Use curly braces in 'if' to make it easier to add logs

Change-Id: I58b33fefdd8bf703647414157a99a3223be3531c

9 files changed, 62 insertions, 34 deletions

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index bc27bac..2fc83cc 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -157,12 +157,15 @@ AudioFlinger::AudioFlinger()
         (void) property_get("af.tee", value, "0");
         teeEnabled = atoi(value);
     }
-    if (teeEnabled & 1)
+    if (teeEnabled & 1) {
         mTeeSinkInputEnabled = true;
-    if (teeEnabled & 2)
+    }
+    if (teeEnabled & 2) {
         mTeeSinkOutputEnabled = true;
-    if (teeEnabled & 4)
+    }
+    if (teeEnabled & 4) {
         mTeeSinkTrackEnabled = true;
+    }
 #endif
 }
 
@@ -1449,8 +1452,9 @@ audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module,
     Mutex::Autolock _l(mLock);
 
     outHwDev = findSuitableHwDev_l(module, *pDevices);
-    if (outHwDev == NULL)
+    if (outHwDev == NULL) {
         return 0;
+    }
 
     audio_hw_device_t *hwDevHal = outHwDev->hwDevice();
     audio_io_handle_t id = nextUniqueId();
@@ -1664,8 +1668,9 @@ audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module,
     Mutex::Autolock _l(mLock);
 
     inHwDev = findSuitableHwDev_l(module, *pDevices);
-    if (inHwDev == NULL)
+    if (inHwDev == NULL) {
         return 0;
+    }
 
     audio_hw_device_t *inHwHal = inHwDev->hwDevice();
     audio_io_handle_t id = nextUniqueId();
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index df4e029..84c7968 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -752,8 +752,7 @@ void AudioMixer::process__validate(state_t* state, int64_t pts)
             const int i = 31 - __builtin_clz(en);
             en &= ~(1<<i);
             track_t& t = state->tracks[i];
-            if (!t.doesResample() && t.volumeRL == 0)
-            {
+            if (!t.doesResample() && t.volumeRL == 0) {
                 t.needs |= NEEDS_MUTE_ENABLED;
                 t.hook = track__nop;
             } else {
@@ -1124,8 +1123,9 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
         t.in = t.buffer.raw;
         // t.in == NULL can happen if the track was flushed just after having
         // been enabled for mixing.
-        if (t.in == NULL)
+        if (t.in == NULL) {
             enabledTracks &= ~(1<<i);
+        }
     }
 
     e0 = enabledTracks;
@@ -1445,8 +1445,9 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state,
 int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
                                        int outputFrameIndex)
 {
-    if (AudioBufferProvider::kInvalidPTS == basePTS)
+    if (AudioBufferProvider::kInvalidPTS == basePTS) {
         return AudioBufferProvider::kInvalidPTS;
+    }
 
     return basePTS + ((outputFrameIndex * sLocalTimeFreq) / t.sampleRate);
 }
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 900b411..d6ea758 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -77,24 +77,28 @@ AudioPolicyService::AudioPolicyService()
     mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this);
     /* instantiate the audio policy manager */
     rc = hw_get_module(AUDIO_POLICY_HARDWARE_MODULE_ID, &module);
-    if (rc)
+    if (rc) {
         return;
+    }
 
     rc = audio_policy_dev_open(module, &mpAudioPolicyDev);
     ALOGE_IF(rc, "couldn't open audio policy device (%s)", strerror(-rc));
-    if (rc)
+    if (rc) {
         return;
+    }
 
     rc = mpAudioPolicyDev->create_audio_policy(mpAudioPolicyDev, &aps_ops, this,
                                                &mpAudioPolicy);
     ALOGE_IF(rc, "couldn't create audio policy (%s)", strerror(-rc));
-    if (rc)
+    if (rc) {
         return;
+    }
 
     rc = mpAudioPolicy->init_check(mpAudioPolicy);
     ALOGE_IF(rc, "couldn't init_check the audio policy (%s)", strerror(-rc));
-    if (rc)
+    if (rc) {
         return;
+    }
 
     ALOGI("Loaded audio policy from %s (%s)", module->name, module->id);
 
@@ -126,10 +130,12 @@ AudioPolicyService::~AudioPolicyService()
     }
     mInputs.clear();
 
-    if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL)
+    if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL) {
         mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy);
-    if (mpAudioPolicyDev != NULL)
+    }
+    if (mpAudioPolicyDev != NULL) {
         audio_policy_dev_close(mpAudioPolicyDev);
+    }
 }
 
 status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
@@ -1105,11 +1111,13 @@ int AudioPolicyService::setStreamVolume(audio_stream_type_t stream,
 int AudioPolicyService::startTone(audio_policy_tone_t tone,
                                   audio_stream_type_t stream)
 {
-    if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION)
+    if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) {
         ALOGE("startTone: illegal tone requested (%d)", tone);
-    if (stream != AUDIO_STREAM_VOICE_CALL)
+    }
+    if (stream != AUDIO_STREAM_VOICE_CALL) {
         ALOGE("startTone: illegal stream (%d) requested for tone %d", stream,
             tone);
+    }
     mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING,
                                           AUDIO_STREAM_VOICE_CALL);
     return 0;
@@ -1500,8 +1508,9 @@ static audio_io_handle_t aps_open_dup_output(void *service,
 static int aps_close_output(void *service, audio_io_handle_t output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0)
+    if (af == 0) {
         return PERMISSION_DENIED;
+    }
 
     return af->closeOutput(output);
 }
@@ -1564,8 +1573,9 @@ static audio_io_handle_t aps_open_input_on_module(void *service,
 static int aps_close_input(void *service, audio_io_handle_t input)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0)
+    if (af == 0) {
         return PERMISSION_DENIED;
+    }
 
     return af->closeInput(input);
 }
@@ -1574,8 +1584,9 @@ static int aps_set_stream_output(void *service, audio_stream_type_t stream,
                                      audio_io_handle_t output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0)
+    if (af == 0) {
         return PERMISSION_DENIED;
+    }
 
     return af->setStreamOutput(stream, output);
 }
@@ -1585,8 +1596,9 @@ static int aps_move_effects(void *service, int session,
                                 audio_io_handle_t dst_output)
 {
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
-    if (af == 0)
+    if (af == 0) {
         return PERMISSION_DENIED;
+    }
 
     return af->moveEffects(session, src_output, dst_output);
 }
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 2c3c719..323f1a4 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -339,8 +339,9 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
             out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
             out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
             Advance(&inputIndex, &phaseFraction, phaseIncrement);
-            if (outputIndex == outputSampleCount)
+            if (outputIndex == outputSampleCount) {
                 break;
+            }
         }
 
         // process input samples
@@ -434,8 +435,9 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
             out[outputIndex++] += vl * sample;
             out[outputIndex++] += vr * sample;
             Advance(&inputIndex, &phaseFraction, phaseIncrement);
-            if (outputIndex == outputSampleCount)
+            if (outputIndex == outputSampleCount) {
                 break;
+            }
         }
 
         // process input samples
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 18e59e9..1f9714b 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -66,8 +66,9 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
     if (mBuffer.frameCount == 0) {
         mBuffer.frameCount = inFrameCount;
         provider->getNextBuffer(&mBuffer, mPTS);
-        if (mBuffer.raw == NULL)
+        if (mBuffer.raw == NULL) {
             return;
+        }
         // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
     }
     int16_t *in = mBuffer.i16;
@@ -97,8 +98,9 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
                 mBuffer.frameCount = inFrameCount;
                 provider->getNextBuffer(&mBuffer,
                                         calculateOutputPTS(outputIndex / 2));
-                if (mBuffer.raw == NULL)
+                if (mBuffer.raw == NULL) {
                     goto save_state;  // ugly, but efficient
+                }
                 in = mBuffer.i16;
                 // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
             }
@@ -132,8 +134,9 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
     if (mBuffer.frameCount == 0) {
         mBuffer.frameCount = inFrameCount;
         provider->getNextBuffer(&mBuffer, mPTS);
-        if (mBuffer.raw == NULL)
+        if (mBuffer.raw == NULL) {
             return;
+        }
         // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
     }
     int16_t *in = mBuffer.i16;
@@ -163,8 +166,9 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
                 mBuffer.frameCount = inFrameCount;
                 provider->getNextBuffer(&mBuffer,
                                         calculateOutputPTS(outputIndex / 2));
-                if (mBuffer.raw == NULL)
+                if (mBuffer.raw == NULL) {
                     goto save_state;  // ugly, but efficient
+                }
                 // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
                 in = mBuffer.i16;
             }
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index d5a21a7..2a18104 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -116,8 +116,9 @@ status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle)
             continue;
         }
         // first non destroyed handle is considered in control
-        if (controlHandle == NULL)
+        if (controlHandle == NULL) {
             controlHandle = h;
+        }
         if (h->priority() <= priority) {
             break;
         }
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index ad9f4f2..4b36987 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -427,8 +427,9 @@ bool FastMixer::threadLoop()
             }
 
             int64_t pts;
-            if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts)))
+            if (outputSink == NULL || (OK != outputSink->getNextWriteTimestamp(&pts))) {
                 pts = AudioBufferProvider::kInvalidPTS;
+            }
 
             // process() is CPU-bound
             mixer->process(pts);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 580db70..862258f 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -4281,8 +4281,9 @@ bool AudioFlinger::RecordThread::threadLoop()
                             int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
                             int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
                                     mActiveTrack->mFrameSize;
-                            if (framesIn > framesOut)
+                            if (framesIn > framesOut) {
                                 framesIn = framesOut;
+                            }
                             mRsmpInIndex += framesIn;
                             framesOut -= framesIn;
                             if (mChannelCount == mReqChannelCount) {
@@ -4989,8 +4990,7 @@ void AudioFlinger::RecordThread::readInputParameters()
     mFrameCount = mBufferSize / mFrameSize;
     mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
 
-    if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
-    {
+    if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) {
         int channelCount;
         // 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.
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 6039078..e173d4b 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -943,15 +943,17 @@ status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
 
         mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
                                               "AudioFlingerTimed");
-        if (mTimedMemoryDealer == NULL)
+        if (mTimedMemoryDealer == NULL) {
             return NO_MEMORY;
+        }
     }
 
     sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
     if (newBuffer == NULL) {
         newBuffer = mTimedMemoryDealer->allocate(size);
-        if (newBuffer == NULL)
+        if (newBuffer == NULL) {
             return NO_MEMORY;
+        }
     }
 
     *buffer = newBuffer;