Android/frameworks_base
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Merge "Use NULL not 0 for pointers"

Browse Source
This commit is contained in:
Glenn Kasten
2011-12-15 16:16:55 -08:00
committed by Android (Google) Code Review
parent f15974a37f c434c90cc2
commit 2efd1184f9
3 changed files with 29 additions and 29 deletions
Download Patch File Download Diff File Expand all files Collapse all files

30
services/audioflinger/AudioFlinger.cpp
View File

@@ -160,7 +160,7 @@ static const char * const audio_interfaces[] = {
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
mBtNrecIsOff(false)
{
}
@@ -1371,7 +1371,7 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
int id,
uint32_t device)
: ThreadBase(audioFlinger, id, device),
mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
{
snprintf(mName, kNameLength, "AudioOut_%d", id);
@@ -1838,7 +1838,7 @@ uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs()
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device),
mAudioMixer(0)
mAudioMixer(NULL)
{
mType = ThreadBase::MIXER;
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
@@ -2772,7 +2772,7 @@ bool AudioFlinger::DirectOutputThread::threadLoop()
while (frameCount) {
buffer.frameCount = frameCount;
activeTrack->getNextBuffer(&buffer);
if (UNLIKELY(buffer.raw == 0)) {
if (UNLIKELY(buffer.raw == NULL)) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
@@ -3270,7 +3270,7 @@ AudioFlinger::ThreadBase::TrackBase::~TrackBase()
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
buffer->raw = 0;
buffer->raw = NULL;
mFrameCount = buffer->frameCount;
step();
buffer->frameCount = 0;
@@ -3463,14 +3463,14 @@ status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider:
}
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
if (buffer->raw == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
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;
@@ -3711,14 +3711,14 @@ status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvi
}
buffer->raw = getBuffer(s, framesReq);
if (buffer->raw == 0) goto getNextBuffer_exit;
if (buffer->raw == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
buffer->raw = 0;
buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
@@ -4223,7 +4223,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
int id,
uint32_t device) :
ThreadBase(audioFlinger, id, device),
mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL)
{
mType = ThreadBase::RECORD;
@@ -4238,7 +4238,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
if (mResampler != 0) {
if (mResampler != NULL) {
delete mResampler;
delete[] mRsmpOutBuffer;
}
@@ -4332,7 +4332,7 @@ bool AudioFlinger::RecordThread::threadLoop()
buffer.frameCount = mFrameCount;
if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
if (mResampler == 0) {
if (mResampler == NULL) {
// no resampling
while (framesOut) {
size_t framesIn = mFrameCount - mRsmpInIndex;
@@ -4590,7 +4590,7 @@ status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
result.append(buffer);
snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
result.append(buffer);
snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
result.append(buffer);
snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
result.append(buffer);
@@ -4625,7 +4625,7 @@ status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer*
mInput->stream->common.standby(&mInput->stream->common);
usleep(kRecordThreadSleepUs);
}
buffer->raw = 0;
buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
@@ -4788,7 +4788,7 @@ void AudioFlinger::RecordThread::readInputParameters()
if (mRsmpInBuffer) delete mRsmpInBuffer;
if (mRsmpOutBuffer) delete mRsmpOutBuffer;
if (mResampler) delete mResampler;
mResampler = 0;
mResampler = NULL;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);

4
services/audioflinger/AudioFlinger.h
View File

@@ -703,7 +703,7 @@ private:
virtual status_t readyToRun();
virtual void onFirstRef();
virtual status_t initCheck() const { return (mOutput == 0) ? NO_INIT : NO_ERROR; }
virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
virtual uint32_t latency() const;
@@ -980,7 +980,7 @@ private:
virtual status_t readyToRun();
virtual void onFirstRef();
virtual status_t initCheck() const { return (mInput == 0) ? NO_INIT : NO_ERROR; }
virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,

24
services/audioflinger/AudioMixer.cpp
View File

@@ -50,8 +50,8 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
mState.outputTemp = 0;
mState.resampleTemp = 0;
mState.outputTemp = NULL;
mState.resampleTemp = NULL;
mState.hook = process__nop;
track_t* t = mState.tracks;
for (int i=0 ; i<32 ; i++) {
@@ -73,11 +73,11 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
t->buffer.raw = 0;
t->bufferProvider = 0;
t->hook = 0;
t->resampler = 0;
t->bufferProvider = NULL;
t->hook = NULL;
t->resampler = NULL;
t->sampleRate = mSampleRate;
t->in = 0;
t->in = NULL;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
t++;
@@ -133,7 +133,7 @@ void AudioMixer::deleteTrackName(int name)
if (track.resampler) {
// delete the resampler
delete track.resampler;
track.resampler = 0;
track.resampler = NULL;
track.sampleRate = mSampleRate;
invalidateState(1<<name);
}
@@ -296,7 +296,7 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
if (value!=devSampleRate || resampler) {
if (sampleRate != value) {
sampleRate = value;
if (resampler == 0) {
if (resampler == NULL) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
}
@@ -308,12 +308,12 @@ bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
bool AudioMixer::track_t::doesResample() const
{
return resampler != 0;
return resampler != NULL;
}
void AudioMixer::track_t::resetResampler()
{
if (resampler != 0) {
if (resampler != NULL) {
resampler->reset();
}
}
@@ -436,11 +436,11 @@ void AudioMixer::process__validate(state_t* state)
} else {
if (state->outputTemp) {
delete [] state->outputTemp;
state->outputTemp = 0;
state->outputTemp = NULL;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
state->resampleTemp = 0;
state->resampleTemp = NULL;
}
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {