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。
firinterpolator.h
- Committer:
- shorie
- Date:
- 2017-02-10
- Revision:
- 6:ed10856c2305
- Parent:
- 5:3d6cf4dbf458
File content as of revision 6:ed10856c2305:
#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] block_size 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 block_size, 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. */ virtual void run( float32_t *pSrc, float32_t *pDst); private: arm_fir_interpolate_instance_f32 state; }; } #endif