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Commit e0feee3d authored by Glenn Kasten's avatar Glenn Kasten
Browse files

Use NULL not 0 for pointers 

Change-Id: Iab3f9abbdab617dc5a599e657ec46a0b0a002eef
parent eebeceec

services/audioflinger/AudioFlinger.cpp

+15 −15
Original line number Diff line number Diff line
@@ -158,7 +158,7 @@ static const char *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)

{

}
@@ -1367,7 +1367,7 @@ 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);
@@ -1832,7 +1832,7 @@ 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);
@@ -2766,7 +2766,7 @@ 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;

                }
@@ -3264,7 +3264,7 @@ 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;
@@ -3457,14 +3457,14 @@ 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;
@@ -3705,14 +3705,14 @@ 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;

}
@@ -4217,7 +4217,7 @@ 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;
@@ -4232,7 +4232,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 
AudioFlinger::RecordThread::~RecordThread()

{

    delete[] mRsmpInBuffer;

    if (mResampler != 0) {

    if (mResampler != NULL) {

        delete mResampler;

        delete[] mRsmpOutBuffer;

    }
@@ -4326,7 +4326,7 @@ 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;
@@ -4584,7 +4584,7 @@ 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);
@@ -4619,7 +4619,7 @@ 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;

        }
@@ -4782,7 +4782,7 @@ 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);

services/audioflinger/AudioFlinger.h

+2 −2
Original line number Diff line number Diff line
@@ -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,

services/audioflinger/AudioMixer.cpp

+12 −12
Original line number Diff line number Diff line
@@ -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++) {
@@ -67,11 +67,11 @@ 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++;
@@ -127,7 +127,7 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate) 
        if (track.resampler) {

            // delete  the resampler

            delete track.resampler;

            track.resampler = 0;

            track.resampler = NULL;

            track.sampleRate = mSampleRate;

            invalidateState(1<<name);

        }
@@ -290,7 +290,7 @@ 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);

            }
@@ -302,12 +302,12 @@ 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();

    }

}
@@ -430,11 +430,11 @@ 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) {