Add and enable floating point option for audio resampler
Can be tested with test-resample.
Change-Id: I8339846d7c647444b6025d33cfa145d5d3658121
Signed-off-by: Andy Hung <hunga@google.com>
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
index df1fdbe..8c56319 100644
--- a/services/audioflinger/AudioResamplerDyn.h
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -25,10 +25,24 @@
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 bitDepth, int inChannelCount, int32_t sampleRate,
- src_quality quality);
+ AudioResamplerDyn(int bitDepth, int inChannelCount,
+ int32_t sampleRate, src_quality quality);
virtual ~AudioResamplerDyn();
@@ -46,46 +60,38 @@
class Constants { // stores the filter constants.
public:
Constants() :
- mL(0), mShift(0), mHalfNumCoefs(0), mFirCoefsS16(NULL)
+ mL(0), mShift(0), mHalfNumCoefs(0), mFirCoefs(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
+ 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;
- };
+ const TC* mFirCoefs; // polyphase filter bank
};
- // 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 {
+ 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);
@@ -94,31 +100,35 @@
// 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.
+ 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.
};
- 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 filter sample rate.
- src_quality mFilterQuality; // designed filter quality.
- void* mCoefBuffer; // if a filter is created, this is not null
+ void setResampler(unsigned resampleType);
+
+ template<int CHANNELS, bool LOCKED, int STRIDE>
+ void resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
+
+ // declare a pointer to member function for resample
+ typedef void (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*/