diff options
Diffstat (limited to 'services/audioflinger/AudioResampler.cpp')
| -rw-r--r-- | services/audioflinger/AudioResampler.cpp | 135 |
1 files changed, 106 insertions, 29 deletions
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp index e5cceb1..1f7a613 100644 --- a/services/audioflinger/AudioResampler.cpp +++ b/services/audioflinger/AudioResampler.cpp @@ -22,9 +22,11 @@ #include <sys/types.h> #include <cutils/log.h> #include <cutils/properties.h> +#include <audio_utils/primitives.h> #include "AudioResampler.h" #include "AudioResamplerSinc.h" #include "AudioResamplerCubic.h" +#include "AudioResamplerDyn.h" #ifdef __arm__ #include <machine/cpu-features.h> @@ -39,8 +41,8 @@ namespace android { class AudioResamplerOrder1 : public AudioResampler { public: - AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) : - AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { + AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) : + AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { } virtual void resample(int32_t* out, size_t outFrameCount, AudioBufferProvider* provider); @@ -77,6 +79,9 @@ private: int mX0R; }; +/*static*/ +const double AudioResampler::kPhaseMultiplier = 1L << AudioResampler::kNumPhaseBits; + bool AudioResampler::qualityIsSupported(src_quality quality) { switch (quality) { @@ -85,6 +90,9 @@ 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 +113,7 @@ 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 +133,12 @@ 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; } } @@ -132,7 +146,7 @@ static const uint32_t maxMHz = 130; // an arbitrary number that permits 3 VHQ, s static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER; static uint32_t currentMHz = 0; -AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount, +AudioResampler* AudioResampler::create(audio_format_t format, int inChannelCount, int32_t sampleRate, src_quality quality) { bool atFinalQuality; @@ -148,6 +162,16 @@ 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 +186,6 @@ 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 +198,15 @@ 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); @@ -183,22 +215,43 @@ 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); + LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT); + resampler = new AudioResamplerOrder1(inChannelCount, sampleRate); break; case MED_QUALITY: ALOGV("Create cubic Resampler"); - resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); + LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT); + resampler = new AudioResamplerCubic(inChannelCount, sampleRate); break; case HIGH_QUALITY: ALOGV("Create HIGH_QUALITY sinc Resampler"); - resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); + LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT); + resampler = new AudioResamplerSinc(inChannelCount, sampleRate); break; case VERY_HIGH_QUALITY: ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality); - resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality); + LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT); + resampler = new AudioResamplerSinc(inChannelCount, sampleRate, quality); + break; + case DYN_LOW_QUALITY: + case DYN_MED_QUALITY: + case DYN_HIGH_QUALITY: + ALOGV("Create dynamic Resampler = %d", quality); + if (format == AUDIO_FORMAT_PCM_FLOAT) { + resampler = new AudioResamplerDyn<float, float, float>(inChannelCount, + sampleRate, quality); + } else { + LOG_ALWAYS_FATAL_IF(format != AUDIO_FORMAT_PCM_16_BIT); + if (quality == DYN_HIGH_QUALITY) { + resampler = new AudioResamplerDyn<int32_t, int16_t, int32_t>(inChannelCount, + sampleRate, quality); + } else { + resampler = new AudioResamplerDyn<int16_t, int16_t, int32_t>(inChannelCount, + sampleRate, quality); + } + } break; } @@ -207,26 +260,26 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount, return resampler; } -AudioResampler::AudioResampler(int bitDepth, int inChannelCount, +AudioResampler::AudioResampler(int inChannelCount, int32_t sampleRate, src_quality quality) : - mBitDepth(bitDepth), mChannelCount(inChannelCount), - mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0), - mPhaseFraction(0), mLocalTimeFreq(0), - mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) { - // sanity check on format - if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) { - ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth, - inChannelCount); - // ALOG_ASSERT(0); + mChannelCount(inChannelCount), + mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0), + mPhaseFraction(0), mLocalTimeFreq(0), + mPTS(AudioBufferProvider::kInvalidPTS), mQuality(quality) { + + const int maxChannels = quality < DYN_LOW_QUALITY ? 2 : 8; + if (inChannelCount < 1 + || inChannelCount > maxChannels) { + LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d channels", + quality, inChannelCount); } if (sampleRate <= 0) { - ALOGE("Unsupported sample rate %d Hz", sampleRate); + LOG_ALWAYS_FATAL("Unsupported sample rate %d Hz", sampleRate); } // initialize common members mVolume[0] = mVolume[1] = 0; mBuffer.frameCount = 0; - } AudioResampler::~AudioResampler() { @@ -246,10 +299,12 @@ void AudioResampler::setSampleRate(int32_t inSampleRate) { mPhaseIncrement = (uint32_t)((kPhaseMultiplier * inSampleRate) / mSampleRate); } -void AudioResampler::setVolume(int16_t left, int16_t right) { +void AudioResampler::setVolume(float left, float right) { // TODO: Implement anti-zipper filter - mVolume[0] = left; - mVolume[1] = right; + // convert to U4.12 for internal integer use (round down) + // integer volume values are clamped to 0 to UNITY_GAIN. + mVolume[0] = u4_12_from_float(clampFloatVol(left)); + mVolume[1] = u4_12_from_float(clampFloatVol(right)); } void AudioResampler::setLocalTimeFreq(uint64_t freq) { @@ -305,7 +360,7 @@ 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 +394,9 @@ 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 +458,7 @@ 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 +490,9 @@ 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 +571,16 @@ 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 +692,16 @@ 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" |