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Diffstat (limited to 'libs/audioflinger/AudioResamplerSinc.h')
-rw-r--r-- | libs/audioflinger/AudioResamplerSinc.h | 88 |
1 files changed, 0 insertions, 88 deletions
diff --git a/libs/audioflinger/AudioResamplerSinc.h b/libs/audioflinger/AudioResamplerSinc.h deleted file mode 100644 index e6cb90b..0000000 --- a/libs/audioflinger/AudioResamplerSinc.h +++ /dev/null @@ -1,88 +0,0 @@ -/* - * 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*/ |