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-rw-r--r--libs/audioflinger/AudioResamplerSinc.h88
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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*/