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configure GPIO and sample rates

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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
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44
main/Application.cc
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@@ -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) {