seiichi horie
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