Merge "AudioResampler: Improve resampling for voice quality devices" (34e548f9) · Commits · e / os / android_frameworks_av

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