 2, channels: 1, sample rate: 48000 
08-25 13:47:12.109 20887-23918/com.justtotaltech.tagove.app D/WebRtcAudioRecord: AcousticEchoCanceler.isAvailable: false 
08-25 13:47:12.129 20887-23918/com.justtotaltech.tagove.app D/AudioRecordJni: frames_per_buffer: 480 
08-25 13:47:12.129 20887-23918/com.justtotaltech.tagove.app D/AudioRecordJni: [email protected][tid=23918] 
08-25 13:47:12.129 20887-23918/com.justtotaltech.tagove.app D/WebRtcAudioRecord: StartRecording 
08-25 13:47:12.159 20887-24120/com.justtotaltech.tagove.app D/WebRtcAudioRecord: [email protected][name=AudioRecordJavaThread, id=1707] 
08-25 13:47:12.159 20887-24120/com.justtotaltech.tagove.app E/AudioRecord: start() status -38 
08-25 13:47:12.159 20887-24120/com.justtotaltech.tagove.app W/dalvikvm: threadid=41: thread exiting with uncaught exception (group=0x41687d58) 
08-25 13:47:15.232 20887-24120/com.justtotaltech.tagove.app E/AndroidRuntime: FATAL EXCEPTION: AudioRecordJavaThread 

       Process: com.justtotaltech.tagove.app, PID: 20887 
          java.lang.AssertionError: Expected condition to be true 
          at org.webrtc.voiceengine.WebRtcAudioRecord.assertTrue(WebRtcAudioRecord.java:259) 
          at org.webrtc.voiceengine.WebRtcAudioRecord.access$300(WebRtcAudioRecord.java:29) 
          at org.webrtc.voiceengine.WebRtcAudioRecord$AudioRecordThread.run(WebRtcAudioRecord.java:79) 

编辑:要断开我有此代码调用。希望你能弄清楚错误是什么。在此之后,我释放peerConnections

if(mediaStream != null && isLocal) { 

     mediaStream.removeTrack(audioTrack); 
     audioTrack.setState(MediaStreamTrack.State.ENDED); 
     audioTrack = null; 
     audioSource = null; 

     if(videoTrack != null && ActivityCall.callTypeGlobal.equals(CallManager.CallType.VIDEO)){ 
      Log.d("TestCallType",String.valueOf(ActivityCall.callTypeGlobal)); 
      mediaStream.removeTrack(videoTrack); 
      videoTrack = null; 

      source.stop(); 

      videoCapturer.dispose(); 
      videoCapturer = null; 
      activeCamera = null; 
      Log.d("TestCallType","Video Track"+String.valueOf(videoTrack)); 
      Log.d("TestCallType","Video Capturer"+String.valueOf(videoCapturer)); 
     } 

     mediaStream = null; 
    } 

回答

0

望着source code(如果这不是仓库我敢肯定这是一个你要根据错误行号使用类)引起的例外是该行: assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING)

确保在通话结束后释放您使用的所有内容。当新的呼叫开始时,每个资源都需要新鲜。然后再重新初始化。

+0

@AkshayBissa我有基本相同的代码,所以我不知道 –

+0

@SamuelMéndez我使用的MediaPlayer播放的铃声任何其他步骤可以说也为DIS – abissa

+0

看到的日志的第一个错误'AudioRecord启动问题( )状态-38'。有关于该主题的一些问题(如http://stackoverflow.com/questions/20460892/audiorecord-start-status-38)。也许解决方案来自 –