forked from xiaozhi/xiaozhi-esp32
* Enhance audio processing and wake word detection - Set task priority in Application::Run to improve responsiveness. - Log detected wake words with their state in HandleWakeWordDetectedEvent. - Streamline audio feeding in AudioService to handle both wake word and audio processor events. - Implement input buffering in AfeAudioProcessor, AfeWakeWord, CustomWakeWord, and EspWakeWord to manage audio data more efficiently. - Clear input buffers on stop to prevent residual data issues. * Refactor audio processing to enhance thread safety and state management - Implement early return checks in Feed methods of AfeAudioProcessor, AfeWakeWord, CustomWakeWord, and EspWakeWord to prevent processing when not running. - Introduce std::atomic for running state in CustomWakeWord and EspWakeWord to ensure thread-safe access. - Consolidate input buffer management with mutex locks to avoid race conditions during Stop and Feed operations. * Refactor listening mode handling and wake word detection configuration - Replace direct mode setting logic with a new GetDefaultListeningMode method for improved clarity and maintainability. - Update HandleToggleChatEvent, HandleWakeWordDetectedEvent, and ContinueWakeWordInvoke to utilize the new method for determining listening mode. - Introduce Kconfig option WAKE_WORD_DETECTION_IN_LISTENING to enable or disable wake word detection during listening mode, enhancing configurability.
805 lines
31 KiB
C++
805 lines
31 KiB
C++
#include "audio_service.h"
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#include <esp_log.h>
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#include <cstring>
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#define RATE_CVT_CFG(_src_rate, _dest_rate, _channel) \
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(esp_ae_rate_cvt_cfg_t) \
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{ \
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.src_rate = (uint32_t)(_src_rate), \
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.dest_rate = (uint32_t)(_dest_rate), \
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.channel = (uint8_t)(_channel), \
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.bits_per_sample = ESP_AUDIO_BIT16, \
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.complexity = 2, \
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.perf_type = ESP_AE_RATE_CVT_PERF_TYPE_SPEED, \
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}
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#define OPUS_DEC_CFG(_sample_rate, _frame_duration_ms) \
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(esp_opus_dec_cfg_t) \
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{ \
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.sample_rate = (uint32_t)(_sample_rate), \
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.channel = ESP_AUDIO_MONO, \
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.frame_duration = (esp_opus_dec_frame_duration_t)AS_OPUS_GET_FRAME_DRU_ENUM(_frame_duration_ms), \
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.self_delimited = false, \
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}
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#if CONFIG_USE_AUDIO_PROCESSOR
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#include "processors/afe_audio_processor.h"
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#else
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#include "processors/no_audio_processor.h"
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#endif
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#if CONFIG_IDF_TARGET_ESP32S3 || CONFIG_IDF_TARGET_ESP32P4
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#include "wake_words/afe_wake_word.h"
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#include "wake_words/custom_wake_word.h"
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#else
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#include "wake_words/esp_wake_word.h"
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#endif
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#define TAG "AudioService"
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AudioService::AudioService() {
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event_group_ = xEventGroupCreate();
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}
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AudioService::~AudioService() {
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if (event_group_ != nullptr) {
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vEventGroupDelete(event_group_);
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}
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if (opus_encoder_ != nullptr) {
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esp_opus_enc_close(opus_encoder_);
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}
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if (opus_decoder_ != nullptr) {
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esp_opus_dec_close(opus_decoder_);
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}
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if (input_resampler_ != nullptr) {
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esp_ae_rate_cvt_close(input_resampler_);
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}
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if (output_resampler_ != nullptr) {
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esp_ae_rate_cvt_close(output_resampler_);
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}
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}
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void AudioService::Initialize(AudioCodec* codec) {
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codec_ = codec;
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codec_->Start();
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esp_opus_dec_cfg_t opus_dec_cfg = OPUS_DEC_CFG(codec->output_sample_rate(), OPUS_FRAME_DURATION_MS);
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auto ret = esp_opus_dec_open(&opus_dec_cfg, sizeof(esp_opus_dec_cfg_t), &opus_decoder_);
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if (opus_decoder_ == nullptr) {
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ESP_LOGE(TAG, "Failed to create audio decoder, error code: %d", ret);
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} else {
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decoder_sample_rate_ = codec->output_sample_rate();
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decoder_duration_ms_ = OPUS_FRAME_DURATION_MS;
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decoder_frame_size_ = decoder_sample_rate_ / 1000 * OPUS_FRAME_DURATION_MS;
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}
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esp_opus_enc_config_t opus_enc_cfg = AS_OPUS_ENC_CONFIG();
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ret = esp_opus_enc_open(&opus_enc_cfg, sizeof(esp_opus_enc_config_t), &opus_encoder_);
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if (opus_encoder_ == nullptr) {
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ESP_LOGE(TAG, "Failed to create audio encoder, error code: %d", ret);
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} else {
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encoder_sample_rate_ = 16000;
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encoder_duration_ms_ = OPUS_FRAME_DURATION_MS;
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esp_opus_enc_get_frame_size(opus_encoder_, &encoder_frame_size_, &encoder_outbuf_size_);
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encoder_frame_size_ = encoder_frame_size_ / sizeof(int16_t);
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}
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if (codec->input_sample_rate() != 16000) {
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esp_ae_rate_cvt_cfg_t input_resampler_cfg = RATE_CVT_CFG(
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codec->input_sample_rate(), ESP_AUDIO_SAMPLE_RATE_16K, codec->input_channels());
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auto resampler_ret = esp_ae_rate_cvt_open(&input_resampler_cfg, &input_resampler_);
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if (input_resampler_ == nullptr) {
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ESP_LOGE(TAG, "Failed to create input resampler, error code: %d", resampler_ret);
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}
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}
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#if CONFIG_USE_AUDIO_PROCESSOR
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audio_processor_ = std::make_unique<AfeAudioProcessor>();
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#else
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audio_processor_ = std::make_unique<NoAudioProcessor>();
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#endif
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audio_processor_->OnOutput([this](std::vector<int16_t>&& data) {
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PushTaskToEncodeQueue(kAudioTaskTypeEncodeToSendQueue, std::move(data));
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});
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audio_processor_->OnVadStateChange([this](bool speaking) {
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voice_detected_ = speaking;
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if (callbacks_.on_vad_change) {
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callbacks_.on_vad_change(speaking);
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}
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});
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esp_timer_create_args_t audio_power_timer_args = {
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.callback = [](void* arg) {
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AudioService* audio_service = (AudioService*)arg;
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audio_service->CheckAndUpdateAudioPowerState();
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},
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.arg = this,
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.dispatch_method = ESP_TIMER_TASK,
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.name = "audio_power_timer",
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.skip_unhandled_events = true,
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};
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esp_timer_create(&audio_power_timer_args, &audio_power_timer_);
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}
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void AudioService::Start() {
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service_stopped_ = false;
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xEventGroupClearBits(event_group_, AS_EVENT_AUDIO_TESTING_RUNNING | AS_EVENT_WAKE_WORD_RUNNING | AS_EVENT_AUDIO_PROCESSOR_RUNNING);
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esp_timer_start_periodic(audio_power_timer_, 1000000);
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#if CONFIG_USE_AUDIO_PROCESSOR
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/* Start the audio input task */
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xTaskCreatePinnedToCore([](void* arg) {
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AudioService* audio_service = (AudioService*)arg;
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audio_service->AudioInputTask();
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vTaskDelete(NULL);
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}, "audio_input", 2048 * 3, this, 8, &audio_input_task_handle_, 0);
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/* Start the audio output task */
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xTaskCreate([](void* arg) {
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AudioService* audio_service = (AudioService*)arg;
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audio_service->AudioOutputTask();
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vTaskDelete(NULL);
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}, "audio_output", 2048 * 2, this, 4, &audio_output_task_handle_);
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#else
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/* Start the audio input task */
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xTaskCreate([](void* arg) {
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AudioService* audio_service = (AudioService*)arg;
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audio_service->AudioInputTask();
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vTaskDelete(NULL);
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}, "audio_input", 2048 * 2, this, 8, &audio_input_task_handle_);
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/* Start the audio output task */
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xTaskCreate([](void* arg) {
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AudioService* audio_service = (AudioService*)arg;
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audio_service->AudioOutputTask();
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vTaskDelete(NULL);
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}, "audio_output", 2048, this, 4, &audio_output_task_handle_);
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#endif
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/* Start the opus codec task */
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xTaskCreate([](void* arg) {
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AudioService* audio_service = (AudioService*)arg;
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audio_service->OpusCodecTask();
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vTaskDelete(NULL);
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}, "opus_codec", 2048 * 12, this, 2, &opus_codec_task_handle_);
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}
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void AudioService::Stop() {
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esp_timer_stop(audio_power_timer_);
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service_stopped_ = true;
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xEventGroupSetBits(event_group_, AS_EVENT_AUDIO_TESTING_RUNNING |
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AS_EVENT_WAKE_WORD_RUNNING |
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AS_EVENT_AUDIO_PROCESSOR_RUNNING);
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std::lock_guard<std::mutex> lock(audio_queue_mutex_);
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audio_encode_queue_.clear();
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audio_decode_queue_.clear();
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audio_playback_queue_.clear();
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audio_testing_queue_.clear();
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audio_queue_cv_.notify_all();
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}
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bool AudioService::ReadAudioData(std::vector<int16_t>& data, int sample_rate, int samples) {
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if (!codec_->input_enabled()) {
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esp_timer_stop(audio_power_timer_);
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esp_timer_start_periodic(audio_power_timer_, AUDIO_POWER_CHECK_INTERVAL_MS * 1000);
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codec_->EnableInput(true);
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}
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if (codec_->input_sample_rate() != sample_rate) {
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data.resize(samples * codec_->input_sample_rate() / sample_rate * codec_->input_channels());
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if (!codec_->InputData(data)) {
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return false;
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}
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if (input_resampler_ != nullptr) {
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std::lock_guard<std::mutex> lock(input_resampler_mutex_);
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uint32_t in_sample_num = data.size() / codec_->input_channels();
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uint32_t output_samples = 0;
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esp_ae_rate_cvt_get_max_out_sample_num(input_resampler_, in_sample_num, &output_samples);
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auto resampled = std::vector<int16_t>(output_samples * codec_->input_channels());
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uint32_t actual_output = output_samples;
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esp_ae_rate_cvt_process(input_resampler_, (esp_ae_sample_t)data.data(), in_sample_num,
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(esp_ae_sample_t)resampled.data(), &actual_output);
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resampled.resize(actual_output * codec_->input_channels());
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data = std::move(resampled);
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}
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} else {
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data.resize(samples * codec_->input_channels());
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if (!codec_->InputData(data)) {
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return false;
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}
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}
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/* Update the last input time */
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last_input_time_ = std::chrono::steady_clock::now();
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debug_statistics_.input_count++;
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#if CONFIG_USE_AUDIO_DEBUGGER
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// 音频调试:发送原始音频数据
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if (audio_debugger_ == nullptr) {
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audio_debugger_ = std::make_unique<AudioDebugger>();
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}
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audio_debugger_->Feed(data);
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#endif
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return true;
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}
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void AudioService::AudioInputTask() {
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while (true) {
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EventBits_t bits = xEventGroupWaitBits(event_group_, AS_EVENT_AUDIO_TESTING_RUNNING |
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AS_EVENT_WAKE_WORD_RUNNING | AS_EVENT_AUDIO_PROCESSOR_RUNNING,
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pdFALSE, pdFALSE, portMAX_DELAY);
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if (service_stopped_) {
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break;
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}
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if (audio_input_need_warmup_) {
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audio_input_need_warmup_ = false;
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vTaskDelay(pdMS_TO_TICKS(120));
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continue;
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}
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/* Used for audio testing in NetworkConfiguring mode by clicking the BOOT button */
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if (bits & AS_EVENT_AUDIO_TESTING_RUNNING) {
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if (audio_testing_queue_.size() >= AUDIO_TESTING_MAX_DURATION_MS / OPUS_FRAME_DURATION_MS) {
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ESP_LOGW(TAG, "Audio testing queue is full, stopping audio testing");
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EnableAudioTesting(false);
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continue;
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}
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std::vector<int16_t> data;
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int samples = OPUS_FRAME_DURATION_MS * 16000 / 1000;
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if (ReadAudioData(data, 16000, samples)) {
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// If input channels is 2, we need to fetch the left channel data
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if (codec_->input_channels() == 2) {
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auto mono_data = std::vector<int16_t>(data.size() / 2);
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for (size_t i = 0, j = 0; i < mono_data.size(); ++i, j += 2) {
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mono_data[i] = data[j];
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}
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data = std::move(mono_data);
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}
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PushTaskToEncodeQueue(kAudioTaskTypeEncodeToTestingQueue, std::move(data));
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continue;
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}
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}
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/* Feed the wake word and/or audio processor */
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if (bits & (AS_EVENT_WAKE_WORD_RUNNING | AS_EVENT_AUDIO_PROCESSOR_RUNNING)) {
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int samples = 160; // 10ms
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std::vector<int16_t> data;
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if (ReadAudioData(data, 16000, samples)) {
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if (bits & AS_EVENT_WAKE_WORD_RUNNING) {
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wake_word_->Feed(data);
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}
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if (bits & AS_EVENT_AUDIO_PROCESSOR_RUNNING) {
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audio_processor_->Feed(std::move(data));
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}
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continue;
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}
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}
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ESP_LOGE(TAG, "Should not be here, bits: %lx", bits);
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break;
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}
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ESP_LOGW(TAG, "Audio input task stopped");
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}
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void AudioService::AudioOutputTask() {
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while (true) {
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std::unique_lock<std::mutex> lock(audio_queue_mutex_);
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audio_queue_cv_.wait(lock, [this]() { return !audio_playback_queue_.empty() || service_stopped_; });
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if (service_stopped_) {
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break;
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}
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auto task = std::move(audio_playback_queue_.front());
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audio_playback_queue_.pop_front();
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audio_queue_cv_.notify_all();
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lock.unlock();
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if (!codec_->output_enabled()) {
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esp_timer_stop(audio_power_timer_);
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esp_timer_start_periodic(audio_power_timer_, AUDIO_POWER_CHECK_INTERVAL_MS * 1000);
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codec_->EnableOutput(true);
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}
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codec_->OutputData(task->pcm);
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/* Update the last output time */
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last_output_time_ = std::chrono::steady_clock::now();
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debug_statistics_.playback_count++;
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#if CONFIG_USE_SERVER_AEC
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/* Record the timestamp for server AEC */
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if (task->timestamp > 0) {
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lock.lock();
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timestamp_queue_.push_back(task->timestamp);
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}
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#endif
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}
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ESP_LOGW(TAG, "Audio output task stopped");
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}
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void AudioService::OpusCodecTask() {
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while (true) {
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std::unique_lock<std::mutex> lock(audio_queue_mutex_);
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audio_queue_cv_.wait(lock, [this]() {
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return service_stopped_ ||
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(!audio_encode_queue_.empty() && audio_send_queue_.size() < MAX_SEND_PACKETS_IN_QUEUE) ||
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(!audio_decode_queue_.empty() && audio_playback_queue_.size() < MAX_PLAYBACK_TASKS_IN_QUEUE);
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});
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if (service_stopped_) {
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break;
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}
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/* Decode the audio from decode queue */
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if (!audio_decode_queue_.empty() && audio_playback_queue_.size() < MAX_PLAYBACK_TASKS_IN_QUEUE) {
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auto packet = std::move(audio_decode_queue_.front());
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audio_decode_queue_.pop_front();
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audio_queue_cv_.notify_all();
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lock.unlock();
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auto task = std::make_unique<AudioTask>();
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task->type = kAudioTaskTypeDecodeToPlaybackQueue;
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task->timestamp = packet->timestamp;
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SetDecodeSampleRate(packet->sample_rate, packet->frame_duration);
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if (opus_decoder_ != nullptr) {
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task->pcm.resize(decoder_frame_size_);
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esp_audio_dec_in_raw_t raw = {
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.buffer = (uint8_t *)(packet->payload.data()),
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.len = (uint32_t)(packet->payload.size()),
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.consumed = 0,
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.frame_recover = ESP_AUDIO_DEC_RECOVERY_NONE,
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};
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esp_audio_dec_out_frame_t out_frame = {
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.buffer = (uint8_t *)(task->pcm.data()),
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.len = (uint32_t)(task->pcm.size() * sizeof(int16_t)),
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.decoded_size = 0,
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};
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esp_audio_dec_info_t dec_info = {};
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std::unique_lock<std::mutex> decoder_lock(decoder_mutex_);
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auto ret = esp_opus_dec_decode(opus_decoder_, &raw, &out_frame, &dec_info);
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decoder_lock.unlock();
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if (ret == ESP_AUDIO_ERR_OK) {
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task->pcm.resize(out_frame.decoded_size / sizeof(int16_t));
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if (decoder_sample_rate_ != codec_->output_sample_rate() && output_resampler_ != nullptr) {
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uint32_t target_size = 0;
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esp_ae_rate_cvt_get_max_out_sample_num(output_resampler_, task->pcm.size(), &target_size);
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std::vector<int16_t> resampled(target_size);
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uint32_t actual_output = target_size;
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esp_ae_rate_cvt_process(output_resampler_, (esp_ae_sample_t)task->pcm.data(), task->pcm.size(),
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(esp_ae_sample_t)resampled.data(), &actual_output);
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resampled.resize(actual_output);
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task->pcm = std::move(resampled);
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}
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lock.lock();
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audio_playback_queue_.push_back(std::move(task));
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audio_queue_cv_.notify_all();
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debug_statistics_.decode_count++;
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} else {
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ESP_LOGE(TAG, "Failed to decode audio after resize, error code: %d", ret);
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lock.lock();
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}
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} else {
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ESP_LOGE(TAG, "Audio decoder is not configured");
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lock.lock();
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}
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debug_statistics_.decode_count++;
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}
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/* Encode the audio to send queue */
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if (!audio_encode_queue_.empty() && audio_send_queue_.size() < MAX_SEND_PACKETS_IN_QUEUE) {
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auto task = std::move(audio_encode_queue_.front());
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audio_encode_queue_.pop_front();
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audio_queue_cv_.notify_all();
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lock.unlock();
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auto packet = std::make_unique<AudioStreamPacket>();
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packet->frame_duration = OPUS_FRAME_DURATION_MS;
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packet->sample_rate = 16000;
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packet->timestamp = task->timestamp;
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if (opus_encoder_ != nullptr && task->pcm.size() == encoder_frame_size_) {
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std::vector<uint8_t> buf(encoder_outbuf_size_);
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esp_audio_enc_in_frame_t in = {
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.buffer = (uint8_t *)(task->pcm.data()),
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.len = (uint32_t)(encoder_frame_size_ * sizeof(int16_t)),
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};
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esp_audio_enc_out_frame_t out = {
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.buffer = buf.data(),
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.len = (uint32_t)encoder_outbuf_size_,
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.encoded_bytes = 0,
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};
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auto ret = esp_opus_enc_process(opus_encoder_, &in, &out);
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if (ret == ESP_AUDIO_ERR_OK) {
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packet->payload.assign(buf.data(), buf.data() + out.encoded_bytes);
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if (task->type == kAudioTaskTypeEncodeToSendQueue) {
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{
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std::lock_guard<std::mutex> lock2(audio_queue_mutex_);
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audio_send_queue_.push_back(std::move(packet));
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}
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if (callbacks_.on_send_queue_available) {
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callbacks_.on_send_queue_available();
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}
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} else if (task->type == kAudioTaskTypeEncodeToTestingQueue) {
|
||
std::lock_guard<std::mutex> lock2(audio_queue_mutex_);
|
||
audio_testing_queue_.push_back(std::move(packet));
|
||
}
|
||
debug_statistics_.encode_count++;
|
||
} else {
|
||
ESP_LOGE(TAG, "Failed to encode audio, error code: %d", ret);
|
||
}
|
||
} else {
|
||
ESP_LOGE(TAG, "Failed to encode audio: encoder not configured or invalid frame size (got %u, expected %u)",
|
||
task->pcm.size(), encoder_frame_size_);
|
||
}
|
||
lock.lock();
|
||
}
|
||
}
|
||
|
||
ESP_LOGW(TAG, "Opus codec task stopped");
|
||
}
|
||
|
||
void AudioService::SetDecodeSampleRate(int sample_rate, int frame_duration) {
|
||
if (decoder_sample_rate_ == sample_rate && decoder_duration_ms_ == frame_duration) {
|
||
return;
|
||
}
|
||
std::unique_lock<std::mutex> decoder_lock(decoder_mutex_);
|
||
if (opus_decoder_ != nullptr) {
|
||
esp_opus_dec_close(opus_decoder_);
|
||
opus_decoder_ = nullptr;
|
||
}
|
||
decoder_lock.unlock();
|
||
esp_opus_dec_cfg_t opus_dec_cfg = OPUS_DEC_CFG(sample_rate, frame_duration);
|
||
auto ret = esp_opus_dec_open(&opus_dec_cfg, sizeof(esp_opus_dec_cfg_t), &opus_decoder_);
|
||
if (opus_decoder_ == nullptr) {
|
||
ESP_LOGE(TAG, "Failed to create audio decoder, error code: %d", ret);
|
||
return;
|
||
}
|
||
decoder_sample_rate_ = sample_rate;
|
||
decoder_duration_ms_ = frame_duration;
|
||
decoder_frame_size_ = decoder_sample_rate_ / 1000 * frame_duration;
|
||
|
||
auto codec = Board::GetInstance().GetAudioCodec();
|
||
if (decoder_sample_rate_ != codec->output_sample_rate()) {
|
||
ESP_LOGI(TAG, "Resampling audio from %d to %d", decoder_sample_rate_, codec->output_sample_rate());
|
||
if (output_resampler_ != nullptr) {
|
||
esp_ae_rate_cvt_close(output_resampler_);
|
||
output_resampler_ = nullptr;
|
||
}
|
||
esp_ae_rate_cvt_cfg_t output_resampler_cfg = RATE_CVT_CFG(
|
||
decoder_sample_rate_, codec->output_sample_rate(), ESP_AUDIO_MONO);
|
||
auto resampler_ret = esp_ae_rate_cvt_open(&output_resampler_cfg, &output_resampler_);
|
||
if (output_resampler_ == nullptr) {
|
||
ESP_LOGE(TAG, "Failed to create output resampler, error code: %d", resampler_ret);
|
||
}
|
||
}
|
||
}
|
||
|
||
void AudioService::PushTaskToEncodeQueue(AudioTaskType type, std::vector<int16_t>&& pcm) {
|
||
auto task = std::make_unique<AudioTask>();
|
||
task->type = type;
|
||
task->pcm = std::move(pcm);
|
||
/* Push the task to the encode queue */
|
||
std::unique_lock<std::mutex> lock(audio_queue_mutex_);
|
||
|
||
/* If the task is to send queue, we need to set the timestamp */
|
||
if (type == kAudioTaskTypeEncodeToSendQueue && !timestamp_queue_.empty()) {
|
||
if (timestamp_queue_.size() <= MAX_TIMESTAMPS_IN_QUEUE) {
|
||
task->timestamp = timestamp_queue_.front();
|
||
} else {
|
||
ESP_LOGW(TAG, "Timestamp queue (%u) is full, dropping timestamp", timestamp_queue_.size());
|
||
}
|
||
timestamp_queue_.pop_front();
|
||
}
|
||
|
||
audio_queue_cv_.wait(lock, [this]() { return audio_encode_queue_.size() < MAX_ENCODE_TASKS_IN_QUEUE; });
|
||
audio_encode_queue_.push_back(std::move(task));
|
||
audio_queue_cv_.notify_all();
|
||
}
|
||
|
||
bool AudioService::PushPacketToDecodeQueue(std::unique_ptr<AudioStreamPacket> packet, bool wait) {
|
||
std::unique_lock<std::mutex> lock(audio_queue_mutex_);
|
||
if (audio_decode_queue_.size() >= MAX_DECODE_PACKETS_IN_QUEUE) {
|
||
if (wait) {
|
||
audio_queue_cv_.wait(lock, [this]() { return audio_decode_queue_.size() < MAX_DECODE_PACKETS_IN_QUEUE; });
|
||
} else {
|
||
return false;
|
||
}
|
||
}
|
||
audio_decode_queue_.push_back(std::move(packet));
|
||
audio_queue_cv_.notify_all();
|
||
return true;
|
||
}
|
||
|
||
std::unique_ptr<AudioStreamPacket> AudioService::PopPacketFromSendQueue() {
|
||
std::lock_guard<std::mutex> lock(audio_queue_mutex_);
|
||
if (audio_send_queue_.empty()) {
|
||
return nullptr;
|
||
}
|
||
auto packet = std::move(audio_send_queue_.front());
|
||
audio_send_queue_.pop_front();
|
||
audio_queue_cv_.notify_all();
|
||
return packet;
|
||
}
|
||
|
||
void AudioService::EncodeWakeWord() {
|
||
if (wake_word_) {
|
||
wake_word_->EncodeWakeWordData();
|
||
}
|
||
}
|
||
|
||
const std::string& AudioService::GetLastWakeWord() const {
|
||
return wake_word_->GetLastDetectedWakeWord();
|
||
}
|
||
|
||
std::unique_ptr<AudioStreamPacket> AudioService::PopWakeWordPacket() {
|
||
auto packet = std::make_unique<AudioStreamPacket>();
|
||
if (wake_word_->GetWakeWordOpus(packet->payload)) {
|
||
return packet;
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
void AudioService::EnableWakeWordDetection(bool enable) {
|
||
if (!wake_word_) {
|
||
return;
|
||
}
|
||
|
||
ESP_LOGD(TAG, "%s wake word detection", enable ? "Enabling" : "Disabling");
|
||
if (enable) {
|
||
if (!wake_word_initialized_) {
|
||
if (!wake_word_->Initialize(codec_, models_list_)) {
|
||
ESP_LOGE(TAG, "Failed to initialize wake word");
|
||
return;
|
||
}
|
||
wake_word_initialized_ = true;
|
||
}
|
||
// Reset input resampler to clear cached data from previous mode (e.g. AudioProcessor)
|
||
// This prevents buffer overflow when switching between different feed sizes
|
||
{
|
||
std::lock_guard<std::mutex> lock(input_resampler_mutex_);
|
||
if (input_resampler_ != nullptr) {
|
||
esp_ae_rate_cvt_reset(input_resampler_);
|
||
}
|
||
}
|
||
wake_word_->Start();
|
||
xEventGroupSetBits(event_group_, AS_EVENT_WAKE_WORD_RUNNING);
|
||
} else {
|
||
wake_word_->Stop();
|
||
xEventGroupClearBits(event_group_, AS_EVENT_WAKE_WORD_RUNNING);
|
||
}
|
||
}
|
||
|
||
void AudioService::EnableVoiceProcessing(bool enable) {
|
||
ESP_LOGD(TAG, "%s voice processing", enable ? "Enabling" : "Disabling");
|
||
if (enable) {
|
||
if (!audio_processor_initialized_) {
|
||
audio_processor_->Initialize(codec_, OPUS_FRAME_DURATION_MS, models_list_);
|
||
audio_processor_initialized_ = true;
|
||
}
|
||
|
||
/* We should make sure no audio is playing */
|
||
ResetDecoder();
|
||
audio_input_need_warmup_ = true;
|
||
// Reset input resampler to clear cached data from previous mode (e.g. WakeWord)
|
||
// This prevents buffer overflow when switching between different feed sizes
|
||
{
|
||
std::lock_guard<std::mutex> lock(input_resampler_mutex_);
|
||
if (input_resampler_ != nullptr) {
|
||
esp_ae_rate_cvt_reset(input_resampler_);
|
||
}
|
||
}
|
||
audio_processor_->Start();
|
||
xEventGroupSetBits(event_group_, AS_EVENT_AUDIO_PROCESSOR_RUNNING);
|
||
} else {
|
||
audio_processor_->Stop();
|
||
xEventGroupClearBits(event_group_, AS_EVENT_AUDIO_PROCESSOR_RUNNING);
|
||
}
|
||
}
|
||
|
||
void AudioService::EnableAudioTesting(bool enable) {
|
||
ESP_LOGI(TAG, "%s audio testing", enable ? "Enabling" : "Disabling");
|
||
if (enable) {
|
||
xEventGroupSetBits(event_group_, AS_EVENT_AUDIO_TESTING_RUNNING);
|
||
} else {
|
||
xEventGroupClearBits(event_group_, AS_EVENT_AUDIO_TESTING_RUNNING);
|
||
/* Copy audio_testing_queue_ to audio_decode_queue_ */
|
||
std::lock_guard<std::mutex> lock(audio_queue_mutex_);
|
||
audio_decode_queue_ = std::move(audio_testing_queue_);
|
||
audio_queue_cv_.notify_all();
|
||
}
|
||
}
|
||
|
||
void AudioService::EnableDeviceAec(bool enable) {
|
||
ESP_LOGI(TAG, "%s device AEC", enable ? "Enabling" : "Disabling");
|
||
if (!audio_processor_initialized_) {
|
||
audio_processor_->Initialize(codec_, OPUS_FRAME_DURATION_MS, models_list_);
|
||
audio_processor_initialized_ = true;
|
||
}
|
||
|
||
audio_processor_->EnableDeviceAec(enable);
|
||
}
|
||
|
||
void AudioService::SetCallbacks(AudioServiceCallbacks& callbacks) {
|
||
callbacks_ = callbacks;
|
||
}
|
||
|
||
void AudioService::PlaySound(const std::string_view& ogg) {
|
||
if (!codec_->output_enabled()) {
|
||
esp_timer_stop(audio_power_timer_);
|
||
esp_timer_start_periodic(audio_power_timer_, AUDIO_POWER_CHECK_INTERVAL_MS * 1000);
|
||
codec_->EnableOutput(true);
|
||
}
|
||
|
||
const uint8_t* buf = reinterpret_cast<const uint8_t*>(ogg.data());
|
||
size_t size = ogg.size();
|
||
size_t offset = 0;
|
||
|
||
auto find_page = [&](size_t start)->size_t {
|
||
for (size_t i = start; i + 4 <= size; ++i) {
|
||
if (buf[i] == 'O' && buf[i+1] == 'g' && buf[i+2] == 'g' && buf[i+3] == 'S') return i;
|
||
}
|
||
return static_cast<size_t>(-1);
|
||
};
|
||
|
||
bool seen_head = false;
|
||
bool seen_tags = false;
|
||
int sample_rate = 16000; // 默认值
|
||
|
||
while (true) {
|
||
size_t pos = find_page(offset);
|
||
if (pos == static_cast<size_t>(-1)) break;
|
||
offset = pos;
|
||
if (offset + 27 > size) break;
|
||
|
||
const uint8_t* page = buf + offset;
|
||
uint8_t page_segments = page[26];
|
||
size_t seg_table_off = offset + 27;
|
||
if (seg_table_off + page_segments > size) break;
|
||
|
||
size_t body_size = 0;
|
||
for (size_t i = 0; i < page_segments; ++i) body_size += page[27 + i];
|
||
|
||
size_t body_off = seg_table_off + page_segments;
|
||
if (body_off + body_size > size) break;
|
||
|
||
// Parse packets using lacing
|
||
size_t cur = body_off;
|
||
size_t seg_idx = 0;
|
||
while (seg_idx < page_segments) {
|
||
size_t pkt_len = 0;
|
||
size_t pkt_start = cur;
|
||
bool continued = false;
|
||
do {
|
||
uint8_t l = page[27 + seg_idx++];
|
||
pkt_len += l;
|
||
cur += l;
|
||
continued = (l == 255);
|
||
} while (continued && seg_idx < page_segments);
|
||
|
||
if (pkt_len == 0) continue;
|
||
const uint8_t* pkt_ptr = buf + pkt_start;
|
||
|
||
if (!seen_head) {
|
||
// 解析OpusHead包
|
||
if (pkt_len >= 19 && std::memcmp(pkt_ptr, "OpusHead", 8) == 0) {
|
||
seen_head = true;
|
||
// OpusHead结构:[0-7] "OpusHead", [8] version, [9] channel_count, [10-11] pre_skip
|
||
// [12-15] input_sample_rate, [16-17] output_gain, [18] mapping_family
|
||
if (pkt_len >= 12) {
|
||
uint8_t version = pkt_ptr[8];
|
||
uint8_t channel_count = pkt_ptr[9];
|
||
if (pkt_len >= 16) {
|
||
// 读取输入采样率 (little-endian)
|
||
sample_rate = pkt_ptr[12] | (pkt_ptr[13] << 8) |
|
||
(pkt_ptr[14] << 16) | (pkt_ptr[15] << 24);
|
||
ESP_LOGI(TAG, "OpusHead: version=%d, channels=%d, sample_rate=%d",
|
||
version, channel_count, sample_rate);
|
||
}
|
||
}
|
||
}
|
||
continue;
|
||
}
|
||
if (!seen_tags) {
|
||
// Expect OpusTags in second packet
|
||
if (pkt_len >= 8 && std::memcmp(pkt_ptr, "OpusTags", 8) == 0) {
|
||
seen_tags = true;
|
||
}
|
||
continue;
|
||
}
|
||
|
||
// Audio packet (Opus)
|
||
auto packet = std::make_unique<AudioStreamPacket>();
|
||
packet->sample_rate = sample_rate;
|
||
packet->frame_duration = 60;
|
||
packet->payload.resize(pkt_len);
|
||
std::memcpy(packet->payload.data(), pkt_ptr, pkt_len);
|
||
PushPacketToDecodeQueue(std::move(packet), true);
|
||
}
|
||
|
||
offset = body_off + body_size;
|
||
}
|
||
}
|
||
|
||
bool AudioService::IsIdle() {
|
||
std::lock_guard<std::mutex> lock(audio_queue_mutex_);
|
||
return audio_encode_queue_.empty() && audio_decode_queue_.empty() && audio_playback_queue_.empty() && audio_testing_queue_.empty();
|
||
}
|
||
|
||
void AudioService::WaitForPlaybackQueueEmpty() {
|
||
std::unique_lock<std::mutex> lock(audio_queue_mutex_);
|
||
audio_queue_cv_.wait(lock, [this]() {
|
||
return service_stopped_ || (audio_decode_queue_.empty() && audio_playback_queue_.empty());
|
||
});
|
||
}
|
||
|
||
void AudioService::ResetDecoder() {
|
||
std::lock_guard<std::mutex> lock(audio_queue_mutex_);
|
||
std::unique_lock<std::mutex> decoder_lock(decoder_mutex_);
|
||
if (opus_decoder_ != nullptr) {
|
||
esp_opus_dec_reset(opus_decoder_);
|
||
}
|
||
decoder_lock.unlock();
|
||
timestamp_queue_.clear();
|
||
audio_decode_queue_.clear();
|
||
audio_playback_queue_.clear();
|
||
audio_testing_queue_.clear();
|
||
audio_queue_cv_.notify_all();
|
||
}
|
||
|
||
void AudioService::CheckAndUpdateAudioPowerState() {
|
||
auto now = std::chrono::steady_clock::now();
|
||
auto input_elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(now - last_input_time_).count();
|
||
auto output_elapsed = std::chrono::duration_cast<std::chrono::milliseconds>(now - last_output_time_).count();
|
||
if (input_elapsed > AUDIO_POWER_TIMEOUT_MS && codec_->input_enabled()) {
|
||
codec_->EnableInput(false);
|
||
}
|
||
if (output_elapsed > AUDIO_POWER_TIMEOUT_MS && codec_->output_enabled()) {
|
||
codec_->EnableOutput(false);
|
||
}
|
||
if (!codec_->input_enabled() && !codec_->output_enabled()) {
|
||
esp_timer_stop(audio_power_timer_);
|
||
}
|
||
}
|
||
|
||
void AudioService::SetModelsList(srmodel_list_t* models_list) {
|
||
models_list_ = models_list;
|
||
|
||
#if CONFIG_IDF_TARGET_ESP32S3 || CONFIG_IDF_TARGET_ESP32P4
|
||
if (esp_srmodel_filter(models_list_, ESP_MN_PREFIX, NULL) != nullptr) {
|
||
wake_word_ = std::make_unique<CustomWakeWord>();
|
||
} else if (esp_srmodel_filter(models_list_, ESP_WN_PREFIX, NULL) != nullptr) {
|
||
wake_word_ = std::make_unique<AfeWakeWord>();
|
||
} else {
|
||
wake_word_ = nullptr;
|
||
}
|
||
#else
|
||
if (esp_srmodel_filter(models_list_, ESP_WN_PREFIX, NULL) != nullptr) {
|
||
wake_word_ = std::make_unique<EspWakeWord>();
|
||
} else {
|
||
wake_word_ = nullptr;
|
||
}
|
||
#endif
|
||
|
||
if (wake_word_) {
|
||
wake_word_->OnWakeWordDetected([this](const std::string& wake_word) {
|
||
if (callbacks_.on_wake_word_detected) {
|
||
callbacks_.on_wake_word_detected(wake_word);
|
||
}
|
||
});
|
||
}
|
||
}
|
||
|
||
bool AudioService::IsAfeWakeWord() {
|
||
#if CONFIG_IDF_TARGET_ESP32S3 || CONFIG_IDF_TARGET_ESP32P4
|
||
return wake_word_ != nullptr && dynamic_cast<AfeWakeWord*>(wake_word_.get()) != nullptr;
|
||
#else
|
||
return false;
|
||
#endif
|
||
}
|