| /* |
| * 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. |
| */ |
| |
| #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 { |
| // ---------------------------------------------------------------------------- |
| |
| class AudioResamplerOrder1 : public AudioResampler { |
| public: |
| AudioResamplerOrder1(int bitDepth, int inChannelCount, int32_t sampleRate) : |
| AudioResampler(bitDepth, inChannelCount, sampleRate), 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); |
| 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; |
| }; |
| |
| // ---------------------------------------------------------------------------- |
| AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount, |
| int32_t sampleRate, int quality) { |
| |
| // can only create low quality resample now |
| AudioResampler* resampler; |
| |
| char value[PROPERTY_VALUE_MAX]; |
| if (property_get("af.resampler.quality", value, 0)) { |
| quality = atoi(value); |
| LOGD("forcing AudioResampler quality to %d", quality); |
| } |
| |
| if (quality == DEFAULT) |
| quality = LOW_QUALITY; |
| |
| switch (quality) { |
| default: |
| case LOW_QUALITY: |
| resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); |
| break; |
| case MED_QUALITY: |
| resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); |
| break; |
| case HIGH_QUALITY: |
| resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); |
| break; |
| } |
| |
| // initialize resampler |
| resampler->init(); |
| return resampler; |
| } |
| |
| AudioResampler::AudioResampler(int bitDepth, int inChannelCount, |
| int32_t sampleRate) : |
| mBitDepth(bitDepth), mChannelCount(inChannelCount), |
| mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0), |
| mPhaseFraction(0) { |
| // sanity check on format |
| if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) { |
| LOGE("Unsupported sample format, %d bits, %d channels", bitDepth, |
| inChannelCount); |
| // LOG_ASSERT(0); |
| } |
| |
| // initialize common members |
| mVolume[0] = mVolume[1] = 0; |
| mBuffer.raw = NULL; |
| |
| // save format for quick lookup |
| if (inChannelCount == 1) { |
| mFormat = MONO_16_BIT; |
| } else { |
| mFormat = STEREO_16_BIT; |
| } |
| } |
| |
| AudioResampler::~AudioResampler() { |
| } |
| |
| 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 AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount, |
| AudioBufferProvider* provider) { |
| |
| // should never happen, but we overflow if it does |
| // LOG_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; |
| |
| // 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) { |
| provider->getNextBuffer(&mBuffer); |
| if (mBuffer.raw == NULL) |
| break; |
| // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount); |
| } |
| int16_t *in = mBuffer.i16; |
| |
| // handle boundary case |
| while (inputIndex == 0) { |
| // LOGE("boundary case\n"); |
| 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 |
| // LOGE("general case\n"); |
| while (outputIndex < outputSampleCount) { |
| 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); |
| |
| 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); |
| } |
| } |
| |
| // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); |
| |
| // 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; |
| |
| // 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) { |
| provider->getNextBuffer(&mBuffer); |
| if (mBuffer.raw == NULL) |
| break; |
| // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount); |
| } |
| int16_t *in = mBuffer.i16; |
| |
| // handle boundary case |
| while (inputIndex == 0) { |
| // LOGE("boundary case\n"); |
| 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 |
| // LOGE("general case\n"); |
| while (outputIndex < outputSampleCount) { |
| 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); |
| |
| mX0L = mBuffer.i16[mBuffer.frameCount-1]; |
| provider->releaseBuffer(&mBuffer); |
| |
| // verify that the releaseBuffer NULLS the buffer pointer |
| // LOG_ASSERT(mBuffer.raw == NULL); |
| } |
| } |
| |
| // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex); |
| |
| // save state |
| mInputIndex = inputIndex; |
| mPhaseFraction = phaseFraction; |
| } |
| |
| // ---------------------------------------------------------------------------- |
| } |
| ; // namespace android |
| |