kanshan/xiaozhi-esp32
1
0
Fork 0
mirror of https://github.com/78/xiaozhi-esp32.git synced 2026-01-14 01:07:30 +08:00
Code Issues Actions Packages Projects Releases Wiki Activity

configure GPIO and sample rates

Browse Source
This commit is contained in:
Terrence 2024-09-01 13:24:45 +08:00
parent 490b8668f6
commit 16334ca75f
4 changed files with 103 additions and 30 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

44
main/Application.cc
View File

@ -7,9 +7,6 @@
#include "silk_resampler.h"
#define TAG "application"
#define INPUT_SAMPLE_RATE 16000
#define DECODE_SAMPLE_RATE 24000
#define OUTPUT_SAMPLE_RATE 24000
Application::Application() {
@ -27,10 +24,10 @@ Application::Application() {
}
}
opus_encoder_.Configure(INPUT_SAMPLE_RATE, 1);
opus_decoder_ = opus_decoder_create(DECODE_SAMPLE_RATE, 1, NULL);
if (DECODE_SAMPLE_RATE != OUTPUT_SAMPLE_RATE) {
assert(0 == silk_resampler_init(&resampler_state_, DECODE_SAMPLE_RATE, OUTPUT_SAMPLE_RATE, 1));
opus_encoder_.Configure(CONFIG_AUDIO_INPUT_SAMPLE_RATE, 1);
opus_decoder_ = opus_decoder_create(opus_decode_sample_rate_, 1, NULL);
if (opus_decode_sample_rate_ != CONFIG_AUDIO_OUTPUT_SAMPLE_RATE) {
assert(0 == silk_resampler_init(&resampler_state_, opus_decode_sample_rate_, CONFIG_AUDIO_OUTPUT_SAMPLE_RATE, 1));
}
}
@ -59,7 +56,7 @@ Application::~Application() {
}
void Application::Start() {
audio_device_.Start(INPUT_SAMPLE_RATE, OUTPUT_SAMPLE_RATE);
audio_device_.Start(CONFIG_AUDIO_INPUT_SAMPLE_RATE, CONFIG_AUDIO_OUTPUT_SAMPLE_RATE);
audio_device_.OnStateChanged([this]() {
if (audio_device_.playing()) {
SetChatState(kChatStateSpeaking);
@ -154,7 +151,7 @@ void Application::StartCommunication() {
.total_ch_num = 1,
.mic_num = 1,
.ref_num = 0,
.sample_rate = INPUT_SAMPLE_RATE
.sample_rate = CONFIG_AUDIO_INPUT_SAMPLE_RATE,
},
.debug_init = false,
.debug_hook = {{ AFE_DEBUG_HOOK_MASE_TASK_IN, NULL }, { AFE_DEBUG_HOOK_FETCH_TASK_IN, NULL }},
@ -195,7 +192,7 @@ void Application::StartDetection() {
.total_ch_num = 1,
.mic_num = 1,
.ref_num = 0,
.sample_rate = INPUT_SAMPLE_RATE
.sample_rate = CONFIG_AUDIO_INPUT_SAMPLE_RATE
},
.debug_init = false,
.debug_hook = {{ AFE_DEBUG_HOOK_MASE_TASK_IN, NULL }, { AFE_DEBUG_HOOK_FETCH_TASK_IN, NULL }},
@ -335,11 +332,11 @@ void Application::AudioEncodeTask() {
}
void Application::AudioDecodeTask() {
int frame_size = DECODE_SAMPLE_RATE / 1000 * opus_duration_ms_;
while (true) {
AudioPacket* packet;
xQueueReceive(audio_decode_queue_, &packet, portMAX_DELAY);
int frame_size = opus_decode_sample_rate_ / 1000 * opus_duration_ms_;
packet->pcm.resize(frame_size);
int ret = opus_decode(opus_decoder_, packet->opus.data(), packet->opus.size(), packet->pcm.data(), frame_size, 0);
@ -349,8 +346,8 @@ void Application::AudioDecodeTask() {
continue;
}
if (DECODE_SAMPLE_RATE != OUTPUT_SAMPLE_RATE) {
int target_size = frame_size * OUTPUT_SAMPLE_RATE / DECODE_SAMPLE_RATE;
if (opus_decode_sample_rate_ != CONFIG_AUDIO_OUTPUT_SAMPLE_RATE) {
int target_size = frame_size * CONFIG_AUDIO_OUTPUT_SAMPLE_RATE / opus_decode_sample_rate_;
std::vector<int16_t> resampled(target_size);
assert(0 == silk_resampler(&resampler_state_, resampled.data(), packet->pcm.data(), frame_size));
packet->pcm = std::move(resampled);
@ -360,6 +357,19 @@ void Application::AudioDecodeTask() {
}
}
void Application::SetDecodeSampleRate(int sample_rate) {
if (opus_decode_sample_rate_ == sample_rate) {
return;
}
opus_decoder_destroy(opus_decoder_);
opus_decode_sample_rate_ = sample_rate;
opus_decoder_ = opus_decoder_create(opus_decode_sample_rate_, 1, NULL);
if (opus_decode_sample_rate_ != CONFIG_AUDIO_OUTPUT_SAMPLE_RATE) {
assert(0 == silk_resampler_init(&resampler_state_, opus_decode_sample_rate_, CONFIG_AUDIO_OUTPUT_SAMPLE_RATE, 1));
}
}
void Application::StartWebSocketClient() {
if (ws_client_ != nullptr) {
delete ws_client_;
@ -379,7 +389,7 @@ void Application::StartWebSocketClient() {
message += "\"type\":\"hello\", \"version\":\"1.0\",";
message += "\"wakeup_model\":\"" + std::string(wakenet_model_) + "\",";
message += "\"audio_params\":{";
message += "\"format\":\"opus\", \"sample_rate\":" + std::to_string(INPUT_SAMPLE_RATE) + ", \"channels\":1";
message += "\"format\":\"opus\", \"sample_rate\":" + std::to_string(CONFIG_AUDIO_INPUT_SAMPLE_RATE) + ", \"channels\":1";
message += "}}";
ws_client_->Send(message);
});
@ -403,6 +413,10 @@ void Application::StartWebSocketClient() {
auto state = cJSON_GetObjectItem(root, "state");
if (strcmp(state->valuestring, "start") == 0) {
packet->type = kAudioPacketTypeStart;
auto sample_rate = cJSON_GetObjectItem(root, "sample_rate");
if (sample_rate != NULL) {
SetDecodeSampleRate(sample_rate->valueint);
}
} else if (strcmp(state->valuestring, "stop") == 0) {
packet->type = kAudioPacketTypeStop;
} else if (strcmp(state->valuestring, "sentence_end") == 0) {

2
main/Application.h
View File

@ -62,8 +62,10 @@ private:
OpusDecoder* opus_decoder_ = nullptr;
int opus_duration_ms_ = 60;
int opus_decode_sample_rate_ = CONFIG_AUDIO_OUTPUT_SAMPLE_RATE;
silk_resampler_state_struct resampler_state_;
void SetDecodeSampleRate(int sample_rate);
void SetChatState(ChatState state);
void StartDetection();
void StartCommunication();

31
main/AudioDevice.cc
View File

@ -23,15 +23,14 @@ AudioDevice::~AudioDevice() {
}
void AudioDevice::Start(int input_sample_rate, int output_sample_rate) {
assert(input_sample_rate == 16000);
input_sample_rate_ = input_sample_rate;
output_sample_rate_ = output_sample_rate;
if (output_sample_rate == 16000) {
CreateDuplexChannels();
} else {
#ifdef CONFIG_AUDIO_DEVICE_I2S_SIMPLEX
CreateSimplexChannels();
}
#else
CreateDuplexChannels();
#endif
ESP_ERROR_CHECK(i2s_channel_enable(tx_handle_));
ESP_ERROR_CHECK(i2s_channel_enable(rx_handle_));
@ -77,10 +76,10 @@ void AudioDevice::CreateDuplexChannels() {
},
.gpio_cfg = {
.mclk = I2S_GPIO_UNUSED,
.bclk = GPIO_NUM_5,
.ws = GPIO_NUM_4,
.dout = GPIO_NUM_6,
.din = GPIO_NUM_3,
.bclk = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_GPIO_BCLK,
.ws = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_GPIO_WS,
.dout = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_GPIO_DOUT,
.din = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_GPIO_DIN,
.invert_flags = {
.mclk_inv = false,
.bclk_inv = false,
@ -93,6 +92,7 @@ void AudioDevice::CreateDuplexChannels() {
ESP_LOGI(TAG, "Duplex channels created");
}
#ifdef CONFIG_AUDIO_DEVICE_I2S_SIMPLEX
void AudioDevice::CreateSimplexChannels() {
// Create a new channel for speaker
i2s_chan_config_t chan_cfg = {
@ -127,9 +127,9 @@ void AudioDevice::CreateSimplexChannels() {
},
.gpio_cfg = {
.mclk = I2S_GPIO_UNUSED,
.bclk = GPIO_NUM_5,
.ws = GPIO_NUM_4,
.dout = GPIO_NUM_6,
.bclk = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_GPIO_BCLK,
.ws = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_GPIO_WS,
.dout = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_GPIO_DOUT,
.din = I2S_GPIO_UNUSED,
.invert_flags = {
.mclk_inv = false,
@ -144,13 +144,14 @@ void AudioDevice::CreateSimplexChannels() {
chan_cfg.id = I2S_NUM_1;
ESP_ERROR_CHECK(i2s_new_channel(&chan_cfg, nullptr, &rx_handle_));
std_cfg.clk_cfg.sample_rate_hz = (uint32_t)input_sample_rate_;
std_cfg.gpio_cfg.bclk = GPIO_NUM_11;
std_cfg.gpio_cfg.ws = GPIO_NUM_10;
std_cfg.gpio_cfg.bclk = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_MIC_GPIO_BCLK;
std_cfg.gpio_cfg.ws = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_MIC_GPIO_WS;
std_cfg.gpio_cfg.dout = I2S_GPIO_UNUSED;
std_cfg.gpio_cfg.din = GPIO_NUM_3;
std_cfg.gpio_cfg.din = (gpio_num_t)CONFIG_AUDIO_DEVICE_I2S_GPIO_DIN;
ESP_ERROR_CHECK(i2s_channel_init_std_mode(rx_handle_, &std_cfg));
ESP_LOGI(TAG, "Simplex channels created");
}
#endif
void AudioDevice::Write(const int16_t* data, int samples) {
int32_t buffer[samples];

56
main/Kconfig.projbuild
View File

@ -19,4 +19,60 @@ config BUILTIN_LED_GPIO
help
GPIO number of the builtin LED.
config AUDIO_INPUT_SAMPLE_RATE
int "Audio Input Sample Rate"
default 16000
help
Audio input sample rate.
config AUDIO_OUTPUT_SAMPLE_RATE
int "Audio Output Sample Rate"
default 24000
help
Audio output sample rate.
config AUDIO_DEVICE_I2S_GPIO_BCLK
int "I2S GPIO BCLK"
default 5
help
GPIO number of the I2S BCLK.
config AUDIO_DEVICE_I2S_GPIO_WS
int "I2S GPIO WS"
default 4
help
GPIO number of the I2S WS.
config AUDIO_DEVICE_I2S_GPIO_DOUT
int "I2S GPIO DOUT"
default 6
help
GPIO number of the I2S DOUT.
config AUDIO_DEVICE_I2S_GPIO_DIN
int "I2S GPIO DIN"
default 3
help
GPIO number of the I2S DIN.
config AUDIO_DEVICE_I2S_SIMPLEX
bool "I2S Simplex"
default n
help
Enable I2S Simplex mode.
config AUDIO_DEVICE_I2S_MIC_GPIO_BCLK
int "I2S MIC GPIO BCLK"
default 11
depends on AUDIO_DEVICE_I2S_SIMPLEX
help
GPIO number of the I2S MIC BCLK.
config AUDIO_DEVICE_I2S_MIC_GPIO_WS
int "I2S MIC GPIO WS"
default 10
depends on AUDIO_DEVICE_I2S_SIMPLEX
help
GPIO number of the I2S MIC WS.
endmenu