diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java index 3df9e160a3..84e3fb8ed7 100644 --- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java +++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java @@ -70,7 +70,7 @@ class WebRtcAudioRecord { @Override public void run() { Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO); - Logging.w(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo()); + Logging.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo()); assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING); @@ -90,7 +90,7 @@ class WebRtcAudioRecord { long durationInMs = TimeUnit.NANOSECONDS.toMillis((nowTime - lastTime)); lastTime = nowTime; - Logging.w(TAG, "bytesRead[" + durationInMs + "] " + bytesRead); + Logging.d(TAG, "bytesRead[" + durationInMs + "] " + bytesRead); } } @@ -114,7 +114,7 @@ class WebRtcAudioRecord { } WebRtcAudioRecord(Context context, long nativeAudioRecord) { - Logging.w(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo()); + Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo()); this.context = context; this.nativeAudioRecord = nativeAudioRecord; if (DEBUG) { @@ -124,7 +124,7 @@ class WebRtcAudioRecord { } private boolean enableBuiltInAEC(boolean enable) { - Logging.w(TAG, "enableBuiltInAEC(" + enable + ')'); + Logging.d(TAG, "enableBuiltInAEC(" + enable + ')'); if (effects == null) { Logging.e(TAG,"Built-in AEC is not supported on this platform"); return false; @@ -133,7 +133,7 @@ class WebRtcAudioRecord { } private boolean enableBuiltInAGC(boolean enable) { - Logging.w(TAG, "enableBuiltInAGC(" + enable + ')'); + Logging.d(TAG, "enableBuiltInAGC(" + enable + ')'); if (effects == null) { Logging.e(TAG,"Built-in AGC is not supported on this platform"); return false; @@ -142,7 +142,7 @@ class WebRtcAudioRecord { } private boolean enableBuiltInNS(boolean enable) { - Logging.w(TAG, "enableBuiltInNS(" + enable + ')'); + Logging.d(TAG, "enableBuiltInNS(" + enable + ')'); if (effects == null) { Logging.e(TAG,"Built-in NS is not supported on this platform"); return false; @@ -151,7 +151,7 @@ class WebRtcAudioRecord { } private int initRecording(int sampleRate, int channels) { - Logging.w(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + + Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + channels + ")"); if (!WebRtcAudioUtils.hasPermission( context, android.Manifest.permission.RECORD_AUDIO)) { @@ -165,7 +165,7 @@ class WebRtcAudioRecord { final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8); final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND; byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer); - Logging.w(TAG, "byteBuffer.capacity: " + byteBuffer.capacity()); + Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity()); // Rather than passing the ByteBuffer with every callback (requiring // the potentially expensive GetDirectBufferAddress) we simply have the // the native class cache the address to the memory once. @@ -183,14 +183,14 @@ class WebRtcAudioRecord { Logging.e(TAG, "AudioRecord.getMinBufferSize failed: " + minBufferSize); return -1; } - Logging.w(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize); + Logging.d(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize); // Use a larger buffer size than the minimum required when creating the // AudioRecord instance to ensure smooth recording under load. It has been // verified that it does not increase the actual recording latency. int bufferSizeInBytes = Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity()); - Logging.w(TAG, "bufferSizeInBytes: " + bufferSizeInBytes); + Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes); try { audioRecord = new AudioRecord(AudioSource.VOICE_COMMUNICATION, sampleRate, @@ -206,7 +206,7 @@ class WebRtcAudioRecord { Logging.e(TAG,"Failed to create a new AudioRecord instance"); return -1; } - Logging.w(TAG, "AudioRecord " + Logging.d(TAG, "AudioRecord " + "session ID: " + audioRecord.getAudioSessionId() + ", " + "audio format: " + audioRecord.getAudioFormat() + ", " + "channels: " + audioRecord.getChannelCount() + ", " @@ -227,7 +227,7 @@ class WebRtcAudioRecord { } private boolean startRecording() { - Logging.w(TAG, "startRecording"); + Logging.d(TAG, "startRecording"); assertTrue(audioRecord != null); assertTrue(audioThread == null); try { @@ -246,7 +246,7 @@ class WebRtcAudioRecord { } private boolean stopRecording() { - Logging.w(TAG, "stopRecording"); + Logging.d(TAG, "stopRecording"); assertTrue(audioThread != null); audioThread.joinThread(); audioThread = null; diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java index e99b9d7493..43c1a19c46 100644 --- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java +++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java @@ -61,7 +61,7 @@ class WebRtcAudioTrack { @Override public void run() { Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO); - Logd("AudioTrackThread" + WebRtcAudioUtils.getThreadInfo()); + Logging.d(TAG, "AudioTrackThread" + WebRtcAudioUtils.getThreadInfo()); try { // In MODE_STREAM mode we can optionally prime the output buffer by @@ -71,7 +71,7 @@ class WebRtcAudioTrack { audioTrack.play(); assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING); } catch (IllegalStateException e) { - Loge("AudioTrack.play failed: " + e.getMessage()); + Logging.e(TAG, "AudioTrack.play failed: " + e.getMessage()); return; } @@ -99,7 +99,7 @@ class WebRtcAudioTrack { sizeInBytes); } if (bytesWritten != sizeInBytes) { - Loge("AudioTrack.write failed: " + bytesWritten); + Logging.e(TAG, "AudioTrack.write failed: " + bytesWritten); if (bytesWritten == AudioTrack.ERROR_INVALID_OPERATION) { keepAlive = false; } @@ -117,7 +117,7 @@ class WebRtcAudioTrack { try { audioTrack.stop(); } catch (IllegalStateException e) { - Loge("AudioTrack.stop failed: " + e.getMessage()); + Logging.e(TAG, "AudioTrack.stop failed: " + e.getMessage()); } assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_STOPPED); audioTrack.flush(); @@ -136,7 +136,7 @@ class WebRtcAudioTrack { } WebRtcAudioTrack(Context context, long nativeAudioTrack) { - Logd("ctor" + WebRtcAudioUtils.getThreadInfo()); + Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo()); this.context = context; this.nativeAudioTrack = nativeAudioTrack; audioManager = (AudioManager) context.getSystemService( @@ -147,12 +147,12 @@ class WebRtcAudioTrack { } private void initPlayout(int sampleRate, int channels) { - Logd("initPlayout(sampleRate=" + sampleRate + ", channels=" + - channels + ")"); + Logging.d(TAG, "initPlayout(sampleRate=" + sampleRate + ", channels=" + + channels + ")"); final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8); byteBuffer = byteBuffer.allocateDirect( bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND)); - Logd("byteBuffer.capacity: " + byteBuffer.capacity()); + Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity()); // Rather than passing the ByteBuffer with every callback (requiring // the potentially expensive GetDirectBufferAddress) we simply have the // the native class cache the address to the memory once. @@ -166,7 +166,7 @@ class WebRtcAudioTrack { sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT); - Logd("AudioTrack.getMinBufferSize: " + minBufferSizeInBytes); + Logging.d(TAG, "AudioTrack.getMinBufferSize: " + minBufferSizeInBytes); assertTrue(audioTrack == null); // For the streaming mode, data must be written to the audio sink in @@ -184,7 +184,7 @@ class WebRtcAudioTrack { minBufferSizeInBytes, AudioTrack.MODE_STREAM); } catch (IllegalArgumentException e) { - Logd(e.getMessage()); + Logging.d(TAG, e.getMessage()); return; } assertTrue(audioTrack.getState() == AudioTrack.STATE_INITIALIZED); @@ -193,7 +193,7 @@ class WebRtcAudioTrack { } private boolean startPlayout() { - Logd("startPlayout"); + Logging.d(TAG, "startPlayout"); assertTrue(audioTrack != null); assertTrue(audioThread == null); audioThread = new AudioTrackThread("AudioTrackJavaThread"); @@ -202,7 +202,7 @@ class WebRtcAudioTrack { } private boolean stopPlayout() { - Logd("stopPlayout"); + Logging.d(TAG, "stopPlayout"); assertTrue(audioThread != null); audioThread.joinThread(); audioThread = null; @@ -215,18 +215,18 @@ class WebRtcAudioTrack { /** Get max possible volume index for a phone call audio stream. */ private int getStreamMaxVolume() { - Logd("getStreamMaxVolume"); + Logging.d(TAG, "getStreamMaxVolume"); assertTrue(audioManager != null); return audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL); } /** Set current volume level for a phone call audio stream. */ private boolean setStreamVolume(int volume) { - Logd("setStreamVolume(" + volume + ")"); + Logging.d(TAG, "setStreamVolume(" + volume + ")"); assertTrue(audioManager != null); if (WebRtcAudioUtils.runningOnLollipopOrHigher()) { if (audioManager.isVolumeFixed()) { - Loge("The device implements a fixed volume policy."); + Logging.e(TAG, "The device implements a fixed volume policy."); return false; } } @@ -236,7 +236,7 @@ class WebRtcAudioTrack { /** Get current volume level for a phone call audio stream. */ private int getStreamVolume() { - Logd("getStreamVolume"); + Logging.d(TAG, "getStreamVolume"); assertTrue(audioManager != null); return audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL); } @@ -248,14 +248,6 @@ class WebRtcAudioTrack { } } - private static void Logd(String msg) { - Logging.d(TAG, msg); - } - - private static void Loge(String msg) { - Logging.e(TAG, msg); - } - private native void nativeCacheDirectBufferAddress( ByteBuffer byteBuffer, long nativeAudioRecord);