mirror of
https://github.com/hathach/tinyusb.git
synced 2025-01-31 05:52:55 +08:00
Merge pull request #2259 from HiFiPhile/uac_interl
Fix UAC encoding, update example.
This commit is contained in:
commit
118823c254
@ -28,6 +28,11 @@ target_include_directories(${PROJECT} PUBLIC
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${CMAKE_CURRENT_SOURCE_DIR}/src
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)
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# Add libm for GCC
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if (CMAKE_C_COMPILER_ID STREQUAL "GNU")
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target_link_libraries(${PROJECT} PUBLIC m)
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endif()
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# Configure compilation flags and libraries for the example without RTOS.
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# See the corresponding function in hw/bsp/FAMILY/family.cmake for details.
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family_configure_device_example(${PROJECT} noos)
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@ -5,7 +5,10 @@ INC += \
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$(TOP)/hw \
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# Example source
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EXAMPLE_SOURCE += $(wildcard src/*.c)
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EXAMPLE_SOURCE += \
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src/main.c \
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src/usb_descriptors.c \
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SRC_C += $(addprefix $(CURRENT_PATH)/, $(EXAMPLE_SOURCE))
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include ../../rules.mk
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@ -1,3 +1,4 @@
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mcu:SAMD11
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mcu:SAME5X
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mcu:SAMG
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family:broadcom_64bit
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@ -34,17 +34,16 @@
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#include <stdlib.h>
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#include <stdio.h>
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#include <string.h>
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#include <math.h>
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#include "bsp/board_api.h"
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#include "tusb.h"
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#include "tusb_config.h"
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//--------------------------------------------------------------------+
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// MACRO CONSTANT TYPEDEF PROTYPES
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//--------------------------------------------------------------------+
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#ifndef AUDIO_SAMPLE_RATE
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#define AUDIO_SAMPLE_RATE 48000
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#endif
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#define AUDIO_SAMPLE_RATE CFG_TUD_AUDIO_FUNC_1_SAMPLE_RATE
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/* Blink pattern
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* - 250 ms : device not mounted
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@ -70,7 +69,7 @@ uint8_t clkValid;
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audio_control_range_2_n_t(1) volumeRng[CFG_TUD_AUDIO_FUNC_1_N_CHANNELS_TX+1]; // Volume range state
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audio_control_range_4_n_t(1) sampleFreqRng; // Sample frequency range state
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// Audio test data
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// Audio test data, each buffer contains 2 channels, buffer[0] for CH0-1, buffer[1] for CH1-2
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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
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void led_blinking_task(void);
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@ -97,6 +96,27 @@ int main(void)
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sampleFreqRng.subrange[0].bMax = AUDIO_SAMPLE_RATE;
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sampleFreqRng.subrange[0].bRes = 0;
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// Generate dummy data
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uint16_t * p_buff = i2s_dummy_buffer[0];
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uint16_t dataVal = 1;
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for (uint16_t cnt = 0; cnt < AUDIO_SAMPLE_RATE/1000; cnt++)
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{
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// CH0 saw wave
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*p_buff++ = dataVal;
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// CH1 inverted saw wave
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*p_buff++ = 60 + AUDIO_SAMPLE_RATE/1000 - dataVal;
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dataVal++;
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}
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p_buff = i2s_dummy_buffer[1];
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for (uint16_t cnt = 0; cnt < AUDIO_SAMPLE_RATE/1000; cnt++)
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{
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// CH3 square wave
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*p_buff++ = cnt < (AUDIO_SAMPLE_RATE/1000/2) ? 120:170;
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// CH4 sinus wave
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float t = 2*3.1415f * cnt / (AUDIO_SAMPLE_RATE/1000);
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*p_buff++ = (uint16_t)(sinf(t) * 25) + 200;
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}
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while (1)
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{
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tud_task(); // tinyusb device task
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@ -400,7 +420,17 @@ bool tud_audio_tx_done_pre_load_cb(uint8_t rhport, uint8_t itf, uint8_t ep_in, u
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(void) ep_in;
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(void) cur_alt_setting;
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for (uint8_t cnt=0; cnt < CFG_TUD_AUDIO_FUNC_1_N_TX_SUPP_SW_FIFO; cnt++)
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// In read world application data flow is driven by I2S clock,
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// both tud_audio_tx_done_pre_load_cb() & tud_audio_tx_done_post_load_cb() are hardly used.
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// For example in your I2S receive callback:
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// void I2S_Rx_Callback(int channel, const void* data, uint16_t samples)
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// {
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// tud_audio_write_support_ff(channel, data, samples * N_BYTES_PER_SAMPLE * N_CHANNEL_PER_FIFO);
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// }
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// Write I2S buffer into FIFO
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for (uint8_t cnt=0; cnt < 2; cnt++)
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{
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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);
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}
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@ -416,22 +446,6 @@ bool tud_audio_tx_done_post_load_cb(uint8_t rhport, uint16_t n_bytes_copied, uin
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(void) ep_in;
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(void) cur_alt_setting;
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uint16_t dataVal;
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// Generate dummy data
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for (uint16_t cnt = 0; cnt < CFG_TUD_AUDIO_FUNC_1_N_TX_SUPP_SW_FIFO; cnt++)
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{
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uint16_t * p_buff = i2s_dummy_buffer[cnt]; // 2 bytes per sample
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dataVal = 1;
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for (uint16_t cnt2 = 0; cnt2 < AUDIO_SAMPLE_RATE/1000; cnt2++)
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{
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for (uint8_t cnt3 = 0; cnt3 < CFG_TUD_AUDIO_FUNC_1_CHANNEL_PER_FIFO_TX; cnt3++)
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{
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*p_buff++ = dataVal;
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}
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dataVal++;
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}
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}
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return true;
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}
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@ -10,7 +10,7 @@ if __name__ == '__main__':
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# print(sd.query_devices())
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fs = 48000 # Sample rate
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duration = 100e-3 # Duration of recording
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duration = 20e-3 # Duration of recording
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if platform.system() == 'Windows':
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# WDM-KS is needed since there are more than one MicNode device APIs (at least in Windows)
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@ -25,9 +25,14 @@ if __name__ == '__main__':
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sd.wait() # Wait until recording is finished
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print('Done!')
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time = np.arange(0, duration, 1 / fs) # time vector
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# strip starting zero
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myrecording = myrecording[100:]
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time = time[100:]
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plt.plot(time, myrecording)
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plt.xlabel('Time [s]')
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plt.ylabel('Amplitude')
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plt.title('MicNode 4 Channel')
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plt.legend(['CH-1', 'CH-2', 'CH-3','CH-4'])
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plt.show()
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@ -103,6 +103,7 @@ extern "C" {
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//--------------------------------------------------------------------
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// Have a look into audio_device.h for all configurations
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#define CFG_TUD_AUDIO_FUNC_1_SAMPLE_RATE 48000
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#define CFG_TUD_AUDIO_FUNC_1_DESC_LEN TUD_AUDIO_MIC_FOUR_CH_DESC_LEN
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@ -112,7 +113,7 @@ extern "C" {
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#define CFG_TUD_AUDIO_ENABLE_EP_IN 1
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#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
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#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
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#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
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#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)
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#define CFG_TUD_AUDIO_FUNC_1_EP_IN_SZ_MAX CFG_TUD_AUDIO_EP_SZ_IN
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#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
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// Decoding according to 2.3.1.5 Audio Streams
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// Helper function
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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)
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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)
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{
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// Due to one FIFO contains 2 channels, data always aligned to (nBytesPerSample * 2)
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uint16_t * dst16 = dst;
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uint16_t * src16 = src;
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const uint16_t * dst_end16 = dst_end;
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uint32_t * dst32 = dst;
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uint32_t * src32 = src;
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const uint32_t * dst_end32 = dst_end;
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// This function is an optimized version of
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// while((uint8_t *)dst < dst_end)
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// {
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// memcpy(dst, src, nBytesToCopy);
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// dst = (uint8_t *)dst + nBytesToCopy;
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// src += nBytesToCopy * n_ff_used;
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// }
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// Optimize for fast half word copies
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typedef struct{
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uint16_t val;
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} __attribute((__packed__)) unaligned_uint16_t;
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// Optimize for fast word copies
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typedef struct{
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uint32_t val;
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} __attribute((__packed__)) unaligned_uint32_t;
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switch (nBytesToCopy)
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if (nBytesPerSample == 1)
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{
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case 1:
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while((uint8_t *)dst < dst_end)
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while(dst16 < dst_end16)
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{
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*(uint8_t *)dst++ = *src;
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src += n_ff_used;
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*dst16++ = *src16++;
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src16 += n_ff_used - 1;
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}
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break;
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case 2:
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while((uint8_t *)dst < dst_end)
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return src16;
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}
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else if (nBytesPerSample == 2)
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{
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*(unaligned_uint16_t*)dst = *(unaligned_uint16_t*)src;
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dst += 2;
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src += 2 * n_ff_used;
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}
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break;
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case 3:
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while((uint8_t *)dst < dst_end)
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while(dst32 < dst_end32)
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{
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// memcpy(dst, src, 3);
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// dst = (uint8_t *)dst + 3;
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// src += 3 * n_ff_used;
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// TODO: Is there a faster way to copy 3 bytes?
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*(uint8_t *)dst++ = *src++;
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*(uint8_t *)dst++ = *src++;
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*(uint8_t *)dst++ = *src++;
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src += 3 * (n_ff_used - 1);
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*dst32++ = *src32++;
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src32 += n_ff_used - 1;
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}
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break;
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case 4:
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while((uint8_t *)dst < dst_end)
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return src32;
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}
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else if (nBytesPerSample == 3)
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{
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*(unaligned_uint32_t*)dst = *(unaligned_uint32_t*)src;
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dst += 4;
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src += 4 * n_ff_used;
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while(dst16 < dst_end16)
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{
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*dst16++ = *src16++;
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*dst16++ = *src16++;
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*dst16++ = *src16++;
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src16 += 3 * (n_ff_used - 1);
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}
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break;
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return src16;
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}
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else // nBytesPerSample == 4
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{
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while(dst32 < dst_end32)
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{
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*dst32++ = *src32++;
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*dst32++ = *src32++;
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src32 += 2 * (n_ff_used - 1);
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}
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return src32;
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}
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return src;
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}
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static bool audiod_decode_type_I_pcm(uint8_t rhport, audiod_function_t* audio, uint16_t n_bytes_received)
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@ -944,64 +926,55 @@ range [-1, +1)
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* */
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// Helper function
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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)
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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)
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{
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// Optimize for fast half word copies
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typedef struct{
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uint16_t val;
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} __attribute((__packed__)) unaligned_uint16_t;
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// Due to one FIFO contains 2 channels, data always aligned to (nBytesPerSample * 2)
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uint16_t * dst16 = dst;
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uint16_t * src16 = src;
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const uint16_t * src_end16 = src_end;
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uint32_t * dst32 = dst;
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uint32_t * src32 = src;
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const uint32_t * src_end32 = src_end;
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// Optimize for fast word copies
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typedef struct{
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uint32_t val;
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} __attribute((__packed__)) unaligned_uint32_t;
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switch (nBytesToCopy)
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if (nBytesPerSample == 1)
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{
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case 1:
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while(src < src_end)
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while(src16 < src_end16)
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{
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*dst = *src++;
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dst += n_ff_used;
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*dst16++ = *src16++;
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dst16 += n_ff_used - 1;
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}
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break;
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case 2:
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while(src < src_end)
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return dst16;
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}
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else if (nBytesPerSample == 2)
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{
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*(unaligned_uint16_t*)dst = *(unaligned_uint16_t*)src;
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src += 2;
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dst += 2 * n_ff_used;
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}
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break;
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case 3:
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while(src < src_end)
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while(src32 < src_end32)
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{
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// memcpy(dst, src, 3);
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// src = (uint8_t *)src + 3;
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// dst += 3 * n_ff_used;
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// TODO: Is there a faster way to copy 3 bytes?
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*dst++ = *src++;
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*dst++ = *src++;
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*dst++ = *src++;
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dst += 3 * (n_ff_used - 1);
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*dst32++ = *src32++;
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dst32 += n_ff_used - 1;
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}
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break;
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case 4:
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while(src < src_end)
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return dst32;
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}
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else if (nBytesPerSample == 3)
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{
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*(unaligned_uint32_t*)dst = *(unaligned_uint32_t*)src;
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src += 4;
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dst += 4 * n_ff_used;
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while(src16 < src_end16)
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{
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*dst16++ = *src16++;
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*dst16++ = *src16++;
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*dst16++ = *src16++;
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dst16 += 3 * (n_ff_used - 1);
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}
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break;
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return dst16;
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}
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else // nBytesPerSample == 4
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{
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while(src32 < src_end32)
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{
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*dst32++ = *src32++;
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*dst32++ = *src32++;
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dst32 += 2 * (n_ff_used - 1);
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}
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return dst32;
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}
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return dst;
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}
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static uint16_t audiod_encode_type_I_pcm(uint8_t rhport, audiod_function_t* audio)
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