R&J
lab 1 code
Dependencies: CMSIS-DSP_for_STM32F746G BSP_DISCO_F746NG
Diff: main.cpp
- Revision:
- 30:debea332cdfe
- Parent:
- 26:023edf8cea6c
- Child:
- 31:5a0235b66851
--- a/main.cpp Wed Jan 01 23:20:43 2020 +0000
+++ b/main.cpp Fri Jan 03 20:15:03 2020 +0000
@@ -21,49 +21,62 @@
#include "stm32746g_discovery_lcd.h"
#include "signal_processing.h"
+/* The following type definitions are used to control the
+ * buffering of the audio data using a double buffering technique.
+ * Most of the transactions between the WM8994 and the microcontroller
+ * are handled by other code - but this signals what the buffering state
+ * is, so the data can be appropriately processed. */
typedef enum {
BUFFER_OFFSET_NONE = 0,
BUFFER_OFFSET_HALF = 1,
BUFFER_OFFSET_FULL = 2,
} BUFFER_StateTypeDef;
+/* These audio block samples define the size of the buffering */
#define AUDIO_BLOCK_SAMPLES ((uint32_t)128) // Number of samples (L and R) in audio block (each samples is 16 bits)
#define AUDIO_BLOCK_SIZE ((uint32_t)512) // Number of bytes in audio block (4 * AUDIO_BLOCK_SAMPLES)
+/* These RAM addresses are important to determine where the audio data is stored. */
#define SDRAM_DEVICE_ADDR_AUDIO_MEM ((uint32_t)0xC0400000)
#define AUDIO_BUFFER_IN SDRAM_DEVICE_ADDR_AUDIO_MEM
#define AUDIO_BUFFER_OUT (AUDIO_BUFFER_IN + (AUDIO_BLOCK_SIZE * 2))
+/* These definitions define the size of the oscilloscope that is used to display data. */
#define OSC_START_X_POS 20
#define OSC_LINE_SIZE 256
#define OSC_Y_POS 110
#define AUDIO_DRAW_LIMIT 30
+/* This define a timer that is then used to record the timing of the different processing stages. */
Timer timer;
+/* This variable is important because it define the audio buffer recording state. */
volatile uint32_t audio_rec_buffer_state = BUFFER_OFFSET_NONE;
+
+/* Function declarations */
static void Erase_Trace(uint16_t Xpos, uint16_t Ypos, uint16_t Length);
static void Draw_Trace(uint16_t Xpos, uint16_t Ypos, uint16_t* Mem_start, uint16_t Length);
static void Audio_to_Float(uint16_t* buffer_in, float* L_out, float* R_out, uint16_t Length);
static void Float_to_Audio(float* L_in, float* R_in, uint16_t* buffer_out, uint16_t Length);
-/* To do conversion to float */
+/* These memory blocks are important for converting to floating point representation. */
float L_channel_float[AUDIO_BLOCK_SAMPLES];
float R_channel_float[AUDIO_BLOCK_SAMPLES];
float *L_channel_float_p = &L_channel_float[0];
float *R_channel_float_p = &R_channel_float[0];
-/* Back conversion to integer */
+/* These memory blocks are where the information is stored to send back out to the WM8994 chip. */
uint16_t Processed_audio[AUDIO_BLOCK_SAMPLES];
uint16_t *Processed_audio_p = &Processed_audio[0];
/* Useful variables during looping */
-uint32_t counter = 0;
-char buf[40];
-int first_half_time = 0;
-int second_half_time = 0;
-int total_time = 0;
+uint32_t counter = 0; // Loop counter
+char buf[40]; // Character buffer for sprintf statements to the LCD
+int first_half_time = 0; // Time of first processing block
+int second_half_time = 0; // Time of second processing block
+int total_time = 0; // Time of total loop (first and second blocks)
+/* Main Function */
int main()
{
/* Initialize the LCD Screen and display information */
@@ -71,41 +84,57 @@
BSP_LCD_LayerDefaultInit(LTDC_ACTIVE_LAYER, LCD_FB_START_ADDRESS);
BSP_LCD_SelectLayer(LTDC_ACTIVE_LAYER);
+ /* Clear the LCD and set the font to be default */
BSP_LCD_Clear(LCD_COLOR_BLACK);
BSP_LCD_SetFont(&LCD_DEFAULT_FONT);
-
+
+ /* Set the backcolor to be black and the textcolor to be orange. */
BSP_LCD_SetBackColor(LCD_COLOR_BLACK);
BSP_LCD_SetTextColor(LCD_COLOR_ORANGE);
- BSP_LCD_DisplayStringAt(0, 0, (uint8_t *)"487 Mic Audio Test Code", LEFT_MODE);
+
+ /* The following are static display elements that will remain on the screen. */
+ BSP_LCD_DisplayStringAt(0, 0, (uint8_t *)"487 Demo Code (Mazzeo)", LEFT_MODE);
+
+ /* Display the L and R colors for the channels */
BSP_LCD_SetTextColor(LCD_COLOR_BLUE);
BSP_LCD_DisplayStringAt(0, OSC_Y_POS - 20, (uint8_t *)"L", LEFT_MODE);
BSP_LCD_SetTextColor(LCD_COLOR_GREEN);
BSP_LCD_DisplayStringAt(0, OSC_Y_POS, (uint8_t *)"R", LEFT_MODE);
-
- /* Initialize the Audio Interface */
- BSP_AUDIO_IN_OUT_Init(INPUT_DEVICE_DIGITAL_MICROPHONE_2, OUTPUT_DEVICE_HEADPHONE, DEFAULT_AUDIO_IN_FREQ, DEFAULT_AUDIO_IN_BIT_RESOLUTION, DEFAULT_AUDIO_IN_CHANNEL_NBR);
-
- /* Initialize SDRAM buffers */
- BSP_SDRAM_Init();
- memset((uint16_t *)AUDIO_BUFFER_IN, 0, AUDIO_BLOCK_SIZE * 2);
- memset((uint16_t *)AUDIO_BUFFER_OUT, 0, AUDIO_BLOCK_SIZE * 2);
+ /* The following code should not be code that you need to worry about - it sets up the audio interfaces. */
+ /* Initialize the Audio Interface */
+ BSP_AUDIO_IN_OUT_Init(INPUT_DEVICE_DIGITAL_MICROPHONE_2, OUTPUT_DEVICE_HEADPHONE, DEFAULT_AUDIO_IN_FREQ, DEFAULT_AUDIO_IN_BIT_RESOLUTION, DEFAULT_AUDIO_IN_CHANNEL_NBR);
+
+ /* Initialize SDRAM buffers */
+ BSP_SDRAM_Init();
+ memset((uint16_t *)AUDIO_BUFFER_IN, 0, AUDIO_BLOCK_SIZE * 2);
+ memset((uint16_t *)AUDIO_BUFFER_OUT, 0, AUDIO_BLOCK_SIZE * 2);
+
+ /* Start Recording */
+ if (BSP_AUDIO_IN_Record((uint16_t *)AUDIO_BUFFER_IN, AUDIO_BLOCK_SIZE) != AUDIO_OK) { printf("BSP_AUDIO_IN_Record error\n"); }
+
+ /* Start Playback */
+ BSP_AUDIO_OUT_SetAudioFrameSlot(CODEC_AUDIOFRAME_SLOT_02);
+ if (BSP_AUDIO_OUT_Play((uint16_t *)AUDIO_BUFFER_OUT, AUDIO_BLOCK_SIZE * 2) != AUDIO_OK) { printf("BSP_AUDIO_OUT_Play error\n"); }
+
+ /* The audio interfaces are all now working.
+ * Importantly - AUDIO_BUFFER_IN is the pointer to the incoming data from the WM8994
+ * AUDIO_BUFFER_OUT is the pointer to the outgoing data to the WM8994 */
- /* Start Recording */
- if (BSP_AUDIO_IN_Record((uint16_t *)AUDIO_BUFFER_IN, AUDIO_BLOCK_SIZE) != AUDIO_OK) { printf("BSP_AUDIO_IN_Record error\n"); }
-
- /* Start Playback */
- BSP_AUDIO_OUT_SetAudioFrameSlot(CODEC_AUDIOFRAME_SLOT_02);
- if (BSP_AUDIO_OUT_Play((uint16_t *)AUDIO_BUFFER_OUT, AUDIO_BLOCK_SIZE * 2) != AUDIO_OK) { printf("BSP_AUDIO_OUT_Play error\n"); }
-
- /* Initialize signal processing filters */
+ /* Initialize signal processing filters - usually there are variables that
+ * need to be set up or that there are arrays you need to precompute - this
+ * function call allows you to do that. */
initalize_signal_processing();
+ /* Hardware timer starts. Set to zero */
timer.start();
+
+ /* Main signal processing while loop */
while (1) {
+
/* First Half */
- /* Wait end of half block recording before going on in the first half cycle*/
+ /* Wait until end of half block recording before going on in the first half cycle*/
while (audio_rec_buffer_state != BUFFER_OFFSET_HALF) {}
/* This captures the time of an entire cycle */
@@ -121,11 +150,13 @@
/* Convert data to floating point representation for processing */
Audio_to_Float((uint16_t *) AUDIO_BUFFER_IN, L_channel_float_p, R_channel_float_p, AUDIO_BLOCK_SAMPLES);
- /* Here is where signal processing can be done on the floating point arrays */
+ /* ------------------------------------------------------------------------ */
+ /* Here is where signal processing can be done on the floating point arrays */
- process_audio_channel_signals(L_channel_float_p, R_channel_float_p, AUDIO_BLOCK_SAMPLES);
+ process_audio_channel_signals(L_channel_float_p, R_channel_float_p, AUDIO_BLOCK_SAMPLES);
- /* Here is where signal processing can end on the floating point arrays */
+ /* Here is where signal processing can end on the floating point arrays */
+ /* -------------------------------------------------------------------- */
/* Covert floating point data back to fixed point audio format */
Float_to_Audio(L_channel_float_p, R_channel_float_p, (uint16_t *) Processed_audio, AUDIO_BLOCK_SAMPLES);
@@ -154,12 +185,14 @@
/* Convert data to floating point representation for processing */
Audio_to_Float((uint16_t *) (AUDIO_BUFFER_IN + (AUDIO_BLOCK_SIZE)), L_channel_float_p, R_channel_float_p, AUDIO_BLOCK_SAMPLES);
- /* Here is where signal processing can be done on the floating point arrays */
+ /* ------------------------------------------------------------------------ */
+ /* Here is where signal processing can be done on the floating point arrays */
- process_audio_channel_signals(L_channel_float_p, R_channel_float_p, AUDIO_BLOCK_SAMPLES);
+ process_audio_channel_signals(L_channel_float_p, R_channel_float_p, AUDIO_BLOCK_SAMPLES);
- /* Here is where signal processing can end on the floating point arrays */
-
+ /* Here is where signal processing can end on the floating point arrays */
+ /* -------------------------------------------------------------------- */
+
/* Covert floating point data back to fixed point audio format */
Float_to_Audio(L_channel_float_p, R_channel_float_p, (uint16_t *) Processed_audio, AUDIO_BLOCK_SAMPLES);