Merge "Fix indentation and whitespace"

5 files changed, 79 insertions, 81 deletions

diff --git a/core/jni/android_media_AudioTrack.cpp b/core/jni/android_media_AudioTrack.cpp
index 84e7432..2573aa6 100644
--- a/core/jni/android_media_AudioTrack.cpp
+++ b/core/jni/android_media_AudioTrack.cpp
@@ -106,7 +106,7 @@ class AudioTrackJniStorage {
 #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 @@ public class AudioTrack
         // 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 class AudioTrack
     public int getSampleRate() {
         return mSampleRate;
     }
-    
+
     /**
      * Returns the current playback rate in Hz.
      */
@@ -590,22 +590,22 @@ public class AudioTrack
     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 @@ public class AudioTrack
             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 class AudioTrack
     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 @@ public class AudioTrack
      * @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 @@ public class AudioTrack
         if (listener != null) {
             mEventHandlerDelegate = new NativeEventHandlerDelegate(this, handler);
         }
-        
+
     }
 
 
@@ -917,7 +917,7 @@ public class AudioTrack
             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 @@ public class AudioTrack
                 && (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 @@ public class AudioTrack
          * 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 @@ public class AudioTrack
     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 @@ public class AudioTrack
                 // 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 @@ public class AudioTrack
                 };
             } else {
                 mHandler = null;
-            } 
+            }
         }
-        
+
         Handler getHandler() {
             return mHandler;
         }
@@ -1133,7 +1133,7 @@ public class AudioTrack
         }
 
         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 05271bc..6868d99 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1467,4 +1467,3 @@ bool audio_track_cblk_t::tryLock()
 // -------------------------------------------------------------------------
 
 }; // namespace android
-
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1883860..fd82306 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1973,7 +1973,7 @@ bool AudioFlinger::MixerThread::threadLoop()
             // 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...
@@ -2018,11 +2018,11 @@ bool AudioFlinger::MixerThread::threadLoop()
         }
         // 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 63d9a88..847f8dc 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -84,19 +84,19 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
     }
 }
 
- 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;
@@ -110,18 +110,18 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
         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);
@@ -141,7 +141,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
         track.volumeInc[1] = 0;
         mTrackNames &= ~(1<<name);
     }
- }
+}
 
 status_t AudioMixer::enable(int name)
 {
@@ -456,33 +456,33 @@ void AudioMixer::process__validate(state_t* state)
         countActiveTracks, state->enabledTracks,
         all16BitsStereoNoResample, resampling, volumeRamp);
 
-   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;
-           }
-       }
-   }
+    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;
+            }
+        }
+    }
 }
 
 static inline
@@ -999,7 +999,7 @@ void AudioMixer::process__genericNoResampling(state_t* state)
 }
 
 
-  // generic code with resampling
+// generic code with resampling
 void AudioMixer::process__genericResampling(state_t* state)
 {
     int32_t* const outTemp = state->outputTemp;
@@ -1179,7 +1179,7 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
                 }
                 in1 = buff;
                 b1.frameCount = numFrames;
-               } else {
+            } else {
                 in1 = b1.i16;
             }
             frameCount1 = b1.frameCount;
@@ -1221,4 +1221,3 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
 
 // ----------------------------------------------------------------------------
 }; // namespace android
-