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-rw-r--r--services/audioflinger/AudioResamplerSinc.cpp75
1 files changed, 42 insertions, 33 deletions
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 7d3681c..952abb4 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -470,6 +470,9 @@ AudioResamplerSinc::AudioResamplerSinc(int bitDepth,
*
*/
+ mVolumeSIMD[0] = 0;
+ mVolumeSIMD[1] = 0;
+
// Load the constants for coefficients
int ok = pthread_once(&once_control, init_routine);
if (ok != 0) {
@@ -494,6 +497,12 @@ void AudioResamplerSinc::init() {
mRingFull = mImpulse + (numCoefs+1)*mChannelCount;
}
+void AudioResamplerSinc::setVolume(int16_t left, int16_t right) {
+ AudioResampler::setVolume(left, right);
+ mVolumeSIMD[0] = int32_t(left)<<16;
+ mVolumeSIMD[1] = int32_t(right)<<16;
+}
+
void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
@@ -568,11 +577,9 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
}
// handle boundary case
- int32_t l, r;
while (CC_LIKELY(outputIndex < outputSampleCount)) {
- filterCoefficient<CHANNELS>(l, r, phaseFraction, impulse, vRL);
- out[outputIndex++] += l;
- out[outputIndex++] += r;
+ filterCoefficient<CHANNELS>(&out[outputIndex], phaseFraction, impulse, vRL);
+ outputIndex += 2;
phaseFraction += phaseIncrement;
const size_t phaseIndex = phaseFraction >> kNumPhaseBits;
@@ -628,7 +635,7 @@ void AudioResamplerSinc::read(
template<int CHANNELS>
void AudioResamplerSinc::filterCoefficient(
- int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples, uint32_t vRL)
+ int32_t* out, uint32_t phase, const int16_t *samples, uint32_t vRL)
{
// compute the index of the coefficient on the positive side and
// negative side
@@ -650,19 +657,19 @@ void AudioResamplerSinc::filterCoefficient(
int16_t const* sP = samples;
int16_t const* sN = samples + CHANNELS;
- l = 0;
- r = 0;
size_t count = offset;
if (!USE_NEON) {
+ int32_t l = 0;
+ int32_t r = 0;
for (size_t i=0 ; i<count ; i++) {
interpolate<CHANNELS>(l, r, coefsP++, offset, lerpP, sP);
sP -= CHANNELS;
interpolate<CHANNELS>(l, r, coefsN++, offset, lerpN, sN);
sN += CHANNELS;
}
- l = 2 * mulRL(1, l, vRL);
- r = 2 * mulRL(0, r, vRL);
+ out[0] += 2 * mulRL(1, l, vRL);
+ out[1] += 2 * mulRL(0, r, vRL);
} else if (CHANNELS == 1) {
int32_t const* coefsP1 = coefsP + offset;
int32_t const* coefsN1 = coefsN + offset;
@@ -674,11 +681,11 @@ void AudioResamplerSinc::filterCoefficient(
"1: \n"
"vld1.16 { d4}, [%[sP]] \n" // load 4 16-bits stereo samples
- "vld1.32 { q8}, [%[coefsP0]]! \n" // load 4 32-bits coefs
- "vld1.32 { q9}, [%[coefsP1]]! \n" // load 4 32-bits coefs for interpolation
+ "vld1.32 { q8}, [%[coefsP0]:128]! \n" // load 4 32-bits coefs
+ "vld1.32 { q9}, [%[coefsP1]:128]! \n" // load 4 32-bits coefs for interpolation
"vld1.16 { d6}, [%[sN]]! \n" // load 4 16-bits stereo samples
- "vld1.32 {q10}, [%[coefsN0]]! \n" // load 4 32-bits coefs
- "vld1.32 {q11}, [%[coefsN1]]! \n" // load 4 32-bits coefs for interpolation
+ "vld1.32 {q10}, [%[coefsN0]:128]! \n" // load 4 32-bits coefs
+ "vld1.32 {q11}, [%[coefsN1]:128]! \n" // load 4 32-bits coefs for interpolation
"vrev64.16 d4, d4 \n" // reverse 2 frames of the positive side
@@ -703,12 +710,16 @@ void AudioResamplerSinc::filterCoefficient(
"bne 1b \n" // loop
+ "vld1.s32 {d2}, [%[vLR]] \n" // load volumes
+ "vld1.s32 {d3}, %[out] \n" // load the output
"vpadd.s32 d0, d0, d1 \n" // add all 4 partial sums
"vpadd.s32 d0, d0, d0 \n" // together
+ "vdup.i32 d0, d0[0] \n" // interleave L,R channels
+ "vqrdmulh.s32 d0, d0, d2 \n" // apply volume
+ "vadd.s32 d3, d3, d0 \n" // accumulate result
+ "vst1.s32 {d0}, %[out] \n" // store result
- "vmov.s32 %[l], d0[0] \n" // save result in ARM register
-
- : [l] "=r" (l),
+ : [out] "=Uv" (out[0]),
[count] "+r" (count),
[coefsP0] "+r" (coefsP),
[coefsP1] "+r" (coefsP1),
@@ -718,14 +729,12 @@ void AudioResamplerSinc::filterCoefficient(
[sN] "+r" (sN)
: [lerpP] "r" (lerpP<<16),
[lerpN] "r" (lerpN<<16),
- [vRL] "r" (vRL)
+ [vLR] "r" (mVolumeSIMD)
: "cc", "memory",
"q0", "q1", "q2", "q3",
"q8", "q9", "q10", "q11",
"q12", "q14"
);
- l = 2 * mulRL(1, l, vRL);
- r = l;
} else if (CHANNELS == 2) {
int32_t const* coefsP1 = coefsP + offset;
int32_t const* coefsN1 = coefsN + offset;
@@ -738,11 +747,11 @@ void AudioResamplerSinc::filterCoefficient(
"1: \n"
"vld2.16 {d4,d5}, [%[sP]] \n" // load 4 16-bits stereo samples
- "vld1.32 { q8}, [%[coefsP0]]! \n" // load 4 32-bits coefs
- "vld1.32 { q9}, [%[coefsP1]]! \n" // load 4 32-bits coefs for interpolation
+ "vld1.32 { q8}, [%[coefsP0]:128]! \n" // load 4 32-bits coefs
+ "vld1.32 { q9}, [%[coefsP1]:128]! \n" // load 4 32-bits coefs for interpolation
"vld2.16 {d6,d7}, [%[sN]]! \n" // load 4 16-bits stereo samples
- "vld1.32 {q10}, [%[coefsN0]]! \n" // load 4 32-bits coefs
- "vld1.32 {q11}, [%[coefsN1]]! \n" // load 4 32-bits coefs for interpolation
+ "vld1.32 {q10}, [%[coefsN0]:128]! \n" // load 4 32-bits coefs
+ "vld1.32 {q11}, [%[coefsN1]:128]! \n" // load 4 32-bits coefs for interpolation
"vrev64.16 d4, d4 \n" // reverse 2 frames of the positive side
"vrev64.16 d5, d5 \n" // reverse 2 frames of the positive side
@@ -774,16 +783,18 @@ void AudioResamplerSinc::filterCoefficient(
"bne 1b \n" // loop
- "vpadd.s32 d0, d0, d1 \n" // add all 4 partial sums
- "vpadd.s32 d8, d8, d9 \n" // add all 4 partial sums
+ "vld1.s32 {d2}, [%[vLR]] \n" // load volumes
+ "vld1.s32 {d3}, %[out] \n" // load the output
+ "vpadd.s32 d0, d0, d1 \n" // add all 4 partial sums from q0
+ "vpadd.s32 d8, d8, d9 \n" // add all 4 partial sums from q4
"vpadd.s32 d0, d0, d0 \n" // together
"vpadd.s32 d8, d8, d8 \n" // together
+ "vtrn.s32 d0, d8 \n" // interlace L,R channels
+ "vqrdmulh.s32 d0, d0, d2 \n" // apply volume
+ "vadd.s32 d3, d3, d0 \n" // accumulate result
+ "vst1.s32 {d0}, %[out] \n" // store result
- "vmov.s32 %[l], d0[0] \n" // save result in ARM register
- "vmov.s32 %[r], d8[0] \n" // save result in ARM register
-
- : [l] "=r" (l),
- [r] "=r" (r),
+ : [out] "=Uv" (out[0]),
[count] "+r" (count),
[coefsP0] "+r" (coefsP),
[coefsP1] "+r" (coefsP1),
@@ -793,14 +804,12 @@ void AudioResamplerSinc::filterCoefficient(
[sN] "+r" (sN)
: [lerpP] "r" (lerpP<<16),
[lerpN] "r" (lerpN<<16),
- [vRL] "r" (vRL)
+ [vLR] "r" (mVolumeSIMD)
: "cc", "memory",
"q0", "q1", "q2", "q3", "q4",
"q8", "q9", "q10", "q11",
"q12", "q13", "q14", "q15"
);
- l = 2 * mulRL(1, l, vRL);
- r = 2 * mulRL(0, r, vRL);
}
}
generated by cgit v1.1 at 2024-05-19 22:43:30 +0000