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Commit 06b40f98 authored by Andy Hung's avatar Andy Hung
Browse files

AudioResampler: Improve resampling for voice quality devices 

Enabled for devices with sampling rates 16000 or less.

Test: instrumented log, resampler_tests
Bug: 123920996
Change-Id: Icb2a410b09eccb97188100fd78217123444b2eff
parent 7eb54802

media/libaudioprocessing/AudioResamplerDyn.cpp

+49 −5
Original line number Diff line number Diff line
@@ -383,8 +383,16 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate) 
                    ? 0.5 : 0.5 * mSampleRate / mInSampleRate;

            fcr *= mPropertyCutoffPercent / 100.;

        } else {

            // Voice quality devices have lower sampling rates

            // (and may be a consequence of downstream AMR-WB / G.722 codecs).

            // For these devices, we ensure a wider resampler passband

            // at the expense of aliasing noise (stopband attenuation

            // and stopband frequency).

            //

            constexpr uint32_t kVoiceDeviceSampleRate = 16000;



            if (mFilterQuality == DYN_HIGH_QUALITY) {

                // 32b coefficients, 64 length

                // float or 32b coefficients

                useS32 = true;

                stopBandAtten = 98.;

                if (inSampleRate >= mSampleRate * 4) {
@@ -394,8 +402,18 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate) 
                } else {

                    halfLength = 32;

                }



                if (mSampleRate <= kVoiceDeviceSampleRate) {

                    if (inSampleRate >= mSampleRate * 2) {

                        halfLength += 16;

                    } else {

                        halfLength += 8;

                    }

                    stopBandAtten = 84.;

                    tbwCheat = 1.05;

                }

            } else if (mFilterQuality == DYN_LOW_QUALITY) {

                // 16b coefficients, 16-32 length

                // float or 16b coefficients

                useS32 = false;

                stopBandAtten = 80.;

                if (inSampleRate >= mSampleRate * 4) {
@@ -405,13 +423,18 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate) 
                } else {

                    halfLength = 8;

                }

                if (inSampleRate <= mSampleRate) {

                if (mSampleRate <= kVoiceDeviceSampleRate) {

                    if (inSampleRate >= mSampleRate * 2) {

                        halfLength += 8;

                    }

                    tbwCheat = 1.05;

                } else if (inSampleRate <= mSampleRate) {

                    tbwCheat = 1.05;

                } else {

                    tbwCheat = 1.03;

                }

            } else { // DYN_MED_QUALITY

                // 16b coefficients, 32-64 length

                // float or 16b coefficients

                // note: > 64 length filters with 16b coefs can have quantization noise problems

                useS32 = false;

                stopBandAtten = 84.;
@@ -422,7 +445,15 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate) 
                } else {

                    halfLength = 16;

                }

                if (inSampleRate <= mSampleRate) {



                if (mSampleRate <= kVoiceDeviceSampleRate) {

                    if (inSampleRate >= mSampleRate * 2) {

                        halfLength += 16;

                    } else {

                        halfLength += 8;

                    }

                    tbwCheat = 1.05;

                } else if (inSampleRate <= mSampleRate) {

                    tbwCheat = 1.03;

                } else {

                    tbwCheat = 1.01;
@@ -430,6 +461,19 @@ void AudioResamplerDyn<TC, TI, TO>::setSampleRate(int32_t inSampleRate) 
            }

        }



        if (fcr > 0.) {

            ALOGV("%s: mFilterQuality:%d inSampleRate:%d mSampleRate:%d halfLength:%d "

                    "stopBandAtten:%lf fcr:%lf",

                    __func__, mFilterQuality, inSampleRate, mSampleRate, halfLength,

                    stopBandAtten, fcr);

        } else {

            ALOGV("%s: mFilterQuality:%d inSampleRate:%d mSampleRate:%d halfLength:%d "

                    "stopBandAtten:%lf tbwCheat:%lf",

                    __func__, mFilterQuality, inSampleRate, mSampleRate, halfLength,

                    stopBandAtten, tbwCheat);

        }





        // determine the number of polyphases in the filterbank.

        // for 16b, it is desirable to have 2^(16/2) = 256 phases.

        // https://ccrma.stanford.edu/~jos/resample/Relation_Interpolation_Error_Quantization.html