Whitespace

Fix indentation, and add blank lines in key places for clarity

Change-Id: I57a0a8142394f83203161aa9b8aa9276abf3ed7c

6 files changed, 28 insertions, 23 deletions

diff --git a/include/common_time/local_clock.h b/include/common_time/local_clock.h
index 845d1c2..384c3de 100644
--- a/include/common_time/local_clock.h
+++ b/include/common_time/local_clock.h
@@ -28,7 +28,7 @@ namespace android {
 
 class LocalClock {
   public:
-     LocalClock();
+    LocalClock();
 
     bool initCheck();
 
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 552e829..ad27a1e 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -169,7 +169,7 @@ public:
                                     callback_t cbf       = 0,
                                     void* user           = 0,
                                     int notificationFrames = 0,
-                                    int sessionId = 0);
+                                    int sessionId        = 0);
 
     /* Creates an audio track and registers it with AudioFlinger. With this constructor,
      * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
@@ -215,7 +215,7 @@ public:
                             int notificationFrames = 0,
                             const sp<IMemory>& sharedBuffer = 0,
                             bool threadCanCallJava = false,
-                            int sessionId = 0);
+                            int sessionId       = 0);
 
 
     /* Result of constructing the AudioTrack. This must be checked
@@ -468,6 +468,7 @@ protected:
 
             // body of AudioTrackThread::threadLoop()
             bool processAudioBuffer(const sp<AudioTrackThread>& thread);
+
             status_t createTrack_l(audio_stream_type_t streamType,
                                  uint32_t sampleRate,
                                  audio_format_t format,
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 048be1d..ebb28cd 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -194,6 +194,7 @@ status_t AudioTrack::set(
     if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
         return NO_INIT;
     }
+
     uint32_t afLatency;
     if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
         return NO_INIT;
@@ -203,9 +204,11 @@ status_t AudioTrack::set(
     if (streamType == AUDIO_STREAM_DEFAULT) {
         streamType = AUDIO_STREAM_MUSIC;
     }
+
     if (sampleRate == 0) {
         sampleRate = afSampleRate;
     }
+
     // these below should probably come from the audioFlinger too...
     if (format == AUDIO_FORMAT_DEFAULT) {
         format = AUDIO_FORMAT_PCM_16_BIT;
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d83d19a..86bb7f9 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1588,7 +1588,7 @@ void AudioFlinger::PlaybackThread::onFirstRef()
 }
 
 // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTrack_l(
+sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
         const sp<AudioFlinger::Client>& client,
         audio_stream_type_t streamType,
         uint32_t sampleRate,
@@ -2337,7 +2337,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
     size_t tracksWithEffect = 0;
 
     float masterVolume = mMasterVolume;
-    bool  masterMute = mMasterMute;
+    bool masterMute = mMasterMute;
 
     if (masterMute) {
         masterVolume = 0;
@@ -2376,7 +2376,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
                 // +1 for rounding and +1 for additional sample needed for interpolation
                 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
                 // add frames already consumed but not yet released by the resampler
-                // because cblk->framesReady() will  include these frames
+                // because cblk->framesReady() will include these frames
                 minFrames += mAudioMixer->getUnreleasedFrames(track->name());
                 // the minimum track buffer size is normally twice the number of frames necessary
                 // to fill one buffer and the resampler should not leave more than one buffer worth
@@ -2514,6 +2514,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
 
             // reset retry count
             track->mRetryCount = kMaxTrackRetries;
+
             // If one track is ready, set the mixer ready if:
             //  - the mixer was not ready during previous round OR
             //  - no other track is not ready
@@ -3372,19 +3373,19 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
         }
     } else {
         mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
-            // construct the shared structure in-place.
-            new(mCblk) audio_track_cblk_t();
-            // clear all buffers
-            mCblk->frameCount = frameCount;
-            mCblk->sampleRate = sampleRate;
-            mChannelCount = channelCount;
-            mChannelMask = channelMask;
-            mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
-            memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
-            // Force underrun condition to avoid false underrun callback until first data is
-            // written to buffer (other flags are cleared)
-            mCblk->flags = CBLK_UNDERRUN_ON;
-            mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+        // construct the shared structure in-place.
+        new(mCblk) audio_track_cblk_t();
+        // clear all buffers
+        mCblk->frameCount = frameCount;
+        mCblk->sampleRate = sampleRate;
+        mChannelCount = channelCount;
+        mChannelMask = channelMask;
+        mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+        memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
+        // Force underrun condition to avoid false underrun callback until first data is
+        // written to buffer (other flags are cleared)
+        mCblk->flags = CBLK_UNDERRUN_ON;
+        mBufferEnd = (uint8_t *)mBuffer + bufferSize;
     }
 }
 
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 1ec238b..6e761ba 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -132,7 +132,7 @@ void AudioMixer::deleteTrackName(int name)
         invalidateState(1<<name);
     }
     if (track.resampler != NULL) {
-        // delete  the resampler
+        // delete the resampler
         delete track.resampler;
         track.resampler = NULL;
         track.sampleRate = mSampleRate;
diff --git a/voip/jni/rtp/AudioGroup.cpp b/voip/jni/rtp/AudioGroup.cpp
index 1139577..b9bbd16 100644
--- a/voip/jni/rtp/AudioGroup.cpp
+++ b/voip/jni/rtp/AudioGroup.cpp
@@ -809,9 +809,9 @@ bool AudioGroup::DeviceThread::threadLoop()
     AudioTrack track;
     AudioRecord record;
     if (track.set(AUDIO_STREAM_VOICE_CALL, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
-        AUDIO_CHANNEL_OUT_MONO, output) != NO_ERROR || record.set(
-        AUDIO_SOURCE_VOICE_COMMUNICATION, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
-        AUDIO_CHANNEL_IN_MONO, input) != NO_ERROR) {
+                AUDIO_CHANNEL_OUT_MONO, output) != NO_ERROR ||
+            record.set(AUDIO_SOURCE_VOICE_COMMUNICATION, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
+                AUDIO_CHANNEL_IN_MONO, input) != NO_ERROR) {
         ALOGE("cannot initialize audio device");
         return false;
     }