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-rw-r--r--libs/audioflinger/AudioResampler.cpp350
1 files changed, 324 insertions, 26 deletions
diff --git a/libs/audioflinger/AudioResampler.cpp b/libs/audioflinger/AudioResampler.cpp
index c93ead3..5dabacb 100644
--- a/libs/audioflinger/AudioResampler.cpp
+++ b/libs/audioflinger/AudioResampler.cpp
@@ -14,17 +14,23 @@
* limitations under the License.
*/
+#define LOG_TAG "AudioResampler"
+//#define LOG_NDEBUG 0
+
#include <stdint.h>
#include <stdlib.h>
#include <sys/types.h>
#include <cutils/log.h>
#include <cutils/properties.h>
-
#include "AudioResampler.h"
#include "AudioResamplerSinc.h"
#include "AudioResamplerCubic.h"
namespace android {
+
+#ifdef __ARM_ARCH_5E__ // optimized asm option
+ #define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
+#endif // __ARM_ARCH_5E__
// ----------------------------------------------------------------------------
class AudioResamplerOrder1 : public AudioResampler {
@@ -46,6 +52,15 @@ private:
AudioBufferProvider* provider);
void resampleStereo16(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
+#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
+ void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+ size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+ uint32_t &phaseFraction, uint32_t phaseIncrement);
+ void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+ size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+ uint32_t &phaseFraction, uint32_t phaseIncrement);
+#endif // ASM_ARM_RESAMP1
+
static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) {
return x0 + (((x1 - x0) * (int32_t)(f >> kPreInterpShift)) >> kNumInterpBits);
}
@@ -73,20 +88,23 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
if (quality == DEFAULT)
quality = LOW_QUALITY;
-
+
switch (quality) {
default:
case LOW_QUALITY:
+ LOGV("Create linear Resampler");
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
break;
case MED_QUALITY:
+ LOGV("Create cubic Resampler");
resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
break;
case HIGH_QUALITY:
+ LOGV("Create sinc Resampler");
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
break;
}
-
+
// initialize resampler
resampler->init();
return resampler;
@@ -103,10 +121,10 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
inChannelCount);
// LOG_ASSERT(0);
}
-
+
// initialize common members
mVolume[0] = mVolume[1] = 0;
- mBuffer.raw = NULL;
+ mBuffer.frameCount = 0;
// save format for quick lookup
if (inChannelCount == 1) {
@@ -160,19 +178,31 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
uint32_t phaseIncrement = mPhaseIncrement;
size_t outputIndex = 0;
size_t outputSampleCount = outFrameCount * 2;
+ size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
// LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
- // outFrameCount, inputIndex, phaseFraction, phaseIncrement);
+ // outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
- if (mBuffer.raw == NULL) {
+ while (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer);
- if (mBuffer.raw == NULL)
- break;
+ if (mBuffer.raw == NULL) {
+ goto resampleStereo16_exit;
+ }
+
// LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ if (mBuffer.frameCount > inputIndex) break;
+
+ inputIndex -= mBuffer.frameCount;
+ mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
+ mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
+ provider->releaseBuffer(&mBuffer);
+ // mBuffer.frameCount == 0 now so we reload a new buffer
}
+
int16_t *in = mBuffer.i16;
// handle boundary case
@@ -187,34 +217,47 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
// process input samples
// LOGE("general case\n");
- while (outputIndex < outputSampleCount) {
+
+#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
+ if (inputIndex + 2 < mBuffer.frameCount) {
+ int32_t* maxOutPt;
+ int32_t maxInIdx;
+
+ maxOutPt = out + (outputSampleCount - 2); // 2 because 2 frames per loop
+ maxInIdx = mBuffer.frameCount - 2;
+ AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
+ phaseFraction, phaseIncrement);
+ }
+#endif // ASM_ARM_RESAMP1
+
+ while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
out[outputIndex++] += vl * Interp(in[inputIndex*2-2],
in[inputIndex*2], phaseFraction);
out[outputIndex++] += vr * Interp(in[inputIndex*2-1],
in[inputIndex*2+1], phaseFraction);
Advance(&inputIndex, &phaseFraction, phaseIncrement);
- if (inputIndex >= mBuffer.frameCount)
- break;
}
+
// LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
- // LOGE("buffer done, new input index", inputIndex);
+ // LOGE("buffer done, new input index %d", inputIndex);
mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
provider->releaseBuffer(&mBuffer);
- // verify that the releaseBuffer NULLS the buffer pointer
- // LOG_ASSERT(mBuffer.raw == NULL);
+ // verify that the releaseBuffer resets the buffer frameCount
+ // LOG_ASSERT(mBuffer.frameCount == 0);
}
}
// LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+resampleStereo16_exit:
// save state
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
@@ -231,18 +274,27 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
uint32_t phaseIncrement = mPhaseIncrement;
size_t outputIndex = 0;
size_t outputSampleCount = outFrameCount * 2;
+ size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
// LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
-
while (outputIndex < outputSampleCount) {
-
// buffer is empty, fetch a new one
- if (mBuffer.raw == NULL) {
+ while (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = inFrameCount;
provider->getNextBuffer(&mBuffer);
- if (mBuffer.raw == NULL)
- break;
+ if (mBuffer.raw == NULL) {
+ mInputIndex = inputIndex;
+ mPhaseFraction = phaseFraction;
+ goto resampleMono16_exit;
+ }
// LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ if (mBuffer.frameCount > inputIndex) break;
+
+ inputIndex -= mBuffer.frameCount;
+ mX0L = mBuffer.i16[mBuffer.frameCount-1];
+ provider->releaseBuffer(&mBuffer);
+ // mBuffer.frameCount == 0 now so we reload a new buffer
}
int16_t *in = mBuffer.i16;
@@ -259,38 +311,284 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
// process input samples
// LOGE("general case\n");
- while (outputIndex < outputSampleCount) {
+
+#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
+ if (inputIndex + 2 < mBuffer.frameCount) {
+ int32_t* maxOutPt;
+ int32_t maxInIdx;
+
+ maxOutPt = out + (outputSampleCount - 2);
+ maxInIdx = (int32_t)mBuffer.frameCount - 2;
+ AsmMono16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
+ phaseFraction, phaseIncrement);
+ }
+#endif // ASM_ARM_RESAMP1
+
+ while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
int32_t sample = Interp(in[inputIndex-1], in[inputIndex],
phaseFraction);
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
Advance(&inputIndex, &phaseFraction, phaseIncrement);
- if (inputIndex >= mBuffer.frameCount)
- break;
}
+
+
// LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
- // LOGE("buffer done, new input index", inputIndex);
+ // LOGE("buffer done, new input index %d", inputIndex);
mX0L = mBuffer.i16[mBuffer.frameCount-1];
provider->releaseBuffer(&mBuffer);
- // verify that the releaseBuffer NULLS the buffer pointer
- // LOG_ASSERT(mBuffer.raw == NULL);
+ // verify that the releaseBuffer resets the buffer frameCount
+ // LOG_ASSERT(mBuffer.frameCount == 0);
}
}
// LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+resampleMono16_exit:
// save state
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
}
+#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
+
+/*******************************************************************
+*
+* AsmMono16Loop
+* asm optimized monotonic loop version; one loop is 2 frames
+* Input:
+* in : pointer on input samples
+* maxOutPt : pointer on first not filled
+* maxInIdx : index on first not used
+* outputIndex : pointer on current output index
+* out : pointer on output buffer
+* inputIndex : pointer on current input index
+* vl, vr : left and right gain
+* phaseFraction : pointer on current phase fraction
+* phaseIncrement
+* Ouput:
+* outputIndex :
+* out : updated buffer
+* inputIndex : index of next to use
+* phaseFraction : phase fraction for next interpolation
+*
+*******************************************************************/
+void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+ size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+ uint32_t &phaseFraction, uint32_t phaseIncrement)
+{
+#define MO_PARAM5 "36" // offset of parameter 5 (outputIndex)
+
+ asm(
+ "stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, lr}\n"
+ // get parameters
+ " ldr r6, [sp, #" MO_PARAM5 " + 20]\n" // &phaseFraction
+ " ldr r6, [r6]\n" // phaseFraction
+ " ldr r7, [sp, #" MO_PARAM5 " + 8]\n" // &inputIndex
+ " ldr r7, [r7]\n" // inputIndex
+ " ldr r8, [sp, #" MO_PARAM5 " + 4]\n" // out
+ " ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex
+ " ldr r0, [r0]\n" // outputIndex
+ " add r8, r0, asl #2\n" // curOut
+ " ldr r9, [sp, #" MO_PARAM5 " + 24]\n" // phaseIncrement
+ " ldr r10, [sp, #" MO_PARAM5 " + 12]\n" // vl
+ " ldr r11, [sp, #" MO_PARAM5 " + 16]\n" // vr
+
+ // r0 pin, x0, Samp
+
+ // r1 in
+ // r2 maxOutPt
+ // r3 maxInIdx
+
+ // r4 x1, i1, i3, Out1
+ // r5 out0
+
+ // r6 frac
+ // r7 inputIndex
+ // r8 curOut
+
+ // r9 inc
+ // r10 vl
+ // r11 vr
+
+ // r12
+ // r13 sp
+ // r14
+
+ // the following loop works on 2 frames
+
+ ".Y4L01:\n"
+ " cmp r8, r2\n" // curOut - maxCurOut
+ " bcs .Y4L02\n"
+
+#define MO_ONE_FRAME \
+ " add r0, r1, r7, asl #1\n" /* in + inputIndex */\
+ " ldrsh r4, [r0]\n" /* in[inputIndex] */\
+ " ldr r5, [r8]\n" /* out[outputIndex] */\
+ " ldrsh r0, [r0, #-2]\n" /* in[inputIndex-1] */\
+ " bic r6, r6, #0xC0000000\n" /* phaseFraction & ... */\
+ " sub r4, r4, r0\n" /* in[inputIndex] - in[inputIndex-1] */\
+ " mov r4, r4, lsl #2\n" /* <<2 */\
+ " smulwt r4, r4, r6\n" /* (x1-x0)*.. */\
+ " add r6, r6, r9\n" /* phaseFraction + phaseIncrement */\
+ " add r0, r0, r4\n" /* x0 - (..) */\
+ " mla r5, r0, r10, r5\n" /* vl*interp + out[] */\
+ " ldr r4, [r8, #4]\n" /* out[outputIndex+1] */\
+ " str r5, [r8], #4\n" /* out[outputIndex++] = ... */\
+ " mla r4, r0, r11, r4\n" /* vr*interp + out[] */\
+ " add r7, r7, r6, lsr #30\n" /* inputIndex + phaseFraction>>30 */\
+ " str r4, [r8], #4\n" /* out[outputIndex++] = ... */
+
+ MO_ONE_FRAME // frame 1
+ MO_ONE_FRAME // frame 2
+
+ " cmp r7, r3\n" // inputIndex - maxInIdx
+ " bcc .Y4L01\n"
+ ".Y4L02:\n"
+
+ " bic r6, r6, #0xC0000000\n" // phaseFraction & ...
+ // save modified values
+ " ldr r0, [sp, #" MO_PARAM5 " + 20]\n" // &phaseFraction
+ " str r6, [r0]\n" // phaseFraction
+ " ldr r0, [sp, #" MO_PARAM5 " + 8]\n" // &inputIndex
+ " str r7, [r0]\n" // inputIndex
+ " ldr r0, [sp, #" MO_PARAM5 " + 4]\n" // out
+ " sub r8, r0\n" // curOut - out
+ " asr r8, #2\n" // new outputIndex
+ " ldr r0, [sp, #" MO_PARAM5 " + 0]\n" // &outputIndex
+ " str r8, [r0]\n" // save outputIndex
+
+ " ldmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, pc}\n"
+ );
+}
+
+/*******************************************************************
+*
+* AsmStereo16Loop
+* asm optimized stereo loop version; one loop is 2 frames
+* Input:
+* in : pointer on input samples
+* maxOutPt : pointer on first not filled
+* maxInIdx : index on first not used
+* outputIndex : pointer on current output index
+* out : pointer on output buffer
+* inputIndex : pointer on current input index
+* vl, vr : left and right gain
+* phaseFraction : pointer on current phase fraction
+* phaseIncrement
+* Ouput:
+* outputIndex :
+* out : updated buffer
+* inputIndex : index of next to use
+* phaseFraction : phase fraction for next interpolation
+*
+*******************************************************************/
+void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+ size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+ uint32_t &phaseFraction, uint32_t phaseIncrement)
+{
+#define ST_PARAM5 "40" // offset of parameter 5 (outputIndex)
+ asm(
+ "stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
+ // get parameters
+ " ldr r6, [sp, #" ST_PARAM5 " + 20]\n" // &phaseFraction
+ " ldr r6, [r6]\n" // phaseFraction
+ " ldr r7, [sp, #" ST_PARAM5 " + 8]\n" // &inputIndex
+ " ldr r7, [r7]\n" // inputIndex
+ " ldr r8, [sp, #" ST_PARAM5 " + 4]\n" // out
+ " ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex
+ " ldr r0, [r0]\n" // outputIndex
+ " add r8, r0, asl #2\n" // curOut
+ " ldr r9, [sp, #" ST_PARAM5 " + 24]\n" // phaseIncrement
+ " ldr r10, [sp, #" ST_PARAM5 " + 12]\n" // vl
+ " ldr r11, [sp, #" ST_PARAM5 " + 16]\n" // vr
+
+ // r0 pin, x0, Samp
+
+ // r1 in
+ // r2 maxOutPt
+ // r3 maxInIdx
+
+ // r4 x1, i1, i3, out1
+ // r5 out0
+
+ // r6 frac
+ // r7 inputIndex
+ // r8 curOut
+
+ // r9 inc
+ // r10 vl
+ // r11 vr
+
+ // r12 temporary
+ // r13 sp
+ // r14
+
+ ".Y5L01:\n"
+ " cmp r8, r2\n" // curOut - maxCurOut
+ " bcs .Y5L02\n"
+
+#define ST_ONE_FRAME \
+ " bic r6, r6, #0xC0000000\n" /* phaseFraction & ... */\
+\
+ " add r0, r1, r7, asl #2\n" /* in + 2*inputIndex */\
+\
+ " ldrsh r4, [r0]\n" /* in[2*inputIndex] */\
+ " ldr r5, [r8]\n" /* out[outputIndex] */\
+ " ldrsh r12, [r0, #-4]\n" /* in[2*inputIndex-2] */\
+ " sub r4, r4, r12\n" /* in[2*InputIndex] - in[2*InputIndex-2] */\
+ " mov r4, r4, lsl #2\n" /* <<2 */\
+ " smulwt r4, r4, r6\n" /* (x1-x0)*.. */\
+ " add r12, r12, r4\n" /* x0 - (..) */\
+ " mla r5, r12, r10, r5\n" /* vl*interp + out[] */\
+ " ldr r4, [r8, #4]\n" /* out[outputIndex+1] */\
+ " str r5, [r8], #4\n" /* out[outputIndex++] = ... */\
+\
+ " ldrsh r12, [r0, #+2]\n" /* in[2*inputIndex+1] */\
+ " ldrsh r0, [r0, #-2]\n" /* in[2*inputIndex-1] */\
+ " sub r12, r12, r0\n" /* in[2*InputIndex] - in[2*InputIndex-2] */\
+ " mov r12, r12, lsl #2\n" /* <<2 */\
+ " smulwt r12, r12, r6\n" /* (x1-x0)*.. */\
+ " add r12, r0, r12\n" /* x0 - (..) */\
+ " mla r4, r12, r11, r4\n" /* vr*interp + out[] */\
+ " str r4, [r8], #4\n" /* out[outputIndex++] = ... */\
+\
+ " add r6, r6, r9\n" /* phaseFraction + phaseIncrement */\
+ " add r7, r7, r6, lsr #30\n" /* inputIndex + phaseFraction>>30 */
+
+ ST_ONE_FRAME // frame 1
+ ST_ONE_FRAME // frame 1
+
+ " cmp r7, r3\n" // inputIndex - maxInIdx
+ " bcc .Y5L01\n"
+ ".Y5L02:\n"
+
+ " bic r6, r6, #0xC0000000\n" // phaseFraction & ...
+ // save modified values
+ " ldr r0, [sp, #" ST_PARAM5 " + 20]\n" // &phaseFraction
+ " str r6, [r0]\n" // phaseFraction
+ " ldr r0, [sp, #" ST_PARAM5 " + 8]\n" // &inputIndex
+ " str r7, [r0]\n" // inputIndex
+ " ldr r0, [sp, #" ST_PARAM5 " + 4]\n" // out
+ " sub r8, r0\n" // curOut - out
+ " asr r8, #2\n" // new outputIndex
+ " ldr r0, [sp, #" ST_PARAM5 " + 0]\n" // &outputIndex
+ " str r8, [r0]\n" // save outputIndex
+
+ " ldmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, pc}\n"
+ );
+}
+
+#endif // ASM_ARM_RESAMP1
+
+
// ----------------------------------------------------------------------------
}
; // namespace android
generated by cgit v1.1 at 2025-08-22 11:11:50 +0000