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Diffstat (limited to 'services/audioflinger/AudioResamplerFirProcess.h')
| -rw-r--r-- | services/audioflinger/AudioResamplerFirProcess.h | 333 |
1 files changed, 333 insertions, 0 deletions
diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h new file mode 100644 index 0000000..76d2d66 --- /dev/null +++ b/services/audioflinger/AudioResamplerFirProcess.h @@ -0,0 +1,333 @@ +/* + * Copyright (C) 2013 The Android Open Source Project + * + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + */ + +#ifndef ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H +#define ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H + +namespace android { + +// depends on AudioResamplerFirOps.h + +/* variant for input type TI = int16_t input samples */ +template<typename TC> +static inline +void mac(int32_t& l, int32_t& r, TC coef, const int16_t* samples) +{ + uint32_t rl = *reinterpret_cast<const uint32_t*>(samples); + l = mulAddRL(1, rl, coef, l); + r = mulAddRL(0, rl, coef, r); +} + +template<typename TC> +static inline +void mac(int32_t& l, TC coef, const int16_t* samples) +{ + l = mulAdd(samples[0], coef, l); +} + +/* variant for input type TI = float input samples */ +template<typename TC> +static inline +void mac(float& l, float& r, TC coef, const float* samples) +{ + l += *samples++ * coef; + r += *samples++ * coef; +} + +template<typename TC> +static inline +void mac(float& l, TC coef, const float* samples) +{ + l += *samples++ * coef; +} + +/* variant for output type TO = int32_t output samples */ +static inline +int32_t volumeAdjust(int32_t value, int32_t volume) +{ + return 2 * mulRL(0, value, volume); // Note: only use top 16b +} + +/* variant for output type TO = float output samples */ +static inline +float volumeAdjust(float value, float volume) +{ + return value * volume; +} + +/* + * Calculates a single output frame (two samples). + * + * This function computes both the positive half FIR dot product and + * the negative half FIR dot product, accumulates, and then applies the volume. + * + * This is a locked phase filter (it does not compute the interpolation). + * + * Use fir() to compute the proper coefficient pointers for a polyphase + * filter bank. + */ + +template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO> +static inline +void ProcessL(TO* const out, + int count, + const TC* coefsP, + const TC* coefsN, + const TI* sP, + const TI* sN, + const TO* const volumeLR) +{ + COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS >= 1 && CHANNELS <= 2) + if (CHANNELS == 2) { + TO l = 0; + TO r = 0; + do { + mac(l, r, *coefsP++, sP); + sP -= CHANNELS; + mac(l, r, *coefsN++, sN); + sN += CHANNELS; + } while (--count > 0); + out[0] += volumeAdjust(l, volumeLR[0]); + out[1] += volumeAdjust(r, volumeLR[1]); + } else { /* CHANNELS == 1 */ + TO l = 0; + do { + mac(l, *coefsP++, sP); + sP -= CHANNELS; + mac(l, *coefsN++, sN); + sN += CHANNELS; + } while (--count > 0); + out[0] += volumeAdjust(l, volumeLR[0]); + out[1] += volumeAdjust(l, volumeLR[1]); + } +} + +/* + * Calculates a single output frame (two samples) interpolating phase. + * + * This function computes both the positive half FIR dot product and + * the negative half FIR dot product, accumulates, and then applies the volume. + * + * This is an interpolated phase filter. + * + * Use fir() to compute the proper coefficient pointers for a polyphase + * filter bank. + */ + +template<typename TC, typename T> +void adjustLerp(T& lerpP __unused) +{ +} + +template<int32_t, typename T> +void adjustLerp(T& lerpP) +{ + lerpP >>= 16; // lerpP is 32bit for NEON int32_t, but always 16 bit for non-NEON path +} + +template<typename TC, typename TINTERP> +static inline +TC interpolate(TC coef_0, TC coef_1, TINTERP lerp) +{ + return lerp * (coef_1 - coef_0) + coef_0; +} + +template<int16_t, uint32_t> +static inline +int16_t interpolate(int16_t coef_0, int16_t coef_1, uint32_t lerp) +{ + return (static_cast<int16_t>(lerp) * ((coef_1-coef_0)<<1)>>16) + coef_0; +} + +template<int32_t, uint32_t> +static inline +int32_t interpolate(int32_t coef_0, int32_t coef_1, uint32_t lerp) +{ + return mulAdd(static_cast<int16_t>(lerp), (coef_1-coef_0)<<1, coef_0); +} + +template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO, typename TINTERP> +static inline +void Process(TO* const out, + int count, + const TC* coefsP, + const TC* coefsN, + const TC* coefsP1 __unused, + const TC* coefsN1 __unused, + const TI* sP, + const TI* sN, + TINTERP lerpP, + const TO* const volumeLR) +{ + COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS >= 1 && CHANNELS <= 2) + adjustLerp<TC, TINTERP>(lerpP); // coefficient type adjustment for interpolation + + if (CHANNELS == 2) { + TO l = 0; + TO r = 0; + for (size_t i = 0; i < count; ++i) { + mac(l, r, interpolate(coefsP[0], coefsP[count], lerpP), sP); + coefsP++; + sP -= CHANNELS; + mac(l, r, interpolate(coefsN[count], coefsN[0], lerpP), sN); + coefsN++; + sN += CHANNELS; + } + out[0] += volumeAdjust(l, volumeLR[0]); + out[1] += volumeAdjust(r, volumeLR[1]); + } else { /* CHANNELS == 1 */ + TO l = 0; + for (size_t i = 0; i < count; ++i) { + mac(l, interpolate(coefsP[0], coefsP[count], lerpP), sP); + coefsP++; + sP -= CHANNELS; + mac(l, interpolate(coefsN[count], coefsN[0], lerpP), sN); + coefsN++; + sN += CHANNELS; + } + out[0] += volumeAdjust(l, volumeLR[0]); + out[1] += volumeAdjust(l, volumeLR[1]); + } +} + +/* + * Calculates a single output frame (two samples) from input sample pointer. + * + * This sets up the params for the accelerated Process() and ProcessL() + * functions to do the appropriate dot products. + * + * @param out should point to the output buffer with space for at least one output frame. + * + * @param phase is the fractional distance between input frames for interpolation: + * phase >= 0 && phase < phaseWrapLimit. It can be thought of as a rational fraction + * of phase/phaseWrapLimit. + * + * @param phaseWrapLimit is #polyphases<<coefShift, where #polyphases is the number of polyphases + * in the polyphase filter. Likewise, #polyphases can be obtained as (phaseWrapLimit>>coefShift). + * + * @param coefShift gives the bit alignment of the polyphase index in the phase parameter. + * + * @param halfNumCoefs is the half the number of coefficients per polyphase filter. Since the + * overall filterbank is odd-length symmetric, only halfNumCoefs need be stored. + * + * @param coefs is the polyphase filter bank, starting at from polyphase index 0, and ranging to + * and including the #polyphases. Each polyphase of the filter has half-length halfNumCoefs + * (due to symmetry). The total size of the filter bank in coefficients is + * (#polyphases+1)*halfNumCoefs. + * + * The filter bank coefs should be aligned to a minimum of 16 bytes (preferrably to cache line). + * + * The coefs should be attenuated (to compensate for passband ripple) + * if storing back into the native format. + * + * @param samples are unaligned input samples. The position is in the "middle" of the + * sample array with respect to the FIR filter: + * the negative half of the filter is dot product from samples+1 to samples+halfNumCoefs; + * the positive half of the filter is dot product from samples to samples-halfNumCoefs+1. + * + * @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel, + * expressed as a S32 integer. A negative value inverts the channel 180 degrees. + * The pointer volumeLR should be aligned to a minimum of 8 bytes. + * A typical value for volume is 0x1000 to align to a unity gain output of 20.12. + * + * In between calls to filterCoefficient, the phase is incremented by phaseIncrement, where + * phaseIncrement is calculated as inputSampling * phaseWrapLimit / outputSampling. + * + * The filter polyphase index is given by indexP = phase >> coefShift. Due to + * odd length symmetric filter, the polyphase index of the negative half depends on + * whether interpolation is used. + * + * The fractional siting between the polyphase indices is given by the bits below coefShift: + * + * lerpP = phase << 32 - coefShift >> 1; // for 32 bit unsigned phase multiply + * lerpP = phase << 32 - coefShift >> 17; // for 16 bit unsigned phase multiply + * + * For integer types, this is expressed as: + * + * lerpP = phase << sizeof(phase)*8 - coefShift + * >> (sizeof(phase)-sizeof(*coefs))*8 + 1; + * + * For floating point, lerpP is the fractional phase scaled to [0.0, 1.0): + * + * lerpP = (phase << 32 - coefShift) / (1 << 32); // floating point equivalent + */ + +template<int CHANNELS, bool LOCKED, int STRIDE, typename TC, typename TI, typename TO> +static inline +void fir(TO* const out, + const uint32_t phase, const uint32_t phaseWrapLimit, + const int coefShift, const int halfNumCoefs, const TC* const coefs, + const TI* const samples, const TO* const volumeLR) +{ + // NOTE: be very careful when modifying the code here. register + // pressure is very high and a small change might cause the compiler + // to generate far less efficient code. + // Always sanity check the result with objdump or test-resample. + + if (LOCKED) { + // locked polyphase (no interpolation) + // Compute the polyphase filter index on the positive and negative side. + uint32_t indexP = phase >> coefShift; + uint32_t indexN = (phaseWrapLimit - phase) >> coefShift; + const TC* coefsP = coefs + indexP*halfNumCoefs; + const TC* coefsN = coefs + indexN*halfNumCoefs; + const TI* sP = samples; + const TI* sN = samples + CHANNELS; + + // dot product filter. + ProcessL<CHANNELS, STRIDE>(out, + halfNumCoefs, coefsP, coefsN, sP, sN, volumeLR); + } else { + // interpolated polyphase + // Compute the polyphase filter index on the positive and negative side. + uint32_t indexP = phase >> coefShift; + uint32_t indexN = (phaseWrapLimit - phase - 1) >> coefShift; // one's complement. + const TC* coefsP = coefs + indexP*halfNumCoefs; + const TC* coefsN = coefs + indexN*halfNumCoefs; + const TC* coefsP1 = coefsP + halfNumCoefs; + const TC* coefsN1 = coefsN + halfNumCoefs; + const TI* sP = samples; + const TI* sN = samples + CHANNELS; + + // Interpolation fraction lerpP derived by shifting all the way up and down + // to clear the appropriate bits and align to the appropriate level + // for the integer multiply. The constants should resolve in compile time. + // + // The interpolated filter coefficient is derived as follows for the pos/neg half: + // + // interpolated[P] = index[P]*lerpP + index[P+1]*(1-lerpP) + // interpolated[N] = index[N+1]*lerpP + index[N]*(1-lerpP) + + // on-the-fly interpolated dot product filter + if (is_same<TC, float>::value || is_same<TC, double>::value) { + static const TC scale = 1. / (65536. * 65536.); // scale phase bits to [0.0, 1.0) + TC lerpP = TC(phase << (sizeof(phase)*8 - coefShift)) * scale; + + Process<CHANNELS, STRIDE>(out, + halfNumCoefs, coefsP, coefsN, coefsP1, coefsN1, sP, sN, lerpP, volumeLR); + } else { + uint32_t lerpP = phase << (sizeof(phase)*8 - coefShift) + >> ((sizeof(phase)-sizeof(*coefs))*8 + 1); + + Process<CHANNELS, STRIDE>(out, + halfNumCoefs, coefsP, coefsN, coefsP1, coefsN1, sP, sN, lerpP, volumeLR); + } + } +} + +}; // namespace android + +#endif /*ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H*/ |