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Commit a4daf0b4 authored by Glenn Kasten's avatar Glenn Kasten
Browse files

Fix build warnings 

Change-Id: Ife5e40264f43fb3ccb40830228701003fe106bf5
parent 5fedaffd

services/audioflinger/AudioMixerOps.h

+1 −1
Original line number Diff line number Diff line
@@ -184,7 +184,7 @@ inline int16_t MixMul<int16_t, float, float>(float value, float volume) { 
template <typename TO, typename TI>

inline void MixAccum(TO *auxaccum, TI value) {

    if (!is_same<TO, TI>::value) {

        LOG_ALWAYS_FATAL("MixAccum type not properly specialized: %d %d\n",

        LOG_ALWAYS_FATAL("MixAccum type not properly specialized: %zu %zu\n",

                sizeof(TO), sizeof(TI));

    }

    *auxaccum += value;

services/audioflinger/AudioResamplerDyn.cpp

+4 −4
Original line number Diff line number Diff line
@@ -82,17 +82,17 @@ template<typename TC, typename TI, typename TO> 
void AudioResamplerDyn<TC, TI, TO>::InBuffer::resize(int CHANNELS, int halfNumCoefs)

{

    // calculate desired state size

    int stateCount = halfNumCoefs * CHANNELS * 2 * kStateSizeMultipleOfFilterLength;

    size_t stateCount = halfNumCoefs * CHANNELS * 2 * kStateSizeMultipleOfFilterLength;



    // check if buffer needs resizing

    if (mState

            && stateCount == mStateCount

            && mRingFull-mState == mStateCount-halfNumCoefs*CHANNELS) {

            && mRingFull-mState == (ssize_t) (mStateCount-halfNumCoefs*CHANNELS)) {

        return;

    }



    // create new buffer

    TI* state;

    TI* state = NULL;

    (void)posix_memalign(reinterpret_cast<void**>(&state), 32, stateCount*sizeof(*state));

    memset(state, 0, stateCount*sizeof(*state));
@@ -213,7 +213,7 @@ template<typename TC, typename TI, typename TO> 
void AudioResamplerDyn<TC, TI, TO>::createKaiserFir(Constants &c,

        double stopBandAtten, int inSampleRate, int outSampleRate, double tbwCheat)

{

    TC* buf;

    TC* buf = NULL;

    static const double atten = 0.9998;   // to avoid ripple overflow

    double fcr;

    double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);

services/audioflinger/AudioResamplerFirProcess.h

+1 −1
Original line number Diff line number Diff line
@@ -177,7 +177,7 @@ struct InterpNull { 
template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO, typename TINTERP>

static inline

void ProcessBase(TO* const out,

        int count,

        size_t count,

        const TC* coefsP,

        const TC* coefsN,

        const TI* sP,

services/audioflinger/tests/resampler_tests.cpp

+1 −1
Original line number Diff line number Diff line
@@ -60,7 +60,7 @@ void buffercmp(const void *reference, const void *test, 
        int check = memcmp((const char*)reference + i * outputFrameSize,

                (const char*)test + i * outputFrameSize, outputFrameSize);

        if (check) {

            ALOGE("Failure at frame %d", i);

            ALOGE("Failure at frame %zu", i);

            ASSERT_EQ(check, 0); /* fails */

        }

    }

services/audioflinger/tests/test-mixer.cpp

+1 −1
Original line number Diff line number Diff line
@@ -62,7 +62,7 @@ static int writeFile(const char *filename, const void *buffer, 
    info.samplerate = sampleRate;

    info.channels = channels;

    info.format = SF_FORMAT_WAV | (isBufferFloat ? SF_FORMAT_FLOAT : SF_FORMAT_PCM_16);

    printf("saving file:%s  channels:%d  samplerate:%d  frames:%d\n",

    printf("saving file:%s  channels:%u  samplerate:%u  frames:%zu\n",

            filename, info.channels, info.samplerate, frames);

    SNDFILE *sf = sf_open(filename, SFM_WRITE, &info);

    if (sf == NULL) {