| The Android Open Source Project | 54b6cfa | 2008-10-21 07:00:00 -0700 | [diff] [blame^] | 1 | /* |
| 2 | * Copyright (C) 2007 The Android Open Source Project |
| 3 | * |
| 4 | * Licensed under the Apache License, Version 2.0 (the "License"); |
| 5 | * you may not use this file except in compliance with the License. |
| 6 | * You may obtain a copy of the License at |
| 7 | * |
| 8 | * http://www.apache.org/licenses/LICENSE-2.0 |
| 9 | * |
| 10 | * Unless required by applicable law or agreed to in writing, software |
| 11 | * distributed under the License is distributed on an "AS IS" BASIS, |
| 12 | * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| 13 | * See the License for the specific language governing permissions and |
| 14 | * limitations under the License. |
| 15 | */ |
| 16 | |
| 17 | #ifndef ANDROID_AUDIO_RESAMPLER_SINC_H |
| 18 | #define ANDROID_AUDIO_RESAMPLER_SINC_H |
| 19 | |
| 20 | #include <stdint.h> |
| 21 | #include <sys/types.h> |
| 22 | #include <cutils/log.h> |
| 23 | |
| 24 | #include "AudioResampler.h" |
| 25 | |
| 26 | namespace android { |
| 27 | // ---------------------------------------------------------------------------- |
| 28 | |
| 29 | class AudioResamplerSinc : public AudioResampler { |
| 30 | public: |
| 31 | AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate); |
| 32 | |
| 33 | ~AudioResamplerSinc(); |
| 34 | |
| 35 | virtual void resample(int32_t* out, size_t outFrameCount, |
| 36 | AudioBufferProvider* provider); |
| 37 | private: |
| 38 | void init(); |
| 39 | |
| 40 | template<int CHANNELS> |
| 41 | void resample(int32_t* out, size_t outFrameCount, |
| 42 | AudioBufferProvider* provider); |
| 43 | |
| 44 | template<int CHANNELS> |
| 45 | inline void filterCoefficient( |
| 46 | int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples); |
| 47 | |
| 48 | template<int CHANNELS> |
| 49 | inline void interpolate( |
| 50 | int32_t& l, int32_t& r, |
| 51 | int32_t const* coefs, int16_t lerp, int16_t const* samples); |
| 52 | |
| 53 | template<int CHANNELS> |
| 54 | inline void read(int16_t*& impulse, uint32_t& phaseFraction, |
| 55 | int16_t const* in, size_t inputIndex); |
| 56 | |
| 57 | int16_t *mState; |
| 58 | int16_t *mImpulse; |
| 59 | int16_t *mRingFull; |
| 60 | |
| 61 | int32_t const * mFirCoefs; |
| 62 | static const int32_t mFirCoefsDown[]; |
| 63 | static const int32_t mFirCoefsUp[]; |
| 64 | |
| 65 | // ---------------------------------------------------------------------------- |
| 66 | static const int32_t RESAMPLE_FIR_NUM_COEF = 8; |
| 67 | static const int32_t RESAMPLE_FIR_LERP_INT_BITS = 4; |
| 68 | |
| 69 | // we have 16 coefs samples per zero-crossing |
| 70 | static const int coefsBits = RESAMPLE_FIR_LERP_INT_BITS; |
| 71 | static const int cShift = kNumPhaseBits - coefsBits; |
| 72 | static const uint32_t cMask = ((1<<coefsBits)-1) << cShift; |
| 73 | |
| 74 | // and we use 15 bits to interpolate between these samples |
| 75 | // this cannot change because the mul below rely on it. |
| 76 | static const int pLerpBits = 15; |
| 77 | static const int pShift = kNumPhaseBits - coefsBits - pLerpBits; |
| 78 | static const uint32_t pMask = ((1<<pLerpBits)-1) << pShift; |
| 79 | |
| 80 | // number of zero-crossing on each side |
| 81 | static const unsigned int halfNumCoefs = RESAMPLE_FIR_NUM_COEF; |
| 82 | }; |
| 83 | |
| 84 | // ---------------------------------------------------------------------------- |
| 85 | }; // namespace android |
| 86 | |
| 87 | #endif /*ANDROID_AUDIO_RESAMPLER_SINC_H*/ |