Fix indentation and whitespace
Use git diff -w to verify.
Change-Id: Ib65be0a1ecf65d6cad516110604e3855bf68a638
diff --git a/core/jni/android_media_AudioTrack.cpp b/core/jni/android_media_AudioTrack.cpp
index 84e7432..2573aa6d 100644
--- a/core/jni/android_media_AudioTrack.cpp
+++ b/core/jni/android_media_AudioTrack.cpp
@@ -106,7 +106,7 @@
#define AUDIOTRACK_ERROR_BAD_VALUE -2
#define AUDIOTRACK_ERROR_INVALID_OPERATION -3
#define AUDIOTRACK_ERROR_SETUP_AUDIOSYSTEM -16
-#define AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELMASK -17
+#define AUDIOTRACK_ERROR_SETUP_INVALIDCHANNELMASK -17
#define AUDIOTRACK_ERROR_SETUP_INVALIDFORMAT -18
#define AUDIOTRACK_ERROR_SETUP_INVALIDSTREAMTYPE -19
#define AUDIOTRACK_ERROR_SETUP_NATIVEINITFAILED -20
diff --git a/media/java/android/media/AudioTrack.java b/media/java/android/media/AudioTrack.java
index 4f9eb2b..c5d17eb 100644
--- a/media/java/android/media/AudioTrack.java
+++ b/media/java/android/media/AudioTrack.java
@@ -449,7 +449,7 @@
// AudioTrack subclasses too.
try {
stop();
- } catch(IllegalStateException ise) {
+ } catch(IllegalStateException ise) {
// don't raise an exception, we're releasing the resources.
}
native_release();
@@ -488,7 +488,7 @@
public int getSampleRate() {
return mSampleRate;
}
-
+
/**
* Returns the current playback rate in Hz.
*/
@@ -590,22 +590,22 @@
static public int getNativeOutputSampleRate(int streamType) {
return native_get_output_sample_rate(streamType);
}
-
+
/**
* Returns the minimum buffer size required for the successful creation of an AudioTrack
* object to be created in the {@link #MODE_STREAM} mode. Note that this size doesn't
* guarantee a smooth playback under load, and higher values should be chosen according to
- * the expected frequency at which the buffer will be refilled with additional data to play.
+ * the expected frequency at which the buffer will be refilled with additional data to play.
* @param sampleRateInHz the sample rate expressed in Hertz.
- * @param channelConfig describes the configuration of the audio channels.
+ * @param channelConfig describes the configuration of the audio channels.
* See {@link AudioFormat#CHANNEL_OUT_MONO} and
* {@link AudioFormat#CHANNEL_OUT_STEREO}
- * @param audioFormat the format in which the audio data is represented.
- * See {@link AudioFormat#ENCODING_PCM_16BIT} and
+ * @param audioFormat the format in which the audio data is represented.
+ * See {@link AudioFormat#ENCODING_PCM_16BIT} and
* {@link AudioFormat#ENCODING_PCM_8BIT}
* @return {@link #ERROR_BAD_VALUE} if an invalid parameter was passed,
- * or {@link #ERROR} if the implementation was unable to query the hardware for its output
- * properties,
+ * or {@link #ERROR} if the implementation was unable to query the hardware for its output
+ * properties,
* or the minimum buffer size expressed in bytes.
*/
static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {
@@ -623,18 +623,18 @@
loge("getMinBufferSize(): Invalid channel configuration.");
return AudioTrack.ERROR_BAD_VALUE;
}
-
- if ((audioFormat != AudioFormat.ENCODING_PCM_16BIT)
+
+ if ((audioFormat != AudioFormat.ENCODING_PCM_16BIT)
&& (audioFormat != AudioFormat.ENCODING_PCM_8BIT)) {
loge("getMinBufferSize(): Invalid audio format.");
return AudioTrack.ERROR_BAD_VALUE;
}
-
+
if ( (sampleRateInHz < 4000) || (sampleRateInHz > 48000) ) {
loge("getMinBufferSize(): " + sampleRateInHz +"Hz is not a supported sample rate.");
return AudioTrack.ERROR_BAD_VALUE;
}
-
+
int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat);
if ((size == -1) || (size == 0)) {
loge("getMinBufferSize(): error querying hardware");
@@ -667,7 +667,7 @@
public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener) {
setPlaybackPositionUpdateListener(listener, null);
}
-
+
/**
* Sets the listener the AudioTrack notifies when a previously set marker is reached or
* for each periodic playback head position update.
@@ -676,7 +676,7 @@
* @param listener
* @param handler the Handler that will receive the event notification messages.
*/
- public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener,
+ public void setPlaybackPositionUpdateListener(OnPlaybackPositionUpdateListener listener,
Handler handler) {
synchronized (mPositionListenerLock) {
mPositionListener = listener;
@@ -684,7 +684,7 @@
if (listener != null) {
mEventHandlerDelegate = new NativeEventHandlerDelegate(this, handler);
}
-
+
}
@@ -917,7 +917,7 @@
return ERROR_INVALID_OPERATION;
}
- if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)
+ if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)
|| (offsetInBytes + sizeInBytes > audioData.length)) {
return ERROR_BAD_VALUE;
}
@@ -948,12 +948,12 @@
&& (sizeInShorts > 0)) {
mState = STATE_INITIALIZED;
}
-
+
if (mState != STATE_INITIALIZED) {
return ERROR_INVALID_OPERATION;
}
- if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0)
+ if ( (audioData == null) || (offsetInShorts < 0 ) || (sizeInShorts < 0)
|| (offsetInShorts + sizeInShorts > audioData.length)) {
return ERROR_BAD_VALUE;
}
@@ -1047,7 +1047,7 @@
* by the playback head.
*/
void onMarkerReached(AudioTrack track);
-
+
/**
* Called on the listener to periodically notify it that the playback head has reached
* a multiple of the notification period.
@@ -1066,7 +1066,7 @@
private class NativeEventHandlerDelegate {
private final AudioTrack mAudioTrack;
private final Handler mHandler;
-
+
NativeEventHandlerDelegate(AudioTrack track, Handler handler) {
mAudioTrack = track;
// find the looper for our new event handler
@@ -1077,7 +1077,7 @@
// no given handler, use the looper the AudioTrack was created in
looper = mInitializationLooper;
}
-
+
// construct the event handler with this looper
if (looper != null) {
// implement the event handler delegate
@@ -1111,9 +1111,9 @@
};
} else {
mHandler = null;
- }
+ }
}
-
+
Handler getHandler() {
return mHandler;
}
@@ -1133,7 +1133,7 @@
}
if (track.mEventHandlerDelegate != null) {
- Message m =
+ Message m =
track.mEventHandlerDelegate.getHandler().obtainMessage(what, arg1, arg2, obj);
track.mEventHandlerDelegate.getHandler().sendMessage(m);
}
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 7e55fbd..2775348 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1469,4 +1469,3 @@
// -------------------------------------------------------------------------
}; // namespace android
-
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index b48f23d..e2aa04e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1967,7 +1967,7 @@
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
lockEffectChains_l(effectChains);
- }
+ }
if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
@@ -2012,11 +2012,11 @@
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
- for (size_t i = 0; i < effectChains.size(); i ++) {
- effectChains[i]->process_l();
- }
- // enable changes in effect chain
- unlockEffectChains(effectChains);
+ for (size_t i = 0; i < effectChains.size(); i ++) {
+ effectChains[i]->process_l();
+ }
+ // enable changes in effect chain
+ unlockEffectChains(effectChains);
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 7c7fa56..8011832 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -78,19 +78,19 @@
}
}
- AudioMixer::~AudioMixer()
- {
- track_t* t = mState.tracks;
- for (int i=0 ; i<32 ; i++) {
- delete t->resampler;
- t++;
- }
- delete [] mState.outputTemp;
- delete [] mState.resampleTemp;
- }
+AudioMixer::~AudioMixer()
+{
+ track_t* t = mState.tracks;
+ for (int i=0 ; i<32 ; i++) {
+ delete t->resampler;
+ t++;
+ }
+ delete [] mState.outputTemp;
+ delete [] mState.resampleTemp;
+}
- int AudioMixer::getTrackName()
- {
+int AudioMixer::getTrackName()
+{
uint32_t names = mTrackNames;
uint32_t mask = 1;
int n = 0;
@@ -104,18 +104,18 @@
return TRACK0 + n;
}
return -1;
- }
+}
- void AudioMixer::invalidateState(uint32_t mask)
- {
+void AudioMixer::invalidateState(uint32_t mask)
+{
if (mask) {
mState.needsChanged |= mask;
mState.hook = process__validate;
}
}
- void AudioMixer::deleteTrackName(int name)
- {
+void AudioMixer::deleteTrackName(int name)
+{
name -= TRACK0;
if (uint32_t(name) < MAX_NUM_TRACKS) {
ALOGV("deleteTrackName(%d)", name);
@@ -135,7 +135,7 @@
track.volumeInc[1] = 0;
mTrackNames &= ~(1<<name);
}
- }
+}
status_t AudioMixer::enable(int name)
{
@@ -450,33 +450,33 @@
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
- state->hook(state);
+ state->hook(state);
- // Now that the volume ramp has been done, set optimal state and
- // track hooks for subsequent mixer process
- if (countActiveTracks) {
- int allMuted = 1;
- uint32_t en = state->enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
- track_t& t = state->tracks[i];
- if (!t.doesResample() && t.volumeRL == 0)
- {
- t.needs |= NEEDS_MUTE_ENABLED;
- t.hook = track__nop;
- } else {
- allMuted = 0;
- }
- }
- if (allMuted) {
- state->hook = process__nop;
- } else if (all16BitsStereoNoResample) {
- if (countActiveTracks == 1) {
- state->hook = process__OneTrack16BitsStereoNoResampling;
- }
- }
- }
+ // Now that the volume ramp has been done, set optimal state and
+ // track hooks for subsequent mixer process
+ if (countActiveTracks) {
+ int allMuted = 1;
+ uint32_t en = state->enabledTracks;
+ while (en) {
+ const int i = 31 - __builtin_clz(en);
+ en &= ~(1<<i);
+ track_t& t = state->tracks[i];
+ if (!t.doesResample() && t.volumeRL == 0)
+ {
+ t.needs |= NEEDS_MUTE_ENABLED;
+ t.hook = track__nop;
+ } else {
+ allMuted = 0;
+ }
+ }
+ if (allMuted) {
+ state->hook = process__nop;
+ } else if (all16BitsStereoNoResample) {
+ if (countActiveTracks == 1) {
+ state->hook = process__OneTrack16BitsStereoNoResampling;
+ }
+ }
+ }
}
static inline
@@ -993,7 +993,7 @@
}
- // generic code with resampling
+// generic code with resampling
void AudioMixer::process__genericResampling(state_t* state)
{
int32_t* const outTemp = state->outputTemp;
@@ -1173,7 +1173,7 @@
}
in1 = buff;
b1.frameCount = numFrames;
- } else {
+ } else {
in1 = b1.i16;
}
frameCount1 = b1.frameCount;
@@ -1215,4 +1215,3 @@
// ----------------------------------------------------------------------------
}; // namespace android
-