AudioFlinger: Split off audio processing library
Test: native AudioResampler test, general playback test
Bug: 31015569
Change-Id: Ifb248f4402a583438d756c014dcd7a4577aef713
diff --git a/media/libaudioprocessing/AudioResamplerDyn.h b/media/libaudioprocessing/AudioResamplerDyn.h
new file mode 100644
index 0000000..1840fc7
--- /dev/null
+++ b/media/libaudioprocessing/AudioResamplerDyn.h
@@ -0,0 +1,134 @@
+/*
+ * 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 <android/log.h>
+
+#include <media/AudioResampler.h>
+
+namespace android {
+
+/* AudioResamplerDyn
+ *
+ * This class template is used for floating point and integer resamplers.
+ *
+ * Type variables:
+ * TC = filter coefficient type (one of int16_t, int32_t, or float)
+ * TI = input data type (one of int16_t or float)
+ * TO = output data type (one of int32_t or float)
+ *
+ * For integer input data types TI, the coefficient type TC is either int16_t or int32_t.
+ * For float input data types TI, the coefficient type TC is float.
+ */
+
+template<typename TC, typename TI, typename TO>
+class AudioResamplerDyn: public AudioResampler {
+public:
+ AudioResamplerDyn(int inChannelCount,
+ int32_t sampleRate, src_quality quality);
+
+ virtual ~AudioResamplerDyn();
+
+ virtual void init();
+
+ virtual void setSampleRate(int32_t inSampleRate);
+
+ virtual void setVolume(float left, float right);
+
+ virtual size_t resample(int32_t* out, size_t outFrameCount,
+ AudioBufferProvider* provider);
+
+private:
+
+ class Constants { // stores the filter constants.
+ public:
+ Constants() :
+ mL(0), mShift(0), mHalfNumCoefs(0), mFirCoefs(NULL)
+ {}
+ void set(int L, int halfNumCoefs,
+ int inSampleRate, int outSampleRate);
+
+ int mL; // interpolation phases in the filter.
+ int mShift; // right shift to get polyphase index
+ unsigned int mHalfNumCoefs; // filter half #coefs
+ const TC* mFirCoefs; // polyphase filter bank
+ };
+
+ class InBuffer { // buffer management for input type TI
+ 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);
+
+ void reset();
+
+ private:
+ // tuning parameter guidelines: 2 <= multiple <= 8
+ static const int kStateSizeMultipleOfFilterLength = 4;
+
+ // in general, mRingFull = mState + mStateSize - halfNumCoefs*CHANNELS.
+ TI* mState; // base pointer for the input buffer storage
+ TI* mImpulse; // current location of the impulse response (centered)
+ TI* mRingFull; // mState <= mImpulse < mRingFull
+ size_t mStateCount; // size of state in units of TI.
+ };
+
+ void createKaiserFir(Constants &c, double stopBandAtten,
+ int inSampleRate, int outSampleRate, double tbwCheat);
+
+ template<int CHANNELS, bool LOCKED, int STRIDE>
+ size_t resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
+
+ // define a pointer to member function type for resample
+ typedef size_t (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
+ size_t outFrameCount, AudioBufferProvider* provider);
+
+ // data - the contiguous storage and layout of these is important.
+ InBuffer mInBuffer;
+ Constants mConstants; // current set of coefficient parameters
+ TO __attribute__ ((aligned (8))) mVolumeSimd[2]; // must be aligned or NEON may crash
+ resample_ABP_t mResampleFunc; // called function for resampling
+ int32_t mFilterSampleRate; // designed filter sample rate.
+ src_quality mFilterQuality; // designed filter quality.
+ void* mCoefBuffer; // if a filter is created, this is not null
+};
+
+} // namespace android
+
+#endif /*ANDROID_AUDIO_RESAMPLER_DYN_H*/