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AIOC/stm32/aioc-fw/Src/usb_audio.c

828 wiersze
28 KiB
C
Czysty Wina Historia

#include "usb_audio.h"
#include "stm32f3xx_hal.h"
#include "aioc.h"
#include "tusb.h"
#include "usb.h"
#include <math.h>
/* The one and only supported sample rate */
#define DEFAULT_SAMPLE_RATE 48000
/* This is feedback average responsivity with a denominator of 65536 */
#define SPEAKER_FEEDBACK_AVG 32
/* This is buffer level average responsivity with a denominator of 65536 */
#define SPEAKER_BUFFERLVL_AVG 64
/* This is the amount of buffer level to feedback coupling with a denominator of 65536 to prevent buffer drift */
#define SPEAKER_BUFLVL_FB_COUPLING 1
/* We try to stay on this target with the buffer level */
#define SPEAKER_BUFFERLVL_TARGET (5 * CFG_TUD_AUDIO_EP_SZ_OUT) /* Keep our buffer at 5 frames, i.e. 5ms at full-speed USB and maximum sample rate */
typedef enum {
SAMPLERATE_48000,
SAMPLERATE_24000,
SAMPLERATE_22050, /* For APRSdroid support. NOTE: Has approx. 90 ppm of clock frequency error (ca. 22052 Hz) */
SAMPLERATE_COUNT /* Has to be last element */
} samplerate_t;
typedef enum {
STATE_OFF,
STATE_START,
STATE_RUN
} state_t;
/* Various state variables. N+1 because 0 is always the master channel */
static bool microphoneMute[CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_TX + 1];
static bool speakerMute[CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_RX + 1];
static int16_t microphoneLogVolume[CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_TX + 1] = { [0 ... CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_TX] = 0 }; /* in dB */
static int16_t speakerLogVolume[CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_RX + 1] = { [0 ... CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_RX] = 0 }; /* in dB */
static uint16_t microphoneLinVolume[CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_RX + 1] = { [0 ... CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_RX] = 65535 }; /* 0.16 format */
static uint16_t speakerLinVolume[CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_RX + 1] = { [0 ... CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_RX] = 65535 }; /* 0.16 format */
static uint32_t microphoneSampleFreq = DEFAULT_SAMPLE_RATE; /* Current (requested) sample rate */
static uint32_t speakerSampleFreq = DEFAULT_SAMPLE_RATE; /* Current (requested) sample rate */
static uint64_t speakerFeedbackAvg; /* 32.32 format */
static uint32_t speakerFeedbackMin;
static uint32_t speakerFeedbackMax;
static uint32_t speakerBufferLvlAvg; /* 16.16 format */
static uint16_t speakerBufferLvlMin;
static uint16_t speakerBufferLvlMax;
static volatile uint32_t microphoneSampleFreqCfg; /* Actual configured sample rate in the timer hardware. May be different from requested for odd sample rates */
static volatile uint32_t speakerSampleFreqCfg; /* Actual configured sample rate in the timer hardware. May be different from requested for odd sample rates */
static volatile state_t microphoneState = STATE_OFF;
static volatile state_t speakerState = STATE_OFF;
static audio_control_range_4_n_t(SAMPLERATE_COUNT) sampleFreqRng = {
.wNumSubRanges = SAMPLERATE_COUNT,
.subrange = {
[SAMPLERATE_48000] = {.bMin = 48000, .bMax = 48000, .bRes = 0},
[SAMPLERATE_24000] = {.bMin = 24000, .bMax = 24000, .bRes = 0},
[SAMPLERATE_22050] = {.bMin = 22050, .bMax = 22050, .bRes = 0},
}
};
/* Prototypes of static functions */
static void Timer_ADC_Init(void);
static void Timer_DAC_Init(void);
static void ADC_Init(void);
static void DAC_Init(void);
//--------------------------------------------------------------------+
// Application Callback API Implementations
//--------------------------------------------------------------------+
// Invoked when audio class specific set request received for an entity
bool tud_audio_set_req_entity_cb(uint8_t rhport, tusb_control_request_t const * p_request, uint8_t *pBuff)
{
(void) rhport;
// Page 91 in UAC2 specification
uint8_t channelNum = TU_U16_LOW(p_request->wValue);
uint8_t ctrlSel = TU_U16_HIGH(p_request->wValue);
uint8_t itf = TU_U16_LOW(p_request->wIndex);
uint8_t entityID = TU_U16_HIGH(p_request->wIndex);
TU_ASSERT(itf == ITF_NUM_AUDIO_CONTROL);
// We do not support any set range requests here, only current value requests
TU_VERIFY(p_request->bRequest == AUDIO_CS_REQ_CUR);
if ( entityID == AUDIO_CTRL_ID_MIC_FUNIT )
{
switch ( ctrlSel )
{
case AUDIO_FU_CTRL_MUTE:
// Request uses format layout 1
TU_VERIFY(p_request->wLength == sizeof(audio_control_cur_1_t));
microphoneMute[channelNum] = ((audio_control_cur_1_t*) pBuff)->bCur;
TU_LOG2(" Set Mute: %d of channel: %u\r\n", microphoneMute[channelNum], channelNum);
return true;
case AUDIO_FU_CTRL_VOLUME:
// Request uses format layout 2
TU_VERIFY(p_request->wLength == sizeof(audio_control_cur_2_t));
microphoneLogVolume[channelNum] = ((audio_control_cur_2_t*) pBuff)->bCur;
double logVolume = microphoneLogVolume[channelNum] / 256; /* format is 7.8 fixed point */
microphoneLinVolume[channelNum] = (microphoneLogVolume[channelNum] != 0x8000) ?
(uint16_t) (65535 * pow(10, logVolume/20) + 0.5) : 0; /* log to linear with rounding */
TU_LOG2(" Set Volume: %u.%u dB of channel: %u\r\n", microphoneLogVolume[channelNum] / 256, microphoneLogVolume[channelNum] % 256, channelNum);
return true;
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
}
if ( entityID == AUDIO_CTRL_ID_SPK_FUNIT )
{
switch ( ctrlSel )
{
case AUDIO_FU_CTRL_MUTE:
// Request uses format layout 1
TU_VERIFY(p_request->wLength == sizeof(audio_control_cur_1_t));
speakerMute[channelNum] = ((audio_control_cur_1_t*) pBuff)->bCur;
TU_LOG2(" Set Mute: %d of channel: %u\r\n", speakerMute[channelNum], channelNum);
return true;
case AUDIO_FU_CTRL_VOLUME:
// Request uses format layout 2
TU_VERIFY(p_request->wLength == sizeof(audio_control_cur_2_t));
speakerLogVolume[channelNum] = ((audio_control_cur_2_t*) pBuff)->bCur;
double logVolume = (double) speakerLogVolume[channelNum] / 256; /* format is 7.8 fixed point */
speakerLinVolume[channelNum] = (speakerLogVolume[channelNum] != 0x8000) ?
(uint16_t) (65535 * pow(10, logVolume/20) + 0.5) : 0; /* log to linear with rounding */
TU_LOG2(" Set Volume: %u.%u dB of channel: %u\r\n", microphoneLogVolume[channelNum] / 256, microphoneLogVolume[channelNum] % 256, channelNum);
return true;
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
}
if ( entityID == AUDIO_CTRL_ID_MIC_CLOCK )
{
switch ( ctrlSel )
{
case AUDIO_CS_CTRL_SAM_FREQ:
// channelNum is always zero in this case
switch ( p_request->bRequest )
{
case AUDIO_CS_REQ_CUR:
TU_VERIFY(p_request->wLength == sizeof(audio_control_cur_4_t));
microphoneSampleFreq = ((audio_control_cur_4_t*) pBuff)->bCur;
TU_LOG2(" Set Mic. Sample Freq: %lu\r\n", microphoneSampleFreq);
Timer_ADC_Init();
return true;
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
break;
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
}
if ( entityID == AUDIO_CTRL_ID_SPK_CLOCK )
{
switch ( ctrlSel )
{
case AUDIO_CS_CTRL_SAM_FREQ:
// channelNum is always zero in this case
switch ( p_request->bRequest )
{
case AUDIO_CS_REQ_CUR:
TU_VERIFY(p_request->wLength == sizeof(audio_control_cur_4_t));
speakerSampleFreq = ((audio_control_cur_4_t*) pBuff)->bCur;
TU_LOG2(" Set Spk. Sample Freq: %lu\r\n", speakerSampleFreq);
Timer_DAC_Init();
return true;
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
break;
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
}
return false; // Yet not implemented
}
// Invoked when audio class specific get request received for an entity
bool tud_audio_get_req_entity_cb(uint8_t rhport, tusb_control_request_t const * p_request)
{
(void) rhport;
// Page 91 in UAC2 specification
uint8_t channelNum = TU_U16_LOW(p_request->wValue);
uint8_t ctrlSel = TU_U16_HIGH(p_request->wValue);
uint8_t itf = TU_U16_LOW(p_request->wIndex);
uint8_t entityID = TU_U16_HIGH(p_request->wIndex);
TU_ASSERT(itf == ITF_NUM_AUDIO_CONTROL);
// Input terminal (Microphone input)
if (entityID == AUDIO_CTRL_ID_MIC_INPUT)
{
switch ( ctrlSel )
{
case AUDIO_TE_CTRL_CONNECTOR:
{
// The terminal connector control only has a get request with only the CUR attribute.
audio_desc_channel_cluster_t ret;
// Those are dummy values for now
ret.bNrChannels = 1;
ret.bmChannelConfig = 0;
ret.iChannelNames = 0;
TU_LOG2(" Get terminal connector\r\n");
return tud_audio_buffer_and_schedule_control_xfer(rhport, p_request, (void*) &ret, sizeof(ret));
}
break;
// Unknown/Unsupported control selector
default:
TU_BREAKPOINT();
return false;
}
}
// Output terminal (Speaker output)
if (entityID == AUDIO_CTRL_ID_SPK_OUTPUT)
{
switch ( ctrlSel )
{
case AUDIO_TE_CTRL_CONNECTOR:
{
// The terminal connector control only has a get request with only the CUR attribute.
audio_desc_channel_cluster_t ret;
// Those are dummy values for now
ret.bNrChannels = 1;
ret.bmChannelConfig = 0;
ret.iChannelNames = 0;
TU_LOG2(" Get terminal connector\r\n");
return tud_audio_buffer_and_schedule_control_xfer(rhport, p_request, (void*) &ret, sizeof(ret));
}
break;
// Unknown/Unsupported control selector
default:
TU_BREAKPOINT();
return false;
}
}
if (entityID == AUDIO_CTRL_ID_SPK_FUNIT)
{
switch ( ctrlSel )
{
case AUDIO_FU_CTRL_MUTE:
// Audio control mute cur parameter block consists of only one byte - we thus can send it right away
// There does not exist a range parameter block for microphoneMute
TU_LOG2(" Get Mute of channel: %u\r\n", channelNum);
return tud_control_xfer(rhport, p_request, &speakerMute[channelNum], 1);
case AUDIO_FU_CTRL_VOLUME:
switch ( p_request->bRequest )
{
case AUDIO_CS_REQ_CUR:
TU_LOG2(" Get Volume of channel: %u\r\n", channelNum);
return tud_control_xfer(rhport, p_request, &speakerLogVolume[channelNum], sizeof(speakerLogVolume[channelNum]));
case AUDIO_CS_REQ_RANGE:
TU_LOG2(" Get Volume range of channel: %u\r\n", channelNum);
/* The Volume Control is one of the building blocks of a Feature Unit. A Volume Control must support the
CUR and RANGE(MIN, MAX, RES) attributes. The settings for the CUR, MIN, and MAX attributes can
range from +127.9961 dB (0x7FFF) down to -127.9961 dB (0x8001) in steps of 1/256 dB or 0.00390625
dB (0x0001). The settings for the RES attribute can only have positive values and range from 1/256 dB
(0x0001) to +127.9961 dB (0x7FFF).
In addition, code 0x8000, representing silence (i.e., -∞ dB), must always be implemented. However, it
must never be reported as the MIN attribute value. */
// Copy values - only for testing - better is version below
audio_control_range_2_n_t(1) ret;
/* From 1 (0dB) down to 1/65536 (-96dB) */
ret.wNumSubRanges = 1;
ret.subrange[0].bMin = -96 * 256;
ret.subrange[0].bMax = 0;
ret.subrange[0].bRes = 1;
return tud_audio_buffer_and_schedule_control_xfer(rhport, p_request, (void*) &ret, sizeof(ret));
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
break;
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
}
// Feature unit
if (entityID == AUDIO_CTRL_ID_MIC_FUNIT)
{
switch ( ctrlSel )
{
case AUDIO_FU_CTRL_MUTE:
// Audio control microphoneMute cur parameter block consists of only one byte - we thus can send it right away
// There does not exist a range parameter block for microphoneMute
TU_LOG2(" Get Mute of channel: %u\r\n", channelNum);
return tud_control_xfer(rhport, p_request, &microphoneMute[channelNum], 1);
case AUDIO_FU_CTRL_VOLUME:
switch ( p_request->bRequest )
{
case AUDIO_CS_REQ_CUR:
TU_LOG2(" Get Volume of channel: %u\r\n", channelNum);
return tud_control_xfer(rhport, p_request, &microphoneLogVolume[channelNum], sizeof(microphoneLogVolume[channelNum]));
case AUDIO_CS_REQ_RANGE:
TU_LOG2(" Get Volume range of channel: %u\r\n", channelNum);
/* The Volume Control is one of the building blocks of a Feature Unit. A Volume Control must support the
CUR and RANGE(MIN, MAX, RES) attributes. The settings for the CUR, MIN, and MAX attributes can
range from +127.9961 dB (0x7FFF) down to -127.9961 dB (0x8001) in steps of 1/256 dB or 0.00390625
dB (0x0001). The settings for the RES attribute can only have positive values and range from 1/256 dB
(0x0001) to +127.9961 dB (0x7FFF).
In addition, code 0x8000, representing silence (i.e., -∞ dB), must always be implemented. However, it
must never be reported as the MIN attribute value. */
// Copy values - only for testing - better is version below
audio_control_range_2_n_t(1) ret;
/* From 1 (0dB) down to 1/65536 (-96dB) */
ret.wNumSubRanges = 1;
ret.subrange[0].bMin = -96 * 256;
ret.subrange[0].bMax = 0;
ret.subrange[0].bRes = 1;
return tud_audio_buffer_and_schedule_control_xfer(rhport, p_request, (void*) &ret, sizeof(ret));
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
break;
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
}
// Clock Source unit
if ( entityID == AUDIO_CTRL_ID_MIC_CLOCK )
{
switch ( ctrlSel )
{
case AUDIO_CS_CTRL_SAM_FREQ:
// channelNum is always zero in this case
switch ( p_request->bRequest )
{
case AUDIO_CS_REQ_CUR:
TU_LOG2(" Get Mic. Sample Freq.\r\n");
return tud_control_xfer(rhport, p_request, &microphoneSampleFreq, sizeof(microphoneSampleFreq));
case AUDIO_CS_REQ_RANGE:
TU_LOG2(" Get Mic. Sample Freq. range\r\n");
return tud_control_xfer(rhport, p_request, &sampleFreqRng, sizeof(sampleFreqRng));
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
break;
case AUDIO_CS_CTRL_CLK_VALID:
// Only cur attribute exists for this request
TU_LOG2(" Get Mic Sample Freq. valid\r\n");
uint8_t clkValid = 1;
return tud_control_xfer(rhport, p_request, &clkValid, sizeof(clkValid));
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
}
// Clock Source unit
if ( entityID == AUDIO_CTRL_ID_SPK_CLOCK )
{
switch ( ctrlSel )
{
case AUDIO_CS_CTRL_SAM_FREQ:
// channelNum is always zero in this case
switch ( p_request->bRequest )
{
case AUDIO_CS_REQ_CUR:
TU_LOG2(" Get Spk. Sample Freq.\r\n");
return tud_control_xfer(rhport, p_request, &speakerSampleFreq, sizeof(speakerSampleFreq));
case AUDIO_CS_REQ_RANGE:
TU_LOG2(" Get Spk. Sample Freq. range\r\n");
return tud_control_xfer(rhport, p_request, &sampleFreqRng, sizeof(sampleFreqRng));
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
break;
case AUDIO_CS_CTRL_CLK_VALID:
// Only cur attribute exists for this request
TU_LOG2(" Get Spk. Sample Freq. valid\r\n");
uint8_t clkValid = 1;
return tud_control_xfer(rhport, p_request, &clkValid, sizeof(clkValid));
// Unknown/Unsupported control
default:
TU_BREAKPOINT();
return false;
}
}
TU_LOG2(" Unsupported entity: %d\r\n", entityID);
return false; // Yet not implemented
}
bool tud_audio_tx_done_pre_load_cb(uint8_t rhport, uint8_t itf, uint8_t ep_in, uint8_t cur_alt_setting) {
(void) rhport;
(void) itf;
(void) ep_in;
(void) cur_alt_setting;
if (microphoneState == STATE_START) {
/* Start ADC sampling as soon as device stacks starts loading data (will be a ZLP for first frame) */
NVIC_EnableIRQ(ADC1_2_IRQn);
microphoneState = STATE_RUN;
}
return true;
}
bool tud_audio_rx_done_post_read_cb(uint8_t rhport, uint16_t n_bytes_received, uint8_t func_id, uint8_t ep_out, uint8_t cur_alt_setting)
{
/* Get number of total bytes available in FIFO */
uint16_t count = tud_audio_available();
/* Calculate min/max/average statistics of buffer fill level */
if ( (count - n_bytes_received) < speakerBufferLvlMin) speakerBufferLvlMin = count - n_bytes_received;
if ( count > speakerBufferLvlMax) speakerBufferLvlMax = count;
speakerBufferLvlAvg = ((uint64_t) speakerBufferLvlAvg * (65536 - SPEAKER_BUFFERLVL_AVG) + ((uint64_t) count << 16) * SPEAKER_BUFFERLVL_AVG) / 65536.0;
if (speakerState == STATE_START) {
if (count >= SPEAKER_BUFFERLVL_TARGET) {
/* Wait until whe are at buffer target fill level, then start DAC output */
speakerState = STATE_RUN;
NVIC_EnableIRQ(TIM6_DAC1_IRQn);
}
/* Initialize/override min/max/avg during startup buffering */
speakerBufferLvlAvg = count;
speakerBufferLvlMin = count;
speakerBufferLvlMax = count;
}
return true;
}
bool tud_audio_set_itf_cb(uint8_t rhport, tusb_control_request_t const * p_request)
{
(void) rhport;
(void) p_request;
uint16_t itf = p_request->wIndex;
uint16_t alt = p_request->wValue;
switch(itf) {
case ITF_NUM_AUDIO_STREAMING_IN:
if (alt == 1) {
/* Microphone channel has been activated */
microphoneState = STATE_START;
}
break;
case ITF_NUM_AUDIO_STREAMING_OUT:
if (alt == 1) {
/* Speaker channel has been activated */
speakerState = STATE_START;
}
break;
default:
TU_ASSERT(0, false);
break;
}
return true;
}
bool tud_audio_set_itf_close_EP_cb(uint8_t rhport, tusb_control_request_t const *p_request) {
(void) rhport;
(void) p_request;
uint16_t itf = p_request->wIndex;
switch (itf) {
case ITF_NUM_AUDIO_STREAMING_IN:
/* Microphone channel has been stopped */
NVIC_DisableIRQ(ADC1_2_IRQn);
microphoneState = STATE_OFF;
break;
case ITF_NUM_AUDIO_STREAMING_OUT:
/* Speaker channel has been stopped */
NVIC_DisableIRQ(TIM6_DAC1_IRQn);
speakerState = STATE_OFF;
break;
default:
TU_ASSERT(0, false);
break;
}
return true;
}
void tud_audio_feedback_params_cb(uint8_t func_id, uint8_t alt_itf, audio_feedback_params_t* feedback_param)
{
/* Configure parameters for feedback endpoint */
feedback_param->frequency.mclk_freq = USB_SOF_TIMER_HZ;
feedback_param->sample_freq = speakerSampleFreqCfg;
feedback_param->method = AUDIO_FEEDBACK_METHOD_FREQUENCY_FIXED;
}
TU_ATTR_FAST_FUNC void tud_audio_feedback_interval_isr(uint8_t func_id, uint32_t frame_number, uint8_t interval_shift)
{
static uint32_t prev_cycles = 0;
uint32_t this_cycles = USB_SOF_TIMER_CNT;
uint32_t feedback;
/* Calculate number of master clock cycles between now and last call */
uint32_t cycles = (uint32_t) (((uint64_t) this_cycles - prev_cycles) & 0xFFFFFFFFUL);
TU_ASSERT(cycles != 0, /**/);
/* Prepare for next time */
prev_cycles = this_cycles;
/* Calculate the feedback value, taken from tinyusb stack */
uint64_t fb64 = (((uint64_t) cycles) * speakerSampleFreqCfg) << 16;
feedback = (uint32_t) (fb64 / USB_SOF_TIMER_HZ);
/* Couple the buffer level bias to the feedback value to avoid buffer drift */
if (speakerState == STATE_RUN) {
int32_t bias = (int32_t) speakerBufferLvlAvg - ((int32_t) SPEAKER_BUFFERLVL_TARGET << 16); /* 16.16 format same as feedback */
feedback -= ((int64_t) bias * SPEAKER_BUFLVL_FB_COUPLING) / 65536;
}
/* The size of isochronous packets created by the device must be within the limits specified in FMT-2.0 section 2.3.1.1.
* This means that the deviation of actual packet size from nominal size must not exceed +/- one audio slot
* (audio slot = channel count samples). */
uint32_t sampleFreq = speakerSampleFreq;
uint32_t min_value = (sampleFreq/1000 - 1) << 16; /* 1000 for full-speed USB */
uint32_t max_value = (sampleFreq/1000 + 1) << 16;
/* Limit */
if ( feedback > max_value ) feedback = max_value;
if ( feedback < min_value ) feedback = min_value;
/* Send to host */
tud_audio_n_fb_set(func_id, feedback);
/* Handle min/max/avg statistics */
if (feedback < speakerFeedbackMin) speakerFeedbackMin = feedback;
if (feedback > speakerFeedbackMax) speakerFeedbackMax = feedback;
speakerFeedbackAvg = (speakerFeedbackAvg * (65536 - SPEAKER_FEEDBACK_AVG) + ((uint64_t) feedback << 16) * SPEAKER_FEEDBACK_AVG) / 65536.0;
if (speakerState == STATE_START) {
/* Initialize/overwrite min/max/avg during start */
speakerFeedbackAvg = (uint64_t) feedback << 16;
speakerFeedbackMin = feedback;
speakerFeedbackMax = feedback;
}
}
void ADC1_2_IRQHandler (void)
{
if (ADC2->ISR & ADC_ISR_EOS) {
ADC2->ISR = ADC_ISR_EOS;
/* Get ADC sample */
int16_t sample = ((int32_t) ADC2->DR - 32768) & 0xFFFFU;
/* Get volume */
uint16_t volume = !microphoneMute[1] ? microphoneLinVolume[1] : 0;
/* Scale with 16-bit unsigned volume and round */
sample = (int16_t) (((int32_t) sample * volume + (sample > 0 ? 32768 : -32768)) / 65536);
/* Store in FIFO */
tud_audio_write (&sample, sizeof(sample));
}
}
void TIM6_DAC_IRQHandler(void)
{
if (TIM6->SR & TIM_SR_UIF) {
TIM6->SR = (uint32_t) ~TIM_SR_UIF;
int16_t sample = 0x0000;
/* Read from FIFO, leave sample at 0 if fifo empty */
tud_audio_read(&sample, sizeof(sample));
/* Get volume */
uint16_t volume = !speakerMute[1] ? speakerLinVolume[1] : 0;
/* Scale with 16-bit unsigned volume and round */
sample = (int16_t) (((int32_t) sample * volume + (sample > 0 ? 32768 : -32768)) / 65536);
/* Load DAC holding register with sample */
DAC1->DHR12L1 = ((int32_t) sample + 32768) & 0xFFFFU;
}
}
static void GPIO_Init(void)
{
__HAL_RCC_GPIOB_CLK_ENABLE();
GPIO_InitTypeDef ADCInGpio;
ADCInGpio.Pin = GPIO_PIN_2;
ADCInGpio.Mode = GPIO_MODE_ANALOG;
ADCInGpio.Pull = GPIO_NOPULL;
ADCInGpio.Speed = GPIO_SPEED_FREQ_LOW;
ADCInGpio.Alternate = 0;
HAL_GPIO_Init(GPIOB, &ADCInGpio);
GPIO_InitTypeDef SamplerateGpio;
SamplerateGpio.Pin = GPIO_PIN_0;
SamplerateGpio.Mode = GPIO_MODE_AF_PP;
SamplerateGpio.Pull = GPIO_NOPULL;
SamplerateGpio.Speed = GPIO_SPEED_FREQ_HIGH;
SamplerateGpio.Alternate = GPIO_AF2_TIM3;
HAL_GPIO_Init(GPIOB, &SamplerateGpio);
GPIO_InitTypeDef DACOutGpio;
DACOutGpio.Pin = GPIO_PIN_4;
DACOutGpio.Mode = GPIO_MODE_ANALOG;
DACOutGpio.Pull = GPIO_NOPULL;
DACOutGpio.Speed = GPIO_SPEED_FREQ_LOW;
DACOutGpio.Alternate = 0;
HAL_GPIO_Init(GPIOA, &DACOutGpio);
}
static void Timer_ADC_Init(void)
{
/* Calculate clock rate divider for requested sample rate with rounding */
uint32_t timerFreq = (HAL_RCC_GetHCLKFreq() == HAL_RCC_GetPCLK1Freq()) ? HAL_RCC_GetPCLK1Freq() : 2 * HAL_RCC_GetPCLK1Freq();
uint32_t rateDivider = (timerFreq + microphoneSampleFreq / 2) / microphoneSampleFreq;
/* Store actually realized samplerate */
microphoneSampleFreqCfg = timerFreq / rateDivider;
/* Enable clock and (re-) initialize timer */
__HAL_RCC_TIM3_CLK_ENABLE();
/* TIM3_TRGO triggers ADC2 */
TIM3->CR1 &= ~TIM_CR1_CEN;
TIM3->CR1 = TIM_CLOCKDIVISION_DIV1 | TIM_COUNTERMODE_UP | TIM_AUTORELOAD_PRELOAD_ENABLE;
TIM3->CR2 = TIM_TRGO_UPDATE;
TIM3->PSC = 0;
TIM3->ARR = rateDivider - 1;
TIM3->EGR = TIM_EGR_UG;
#if 1 /* Output sample rate on compare channel 3 */
TIM3->CCMR2 = TIM_OCMODE_PWM1 | TIM_CCMR2_OC3PE;
TIM3->CCER = (0 << TIM_CCER_CC3P_Pos) | TIM_CCER_CC3E;
TIM3->CCR3 = rateDivider/2 - 1;
#endif
TIM3->CR1 |= TIM_CR1_CEN;
}
static void Timer_DAC_Init(void)
{
/* Calculate clock rate divider for requested sample rate with rounding */
uint32_t timerFreq = (HAL_RCC_GetHCLKFreq() == HAL_RCC_GetPCLK1Freq()) ? HAL_RCC_GetPCLK1Freq() : 2 * HAL_RCC_GetPCLK1Freq();
uint32_t rateDivider = (timerFreq + speakerSampleFreq / 2) / speakerSampleFreq;
/* Store actually realized samplerate for feedback algorithm to use */
speakerSampleFreqCfg = timerFreq / rateDivider;
/* Enable clock and (re-) initialize timer */
__HAL_RCC_TIM6_CLK_ENABLE();
/* TIM6_TRGO triggers DAC */
TIM6->CR1 &= ~TIM_CR1_CEN;
TIM6->CR1 = TIM_CLOCKDIVISION_DIV1 | TIM_COUNTERMODE_UP | TIM_AUTORELOAD_PRELOAD_ENABLE;
TIM6->CR2 = TIM_TRGO_UPDATE;
TIM6->PSC = 0;
TIM6->ARR = rateDivider - 1;
TIM6->EGR = TIM_EGR_UG;
TIM6->DIER = TIM_DIER_UIE;
TIM6->CR1 |= TIM_CR1_CEN;
NVIC_SetPriority(TIM6_DAC1_IRQn, AIOC_IRQ_PRIO_AUDIO);
}
static void ADC_Init(void)
{
__HAL_RCC_ADC2_CLK_ENABLE();
ADC2->CR = 0x00 << ADC_CR_ADVREGEN_Pos;
ADC2->CR = 0x01 << ADC_CR_ADVREGEN_Pos;
for (uint32_t i=0; i<200; i++) {
asm volatile ("nop");
}
/* Select AHB clock */
ADC12_COMMON->CCR = (0x1 << ADC12_CCR_CKMODE_Pos) | (0x00 << ADC12_CCR_MULTI_Pos);
ADC2->CR |= ADC_CR_ADCAL;
while (ADC2->CR & ADC_CR_ADCAL)
;
ADC2->CR |= ADC_CR_ADEN;
/* Wait for ADC to be ready */
while (!(ADC2->ISR & ADC_ISR_ADRDY))
;
/* External Trigger on TIM3_TRGO, left aligned data with 12 bit resolution */
ADC2->CFGR = (0x01 << ADC_CFGR_EXTEN_Pos) | (0x04 << ADC_CFGR_EXTSEL_Pos) | (ADC_CFGR_ALIGN) | (0x00 << ADC_CFGR_RES_Pos);
/* Maximum sample time of 601.5 cycles for channel 12. */
ADC2->SMPR2 = 0x7 << ADC_SMPR2_SMP12_Pos;
/* Sample only channel 12 in a regular sequence */
ADC2->SQR1 = (12 << ADC_SQR1_SQ1_Pos) | (0 << ADC_SQR1_L_Pos);
/* Enable Interrupt Request */
ADC2->IER = ADC_IER_EOSIE;
/* Start ADC */
ADC2->CR |= ADC_CR_ADSTART;
NVIC_SetPriority(ADC1_2_IRQn, AIOC_IRQ_PRIO_AUDIO);
}
static void DAC_Init(void)
{
__HAL_RCC_DAC1_CLK_ENABLE();
/* Select TIM6 TRGO as trigger and enable DAC */
DAC->CR = (0x0 << DAC_CR_TSEL1_Pos) | DAC_CR_TEN1 | DAC_CR_EN1;
}
void USB_AudioInit(void)
{
GPIO_Init();
Timer_ADC_Init();
Timer_DAC_Init();
ADC_Init();
DAC_Init();
}
void USB_AudioGetSpeakerFeedbackStats(usb_audio_fbstats_t * status)
{
*status = (usb_audio_fbstats_t) {
.feedbackMin = speakerFeedbackMin,
.feedbackMax = speakerFeedbackMax,
.feedbackAvg = (uint32_t) (speakerFeedbackAvg >> 16)
};
}
void USB_AudioGetSpeakerBufferStats(usb_audio_bufstats_t * status)
{
*status = (usb_audio_bufstats_t) {
.bufLevelMin = speakerBufferLvlMin,
.bufLevelMax = speakerBufferLvlMax,
.bufLevelAvg = (uint16_t) (speakerBufferLvlAvg >> 16)
};
}
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