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Commit 7eb54802 authored by Andy Hung's avatar Andy Hung Committed by Android (Google) Code Review
Browse files

Merge changes I835ec033,I11aa8f88 

* changes:
  AudioResampler: Add downsampling filter response test
  AudioResampler: Minor test code fixes
parents 1dd3c386 2eaa1ee2

media/libaudioprocessing/AudioResamplerDyn.cpp

+3 −3
Original line number Diff line number Diff line
@@ -243,7 +243,7 @@ void AudioResamplerDyn<TC, TI, TO>::createKaiserFir(Constants &c, 
{

    // compute the normalized transition bandwidth

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

    const double halfbw = tbw / 2.;

    const double halfbw = tbw * 0.5;



    double fcr; // compute fcr, the 3 dB amplitude cut-off.

    if (inSampleRate < outSampleRate) { // upsample
@@ -290,7 +290,7 @@ void AudioResamplerDyn<TC, TI, TO>::createKaiserFir(Constants &c, 


#if 0

    // Keep this debug code in case an app causes resampler design issues.

    const double halfbw = tbw / 2.;

    const double halfbw = tbw * 0.5;

    // print basic filter stats

    ALOGD("L:%d  hnc:%d  stopBandAtten:%lf  fcr:%lf  atten:%lf  tbw:%lf\n",

            c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, attenuation, tbw);
@@ -305,7 +305,7 @@ void AudioResamplerDyn<TC, TI, TO>::createKaiserFir(Constants &c, 


    const int32_t passSteps = 1000;



    testFir(coefs, c.mL, c.mHalfNumCoefs, fp, fs, passSteps, passSteps * c.ML /*stopSteps*/,

    testFir(coefs, c.mL, c.mHalfNumCoefs, fp, fs, passSteps, passSteps * c.mL /*stopSteps*/,

            passMin, passMax, passRipple, stopMax, stopRipple);

    ALOGD("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);

    ALOGD("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);

media/libaudioprocessing/AudioResamplerFirGen.h

+2 −2
Original line number Diff line number Diff line
@@ -547,8 +547,8 @@ static void testFir(const T* coef, int L, int halfNumCoef, 
        wstart += wstep;

    }

    // renormalize - this is needed for integer filter types, use 1 for float or double.

    constexpr int64_t integralShift = std::is_integral<T>::value ? (sizeof(T) * 8 - 1) : 0;

    const double norm = 1. / (L << integralShift);

    constexpr int integralShift = std::is_integral<T>::value ? (sizeof(T) * CHAR_BIT - 1) : 0;

    const double norm = 1. / (int64_t{L} << integralShift);



    firMin = fmin * norm;

    firMax = fmax * norm;

media/libaudioprocessing/tests/Android.bp

+1 −1
Original line number Diff line number Diff line
@@ -5,8 +5,8 @@ cc_defaults { 


    header_libs: ["libbase_headers"],

    shared_libs: [

        "libaudioutils",

        "libaudioprocessing",

        "libaudioutils",

        "libcutils",

        "liblog",

        "libutils",

media/libaudioprocessing/tests/resampler_tests.cpp

+44 −11
Original line number Diff line number Diff line
@@ -246,7 +246,8 @@ void testStopbandDownconversion(size_t channels, 
}



void testFilterResponse(

        size_t channels, unsigned inputFreq, unsigned outputFreq)

        size_t channels, unsigned inputFreq, unsigned outputFreq,

        android::AudioResampler::src_quality quality = android::AudioResampler::DYN_HIGH_QUALITY)

{

    // create resampler

    using ResamplerType = android::AudioResamplerDyn<float, float, float>;
@@ -256,7 +257,7 @@ void testFilterResponse( 
                            AUDIO_FORMAT_PCM_FLOAT,

                            channels,

                            outputFreq,

                            android::AudioResampler::DYN_HIGH_QUALITY)));

                            quality)));

    rdyn->setSampleRate(inputFreq);



    // get design parameters
@@ -268,17 +269,20 @@ void testFilterResponse( 
    const double attenuation = rdyn->getFilterAttenuation();

    const double stopbandDb = rdyn->getStopbandAttenuationDb();

    const double passbandDb = rdyn->getPassbandRippleDb();

    const double fp = fcr - tbw / 2;

    const double fs = fcr + tbw / 2;

    const double fp = fcr - tbw * 0.5;

    const double fs = fcr + tbw * 0.5;

    const double idealfs = inputFreq <= outputFreq

        ? 0.5                            // upsample

        : 0.5 * outputFreq  / inputFreq; // downsample



    printf("inputFreq:%d outputFreq:%d design"

    printf("inputFreq:%d outputFreq:%d design quality %d"

            " phases:%d halfLength:%d"

            " fcr:%lf fp:%lf fs:%lf tbw:%lf"

            " fcr:%lf fp:%lf fs:%lf tbw:%lf fcrp:%lf"

            " attenuation:%lf stopRipple:%.lf passRipple:%lf"

            "\n",

            inputFreq, outputFreq,

            inputFreq, outputFreq, quality,

            phases, halfLength,

            fcr, fp, fs, tbw,

            fcr, fp, fs, tbw, fcr * 100. / idealfs,

            attenuation, stopbandDb, passbandDb);



    // verify design parameters
@@ -541,8 +545,36 @@ TEST(audioflinger_resampler, stopbandresponse_float_multichannel) { 
    }

}



TEST(audioflinger_resampler, filterresponse) {

    std::vector<int> inSampleRates{

// Selected downsampling responses for various frequencies relating to hearing aid.

TEST(audioflinger_resampler, downsamplingresponse) {

    static constexpr android::AudioResampler::src_quality qualities[] = {

        android::AudioResampler::DYN_LOW_QUALITY,

        android::AudioResampler::DYN_MED_QUALITY,

        android::AudioResampler::DYN_HIGH_QUALITY,

    };

    static constexpr int inSampleRates[] = {

        32000,

        44100,

        48000,

    };

    static constexpr int outSampleRates[] = {

        16000,

        24000,

    };



    for (auto quality : qualities) {

        for (int outSampleRate : outSampleRates) {

            for (int inSampleRate : inSampleRates) {

                testFilterResponse(2 /* channels */, inSampleRate, outSampleRate, quality);

            }

        }

    }

}



// General responses for typical output device scenarios - 44.1, 48, 96 kHz

// (48, 96 are part of the same resampler generation family).

TEST(audioflinger_resampler, generalresponse) {

    static constexpr int inSampleRates[] = {

        8000,

        11025,

        12000,
@@ -557,7 +589,8 @@ TEST(audioflinger_resampler, filterresponse) { 
        176400,

        192000,

    };

    std::vector<int> outSampleRates{

    static constexpr int outSampleRates[] = {

        44100,

        48000,

        96000,

    };