Initial Contribution
diff --git a/libs/audioflinger/AudioResampler.cpp b/libs/audioflinger/AudioResampler.cpp
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+++ b/libs/audioflinger/AudioResampler.cpp
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+/*
+ * 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
+