Re-format to decrease the maximum line length of files to 100 characters
Test: compiles OK
Change-Id: Ibe663032cd390ed2bcca6dc921d47732e6e15e21
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 4c1fbd7..6c7cdde 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -66,7 +66,8 @@
// ---------------------------------------------------------------------------
AudioRecord::AudioRecord(const String16 &opPackageName)
- : mActive(false), mStatus(NO_INIT), mOpPackageName(opPackageName), mSessionId(AUDIO_SESSION_ALLOCATE),
+ : mActive(false), mStatus(NO_INIT), mOpPackageName(opPackageName),
+ mSessionId(AUDIO_SESSION_ALLOCATE),
mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT),
mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE), mPortId(AUDIO_PORT_HANDLE_NONE)
{
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 1908f0e..5cd2789 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -554,7 +554,8 @@
"channel mask %#x frameCount %zu frameCountHAL %zu deviceId %d",
event == AUDIO_OUTPUT_CONFIG_CHANGED ? "output" : "input",
ioDesc->mIoHandle, ioDesc->mSamplingRate, ioDesc->mFormat,
- ioDesc->mChannelMask, ioDesc->mFrameCount, ioDesc->mFrameCountHAL, ioDesc->getDeviceId());
+ ioDesc->mChannelMask, ioDesc->mFrameCount, ioDesc->mFrameCountHAL,
+ ioDesc->getDeviceId());
} break;
}
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index f878be9..d590cb7 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -917,7 +917,8 @@
}
// Check resampler ratios are within bounds
- if ((uint64_t)effectiveRate > (uint64_t)mSampleRate * (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
+ if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
+ (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value",
playbackRate.mSpeed, playbackRate.mPitch);
return BAD_VALUE;
@@ -1274,9 +1275,10 @@
mFlags, mSelectedDeviceId, &mPortId);
if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
- ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x,"
- " channel mask %#x, flags %#x",
- mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
+ ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u,"
+ " format %#x, channel mask %#x, flags %#x",
+ mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask,
+ mFlags);
return BAD_VALUE;
}
{
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 3d1f268..fb18b05 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -289,7 +289,8 @@
sessionId,
&streamType, client.clientUid,
&fullConfig,
- (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT),
+ (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
+ AUDIO_OUTPUT_FLAG_DIRECT),
*deviceId, &portId);
} else {
ret = AudioSystem::getInputForAttr(attr, &io,
@@ -1979,7 +1980,8 @@
uint32_t *latencyMs,
audio_output_flags_t flags)
{
- ALOGI("openOutput() this %p, module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
+ ALOGI("openOutput() this %p, module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, "
+ "flags %x",
this, module,
(devices != NULL) ? *devices : 0,
config->sample_rate,
@@ -2077,13 +2079,13 @@
if (mPlaybackThreads.size()) {
PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
if (dstThread != NULL) {
- // audioflinger lock is held here so the acquisition order of thread locks does not
- // matter
+ // audioflinger lock is held so order of thread lock acquisition doesn't matter
Mutex::Autolock _dl(dstThread->mLock);
Mutex::Autolock _sl(playbackThread->mLock);
Vector< sp<EffectChain> > effectChains = playbackThread->getEffectChains_l();
for (size_t i = 0; i < effectChains.size(); i ++) {
- moveEffectChain_l(effectChains[i]->sessionId(), playbackThread.get(), dstThread, true);
+ moveEffectChain_l(effectChains[i]->sessionId(), playbackThread.get(),
+ dstThread, true);
}
}
}
@@ -2255,7 +2257,8 @@
inHwDev, inputStream,
primaryOutputDevice_l(), devices, mSystemReady);
mMmapThreads.add(*input, thread);
- ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input, thread.get());
+ ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input,
+ thread.get());
return thread;
} else {
#ifdef TEE_SINK
@@ -2361,7 +2364,7 @@
}
}
if (chain != 0) {
- // first check if a record thread is already opened with a client on the same session.
+ // first check if a record thread is already opened with a client on same session.
// This should only happen in case of overlap between one thread tear down and the
// creation of its replacement
size_t i;
@@ -2378,7 +2381,7 @@
break;
}
}
- // put the chain aside if we could not find a record thread with the same session id.
+ // put the chain aside if we could not find a record thread with the same session id
if (i == mRecordThreads.size()) {
putOrphanEffectChain_l(chain);
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 44fd512..420f7e8 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -561,7 +561,8 @@
virtual status_t createMmapBuffer(int32_t minSizeFrames,
struct audio_mmap_buffer_info *info);
virtual status_t getMmapPosition(struct audio_mmap_position *position);
- virtual status_t start(const MmapStreamInterface::Client& client, audio_port_handle_t *handle);
+ virtual status_t start(const MmapStreamInterface::Client& client,
+ audio_port_handle_t *handle);
virtual status_t stop(audio_port_handle_t handle);
private:
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b10e42c..cf7ba43 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5599,7 +5599,7 @@
ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list",
track->name());
tracksToRemove->add(track);
- // indicate to client process that the track was disabled because of underrun;
+ // tell client process that the track was disabled because of underrun;
// it will then automatically call start() when data is available
track->disable();
}
@@ -7215,7 +7215,8 @@
// Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
mRsmpInFramesOA = mRsmpInFramesP2 + mFrameCount - 1;
(void)posix_memalign(&mRsmpInBuffer, 32, mRsmpInFramesOA * mFrameSize);
- memset(mRsmpInBuffer, 0, mRsmpInFramesOA * mFrameSize); // if posix_memalign fails, will segv here.
+ // if posix_memalign fails, will segv here.
+ memset(mRsmpInBuffer, 0, mRsmpInFramesOA * mFrameSize);
// AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
// But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
@@ -7459,7 +7460,8 @@
return mThread->getMmapPosition(position);
}
-status_t AudioFlinger::MmapThreadHandle::start(const MmapStreamInterface::Client& client, audio_port_handle_t *handle)
+status_t AudioFlinger::MmapThreadHandle::start(const MmapStreamInterface::Client& client,
+ audio_port_handle_t *handle)
{
if (mThread == 0) {
@@ -8051,8 +8053,8 @@
return BAD_VALUE;
}
if (isOutput() && ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
- ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback mmap thread",
- desc->name);
+ ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback mmap "
+ "thread", desc->name);
return BAD_VALUE;
}
@@ -8284,7 +8286,8 @@
{
MmapThread::dumpInternals(fd, args);
- dprintf(fd, " Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n", mStreamType, mStreamVolume, mHalVolFloat, mStreamMute);
+ dprintf(fd, " Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n",
+ mStreamType, mStreamVolume, mHalVolFloat, mStreamMute);
dprintf(fd, " Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute);
}