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-rw-r--r--services/audioflinger/AudioResampler.cpp81
1 files changed, 74 insertions, 7 deletions
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index e5cceb1..562c4ea 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -25,6 +25,7 @@
#include "AudioResampler.h"
#include "AudioResamplerSinc.h"
#include "AudioResamplerCubic.h"
+#include "AudioResamplerDyn.h"
#ifdef __arm__
#include <machine/cpu-features.h>
@@ -77,6 +78,9 @@ private:
int mX0R;
};
+/*static*/
+const double AudioResampler::kPhaseMultiplier = 1L << AudioResampler::kNumPhaseBits;
+
bool AudioResampler::qualityIsSupported(src_quality quality)
{
switch (quality) {
@@ -85,6 +89,9 @@ bool AudioResampler::qualityIsSupported(src_quality quality)
case MED_QUALITY:
case HIGH_QUALITY:
case VERY_HIGH_QUALITY:
+ case DYN_LOW_QUALITY:
+ case DYN_MED_QUALITY:
+ case DYN_HIGH_QUALITY:
return true;
default:
return false;
@@ -105,7 +112,7 @@ void AudioResampler::init_routine()
if (*endptr == '\0') {
defaultQuality = (src_quality) l;
ALOGD("forcing AudioResampler quality to %d", defaultQuality);
- if (defaultQuality < DEFAULT_QUALITY || defaultQuality > VERY_HIGH_QUALITY) {
+ if (defaultQuality < DEFAULT_QUALITY || defaultQuality > DYN_HIGH_QUALITY) {
defaultQuality = DEFAULT_QUALITY;
}
}
@@ -125,6 +132,12 @@ uint32_t AudioResampler::qualityMHz(src_quality quality)
return 20;
case VERY_HIGH_QUALITY:
return 34;
+ case DYN_LOW_QUALITY:
+ return 4;
+ case DYN_MED_QUALITY:
+ return 6;
+ case DYN_HIGH_QUALITY:
+ return 12;
}
}
@@ -148,6 +161,16 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
atFinalQuality = true;
}
+ /* if the caller requests DEFAULT_QUALITY and af.resampler.property
+ * has not been set, the target resampler quality is set to DYN_MED_QUALITY,
+ * and allowed to "throttle" down to DYN_LOW_QUALITY if necessary
+ * due to estimated CPU load of having too many active resamplers
+ * (the code below the if).
+ */
+ if (quality == DEFAULT_QUALITY) {
+ quality = DYN_MED_QUALITY;
+ }
+
// naive implementation of CPU load throttling doesn't account for whether resampler is active
pthread_mutex_lock(&mutex);
for (;;) {
@@ -162,7 +185,6 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
// not enough CPU available for proposed quality level, so try next lowest level
switch (quality) {
default:
- case DEFAULT_QUALITY:
case LOW_QUALITY:
atFinalQuality = true;
break;
@@ -175,6 +197,15 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
case VERY_HIGH_QUALITY:
quality = HIGH_QUALITY;
break;
+ case DYN_LOW_QUALITY:
+ atFinalQuality = true;
+ break;
+ case DYN_MED_QUALITY:
+ quality = DYN_LOW_QUALITY;
+ break;
+ case DYN_HIGH_QUALITY:
+ quality = DYN_MED_QUALITY;
+ break;
}
}
pthread_mutex_unlock(&mutex);
@@ -183,7 +214,6 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
switch (quality) {
default:
- case DEFAULT_QUALITY:
case LOW_QUALITY:
ALOGV("Create linear Resampler");
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
@@ -200,6 +230,21 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality);
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality);
break;
+ case DYN_LOW_QUALITY:
+ case DYN_MED_QUALITY:
+ case DYN_HIGH_QUALITY:
+ ALOGV("Create dynamic Resampler = %d", quality);
+ if (bitDepth == 32) { /* bitDepth == 32 signals float precision */
+ resampler = new AudioResamplerDyn<float, float, float>(bitDepth, inChannelCount,
+ sampleRate, quality);
+ } else if (quality == DYN_HIGH_QUALITY) {
+ resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(bitDepth, inChannelCount,
+ sampleRate, quality);
+ } else {
+ resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(bitDepth, inChannelCount,
+ sampleRate, quality);
+ }
+ break;
}
// initialize resampler
@@ -305,7 +350,7 @@ 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;
+ size_t inFrameCount = getInFrameCountRequired(outFrameCount);
// ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
@@ -339,8 +384,9 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
Advance(&inputIndex, &phaseFraction, phaseIncrement);
- if (outputIndex == outputSampleCount)
+ if (outputIndex == outputSampleCount) {
break;
+ }
}
// process input samples
@@ -402,7 +448,7 @@ 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;
+ size_t inFrameCount = getInFrameCountRequired(outFrameCount);
// ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
@@ -434,8 +480,9 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
Advance(&inputIndex, &phaseFraction, phaseIncrement);
- if (outputIndex == outputSampleCount)
+ if (outputIndex == outputSampleCount) {
break;
+ }
}
// process input samples
@@ -514,6 +561,16 @@ void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
{
+ (void)maxOutPt; // remove unused parameter warnings
+ (void)maxInIdx;
+ (void)outputIndex;
+ (void)out;
+ (void)inputIndex;
+ (void)vl;
+ (void)vr;
+ (void)phaseFraction;
+ (void)phaseIncrement;
+ (void)in;
#define MO_PARAM5 "36" // offset of parameter 5 (outputIndex)
asm(
@@ -625,6 +682,16 @@ void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
{
+ (void)maxOutPt; // remove unused parameter warnings
+ (void)maxInIdx;
+ (void)outputIndex;
+ (void)out;
+ (void)inputIndex;
+ (void)vl;
+ (void)vr;
+ (void)phaseFraction;
+ (void)phaseIncrement;
+ (void)in;
#define ST_PARAM5 "40" // offset of parameter 5 (outputIndex)
asm(
"stmfd sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
generated by cgit v1.1 at 2024-05-13 23:55:44 +0000