| /* |
| * Copyright (C) 2007 The Android Open Source Project |
| * |
| * Licensed under the Apache License, Version 2.0 (the "License"); |
| * you may not use this file except in compliance with the License. |
| * You may obtain a copy of the License at |
| * |
| * http://www.apache.org/licenses/LICENSE-2.0 |
| * |
| * Unless required by applicable law or agreed to in writing, software |
| * distributed under the License is distributed on an "AS IS" BASIS, |
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| * See the License for the specific language governing permissions and |
| * 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" |
| |
| #ifdef __arm__ |
| #include <machine/cpu-features.h> |
| #endif |
| |
| namespace android { |
| |
| #ifdef __ARM_HAVE_HALFWORD_MULTIPLY // optimized asm option |
| #define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1 |
| #endif // __ARM_HAVE_HALFWORD_MULTIPLY |
| // ---------------------------------------------------------------------------- |
| |
| class AudioResamplerOrder1 : public AudioResampler { |
| public: |
| AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) : |
| AudioResampler(bitDepth, inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) { |
| } |
| virtual void resample(int32_t* out, size_t outFrameCount, |
| AudioBufferProvider* provider); |
| private: |
| // number of bits used in interpolation multiply - 15 bits avoids overflow |
| static const int kNumInterpBits = 15; |
| |
| // bits to shift the phase fraction down to avoid overflow |
| static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits; |
| |
| void init() {} |
| void resampleMono16(int32_t* out, size_t outFrameCount, |
| 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); |
| } |
| static inline void Advance(size_t* index, uint32_t* frac, uint32_t inc) { |
| *frac += inc; |
| *index += (size_t)(*frac >> kNumPhaseBits); |
| *frac &= kPhaseMask; |
| } |
| int mX0L; |
| int mX0R; |
| }; |
| |
| bool AudioResampler::qualityIsSupported(src_quality quality) |
| { |
| switch (quality) { |
| case DEFAULT_QUALITY: |
| case LOW_QUALITY: |
| case MED_QUALITY: |
| case HIGH_QUALITY: |
| case VERY_HIGH_QUALITY: |
| return true; |
| default: |
| return false; |
| } |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| static pthread_once_t once_control = PTHREAD_ONCE_INIT; |
| static AudioResampler::src_quality defaultQuality = AudioResampler::DEFAULT_QUALITY; |
| |
| void AudioResampler::init_routine() |
| { |
| char value[PROPERTY_VALUE_MAX]; |
| if (property_get("af.resampler.quality", value, NULL) > 0) { |
| char *endptr; |
| unsigned long l = strtoul(value, &endptr, 0); |
| if (*endptr == '\0') { |
| defaultQuality = (src_quality) l; |
| ALOGD("forcing AudioResampler quality to %d", defaultQuality); |
| if (defaultQuality < DEFAULT_QUALITY || defaultQuality > VERY_HIGH_QUALITY) { |
| defaultQuality = DEFAULT_QUALITY; |
| } |
| } |
| } |
| } |
| |
| uint32_t AudioResampler::qualityMHz(src_quality quality) |
| { |
| switch (quality) { |
| default: |
| case DEFAULT_QUALITY: |
| case LOW_QUALITY: |
| return 3; |
| case MED_QUALITY: |
| return 6; |
| case HIGH_QUALITY: |
| return 20; |
| case VERY_HIGH_QUALITY: |
| return 34; |
| } |
| } |
| |
| static const uint32_t maxMHz = 130; // an arbitrary number that permits 3 VHQ, should be tunable |
| static pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER; |
| static uint32_t currentMHz = 0; |
| |
| AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount, |
| int32_t sampleRate, src_quality quality) { |
| |
| bool atFinalQuality; |
| if (quality == DEFAULT_QUALITY) { |
| // read the resampler default quality property the first time it is needed |
| int ok = pthread_once(&once_control, init_routine); |
| if (ok != 0) { |
| ALOGE("%s pthread_once failed: %d", __func__, ok); |
| } |
| quality = defaultQuality; |
| atFinalQuality = false; |
| } else { |
| atFinalQuality = true; |
| } |
| |
| // naive implementation of CPU load throttling doesn't account for whether resampler is active |
| pthread_mutex_lock(&mutex); |
| for (;;) { |
| uint32_t deltaMHz = qualityMHz(quality); |
| uint32_t newMHz = currentMHz + deltaMHz; |
| if ((qualityIsSupported(quality) && newMHz <= maxMHz) || atFinalQuality) { |
| ALOGV("resampler load %u -> %u MHz due to delta +%u MHz from quality %d", |
| currentMHz, newMHz, deltaMHz, quality); |
| currentMHz = newMHz; |
| break; |
| } |
| // 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; |
| case MED_QUALITY: |
| quality = LOW_QUALITY; |
| break; |
| case HIGH_QUALITY: |
| quality = MED_QUALITY; |
| break; |
| case VERY_HIGH_QUALITY: |
| quality = HIGH_QUALITY; |
| break; |
| } |
| } |
| pthread_mutex_unlock(&mutex); |
| |
| AudioResampler* resampler; |
| |
| switch (quality) { |
| default: |
| case DEFAULT_QUALITY: |
| case LOW_QUALITY: |
| ALOGV("Create linear Resampler"); |
| resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); |
| break; |
| case MED_QUALITY: |
| ALOGV("Create cubic Resampler"); |
| resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); |
| break; |
| case HIGH_QUALITY: |
| ALOGV("Create HIGH_QUALITY sinc Resampler"); |
| resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); |
| break; |
| case VERY_HIGH_QUALITY: |
| ALOGV("Create VERY_HIGH_QUALITY sinc Resampler = %d", quality); |
| resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate, quality); |
| break; |
| } |
| |
| // initialize resampler |
| resampler->init(); |
| return resampler; |
| } |
| |
| AudioResampler::AudioResampler(int bitDepth, 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); |
| } |
| if (sampleRate <= 0) { |
| ALOGE("Unsupported sample rate %d Hz", sampleRate); |
| } |
| |
| // initialize common members |
| mVolume[0] = mVolume[1] = 0; |
| mBuffer.frameCount = 0; |
| |
| } |
| |
| AudioResampler::~AudioResampler() { |
| pthread_mutex_lock(&mutex); |
| src_quality quality = getQuality(); |
| uint32_t deltaMHz = qualityMHz(quality); |
| int32_t newMHz = currentMHz - deltaMHz; |
| ALOGV("resampler load %u -> %d MHz due to delta -%u MHz from quality %d", |
| currentMHz, newMHz, deltaMHz, quality); |
| LOG_ALWAYS_FATAL_IF(newMHz < 0, "negative resampler load %d MHz", newMHz); |
| currentMHz = newMHz; |
| pthread_mutex_unlock(&mutex); |
| } |
| |
| void AudioResampler::setSampleRate(int32_t inSampleRate) { |
| mInSampleRate = inSampleRate; |
| mPhaseIncrement = (uint32_t)((kPhaseMultiplier * inSampleRate) / mSampleRate); |
| } |
| |
| void AudioResampler::setVolume(int16_t left, int16_t right) { |
| // TODO: Implement anti-zipper filter |
| mVolume[0] = left; |
| mVolume[1] = right; |
| } |
| |
| void AudioResampler::setLocalTimeFreq(uint64_t freq) { |
| mLocalTimeFreq = freq; |
| } |
| |
| void AudioResampler::setPTS(int64_t pts) { |
| mPTS = pts; |
| } |
| |
| int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) { |
| |
| if (mPTS == AudioBufferProvider::kInvalidPTS) { |
| return AudioBufferProvider::kInvalidPTS; |
| } else { |
| return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate); |
| } |
| } |
| |
| void AudioResampler::reset() { |
| mInputIndex = 0; |
| mPhaseFraction = 0; |
| mBuffer.frameCount = 0; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| |
| void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount, |
| AudioBufferProvider* provider) { |
| |
| // should never happen, but we overflow if it does |
| // ALOG_ASSERT(outFrameCount < 32767); |
| |
| // select the appropriate resampler |
| switch (mChannelCount) { |
| case 1: |
| resampleMono16(out, outFrameCount, provider); |
| break; |
| case 2: |
| resampleStereo16(out, outFrameCount, provider); |
| break; |
| } |
| } |
| |
| void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, |
| AudioBufferProvider* provider) { |
| |
| int32_t vl = mVolume[0]; |
| int32_t vr = mVolume[1]; |
| |
| size_t inputIndex = mInputIndex; |
| uint32_t phaseFraction = mPhaseFraction; |
| uint32_t phaseIncrement = mPhaseIncrement; |
| size_t outputIndex = 0; |
| size_t outputSampleCount = outFrameCount * 2; |
| size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; |
| |
| // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d", |
| // outFrameCount, inputIndex, phaseFraction, phaseIncrement); |
| |
| while (outputIndex < outputSampleCount) { |
| |
| // buffer is empty, fetch a new one |
| while (mBuffer.frameCount == 0) { |
| mBuffer.frameCount = inFrameCount; |
| provider->getNextBuffer(&mBuffer, |
| calculateOutputPTS(outputIndex / 2)); |
| if (mBuffer.raw == NULL) { |
| goto resampleStereo16_exit; |
| } |
| |
| // ALOGE("New buffer fetched: %d frames", 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 |
| while (inputIndex == 0) { |
| // ALOGE("boundary case"); |
| out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction); |
| out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction); |
| Advance(&inputIndex, &phaseFraction, phaseIncrement); |
| if (outputIndex == outputSampleCount) |
| break; |
| } |
| |
| // process input samples |
| // ALOGE("general case"); |
| |
| #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); |
| } |
| |
| // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex); |
| |
| // if done with buffer, save samples |
| if (inputIndex >= mBuffer.frameCount) { |
| inputIndex -= mBuffer.frameCount; |
| |
| // ALOGE("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 resets the buffer frameCount |
| // ALOG_ASSERT(mBuffer.frameCount == 0); |
| } |
| } |
| |
| // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex); |
| |
| resampleStereo16_exit: |
| // save state |
| mInputIndex = inputIndex; |
| mPhaseFraction = phaseFraction; |
| } |
| |
| void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, |
| AudioBufferProvider* provider) { |
| |
| int32_t vl = mVolume[0]; |
| int32_t vr = mVolume[1]; |
| |
| size_t inputIndex = mInputIndex; |
| uint32_t phaseFraction = mPhaseFraction; |
| uint32_t phaseIncrement = mPhaseIncrement; |
| size_t outputIndex = 0; |
| size_t outputSampleCount = outFrameCount * 2; |
| size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate; |
| |
| // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d", |
| // outFrameCount, inputIndex, phaseFraction, phaseIncrement); |
| while (outputIndex < outputSampleCount) { |
| // buffer is empty, fetch a new one |
| while (mBuffer.frameCount == 0) { |
| mBuffer.frameCount = inFrameCount; |
| provider->getNextBuffer(&mBuffer, |
| calculateOutputPTS(outputIndex / 2)); |
| if (mBuffer.raw == NULL) { |
| mInputIndex = inputIndex; |
| mPhaseFraction = phaseFraction; |
| goto resampleMono16_exit; |
| } |
| // ALOGE("New buffer fetched: %d frames", 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; |
| |
| // handle boundary case |
| while (inputIndex == 0) { |
| // ALOGE("boundary case"); |
| int32_t sample = Interp(mX0L, in[0], phaseFraction); |
| out[outputIndex++] += vl * sample; |
| out[outputIndex++] += vr * sample; |
| Advance(&inputIndex, &phaseFraction, phaseIncrement); |
| if (outputIndex == outputSampleCount) |
| break; |
| } |
| |
| // process input samples |
| // ALOGE("general case"); |
| |
| #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); |
| } |
| |
| |
| // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex); |
| |
| // if done with buffer, save samples |
| if (inputIndex >= mBuffer.frameCount) { |
| inputIndex -= mBuffer.frameCount; |
| |
| // ALOGE("buffer done, new input index %d", inputIndex); |
| |
| mX0L = mBuffer.i16[mBuffer.frameCount-1]; |
| provider->releaseBuffer(&mBuffer); |
| |
| // verify that the releaseBuffer resets the buffer frameCount |
| // ALOG_ASSERT(mBuffer.frameCount == 0); |
| } |
| } |
| |
| // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", 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 |
| * |
| *******************************************************************/ |
| __attribute__((noinline)) |
| 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 |
| |
| "1:\n" |
| " cmp r8, r2\n" // curOut - maxCurOut |
| " bcs 2f\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 1b\n" |
| "2:\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 |
| * |
| *******************************************************************/ |
| __attribute__((noinline)) |
| 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 |
| |
| "3:\n" |
| " cmp r8, r2\n" // curOut - maxCurOut |
| " bcs 4f\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 3b\n" |
| "4:\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 |
