Audio resampler update to add S16 filters
This does not affect the existing resamplers.
New resampler accessed through additional quality settings:
DYN_LOW_QUALITY = 5
DYN_MED_QUALITY = 6
DYN_HIGH_QUALITY = 7
Change-Id: Iebbd31871e808a4a6dee3f3abfd7e9dcf77c48e1
Signed-off-by: Andy Hung <hunga@google.com>
diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h
new file mode 100644
index 0000000..38e387c
--- /dev/null
+++ b/services/audioflinger/AudioResamplerFirProcess.h
@@ -0,0 +1,256 @@
+/*
+ * 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
+
+template<int CHANNELS, typename TC>
+static inline
+void mac(
+ int32_t& l, int32_t& r,
+ const TC coef,
+ const int16_t* samples)
+{
+ if (CHANNELS == 2) {
+ uint32_t rl = *reinterpret_cast<const uint32_t*>(samples);
+ l = mulAddRL(1, rl, coef, l);
+ r = mulAddRL(0, rl, coef, r);
+ } else {
+ r = l = mulAdd(samples[0], coef, l);
+ }
+}
+
+template<int CHANNELS, typename TC>
+static inline
+void interpolate(
+ int32_t& l, int32_t& r,
+ const TC coef_0, const TC coef_1,
+ const int16_t lerp, const int16_t* samples)
+{
+ TC sinc;
+
+ if (is_same<TC, int16_t>::value) {
+ sinc = (lerp * ((coef_1-coef_0)<<1)>>16) + coef_0;
+ } else {
+ sinc = mulAdd(lerp, (coef_1-coef_0)<<1, coef_0);
+ }
+ if (CHANNELS == 2) {
+ uint32_t rl = *reinterpret_cast<const uint32_t*>(samples);
+ l = mulAddRL(1, rl, sinc, l);
+ r = mulAddRL(0, rl, sinc, r);
+ } else {
+ r = l = mulAdd(samples[0], sinc, l);
+ }
+}
+
+/*
+ * Calculates a single output sample (two stereo frames).
+ *
+ * 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>
+static inline
+void ProcessL(int32_t* const out,
+ int count,
+ const TC* coefsP,
+ const TC* coefsN,
+ const int16_t* sP,
+ const int16_t* sN,
+ const int32_t* const volumeLR)
+{
+ int32_t l = 0;
+ int32_t r = 0;
+ do {
+ mac<CHANNELS>(l, r, *coefsP++, sP);
+ sP -= CHANNELS;
+ mac<CHANNELS>(l, r, *coefsN++, sN);
+ sN += CHANNELS;
+ } while (--count > 0);
+ out[0] += 2 * mulRL(0, l, volumeLR[0]); // Note: only use top 16b
+ out[1] += 2 * mulRL(0, r, volumeLR[1]); // Note: only use top 16b
+}
+
+/*
+ * Calculates a single output sample (two stereo frames) 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 <int CHANNELS, int STRIDE, typename TC>
+static inline
+void Process(int32_t* const out,
+ int count,
+ const TC* coefsP,
+ const TC* coefsN,
+ const TC* coefsP1,
+ const TC* coefsN1,
+ const int16_t* sP,
+ const int16_t* sN,
+ uint32_t lerpP,
+ const int32_t* const volumeLR)
+{
+ (void) coefsP1; // suppress unused parameter warning
+ (void) coefsN1;
+ if (sizeof(*coefsP)==4) {
+ lerpP >>= 16; // ensure lerpP is 16b
+ }
+ int32_t l = 0;
+ int32_t r = 0;
+ for (size_t i = 0; i < count; ++i) {
+ interpolate<CHANNELS>(l, r, coefsP[0], coefsP[count], lerpP, sP);
+ coefsP++;
+ sP -= CHANNELS;
+ interpolate<CHANNELS>(l, r, coefsN[count], coefsN[0], lerpP, sN);
+ coefsN++;
+ sN += CHANNELS;
+ }
+ out[0] += 2 * mulRL(0, l, volumeLR[0]); // Note: only use top 16b
+ out[1] += 2 * mulRL(0, r, volumeLR[1]); // Note: only use top 16b
+}
+
+/*
+ * Calculates a single output sample (two stereo frames) 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 at least enough space for 2 output frames.
+ *
+ * @param phase is the fractional distance between input samples 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;
+ *
+ */
+
+template<int CHANNELS, bool LOCKED, int STRIDE, typename TC>
+static inline
+void fir(int32_t* const out,
+ const uint32_t phase, const uint32_t phaseWrapLimit,
+ const int coefShift, const int halfNumCoefs, const TC* const coefs,
+ const int16_t* const samples, const int32_t* 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 int16_t* sP = samples;
+ const int16_t* 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 int16_t* sP = samples;
+ const int16_t* 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)
+ uint32_t lerpP = phase << (sizeof(phase)*8 - coefShift)
+ >> ((sizeof(phase)-sizeof(*coefs))*8 + 1);
+
+ // on-the-fly interpolated dot product filter
+ Process<CHANNELS, STRIDE>(out,
+ halfNumCoefs, coefsP, coefsN, coefsP1, coefsN1, sP, sN, lerpP, volumeLR);
+ }
+}
+
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_FIR_PROCESS_H*/