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Commit 2eaa1ee2 authored by Andy Hung's avatar Andy Hung
Browse files

AudioResampler: Add downsampling filter response test 

Test: resampler_tests
Bug: 123920996
Change-Id: I835ec033bd1fc6fd6905e32c660c779fafb853cb
parent 3f692411

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,

    };