/*
* 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;
}
/*
* Helper template functions for loop unrolling accumulator operations.
*
* Unrolling the loops achieves about 2x gain.
* Using a recursive template rather than an array of TO[] for the accumulator
* values is an additional 10-20% gain.
*/
template<int CHANNELS, typename TO>
class Accumulator : public Accumulator<CHANNELS-1, TO> // recursive
{
public:
inline void clear() {
value = 0;
Accumulator<CHANNELS-1, TO>::clear();
}
template<typename TC, typename TI>
inline void acc(TC coef, const TI*& data) {
mac(value, coef, data++);
Accumulator<CHANNELS-1, TO>::acc(coef, data);
}
inline void volume(TO*& out, TO gain) {
*out++ = volumeAdjust(value, gain);
Accumulator<CHANNELS-1, TO>::volume(out, gain);
}
TO value; // one per recursive inherited base class
};
template<typename TO>
class Accumulator<0, TO> {
public:
inline void clear() {
}
template<typename TC, typename TI>
inline void acc(TC coef __unused, const TI*& data __unused) {
}
inline void volume(TO*& out __unused, TO gain __unused) {
}
};
template<typename TC, typename TINTERP>
inline
TC interpolate(TC coef_0, TC coef_1, TINTERP lerp)
{
return lerp * (coef_1 - coef_0) + coef_0;
}
template<>
inline
int16_t interpolate<int16_t, uint32_t>(int16_t coef_0, int16_t coef_1, uint32_t lerp)
{ // in some CPU architectures 16b x 16b multiplies are faster.
return (static_cast<int16_t>(lerp) * static_cast<int16_t>(coef_1 - coef_0) >> 15) + coef_0;
}
template<>
inline
int32_t interpolate<int32_t, uint32_t>(int32_t coef_0, int32_t coef_1, uint32_t lerp)
{
return (lerp * static_cast<int64_t>(coef_1 - coef_0) >> 31) + coef_0;
}
/* class scope for passing in functions into templates */
struct InterpCompute {
template<typename TC, typename TINTERP>
static inline
TC interpolatep(TC coef_0, TC coef_1, TINTERP lerp) {
return interpolate(coef_0, coef_1, lerp);
}
template<typename TC, typename TINTERP>
static inline
TC interpolaten(TC coef_0, TC coef_1, TINTERP lerp) {
return interpolate(coef_0, coef_1, lerp);
}
};
struct InterpNull {
template<typename TC, typename TINTERP>
static inline
TC interpolatep(TC coef_0, TC coef_1 __unused, TINTERP lerp __unused) {
return coef_0;
}
template<typename TC, typename TINTERP>
static inline
TC interpolaten(TC coef_0 __unused, TC coef_1, TINTERP lerp __unused) {
return coef_1;
}
};
/*
* Calculates a single output frame (two samples).
*
* The Process*() functions compute both the positive half FIR dot product and
* the negative half FIR dot product, accumulates, and then applies the volume.
*
* Use fir() to compute the proper coefficient pointers for a polyphase
* filter bank.
*
* ProcessBase() is the fundamental processing template function.
*
* ProcessL() calls ProcessBase() with TFUNC = InterpNull, for fixed/locked phase.
* Process() calls ProcessBase() with TFUNC = InterpCompute, for interpolated phase.
*/
template <int CHANNELS, int STRIDE, typename TFUNC, typename TC, typename TI, typename TO,
typename TINTERP>
static inline
void ProcessBase(TO* const out,
size_t count,
const TC* coefsP,
const TC* coefsN,
const TI* sP,
const TI* sN,
TINTERP lerpP,
const TO* const volumeLR)
{
COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS > 0)
if (CHANNELS > 2) {
// TO accum[CHANNELS];
Accumulator<CHANNELS, TO> accum;
// for (int j = 0; j < CHANNELS; ++j) accum[j] = 0;
accum.clear();
for (size_t i = 0; i < count; ++i) {
TC c = TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP);
// for (int j = 0; j < CHANNELS; ++j) mac(accum[j], c, sP + j);
const TI *tmp_data = sP; // tmp_ptr seems to work better
accum.acc(c, tmp_data);
coefsP++;
sP -= CHANNELS;
c = TFUNC::interpolaten(coefsN[count], coefsN[0], lerpP);
// for (int j = 0; j < CHANNELS; ++j) mac(accum[j], c, sN + j);
tmp_data = sN; // tmp_ptr seems faster than directly using sN
accum.acc(c, tmp_data);
coefsN++;
sN += CHANNELS;
}
// for (int j = 0; j < CHANNELS; ++j) out[j] += volumeAdjust(accum[j], volumeLR[0]);
TO *tmp_out = out; // may remove if const out definition changes.
accum.volume(tmp_out, volumeLR[0]);
} else if (CHANNELS == 2) {
TO l = 0;
TO r = 0;
for (size_t i = 0; i < count; ++i) {
mac(l, r, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP), sP);
coefsP++;
sP -= CHANNELS;
mac(l, r, TFUNC::interpolaten(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, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP), sP);
coefsP++;
sP -= CHANNELS;
mac(l, TFUNC::interpolaten(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 from a polyphase resampling filter.
* See Process() for parameter details.
*/
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)
{
ProcessBase<CHANNELS, STRIDE, InterpNull>(out, count, coefsP, coefsN, sP, sN, 0, volumeLR);
}
/*
* Calculates a single output frame from a polyphase resampling filter,
* with filter phase interpolation.
*
* @param out should point to the output buffer with space for at least one output frame.
*
* @param count should be half the size of the total filter length (halfNumCoefs), as we
* use symmetry in filter coefficients to evaluate two dot products.
*
* @param coefsP is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
* to the positive sP.
*
* @param coefsN is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
* to the negative sN.
*
* @param coefsP1 is the next phase of coefsP (used for interpolation).
*
* @param coefsN1 is the next phase of coefsN (used for interpolation).
*
* @param sP is the positive half of the coefficients (as viewed by a convolution),
* starting at the original samples pointer and decrementing (by CHANNELS).
*
* @param sN is the negative half of the samples (as viewed by a convolution),
* starting at the original samples pointer + CHANNELS and incrementing (by CHANNELS).
*
* @param lerpP The fractional siting between the polyphase indices is given by the bits
* below coefShift. See fir() for details.
*
* @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel,
* expressed as a S32 integer or float. 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.
*/
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)
{
ProcessBase<CHANNELS, STRIDE, InterpCompute>(out, count, coefsP, coefsN, sP, sN, lerpP,
volumeLR);
}
/*
* Calculates a single output frame 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 or float. 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*/