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:
- 5:3d6cf4dbf458
- Parent:
- 1:0a37bce4f985
- Child:
- 6:ed10856c2305
--- a/firinterpolator.h Thu Jan 26 03:42:05 2017 +0000 +++ b/firinterpolator.h Fri Feb 10 13:24:30 2017 +0000 @@ -16,10 +16,10 @@ * @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] 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 blockSize, uint8_t L); + FIRInterpolator(uint16_t taps, float32_t *pCoeff, uint32_t block_size, uint8_t L); /** * Destructor */ @@ -28,9 +28,9 @@ * @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. + * @param[in] block_size Number of the sample to be filitered. If skipped or zero, block_size is assmued the one passed to the constructor. */ - virtual void run( float32_t *pSrc, float32_t *pDst, uint32_t blockSize = 0 ); + virtual void run( float32_t *pSrc, float32_t *pDst); private: arm_fir_interpolate_instance_f32 state; };