Add and enable multichannel for audio resampler
Change-Id: I2b86fb73d70abc4c456f7567270a888086b301d4
Signed-off-by: Andy Hung <hunga@google.com>
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 318eb57..7ca10c1 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -38,11 +38,6 @@
namespace android {
-// generate a unique resample type compile-time constant (constexpr)
-#define RESAMPLETYPE(CHANNELS, LOCKED, STRIDE) \
- ((((CHANNELS)-1)&1) | !!(LOCKED)<<1 \
- | ((STRIDE)==8 ? 1 : (STRIDE)==16 ? 2 : 0)<<2)
-
/*
* InBuffer is a type agnostic input buffer.
*
@@ -403,12 +398,76 @@
// determine which resampler to use
// check if locked phase (works only if mPhaseIncrement has no "fractional phase bits")
int locked = (mPhaseIncrement << (sizeof(mPhaseIncrement)*8 - c.mShift)) == 0;
- int stride = (c.mHalfNumCoefs&7)==0 ? 16 : (c.mHalfNumCoefs&3)==0 ? 8 : 2;
if (locked) {
mPhaseFraction = mPhaseFraction >> c.mShift << c.mShift; // remove fractional phase
}
- setResampler(RESAMPLETYPE(mChannelCount, locked, stride));
+ // stride is the minimum number of filter coefficients processed per loop iteration.
+ // We currently only allow a stride of 16 to match with SIMD processing.
+ // This means that the filter length must be a multiple of 16,
+ // or half the filter length (mHalfNumCoefs) must be a multiple of 8.
+ //
+ // Note: A stride of 2 is achieved with non-SIMD processing.
+ int stride = ((c.mHalfNumCoefs & 7) == 0) ? 16 : 2;
+ LOG_ALWAYS_FATAL_IF(stride < 16, "Resampler stride must be 16 or more");
+ LOG_ALWAYS_FATAL_IF(mChannelCount > 8 || mChannelCount < 1,
+ "Resampler channels(%d) must be between 1 to 8", mChannelCount);
+ // stride 16 (falls back to stride 2 for machines that do not support NEON)
+ if (locked) {
+ switch (mChannelCount) {
+ case 1:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, true, 16>;
+ break;
+ case 2:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, true, 16>;
+ break;
+ case 3:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, true, 16>;
+ break;
+ case 4:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, true, 16>;
+ break;
+ case 5:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, true, 16>;
+ break;
+ case 6:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, true, 16>;
+ break;
+ case 7:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, true, 16>;
+ break;
+ case 8:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, true, 16>;
+ break;
+ }
+ } else {
+ switch (mChannelCount) {
+ case 1:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, false, 16>;
+ break;
+ case 2:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, false, 16>;
+ break;
+ case 3:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, false, 16>;
+ break;
+ case 4:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, false, 16>;
+ break;
+ case 5:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, false, 16>;
+ break;
+ case 6:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, false, 16>;
+ break;
+ case 7:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, false, 16>;
+ break;
+ case 8:
+ mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, false, 16>;
+ break;
+ }
+ }
#ifdef DEBUG_RESAMPLER
printf("channels:%d %s stride:%d %s coef:%d shift:%d\n",
mChannelCount, locked ? "locked" : "interpolated",
@@ -424,34 +483,12 @@
}
template<typename TC, typename TI, typename TO>
-void AudioResamplerDyn<TC, TI, TO>::setResampler(unsigned resampleType)
-{
- // stride 16 (falls back to stride 2 for machines that do not support NEON)
- switch (resampleType) {
- case RESAMPLETYPE(1, true, 16):
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, true, 16>;
- return;
- case RESAMPLETYPE(2, true, 16):
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, true, 16>;
- return;
- case RESAMPLETYPE(1, false, 16):
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, false, 16>;
- return;
- case RESAMPLETYPE(2, false, 16):
- mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, false, 16>;
- return;
- default:
- LOG_ALWAYS_FATAL("Invalid resampler type: %u", resampleType);
- mResampleFunc = NULL;
- return;
- }
-}
-
-template<typename TC, typename TI, typename TO>
template<int CHANNELS, bool LOCKED, int STRIDE>
void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
AudioBufferProvider* provider)
{
+ // TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out.
+ const int OUTPUT_CHANNELS = (CHANNELS < 2) ? 2 : CHANNELS;
const Constants& c(mConstants);
const TC* const coefs = mConstants.mFirCoefs;
TI* impulse = mInBuffer.getImpulse();
@@ -459,7 +496,7 @@
uint32_t phaseFraction = mPhaseFraction;
const uint32_t phaseIncrement = mPhaseIncrement;
size_t outputIndex = 0;
- size_t outputSampleCount = outFrameCount * 2; // stereo output
+ size_t outputSampleCount = outFrameCount * OUTPUT_CHANNELS;
const uint32_t phaseWrapLimit = c.mL << c.mShift;
size_t inFrameCount = (phaseIncrement * (uint64_t)outFrameCount + phaseFraction)
/ phaseWrapLimit;
@@ -490,7 +527,7 @@
while (mBuffer.frameCount == 0 && inFrameCount > 0) {
mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer,
- calculateOutputPTS(outputIndex / 2));
+ calculateOutputPTS(outputIndex / OUTPUT_CHANNELS));
if (mBuffer.raw == NULL) {
goto resample_exit;
}
@@ -538,7 +575,8 @@
phaseFraction, phaseWrapLimit,
coefShift, halfNumCoefs, coefs,
impulse, volumeSimd);
- outputIndex += 2;
+
+ outputIndex += OUTPUT_CHANNELS;
phaseFraction += phaseIncrement;
while (phaseFraction >= phaseWrapLimit) {