Basic Audio Signal Processing Library
Dependents: unzen_sample_nucleo_f746 skeleton_unzen_nucleo_f746 ifmag_noise_canceller synthesizer_f746
オーディオ信号処理用のライブラリです。
mbed-dspのフィルタ群向けに作ったクラス・ラッパーのほか、以下のクラスを用意しています。
- ヒステリシス
- sin/cosオシレータ
- リミッター
クラスは全て名前空間amakusaに含まれます。
Diff: firinterpolator.h
- Revision:
- 1:0a37bce4f985
- Child:
- 5:3d6cf4dbf458
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/firinterpolator.h Mon Jan 02 11:04:49 2017 +0000 @@ -0,0 +1,40 @@ +#ifndef _firinterpolator_h_ +#define _firinterpolator_h_ + +#include "abstractfilter.h" +namespace amakusa +{ +/** +* @brief Wrapper class of the arm_fir_interpolate_f32() and the arm_fir_interpolate_init_f32(). +* @details +* To use this class, include amakusa.h +*/ + class FIRInterpolator : public AbstractFilter + { + public: + /** + * @brief Constructor + * @param[in] taps Number of the elements in the coeffisients array. Or length of the impuls response. The taps must be integer multiple of L + * @param[in] pCoeff Ponter to the coefficients array ( Impuls response ). + * @param[in] blockSize Maximum number of the input samples to be given to run() method at onece. + * @param[in] L Up sampling ratio + */ + FIRInterpolator(uint16_t taps, float32_t *pCoeff, uint32_t blockSize, uint8_t L); + /** + * Destructor + */ + virtual ~FIRInterpolator(); + /** + * @brief Run the filter. + * @param[in] pSrc Pointer to the source buffer to be filtered. + * @param[out] pDst Pointer to the destination buffer to store the filtered signal. + * @param[in] blockSize Number of the sample to be filitered. If skipped or zero, blockSize is assmued the one passed to the constructor. + */ + virtual void run( float32_t *pSrc, float32_t *pDst, uint32_t blockSize = 0 ); + private: + arm_fir_interpolate_instance_f32 state; + }; + +} + +#endif