Whitespace and indentation

Fix indentation to be multiple of 4.
Make it easier to search:
  sp< not sp < to
  "switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.

Use git diff -b or -w to verify.

Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e

8 files changed, 143 insertions, 142 deletions

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index fad7087..bfa4a49 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -190,13 +190,13 @@ void AudioFlinger::onFirstRef()
             continue;
 
         ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
-             mod->name, mod->id);
+            mod->name, mod->id);
         mAudioHwDevs.push(dev);
 
         if (mPrimaryHardwareDev == NULL) {
             mPrimaryHardwareDev = dev;
             ALOGI("Using '%s' (%s.%s) as the primary audio interface",
-                 mod->name, mod->id, audio_interfaces[i]);
+                mod->name, mod->id, audio_interfaces[i]);
         }
     }
 
@@ -515,7 +515,7 @@ sp<IAudioTrack> AudioFlinger::createTrack(
     }
 
 Exit:
-    if(status) {
+    if (status != NULL) {
         *status = lStatus;
     }
     return trackHandle;
@@ -610,7 +610,7 @@ status_t AudioFlinger::setMasterVolume(float value)
     mMasterVolume   = value;
     mMasterVolumeSW = swmv;
     for (size_t i = 0; i < mPlaybackThreads.size(); i++)
-       mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
+        mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
 
     return NO_ERROR;
 }
@@ -642,7 +642,7 @@ status_t AudioFlinger::setMode(audio_mode_t mode)
         Mutex::Autolock _l(mLock);
         mMode = mode;
         for (size_t i = 0; i < mPlaybackThreads.size(); i++)
-           mPlaybackThreads.valueAt(i)->setMode(mode);
+            mPlaybackThreads.valueAt(i)->setMode(mode);
     }
 
     return ret;
@@ -693,7 +693,7 @@ status_t AudioFlinger::setMasterMute(bool muted)
     // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger
     mMasterMute = muted;
     for (size_t i = 0; i < mPlaybackThreads.size(); i++)
-       mPlaybackThreads.valueAt(i)->setMasterMute(muted);
+        mPlaybackThreads.valueAt(i)->setMasterMute(muted);
 
     return NO_ERROR;
 }
@@ -761,7 +761,7 @@ status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
 
     if (thread == NULL) {
         for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
-           mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
+            mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
         }
     } else {
         thread->setStreamVolume(stream, value);
@@ -786,7 +786,7 @@ status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
     AutoMutex lock(mLock);
     mStreamTypes[stream].mute = muted;
     for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
-       mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
+        mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
 
     return NO_ERROR;
 }
@@ -1161,7 +1161,7 @@ void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
 void AudioFlinger::ThreadBase::processConfigEvents()
 {
     mLock.lock();
-    while(!mConfigEvents.isEmpty()) {
+    while (!mConfigEvents.isEmpty()) {
         ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
         ConfigEvent configEvent = mConfigEvents[0];
         mConfigEvents.removeAt(0);
@@ -1345,13 +1345,13 @@ void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectCha
             mSuspendedSessions.editValueAt(index);
 
     for (size_t i = 0; i < sessionEffects.size(); i++) {
-        sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
+        sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
         for (int j = 0; j < desc->mRefCount; j++) {
             if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
                 chain->setEffectSuspendedAll_l(true);
             } else {
                 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
-                     desc->mType.timeLow);
+                    desc->mType.timeLow);
                 chain->setEffectSuspended_l(&desc->mType, true);
             }
         }
@@ -1386,7 +1386,7 @@ void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *ty
     }
     index = sessionEffects.indexOfKey(key);
 
-    sp <SuspendedSessionDesc> desc;
+    sp<SuspendedSessionDesc> desc;
     if (suspend) {
         if (index >= 0) {
             desc = sessionEffects.valueAt(index);
@@ -1659,14 +1659,14 @@ sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTra
         // createTrack() was called by the client process.
         if (!mStreamTypes[streamType].valid) {
             ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
-                 this, streamType);
+                this, streamType);
             android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
         }
     }
     lStatus = NO_ERROR;
 
 Exit:
-    if(status) {
+    if (status) {
         *status = lStatus;
     }
     return track;
@@ -2645,10 +2645,10 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l()
             status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
                                                     keyValuePair.string());
             if (!mStandby && status == INVALID_OPERATION) {
-               mOutput->stream->common.standby(&mOutput->stream->common);
-               mStandby = true;
-               mBytesWritten = 0;
-               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                mOutput->stream->common.standby(&mOutput->stream->common);
+                mStandby = true;
+                mBytesWritten = 0;
+                status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
                                                        keyValuePair.string());
             }
             if (status == NO_ERROR && reconfig) {
@@ -2902,7 +2902,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix()
         size_t count = mFrameCount * mChannelCount;
         uint8_t *src = (uint8_t *)mMixBuffer + count-1;
         int16_t *dst = mMixBuffer + count-1;
-        while(count--) {
+        while (count--) {
             *dst-- = (int16_t)(*src--^0x80) << 8;
         }
     }
@@ -2955,7 +2955,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix()
         size_t count = mFrameCount * mChannelCount;
         int16_t *src = mMixBuffer;
         uint8_t *dst = (uint8_t *)mMixBuffer;
-        while(count--) {
+        while (count--) {
             *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
         }
     }
@@ -3014,10 +3014,10 @@ bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
             status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
                                                     keyValuePair.string());
             if (!mStandby && status == INVALID_OPERATION) {
-               mOutput->stream->common.standby(&mOutput->stream->common);
-               mStandby = true;
-               mBytesWritten = 0;
-               status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+                mOutput->stream->common.standby(&mOutput->stream->common);
+                mStandby = true;
+                mBytesWritten = 0;
+                status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
                                                        keyValuePair.string());
             }
             if (status == NO_ERROR && reconfig) {
@@ -3208,7 +3208,7 @@ void AudioFlinger::DuplicatingThread::updateWaitTime_l()
 bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks)
 {
     for (size_t i = 0; i < outputTracks.size(); i++) {
-        sp <ThreadBase> thread = outputTracks[i]->thread().promote();
+        sp<ThreadBase> thread = outputTracks[i]->thread().promote();
         if (thread == 0) {
             ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
             return false;
@@ -3264,14 +3264,14 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
     ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
 
     // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
-   size_t size = sizeof(audio_track_cblk_t);
-   uint8_t channelCount = popcount(channelMask);
-   size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
-   if (sharedBuffer == 0) {
-       size += bufferSize;
-   }
-
-   if (client != NULL) {
+    size_t size = sizeof(audio_track_cblk_t);
+    uint8_t channelCount = popcount(channelMask);
+    size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
+    if (sharedBuffer == 0) {
+        size += bufferSize;
+    }
+
+    if (client != NULL) {
         mCblkMemory = client->heap()->allocate(size);
         if (mCblkMemory != 0) {
             mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
@@ -3298,22 +3298,22 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
             client->heap()->dump("AudioTrack");
             return;
         }
-   } 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;
-   }
+    } 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;
+    }
 }
 
 AudioFlinger::ThreadBase::TrackBase::~TrackBase()
@@ -3491,22 +3491,22 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
 
 // AudioBufferProvider interface
 status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
-    AudioBufferProvider::Buffer* buffer, int64_t pts)
+        AudioBufferProvider::Buffer* buffer, int64_t pts)
 {
-     audio_track_cblk_t* cblk = this->cblk();
-     uint32_t framesReady;
-     uint32_t framesReq = buffer->frameCount;
+    audio_track_cblk_t* cblk = this->cblk();
+    uint32_t framesReady;
+    uint32_t framesReq = buffer->frameCount;
 
-     // Check if last stepServer failed, try to step now
-     if (mStepServerFailed) {
-         if (!step())  goto getNextBuffer_exit;
-         ALOGV("stepServer recovered");
-         mStepServerFailed = false;
-     }
+    // Check if last stepServer failed, try to step now
+    if (mStepServerFailed) {
+        if (!step())  goto getNextBuffer_exit;
+        ALOGV("stepServer recovered");
+        mStepServerFailed = false;
+    }
 
-     framesReady = cblk->framesReady();
+    framesReady = cblk->framesReady();
 
-     if (CC_LIKELY(framesReady)) {
+    if (CC_LIKELY(framesReady)) {
         uint32_t s = cblk->server;
         uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
 
@@ -3518,21 +3518,21 @@ status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
             framesReq = bufferEnd - s;
         }
 
-         buffer->raw = getBuffer(s, framesReq);
-         if (buffer->raw == NULL) goto getNextBuffer_exit;
+        buffer->raw = getBuffer(s, framesReq);
+        if (buffer->raw == NULL) goto getNextBuffer_exit;
 
-         buffer->frameCount = framesReq;
+        buffer->frameCount = framesReq;
         return NO_ERROR;
-     }
+    }
 
 getNextBuffer_exit:
-     buffer->raw = NULL;
-     buffer->frameCount = 0;
-     ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
-     return NOT_ENOUGH_DATA;
+    buffer->raw = NULL;
+    buffer->frameCount = 0;
+    ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
+    return NOT_ENOUGH_DATA;
 }
 
-uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{
+uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const {
     return mCblk->framesReady();
 }
 
@@ -3684,8 +3684,8 @@ status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
     status_t status = DEAD_OBJECT;
     sp<ThreadBase> thread = mThread.promote();
     if (thread != 0) {
-       PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
-       status = playbackThread->attachAuxEffect(this, EffectId);
+        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+        status = playbackThread->attachAuxEffect(this, EffectId);
     }
     return status;
 }
@@ -4104,14 +4104,14 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
         mOverflow(false)
 {
     if (mCblk != NULL) {
-       ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
-       if (format == AUDIO_FORMAT_PCM_16_BIT) {
-           mCblk->frameSize = mChannelCount * sizeof(int16_t);
-       } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
-           mCblk->frameSize = mChannelCount * sizeof(int8_t);
-       } else {
-           mCblk->frameSize = sizeof(int8_t);
-       }
+        ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+        if (format == AUDIO_FORMAT_PCM_16_BIT) {
+            mCblk->frameSize = mChannelCount * sizeof(int16_t);
+        } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
+            mCblk->frameSize = mChannelCount * sizeof(int8_t);
+        } else {
+            mCblk->frameSize = sizeof(int8_t);
+        }
     }
 }
 
@@ -4130,7 +4130,7 @@ status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvi
     uint32_t framesAvail;
     uint32_t framesReq = buffer->frameCount;
 
-     // Check if last stepServer failed, try to step now
+    // Check if last stepServer failed, try to step now
     if (mStepServerFailed) {
         if (!step()) goto getNextBuffer_exit;
         ALOGV("stepServer recovered");
@@ -4976,7 +4976,7 @@ Exit:
 status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid)
 {
     ALOGV("RecordThread::start tid=%d", tid);
-    sp <ThreadBase> strongMe = this;
+    sp<ThreadBase> strongMe = this;
     status_t status = NO_ERROR;
     {
         AutoMutex lock(mLock);
@@ -5029,7 +5029,7 @@ startError:
 
 void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
     ALOGV("RecordThread::stop");
-    sp <ThreadBase> strongMe = this;
+    sp<ThreadBase> strongMe = this;
     {
         AutoMutex lock(mLock);
         if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
@@ -5196,8 +5196,9 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
         if (status == NO_ERROR) {
             status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
             if (status == INVALID_OPERATION) {
-               mInput->stream->common.standby(&mInput->stream->common);
-               status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
+                mInput->stream->common.standby(&mInput->stream->common);
+                status = mInput->stream->common.set_parameters(&mInput->stream->common,
+                        keyValuePair.string());
             }
             if (reconfig) {
                 if (status == BAD_VALUE &&
@@ -5285,8 +5286,8 @@ void AudioFlinger::RecordThread::readInputParameters()
     if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
     {
         int channelCount;
-         // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
-         // stereo to mono post process as the resampler always outputs stereo.
+        // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+        // stereo to mono post process as the resampler always outputs stereo.
         if (mChannelCount == 1 && mReqChannelCount == 2) {
             channelCount = 1;
         } else {
@@ -5460,7 +5461,7 @@ status_t AudioFlinger::closeOutput(audio_io_handle_t output)
 {
     // keep strong reference on the playback thread so that
     // it is not destroyed while exit() is executed
-    sp <PlaybackThread> thread;
+    sp<PlaybackThread> thread;
     {
         Mutex::Autolock _l(mLock);
         thread = checkPlaybackThread_l(output);
@@ -5613,7 +5614,7 @@ status_t AudioFlinger::closeInput(audio_io_handle_t input)
 {
     // keep strong reference on the record thread so that
     // it is not destroyed while exit() is executed
-    sp <RecordThread> thread;
+    sp<RecordThread> thread;
     {
         Mutex::Autolock _l(mLock);
         thread = checkRecordThread_l(input);
@@ -5746,7 +5747,7 @@ void AudioFlinger::purgeStaleEffects_l() {
             AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
             if (ref->mSessionid == sessionid) {
                 ALOGV(" session %d still exists for %d with %d refs",
-                     sessionid, ref->mPid, ref->mCnt);
+                    sessionid, ref->mPid, ref->mCnt);
                 found = true;
                 break;
             }
@@ -5979,7 +5980,7 @@ sp<IEffect> AudioFlinger::createEffect(pid_t pid,
         // because of code checking output when entering the function.
         // Note: io is never 0 when creating an effect on an input
         if (io == 0) {
-             // look for the thread where the specified audio session is present
+            // look for the thread where the specified audio session is present
             for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
                 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
                     io = mPlaybackThreads.keyAt(i);
@@ -5987,12 +5988,12 @@ sp<IEffect> AudioFlinger::createEffect(pid_t pid,
                 }
             }
             if (io == 0) {
-               for (size_t i = 0; i < mRecordThreads.size(); i++) {
-                   if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
-                       io = mRecordThreads.keyAt(i);
-                       break;
-                   }
-               }
+                for (size_t i = 0; i < mRecordThreads.size(); i++) {
+                    if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
+                        io = mRecordThreads.keyAt(i);
+                        break;
+                    }
+                }
             }
             // If no output thread contains the requested session ID, default to
             // first output. The effect chain will be moved to the correct output
@@ -6023,7 +6024,7 @@ sp<IEffect> AudioFlinger::createEffect(pid_t pid,
     }
 
 Exit:
-    if(status) {
+    if (status != NULL) {
         *status = lStatus;
     }
     return handle;
@@ -6226,7 +6227,7 @@ Exit:
         handle.clear();
     }
 
-    if(status) {
+    if (status != NULL) {
         *status = lStatus;
     }
     return handle;
@@ -6296,7 +6297,7 @@ void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
 }
 
 void AudioFlinger::ThreadBase::lockEffectChains_l(
-        Vector<sp <AudioFlinger::EffectChain> >& effectChains)
+        Vector< sp<AudioFlinger::EffectChain> >& effectChains)
 {
     effectChains = mEffectChains;
     for (size_t i = 0; i < mEffectChains.size(); i++) {
@@ -6305,7 +6306,7 @@ void AudioFlinger::ThreadBase::lockEffectChains_l(
 }
 
 void AudioFlinger::ThreadBase::unlockEffectChains(
-        const Vector<sp <AudioFlinger::EffectChain> >& effectChains)
+        const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
 {
     for (size_t i = 0; i < effectChains.size(); i++) {
         effectChains[i]->unlock();
@@ -6481,7 +6482,7 @@ status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
 
 void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
 {
-     for (size_t i = 0; i < mTracks.size(); ++i) {
+    for (size_t i = 0; i < mTracks.size(); ++i) {
         sp<Track> track = mTracks[i];
         if (track->auxEffectId() == effectId) {
             attachAuxEffect_l(track, 0);
@@ -7265,7 +7266,7 @@ AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
         if (mCblk != NULL) {
             new(mCblk) effect_param_cblk_t();
             mBuffer = (uint8_t *)mCblk + bufOffset;
-         }
+        }
     } else {
         ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
         return;
@@ -8032,7 +8033,7 @@ void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModul
             }
         }
         ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
-             effect->desc().type.timeLow);
+            effect->desc().type.timeLow);
         sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
         // if effect is requested to suspended but was not yet enabled, supend it now.
         if (desc->mEffect == 0) {
@@ -8045,7 +8046,7 @@ void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModul
             return;
         }
         ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
-             effect->desc().type.timeLow);
+            effect->desc().type.timeLow);
         sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
         desc->mEffect.clear();
         effect->setSuspended(false);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index f3c8dd2..0e4b24a 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -465,9 +465,9 @@ private:
                     // ThreadBase mutex before processing the mixer and effects. This guarantees the
                     // integrity of the chains during the process.
                     // Also sets the parameter 'effectChains' to current value of mEffectChains.
-                    void lockEffectChains_l(Vector<sp <EffectChain> >& effectChains);
+                    void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains);
                     // unlock effect chains after process
-                    void unlockEffectChains(const Vector<sp<EffectChain> >& effectChains);
+                    void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains);
                     // set audio mode to all effect chains
                     void setMode(audio_mode_t mode);
                     // get effect module with corresponding ID on specified audio session
@@ -1056,7 +1056,7 @@ private:
         virtual     uint32_t    activeSleepTimeUs();
 
     private:
-                    bool        outputsReady(const SortedVector<sp<OutputTrack> > &outputTracks);
+                    bool        outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
     protected:
         // threadLoop snippets
         virtual     void        threadLoop_mix();
@@ -1504,7 +1504,7 @@ mutable Mutex               mLock;      // mutex for process, commands and handl
 
         uint32_t strategy() const { return mStrategy; }
         void setStrategy(uint32_t strategy)
-                 { mStrategy = strategy; }
+                { mStrategy = strategy; }
 
         // suspend effect of the given type
         void setEffectSuspended_l(const effect_uuid_t *type,
@@ -1544,7 +1544,7 @@ mutable Mutex               mLock;      // mutex for process, commands and handl
 
         wp<ThreadBase> mThread;     // parent mixer thread
         Mutex mLock;                // mutex protecting effect list
-        Vector<sp<EffectModule> > mEffects; // list of effect modules
+        Vector< sp<EffectModule> > mEffects; // list of effect modules
         int mSessionId;             // audio session ID
         int16_t *mInBuffer;         // chain input buffer
         int16_t *mOutBuffer;        // chain output buffer
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index d57326b..c23eb04 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -649,7 +649,7 @@ bool AudioPolicyService::AudioCommandThread::threadLoop()
     mLock.lock();
     while (!exitPending())
     {
-        while(!mAudioCommands.isEmpty()) {
+        while (!mAudioCommands.isEmpty()) {
             nsecs_t curTime = systemTime();
             // commands are sorted by increasing time stamp: execute them from index 0 and up
             if (mAudioCommands[0]->mTime <= curTime) {
@@ -693,16 +693,16 @@ bool AudioPolicyService::AudioCommandThread::threadLoop()
                     delete data;
                     }break;
                 case SET_PARAMETERS: {
-                     ParametersData *data = (ParametersData *)command->mParam;
-                     ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
-                             data->mKeyValuePairs.string(), data->mIO);
-                     command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
-                     if (command->mWaitStatus) {
-                         command->mCond.signal();
-                         mWaitWorkCV.wait(mLock);
-                     }
-                     delete data;
-                     }break;
+                    ParametersData *data = (ParametersData *)command->mParam;
+                    ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
+                            data->mKeyValuePairs.string(), data->mIO);
+                    command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
+                    if (command->mWaitStatus) {
+                        command->mCond.signal();
+                        mWaitWorkCV.wait(mLock);
+                    }
+                    delete data;
+                    }break;
                 case SET_VOICE_VOLUME: {
                     VoiceVolumeData *data = (VoiceVolumeData *)command->mParam;
                     ALOGV("AudioCommandThread() processing set voice volume volume %f",
@@ -916,19 +916,19 @@ void AudioPolicyService::AudioCommandThread::insertCommand_l(AudioCommand *comma
             AudioParameter param = AudioParameter(data->mKeyValuePairs);
             AudioParameter param2 = AudioParameter(data2->mKeyValuePairs);
             for (size_t j = 0; j < param.size(); j++) {
-               String8 key;
-               String8 value;
-               param.getAt(j, key, value);
-               for (size_t k = 0; k < param2.size(); k++) {
-                  String8 key2;
-                  String8 value2;
-                  param2.getAt(k, key2, value2);
-                  if (key2 == key) {
-                      param2.remove(key2);
-                      ALOGV("Filtering out parameter %s", key2.string());
-                      break;
-                  }
-               }
+                String8 key;
+                String8 value;
+                param.getAt(j, key, value);
+                for (size_t k = 0; k < param2.size(); k++) {
+                    String8 key2;
+                    String8 value2;
+                    param2.getAt(k, key2, value2);
+                    if (key2 == key) {
+                        param2.remove(key2);
+                        ALOGV("Filtering out parameter %s", key2.string());
+                        break;
+                    }
+                }
             }
             // if all keys have been filtered out, remove the command.
             // otherwise, update the key value pairs
@@ -1020,7 +1020,7 @@ int AudioPolicyService::startTone(audio_policy_tone_t tone,
         ALOGE("startTone: illegal tone requested (%d)", tone);
     if (stream != AUDIO_STREAM_VOICE_CALL)
         ALOGE("startTone: illegal stream (%d) requested for tone %d", stream,
-             tone);
+            tone);
     mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING,
                                           AUDIO_STREAM_VOICE_CALL);
     return 0;
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index 7119b90..9ed905d 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -311,8 +311,8 @@ private:
 
     mutable Mutex mLock;    // prevents concurrent access to AudioPolicy manager functions changing
                             // device connection state  or routing
-    sp <AudioCommandThread> mAudioCommandThread;    // audio commands thread
-    sp <AudioCommandThread> mTonePlaybackThread;     // tone playback thread
+    sp<AudioCommandThread> mAudioCommandThread;     // audio commands thread
+    sp<AudioCommandThread> mTonePlaybackThread;     // tone playback thread
     struct audio_policy_device *mpAudioPolicyDev;
     struct audio_policy *mpAudioPolicy;
     KeyedVector< audio_source_t, InputSourceDesc* > mInputSources;
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 398ba0b..fbb54cf 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -227,7 +227,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
             mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
             mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
             provider->releaseBuffer(&mBuffer);
-             // mBuffer.frameCount == 0 now so we reload a new buffer
+            // mBuffer.frameCount == 0 now so we reload a new buffer
         }
 
         int16_t *in = mBuffer.i16;
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 9deb796..1610e00 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -33,7 +33,7 @@ public:
     //  HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
     // NOTE: high quality SRC will only be supported for
     // certain fixed rate conversions. Sample rate cannot be
-    // changed dynamically. 
+    // changed dynamically.
     enum src_quality {
         DEFAULT=0,
         LOW_QUALITY=1,
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index b72b62a..892785a 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -55,7 +55,7 @@ private:
         p->y1 = p->y2;
         p->y2 = p->y3;
         p->y3 = in;
-        p->a = (3 * (p->y1 - p->y2) - p->y0 + p->y3) >> 1;            
+        p->a = (3 * (p->y1 - p->y2) - p->y0 + p->y3) >> 1;
         p->b = (p->y2 << 1) + p->y0 - (((5 * p->y1 + p->y3)) >> 1);
         p->c = (p->y2 - p->y0) >> 1;
     }
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index d373c08..76662d8 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -222,7 +222,7 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
                 } else {
                     read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
                 }
-           }
+            }
         }
         int16_t *in = mBuffer.i16;
         const size_t frameCount = mBuffer.frameCount;
@@ -247,7 +247,7 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
                 if (inputIndex >= frameCount)
                     break;  // need a new buffer
                 read<CHANNELS>(impulse, phaseFraction, in, inputIndex);
-            } else if(phaseIndex == 2) {    // maximum value
+            } else if (phaseIndex == 2) {    // maximum value
                 inputIndex++;
                 if (inputIndex >= frameCount)
                     break;  // 0 frame available, 2 frames needed