Merge pull request #2259 from HiFiPhile/uac_interl

Fix UAC encoding, update example.
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PanRe 2023-10-11 19:18:35 +02:00 committed by GitHub
commit 118823c254
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7 changed files with 144 additions and 142 deletions

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@ -28,6 +28,11 @@ target_include_directories(${PROJECT} PUBLIC
${CMAKE_CURRENT_SOURCE_DIR}/src
)
# Add libm for GCC
if (CMAKE_C_COMPILER_ID STREQUAL "GNU")
target_link_libraries(${PROJECT} PUBLIC m)
endif()
# Configure compilation flags and libraries for the example without RTOS.
# See the corresponding function in hw/bsp/FAMILY/family.cmake for details.
family_configure_device_example(${PROJECT} noos)

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@ -5,7 +5,10 @@ INC += \
$(TOP)/hw \
# Example source
EXAMPLE_SOURCE += $(wildcard src/*.c)
EXAMPLE_SOURCE += \
src/main.c \
src/usb_descriptors.c \
SRC_C += $(addprefix $(CURRENT_PATH)/, $(EXAMPLE_SOURCE))
include ../../rules.mk

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@ -1,3 +1,4 @@
mcu:SAMD11
mcu:SAME5X
mcu:SAMG
family:broadcom_64bit

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@ -34,17 +34,16 @@
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include "bsp/board_api.h"
#include "tusb.h"
#include "tusb_config.h"
//--------------------------------------------------------------------+
// MACRO CONSTANT TYPEDEF PROTYPES
//--------------------------------------------------------------------+
#ifndef AUDIO_SAMPLE_RATE
#define AUDIO_SAMPLE_RATE 48000
#endif
#define AUDIO_SAMPLE_RATE CFG_TUD_AUDIO_FUNC_1_SAMPLE_RATE
/* Blink pattern
* - 250 ms : device not mounted
@ -70,7 +69,7 @@ uint8_t clkValid;
audio_control_range_2_n_t(1) volumeRng[CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_TX+1]; // Volume range state
audio_control_range_4_n_t(1) sampleFreqRng; // Sample frequency range state
// Audio test data
// Audio test data, each buffer contains 2 channels, buffer[0] for CH0-1, buffer[1] for CH1-2
uint16_t i2s_dummy_buffer[CFG_TUD_AUDIO_FUNC_1_N_TX_SUPP_SW_FIFO][CFG_TUD_AUDIO_FUNC_1_TX_SUPP_SW_FIFO_SZ/2]; // Ensure half word aligned
void led_blinking_task(void);
@ -97,6 +96,27 @@ int main(void)
sampleFreqRng.subrange[0].bMax = AUDIO_SAMPLE_RATE;
sampleFreqRng.subrange[0].bRes = 0;
// Generate dummy data
uint16_t * p_buff = i2s_dummy_buffer[0];
uint16_t dataVal = 1;
for (uint16_t cnt = 0; cnt < AUDIO_SAMPLE_RATE/1000; cnt++)
{
// CH0 saw wave
*p_buff++ = dataVal;
// CH1 inverted saw wave
*p_buff++ = 60 + AUDIO_SAMPLE_RATE/1000 - dataVal;
dataVal++;
}
p_buff = i2s_dummy_buffer[1];
for (uint16_t cnt = 0; cnt < AUDIO_SAMPLE_RATE/1000; cnt++)
{
// CH3 square wave
*p_buff++ = cnt < (AUDIO_SAMPLE_RATE/1000/2) ? 120:170;
// CH4 sinus wave
float t = 2*3.1415f * cnt / (AUDIO_SAMPLE_RATE/1000);
*p_buff++ = (uint16_t)(sinf(t) * 25) + 200;
}
while (1)
{
tud_task(); // tinyusb device task
@ -400,7 +420,17 @@ bool tud_audio_tx_done_pre_load_cb(uint8_t rhport, uint8_t itf, uint8_t ep_in, u
(void) ep_in;
(void) cur_alt_setting;
for (uint8_t cnt=0; cnt < CFG_TUD_AUDIO_FUNC_1_N_TX_SUPP_SW_FIFO; cnt++)
// In read world application data flow is driven by I2S clock,
// both tud_audio_tx_done_pre_load_cb() & tud_audio_tx_done_post_load_cb() are hardly used.
// For example in your I2S receive callback:
// void I2S_Rx_Callback(int channel, const void* data, uint16_t samples)
// {
// tud_audio_write_support_ff(channel, data, samples * N_BYTES_PER_SAMPLE * N_CHANNEL_PER_FIFO);
// }
// Write I2S buffer into FIFO
for (uint8_t cnt=0; cnt < 2; cnt++)
{
tud_audio_write_support_ff(cnt, i2s_dummy_buffer[cnt], AUDIO_SAMPLE_RATE/1000 * CFG_TUD_AUDIO_FUNC_1_N_BYTES_PER_SAMPLE_TX * CFG_TUD_AUDIO_FUNC_1_CHANNEL_PER_FIFO_TX);
}
@ -416,22 +446,6 @@ bool tud_audio_tx_done_post_load_cb(uint8_t rhport, uint16_t n_bytes_copied, uin
(void) ep_in;
(void) cur_alt_setting;
uint16_t dataVal;
// Generate dummy data
for (uint16_t cnt = 0; cnt < CFG_TUD_AUDIO_FUNC_1_N_TX_SUPP_SW_FIFO; cnt++)
{
uint16_t * p_buff = i2s_dummy_buffer[cnt]; // 2 bytes per sample
dataVal = 1;
for (uint16_t cnt2 = 0; cnt2 < AUDIO_SAMPLE_RATE/1000; cnt2++)
{
for (uint8_t cnt3 = 0; cnt3 < CFG_TUD_AUDIO_FUNC_1_CHANNEL_PER_FIFO_TX; cnt3++)
{
*p_buff++ = dataVal;
}
dataVal++;
}
}
return true;
}

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@ -10,7 +10,7 @@ if __name__ == '__main__':
# print(sd.query_devices())
fs = 48000 # Sample rate
duration = 100e-3 # Duration of recording
duration = 20e-3 # Duration of recording
if platform.system() == 'Windows':
# WDM-KS is needed since there are more than one MicNode device APIs (at least in Windows)
@ -25,9 +25,14 @@ if __name__ == '__main__':
sd.wait() # Wait until recording is finished
print('Done!')
time = np.arange(0, duration, 1 / fs) # time vector
# strip starting zero
myrecording = myrecording[100:]
time = time[100:]
plt.plot(time, myrecording)
plt.xlabel('Time [s]')
plt.ylabel('Amplitude')
plt.title('MicNode 4 Channel')
plt.legend(['CH-1', 'CH-2', 'CH-3','CH-4'])
plt.show()

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@ -103,6 +103,7 @@ extern "C" {
//--------------------------------------------------------------------
// Have a look into audio_device.h for all configurations
#define CFG_TUD_AUDIO_FUNC_1_SAMPLE_RATE 48000
#define CFG_TUD_AUDIO_FUNC_1_DESC_LEN TUD_AUDIO_MIC_FOUR_CH_DESC_LEN
@ -112,7 +113,7 @@ extern "C" {
#define CFG_TUD_AUDIO_ENABLE_EP_IN 1
#define CFG_TUD_AUDIO_FUNC_1_N_BYTES_PER_SAMPLE_TX 2 // This value is not required by the driver, it parses this information from the descriptor once the alternate interface is set by the host - we use it for the setup
#define CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_TX 4 // This value is not required by the driver, it parses this information from the descriptor once the alternate interface is set by the host - we use it for the setup
#define CFG_TUD_AUDIO_EP_SZ_IN (48 + 1) * CFG_TUD_AUDIO_FUNC_1_N_BYTES_PER_SAMPLE_TX * CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_TX // 48 Samples (48 kHz) x 2 Bytes/Sample x CFG_TUD_AUDIO_N_CHANNELS_TX Channels - the Windows driver always needs an extra sample per channel of space more, otherwise it complains... found by trial and error
#define CFG_TUD_AUDIO_EP_SZ_IN TUD_AUDIO_EP_SIZE(CFG_TUD_AUDIO_FUNC_1_SAMPLE_RATE, CFG_TUD_AUDIO_FUNC_1_N_BYTES_PER_SAMPLE_TX, CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_TX)
#define CFG_TUD_AUDIO_FUNC_1_EP_IN_SZ_MAX CFG_TUD_AUDIO_EP_SZ_IN
#define CFG_TUD_AUDIO_FUNC_1_EP_IN_SW_BUF_SZ CFG_TUD_AUDIO_EP_SZ_IN

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@ -631,73 +631,55 @@ static bool audiod_rx_done_cb(uint8_t rhport, audiod_function_t* audio, uint16_t
// Decoding according to 2.3.1.5 Audio Streams
// Helper function
static inline uint8_t * audiod_interleaved_copy_bytes_fast_decode(uint16_t const nBytesToCopy, void * dst, uint8_t * dst_end, uint8_t * src, uint8_t const n_ff_used)
static inline void * audiod_interleaved_copy_bytes_fast_decode(uint16_t const nBytesPerSample, void * dst, const void * dst_end, void * src, uint8_t const n_ff_used)
{
// Due to one FIFO contains 2 channels, data always aligned to (nBytesPerSample * 2)
uint16_t * dst16 = dst;
uint16_t * src16 = src;
const uint16_t * dst_end16 = dst_end;
uint32_t * dst32 = dst;
uint32_t * src32 = src;
const uint32_t * dst_end32 = dst_end;
// This function is an optimized version of
// while((uint8_t *)dst < dst_end)
// {
// memcpy(dst, src, nBytesToCopy);
// dst = (uint8_t *)dst + nBytesToCopy;
// src += nBytesToCopy * n_ff_used;
// }
// Optimize for fast half word copies
typedef struct{
uint16_t val;
} __attribute((__packed__)) unaligned_uint16_t;
// Optimize for fast word copies
typedef struct{
uint32_t val;
} __attribute((__packed__)) unaligned_uint32_t;
switch (nBytesToCopy)
if (nBytesPerSample == 1)
{
case 1:
while((uint8_t *)dst < dst_end)
{
*(uint8_t *)dst++ = *src;
src += n_ff_used;
}
break;
case 2:
while((uint8_t *)dst < dst_end)
{
*(unaligned_uint16_t*)dst = *(unaligned_uint16_t*)src;
dst += 2;
src += 2 * n_ff_used;
}
break;
case 3:
while((uint8_t *)dst < dst_end)
{
// memcpy(dst, src, 3);
// dst = (uint8_t *)dst + 3;
// src += 3 * n_ff_used;
// TODO: Is there a faster way to copy 3 bytes?
*(uint8_t *)dst++ = *src++;
*(uint8_t *)dst++ = *src++;
*(uint8_t *)dst++ = *src++;
src += 3 * (n_ff_used - 1);
}
break;
case 4:
while((uint8_t *)dst < dst_end)
{
*(unaligned_uint32_t*)dst = *(unaligned_uint32_t*)src;
dst += 4;
src += 4 * n_ff_used;
}
break;
while(dst16 < dst_end16)
{
*dst16++ = *src16++;
src16 += n_ff_used - 1;
}
return src16;
}
else if (nBytesPerSample == 2)
{
while(dst32 < dst_end32)
{
*dst32++ = *src32++;
src32 += n_ff_used - 1;
}
return src32;
}
else if (nBytesPerSample == 3)
{
while(dst16 < dst_end16)
{
*dst16++ = *src16++;
*dst16++ = *src16++;
*dst16++ = *src16++;
src16 += 3 * (n_ff_used - 1);
}
return src16;
}
else // nBytesPerSample == 4
{
while(dst32 < dst_end32)
{
*dst32++ = *src32++;
*dst32++ = *src32++;
src32 += 2 * (n_ff_used - 1);
}
return src32;
}
return src;
}
static bool audiod_decode_type_I_pcm(uint8_t rhport, audiod_function_t* audio, uint16_t n_bytes_received)
@ -944,64 +926,55 @@ range [-1, +1)
* */
// Helper function
static inline uint8_t * audiod_interleaved_copy_bytes_fast_encode(uint16_t const nBytesToCopy, uint8_t * src, uint8_t * src_end, uint8_t * dst, uint8_t const n_ff_used)
static inline void * audiod_interleaved_copy_bytes_fast_encode(uint16_t const nBytesPerSample, void * src, const void * src_end, void * dst, uint8_t const n_ff_used)
{
// Optimize for fast half word copies
typedef struct{
uint16_t val;
} __attribute((__packed__)) unaligned_uint16_t;
// Due to one FIFO contains 2 channels, data always aligned to (nBytesPerSample * 2)
uint16_t * dst16 = dst;
uint16_t * src16 = src;
const uint16_t * src_end16 = src_end;
uint32_t * dst32 = dst;
uint32_t * src32 = src;
const uint32_t * src_end32 = src_end;
// Optimize for fast word copies
typedef struct{
uint32_t val;
} __attribute((__packed__)) unaligned_uint32_t;
switch (nBytesToCopy)
if (nBytesPerSample == 1)
{
case 1:
while(src < src_end)
{
*dst = *src++;
dst += n_ff_used;
}
break;
case 2:
while(src < src_end)
{
*(unaligned_uint16_t*)dst = *(unaligned_uint16_t*)src;
src += 2;
dst += 2 * n_ff_used;
}
break;
case 3:
while(src < src_end)
{
// memcpy(dst, src, 3);
// src = (uint8_t *)src + 3;
// dst += 3 * n_ff_used;
// TODO: Is there a faster way to copy 3 bytes?
*dst++ = *src++;
*dst++ = *src++;
*dst++ = *src++;
dst += 3 * (n_ff_used - 1);
}
break;
case 4:
while(src < src_end)
{
*(unaligned_uint32_t*)dst = *(unaligned_uint32_t*)src;
src += 4;
dst += 4 * n_ff_used;
}
break;
while(src16 < src_end16)
{
*dst16++ = *src16++;
dst16 += n_ff_used - 1;
}
return dst16;
}
else if (nBytesPerSample == 2)
{
while(src32 < src_end32)
{
*dst32++ = *src32++;
dst32 += n_ff_used - 1;
}
return dst32;
}
else if (nBytesPerSample == 3)
{
while(src16 < src_end16)
{
*dst16++ = *src16++;
*dst16++ = *src16++;
*dst16++ = *src16++;
dst16 += 3 * (n_ff_used - 1);
}
return dst16;
}
else // nBytesPerSample == 4
{
while(src32 < src_end32)
{
*dst32++ = *src32++;
*dst32++ = *src32++;
dst32 += 2 * (n_ff_used - 1);
}
return dst32;
}
return dst;
}
static uint16_t audiod_encode_type_I_pcm(uint8_t rhport, audiod_function_t* audio)