Audio resampler update to add S16 filters
This does not affect the existing resamplers.
New resampler accessed through additional quality settings:
DYN_LOW_QUALITY = 5
DYN_MED_QUALITY = 6
DYN_HIGH_QUALITY = 7
Change-Id: Iebbd31871e808a4a6dee3f3abfd7e9dcf77c48e1
Signed-off-by: Andy Hung <hunga@google.com>
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
new file mode 100644
index 0000000..85a01ab
--- /dev/null
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -0,0 +1,123 @@
+/*
+ * Copyright (C) 2013 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_DYN_H
+#define ANDROID_AUDIO_RESAMPLER_DYN_H
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/log.h>
+
+#include "AudioResampler.h"
+
+namespace android {
+
+class AudioResamplerDyn: public AudioResampler {
+public:
+ AudioResamplerDyn(int bitDepth, int inChannelCount, int32_t sampleRate,
+ src_quality quality);
+
+ virtual ~AudioResamplerDyn();
+
+ virtual void init();
+
+ virtual void setSampleRate(int32_t inSampleRate);
+
+ virtual void setVolume(int16_t left, int16_t right);
+
+ virtual void resample(int32_t* out, size_t outFrameCount,
+ AudioBufferProvider* provider);
+
+private:
+
+ class Constants { // stores the filter constants.
+ public:
+ Constants() :
+ mL(0), mShift(0), mHalfNumCoefs(0), mFirCoefsS16(NULL)
+ {}
+ void set(int L, int halfNumCoefs,
+ int inSampleRate, int outSampleRate);
+ inline void setBuf(int16_t* buf) {
+ mFirCoefsS16 = buf;
+ }
+ inline void setBuf(int32_t* buf) {
+ mFirCoefsS32 = buf;
+ }
+
+ int mL; // interpolation phases in the filter.
+ int mShift; // right shift to get polyphase index
+ unsigned int mHalfNumCoefs; // filter half #coefs
+ union { // polyphase filter bank
+ const int16_t* mFirCoefsS16;
+ const int32_t* mFirCoefsS32;
+ };
+ };
+
+ // Input buffer management for a given input type TI, now (int16_t)
+ // Is agnostic of the actual type, can work with int32_t and float.
+ template<typename TI>
+ class InBuffer {
+ public:
+ InBuffer();
+ ~InBuffer();
+ void init();
+ void resize(int CHANNELS, int halfNumCoefs);
+
+ // used for direct management of the mImpulse pointer
+ inline TI* getImpulse() {
+ return mImpulse;
+ }
+ inline void setImpulse(TI *impulse) {
+ mImpulse = impulse;
+ }
+ template<int CHANNELS>
+ inline void readAgain(TI*& impulse, const int halfNumCoefs,
+ const TI* const in, const size_t inputIndex);
+ template<int CHANNELS>
+ inline void readAdvance(TI*& impulse, const int halfNumCoefs,
+ const TI* const in, const size_t inputIndex);
+
+ private:
+ // tuning parameter guidelines: 2 <= multiple <= 8
+ static const int kStateSizeMultipleOfFilterLength = 4;
+
+ TI* mState; // base pointer for the input buffer storage
+ TI* mImpulse; // current location of the impulse response (centered)
+ TI* mRingFull; // mState <= mImpulse < mRingFull
+ // in general, mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS.
+ size_t mStateSize; // in units of TI.
+ };
+
+ template<int CHANNELS, bool LOCKED, int STRIDE, typename TC>
+ void resample(int32_t* out, size_t outFrameCount,
+ const TC* const coefs, AudioBufferProvider* provider);
+
+ template<typename T>
+ void createKaiserFir(Constants &c, double stopBandAtten,
+ int inSampleRate, int outSampleRate, double tbwCheat);
+
+ InBuffer<int16_t> mInBuffer;
+ Constants mConstants; // current set of coefficient parameters
+ int32_t __attribute__ ((aligned (8))) mVolumeSimd[2];
+ int32_t mResampleType; // contains the resample type.
+ int32_t mFilterSampleRate; // designed sample rate for the filter
+ void* mCoefBuffer; // if a filter is created, this is not null
+};
+
+// ----------------------------------------------------------------------------
+}; // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_DYN_H*/