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authorThe Android Open Source Project <initial-contribution@android.com>2009-03-03 19:31:44 -0800
committerThe Android Open Source Project <initial-contribution@android.com>2009-03-03 19:31:44 -0800
commit9066cfe9886ac131c34d59ed0e2d287b0e3c0087 (patch)
treed88beb88001f2482911e3d28e43833b50e4b4e97 /libs/audioflinger/AudioResamplerSinc.h
parentd83a98f4ce9cfa908f5c54bbd70f03eec07e7553 (diff)
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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.
+ */
+
+#ifndef ANDROID_AUDIO_RESAMPLER_SINC_H
+#define ANDROID_AUDIO_RESAMPLER_SINC_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/log.h>
+
+#include "AudioResampler.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+class AudioResamplerSinc : public AudioResampler {
+public:
+ AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate);
+
+ ~AudioResamplerSinc();
+
+ virtual void resample(int32_t* out, size_t outFrameCount,
+ AudioBufferProvider* provider);
+private:
+ void init();
+
+ template<int CHANNELS>
+ void resample(int32_t* out, size_t outFrameCount,
+ AudioBufferProvider* provider);
+
+ template<int CHANNELS>
+ inline void filterCoefficient(
+ int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples);
+
+ template<int CHANNELS>
+ inline void interpolate(
+ int32_t& l, int32_t& r,
+ int32_t const* coefs, int16_t lerp, int16_t const* samples);
+
+ template<int CHANNELS>
+ inline void read(int16_t*& impulse, uint32_t& phaseFraction,
+ int16_t const* in, size_t inputIndex);
+
+ int16_t *mState;
+ int16_t *mImpulse;
+ int16_t *mRingFull;
+
+ int32_t const * mFirCoefs;
+ static const int32_t mFirCoefsDown[];
+ static const int32_t mFirCoefsUp[];
+
+ // ----------------------------------------------------------------------------
+ static const int32_t RESAMPLE_FIR_NUM_COEF = 8;
+ static const int32_t RESAMPLE_FIR_LERP_INT_BITS = 4;
+
+ // we have 16 coefs samples per zero-crossing
+ static const int coefsBits = RESAMPLE_FIR_LERP_INT_BITS; // 4
+ static const int cShift = kNumPhaseBits - coefsBits; // 26
+ static const uint32_t cMask = ((1<<coefsBits)-1) << cShift; // 0xf<<26 = 3c00 0000
+
+ // and we use 15 bits to interpolate between these samples
+ // this cannot change because the mul below rely on it.
+ static const int pLerpBits = 15;
+ static const int pShift = kNumPhaseBits - coefsBits - pLerpBits; // 11
+ static const uint32_t pMask = ((1<<pLerpBits)-1) << pShift; // 0x7fff << 11
+
+ // number of zero-crossing on each side
+ static const unsigned int halfNumCoefs = RESAMPLE_FIR_NUM_COEF;
+};
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_SINC_H*/