java.lang.AssertionError：预期条件为真 webrtc android

Question

我正在使用 libjingle (version-9127) 在 android 中实现 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，第一次通话成功建立然后结束。但是，当拨打第二个电话时，当接听电话时，会显示视频，但不会传输音频流，然后在 3-7 秒后应用程序崩溃并出现此错误。 完整的logcat错误如下：

08-25 13:47:12.099 20887-23918/com.justtotaltech.tagove.app D/AudioRecordJni: BuiltInAECIsAvailable@[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: InitRecording@[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: StartRecording@[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: AudioRecordThread@[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：要断开通话，我已经编写了此代码。希望你能弄清楚错误是什么。在此之后我将释放 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;
    }

Answer 1

查看source code（如果这不是存储库，我确定这是您根据错误行号使用的类）导致异常的行是这样的： assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING)

确保在通话结束后释放您使用的所有内容。当一个新的调用开始时，每个资源都需要是新鲜的。并重新初始化。