initial
Dependencies: mbed BSP_DISCO_F746NG mbed-dsp
signal_processing.cpp
- Committer:
- justenmg
- Date:
- 2020-02-12
- Revision:
- 1:103e3e426b55
- Parent:
- 0:c0f52e8223fe
- Child:
- 2:89234085faae
File content as of revision 1:103e3e426b55:
/** ****************************************************************************** * @file signal_processing.c * @author Brian Mazzeo * @date 2020 * @brief This file provides a set of code for signal processing in 487. * Parts are taken from example code from STMIcroelectronics ****************************************************************************** * @attention * This code was specifically developed for BYU ECEn 487 course * Introduction to Digital Signal Processing. * * ****************************************************************************** */ #include "mbed.h" #include "stm32746g_discovery_lcd.h" #include "arm_math.h" #include "arm_const_structs.h" #include "filter_coefficients.h" #include "our_filter.h" /* ---------------------------------------------------------------------- ** Defines for signal processing ** ------------------------------------------------------------------- */ #define AUDIO_BLOCK_SAMPLES ((uint32_t)128) // Number of samples (L and R) in audio block (each samples is 16 bits) #define BUFFER_LENGTH (WIN_NUM_TAPS + AUDIO_BLOCK_SAMPLES - 1) /* For Lab Exercise */ #define Lab_Execution_Type 2 float32_t lState[NUM_TAPS + AUDIO_BLOCK_SAMPLES - 1]; float32_t rState[NUM_TAPS + AUDIO_BLOCK_SAMPLES - 1]; float32_t l_buf[BUFFER_LENGTH]; float32_t r_buf[BUFFER_LENGTH]; float32_t* l_buf_head = &l_buf; float32_t* r_buf_head = &r_buf; arm_fir_instance_f32 filter_left; arm_fir_instance_f32 filter_right; /* FUNCTION DEFINITIONS BELOW */ /** * @brief Initialize filter structures to be used in loops later * @retval None */ void initalize_signal_processing(void) { switch (Lab_Execution_Type) { case 0: // Passthrough case break; case 1: // FIR case (ARM) arm_fir_init_f32(&filter_left, NUM_TAPS, (float32_t *)&Filter_coeffs, (float32_t *)&lState, AUDIO_BLOCK_SAMPLES); arm_fir_init_f32(&filter_right, NUM_TAPS, (float32_t *)&Filter_coeffs, (float32_t *)&rState, AUDIO_BLOCK_SAMPLES); break; case 2: // FIR case (student) arm_fir_init_f32(&filter_left, OUR_NUM_TAPS, (float32_t *)&our_Filter_coeffs, (float32_t *)&lState, AUDIO_BLOCK_SAMPLES); arm_fir_init_f32(&filter_right, OUR_NUM_TAPS, (float32_t *)&our_Filter_coeffs, (float32_t *)&rState, AUDIO_BLOCK_SAMPLES); break; case 3: // FFT Overlap-add break; case 4: // FFT Overlap-add with real-imag efficiency break; } } /** * @brief Process audio channel signals * @param L_channel_in: Pointer to Left channel data input (float32_t) * @param R_channel_in: Pointer to Right channel data input (float32_t) * @param L_channel_out: Pointer to Left channel data output (float32_t) * @param R_channel_out: Pointer to Right channel data output (float32_t) * @param Signal_Length: length of data to process * @retval None */ void process_audio_channel_signals(float32_t* L_channel_in, float32_t* R_channel_in, float32_t* L_channel_out, float32_t* R_channel_out, uint16_t Signal_Length) { char buf[70]; BSP_LCD_SetFont(&Font8); BSP_LCD_SetTextColor(LCD_COLOR_CYAN); sprintf(buf, "Processing Signals" ); BSP_LCD_DisplayStringAt(0, 200, (uint8_t *) buf, LEFT_MODE); switch(Lab_Execution_Type) { case 0: // Passthrough case arm_copy_f32(L_channel_in, L_channel_out, AUDIO_BLOCK_SAMPLES); arm_copy_f32(R_channel_in, R_channel_out, AUDIO_BLOCK_SAMPLES); break; case 1: // FIR case (ARM) arm_fir_f32(&filter_left, L_channel_in, L_channel_out, AUDIO_BLOCK_SAMPLES); arm_fir_f32(&filter_right, R_channel_in, R_channel_out, AUDIO_BLOCK_SAMPLES); break; case 2: // FIR case (student) arm_fir_f32(&filter_left, L_channel_in, L_channel_out, AUDIO_BLOCK_SAMPLES); arm_fir_f32(&filter_right, R_channel_in, R_channel_out, AUDIO_BLOCK_SAMPLES); break; case 3: // FFT Overlap-add break; case 4: // FFT Overlap-add with real-imag efficiency break; } /* Change font back */ BSP_LCD_SetFont(&Font16); } void filter(float32_t* buffer, float32_t* d_in, float32_t* d_out, uint16_t buf_length) { float32_t result = 0; float32_t* data_sample = d_out; for(i=0; i<buf_length; i++) { *data_sample = convolve(d_in, win_filter_coeffs, AUDIO_BLOCK_SAMPLES, BUFFER_LENGTH); data_sample++; //******************************************************* } } float32_t convolve(float32_t* data, float32_t* filter, uint16_t sig_length, uint16_t buf_length) { float32_t* data_sample = data+buf_length-1; float32_t* filter_sample = filter; float32_t result = 0; for(i=0; i<buf_length; i++) { result += (*filter_sample) * (*data_sample); } return result; }