| Index: webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
|
| 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 4ce35c66c3d52c0d4ceec3a81e67834c72b454f4..c2874316a5ddac0e6a0c03a78f810a918f420a38 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
|
| @@ -77,7 +77,7 @@ public class WebRtcAudioTrack {
|
| audioTrack.play();
|
| assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING);
|
| } catch (IllegalStateException e) {
|
| - Logging.e(TAG, "AudioTrack.play failed: " + e.getMessage());
|
| + Logging.e(TAG, "AudioTrack.play failed: " + e.getMessage());
|
| return;
|
| }
|
|
|
| @@ -155,19 +155,16 @@ public class WebRtcAudioTrack {
|
| Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
|
| this.context = context;
|
| this.nativeAudioTrack = nativeAudioTrack;
|
| - audioManager = (AudioManager) context.getSystemService(
|
| - Context.AUDIO_SERVICE);
|
| + audioManager = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
|
| if (DEBUG) {
|
| WebRtcAudioUtils.logDeviceInfo(TAG);
|
| }
|
| }
|
|
|
| private boolean initPlayout(int sampleRate, int channels) {
|
| - Logging.d(TAG, "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));
|
| + byteBuffer = byteBuffer.allocateDirect(bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
|
| Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
|
| emptyBytes = new byte[byteBuffer.capacity()];
|
| // Rather than passing the ByteBuffer with every callback (requiring
|
| @@ -180,9 +177,7 @@ public class WebRtcAudioTrack {
|
| // Note that this size doesn't guarantee a smooth playback under load.
|
| // TODO(henrika): should we extend the buffer size to avoid glitches?
|
| final int minBufferSizeInBytes = AudioTrack.getMinBufferSize(
|
| - sampleRate,
|
| - AudioFormat.CHANNEL_OUT_MONO,
|
| - AudioFormat.ENCODING_PCM_16BIT);
|
| + sampleRate, AudioFormat.CHANNEL_OUT_MONO, AudioFormat.ENCODING_PCM_16BIT);
|
| Logging.d(TAG, "AudioTrack.getMinBufferSize: " + minBufferSizeInBytes);
|
| // For the streaming mode, data must be written to the audio sink in
|
| // chunks of size (given by byteBuffer.capacity()) less than or equal
|
| @@ -204,12 +199,9 @@ public class WebRtcAudioTrack {
|
| // Create an AudioTrack object and initialize its associated audio buffer.
|
| // The size of this buffer determines how long an AudioTrack can play
|
| // before running out of data.
|
| - audioTrack = new AudioTrack(AudioManager.STREAM_VOICE_CALL,
|
| - sampleRate,
|
| - AudioFormat.CHANNEL_OUT_MONO,
|
| - AudioFormat.ENCODING_PCM_16BIT,
|
| - minBufferSizeInBytes,
|
| - AudioTrack.MODE_STREAM);
|
| + audioTrack =
|
| + new AudioTrack(AudioManager.STREAM_VOICE_CALL, sampleRate, AudioFormat.CHANNEL_OUT_MONO,
|
| + AudioFormat.ENCODING_PCM_16BIT, minBufferSizeInBytes, AudioTrack.MODE_STREAM);
|
| } catch (IllegalArgumentException e) {
|
| Logging.d(TAG, e.getMessage());
|
| return false;
|
| @@ -290,8 +282,7 @@ public class WebRtcAudioTrack {
|
| }
|
| }
|
|
|
| - private native void nativeCacheDirectBufferAddress(
|
| - ByteBuffer byteBuffer, long nativeAudioRecord);
|
| + private native void nativeCacheDirectBufferAddress(ByteBuffer byteBuffer, long nativeAudioRecord);
|
|
|
| private native void nativeGetPlayoutData(int bytes, long nativeAudioRecord);
|
|
|
|
|