diff options
Diffstat (limited to 'services/audioflinger/AudioResampler.cpp')
| -rw-r--r-- | services/audioflinger/AudioResampler.cpp | 65 |
1 files changed, 60 insertions, 5 deletions
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index e5cceb1..07574ab 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> @@ -85,6 +86,9 @@ 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 +109,7 @@ 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 +129,12 @@ 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 +158,16 @@ 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 +182,6 @@ 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 +194,15 @@ 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 +211,6 @@ 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 +227,12 @@ 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); + resampler = new AudioResamplerDyn(bitDepth, inChannelCount, sampleRate, quality); + break; } // initialize resampler @@ -339,8 +372,9 @@ 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 @@ -434,8 +468,9 @@ 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 +549,16 @@ 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 +670,16 @@ 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" |