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Unified Diff: webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java

Issue 2377003002: Format all Java in WebRTC. (Closed)
Patch Set: Rebase. Created 4 years, 3 months ago
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Index: webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
diff --git a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
index 64c9c7c6f8aeaec27548dae5b32169c2a138209e..19ee09a46678bef7a29161d344a8b0135230140d 100644
--- a/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
+++ b/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
@@ -45,8 +45,7 @@ public class WebRtcAudioManager {
// specified in WebRtcAudioUtils.BLACKLISTED_OPEN_SL_ES_MODELS.
// Allows an app to take control over which devices to exlude from using
// the OpenSL ES audio output path
- public static synchronized void setBlacklistDeviceForOpenSLESUsage(
- boolean enable) {
+ public static synchronized void setBlacklistDeviceForOpenSLESUsage(boolean enable) {
blacklistDeviceForOpenSLESUsageIsOverridden = true;
blacklistDeviceForOpenSLESUsage = enable;
}
@@ -62,10 +61,7 @@ public class WebRtcAudioManager {
// List of possible audio modes.
private static final String[] AUDIO_MODES = new String[] {
- "MODE_NORMAL",
- "MODE_RINGTONE",
- "MODE_IN_CALL",
- "MODE_IN_COMMUNICATION",
+ "MODE_NORMAL", "MODE_RINGTONE", "MODE_IN_CALL", "MODE_IN_COMMUNICATION",
};
// Private utility class that periodically checks and logs the volume level
@@ -85,9 +81,8 @@ public class WebRtcAudioManager {
public void start() {
timer = new Timer(THREAD_NAME);
- timer.schedule(new LogVolumeTask(
- audioManager.getStreamMaxVolume(AudioManager.STREAM_RING),
- audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL)),
+ timer.schedule(new LogVolumeTask(audioManager.getStreamMaxVolume(AudioManager.STREAM_RING),
+ audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL)),
0, TIMER_PERIOD_IN_SECONDS * 1000);
}
@@ -104,12 +99,12 @@ public class WebRtcAudioManager {
final int mode = audioManager.getMode();
if (mode == AudioManager.MODE_RINGTONE) {
Logging.d(TAG, "STREAM_RING stream volume: "
- + audioManager.getStreamVolume(AudioManager.STREAM_RING)
- + " (max=" + maxRingVolume + ")");
+ + audioManager.getStreamVolume(AudioManager.STREAM_RING) + " (max="
+ + maxRingVolume + ")");
} else if (mode == AudioManager.MODE_IN_COMMUNICATION) {
Logging.d(TAG, "VOICE_CALL stream volume: "
- + audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL)
- + " (max=" + maxVoiceCallVolume + ")");
+ + audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL) + " (max="
+ + maxVoiceCallVolume + ")");
}
}
}
@@ -147,16 +142,15 @@ public class WebRtcAudioManager {
Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
this.context = context;
this.nativeAudioManager = nativeAudioManager;
- audioManager = (AudioManager) context.getSystemService(
- Context.AUDIO_SERVICE);
+ audioManager = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
if (DEBUG) {
WebRtcAudioUtils.logDeviceInfo(TAG);
}
volumeLogger = new VolumeLogger(audioManager);
storeAudioParameters();
nativeCacheAudioParameters(sampleRate, channels, hardwareAEC, hardwareAGC, hardwareNS,
- lowLatencyOutput, lowLatencyInput, proAudio, outputBufferSize, inputBufferSize,
- nativeAudioManager);
+ lowLatencyOutput, lowLatencyInput, proAudio, outputBufferSize, inputBufferSize,
+ nativeAudioManager);
}
private boolean init() {
@@ -183,9 +177,9 @@ public class WebRtcAudioManager {
}
private boolean isDeviceBlacklistedForOpenSLESUsage() {
- boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden ?
- blacklistDeviceForOpenSLESUsage :
- WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
+ boolean blacklisted = blacklistDeviceForOpenSLESUsageIsOverridden
+ ? blacklistDeviceForOpenSLESUsage
+ : WebRtcAudioUtils.deviceIsBlacklistedForOpenSLESUsage();
if (blacklisted) {
Logging.e(TAG, Build.MODEL + " is blacklisted for OpenSL ES usage!");
}
@@ -203,24 +197,21 @@ public class WebRtcAudioManager {
lowLatencyOutput = isLowLatencyOutputSupported();
lowLatencyInput = isLowLatencyInputSupported();
proAudio = isProAudioSupported();
- outputBufferSize = lowLatencyOutput ?
- getLowLatencyOutputFramesPerBuffer() :
- getMinOutputFrameSize(sampleRate, channels);
+ outputBufferSize = lowLatencyOutput ? getLowLatencyOutputFramesPerBuffer()
+ : getMinOutputFrameSize(sampleRate, channels);
inputBufferSize = lowLatencyInput ? getLowLatencyInputFramesPerBuffer()
: getMinInputFrameSize(sampleRate, channels);
}
// Gets the current earpiece state.
private boolean hasEarpiece() {
- return context.getPackageManager().hasSystemFeature(
- PackageManager.FEATURE_TELEPHONY);
+ return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_TELEPHONY);
}
// Returns true if low-latency audio output is supported.
private boolean isLowLatencyOutputSupported() {
- return isOpenSLESSupported() &&
- context.getPackageManager().hasSystemFeature(
- PackageManager.FEATURE_AUDIO_LOW_LATENCY);
+ return isOpenSLESSupported()
+ && context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY);
}
// Returns true if low-latency audio input is supported.
@@ -231,16 +222,14 @@ public class WebRtcAudioManager {
// as well. The NDK doc states that: "As of API level 21, lower latency
// audio input is supported on select devices. To take advantage of this
// feature, first confirm that lower latency output is available".
- return WebRtcAudioUtils.runningOnLollipopOrHigher() &&
- isLowLatencyOutputSupported();
+ return WebRtcAudioUtils.runningOnLollipopOrHigher() && isLowLatencyOutputSupported();
}
// Returns true if the device has professional audio level of functionality
// and therefore supports the lowest possible round-trip latency.
private boolean isProAudioSupported() {
return WebRtcAudioUtils.runningOnMarshmallowOrHigher()
- && context.getPackageManager().hasSystemFeature(
- PackageManager.FEATURE_AUDIO_PRO);
+ && context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_PRO);
}
// Returns the native output sample rate for this device's output stream.
@@ -254,8 +243,8 @@ public class WebRtcAudioManager {
// Default can be overriden by WebRtcAudioUtils.setDefaultSampleRateHz().
// If so, use that value and return here.
if (WebRtcAudioUtils.isDefaultSampleRateOverridden()) {
- Logging.d(TAG, "Default sample rate is overriden to " +
- WebRtcAudioUtils.getDefaultSampleRateHz() + " Hz");
+ Logging.d(TAG, "Default sample rate is overriden to "
+ + WebRtcAudioUtils.getDefaultSampleRateHz() + " Hz");
return WebRtcAudioUtils.getDefaultSampleRateHz();
}
// No overrides available. Deliver best possible estimate based on default
@@ -272,11 +261,9 @@ public class WebRtcAudioManager {
@TargetApi(17)
private int getSampleRateOnJellyBeanMR10OrHigher() {
- String sampleRateString = audioManager.getProperty(
- AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
- return (sampleRateString == null)
- ? WebRtcAudioUtils.getDefaultSampleRateHz()
- : Integer.parseInt(sampleRateString);
+ String sampleRateString = audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
+ return (sampleRateString == null) ? WebRtcAudioUtils.getDefaultSampleRateHz()
+ : Integer.parseInt(sampleRateString);
}
// Returns the native output buffer size for low-latency output streams.
@@ -286,10 +273,9 @@ public class WebRtcAudioManager {
if (!WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
return DEFAULT_FRAME_PER_BUFFER;
}
- String framesPerBuffer = audioManager.getProperty(
- AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
- return framesPerBuffer == null ?
- DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer);
+ String framesPerBuffer =
+ audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
+ return framesPerBuffer == null ? DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer);
}
// Returns true if the device supports an audio effect (AEC, AGC or NS).
@@ -322,8 +308,8 @@ public class WebRtcAudioManager {
return -1;
}
return AudioTrack.getMinBufferSize(
- sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT) /
- bytesPerFrame;
+ sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
+ / bytesPerFrame;
}
// Returns the native input buffer size for input streams.
@@ -338,9 +324,9 @@ public class WebRtcAudioManager {
private static int getMinInputFrameSize(int sampleRateInHz, int numChannels) {
final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
assertTrue(numChannels == CHANNELS);
- return AudioRecord.getMinBufferSize(sampleRateInHz,
- AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT) /
- bytesPerFrame;
+ return AudioRecord.getMinBufferSize(
+ sampleRateInHz, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT)
+ / bytesPerFrame;
}
// Returns true if OpenSL ES audio is supported.
@@ -357,7 +343,6 @@ public class WebRtcAudioManager {
}
private native void nativeCacheAudioParameters(int sampleRate, int channels, boolean hardwareAEC,
- boolean hardwareAGC, boolean hardwareNS, boolean lowLatencyOutput,
- boolean lowLatencyInput, boolean proAudio, int outputBufferSize, int inputBufferSize,
- long nativeAudioManager);
+ boolean hardwareAGC, boolean hardwareNS, boolean lowLatencyOutput, boolean lowLatencyInput,
+ boolean proAudio, int outputBufferSize, int inputBufferSize, long nativeAudioManager);
}

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