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Fix frame samples for server AEC

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This commit is contained in:
Terrence
2025-07-20 07:53:52 +08:00
parent efc6f238e7
commit a35a344f42
7 changed files with 12 additions and 12 deletions
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6
main/audio/audio_service.cc
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@@ -288,11 +288,13 @@ void AudioService::AudioOutputTask() {
last_output_time_ = std::chrono::steady_clock::now();
debug_statistics_.playback_count++;
/* Record the timestamp */
#if CONFIG_USE_SERVER_AEC
/* Record the timestamp for server AEC */
if (task->timestamp > 0) {
lock.lock();
timestamp_queue_.push_back(task->timestamp);
}
#endif
}
ESP_LOGW(TAG, "Audio output task stopped");
@@ -405,7 +407,7 @@ void AudioService::PushTaskToEncodeQueue(AudioTaskType type, std::vector<int16_t
if (timestamp_queue_.size() <= MAX_TIMESTAMPS_IN_QUEUE) {
task->timestamp = timestamp_queue_.front();
} else {
ESP_LOGW(TAG, "Timestamp queue is full, dropping timestamp");
ESP_LOGW(TAG, "Timestamp queue (%u) is full, dropping timestamp", timestamp_queue_.size());
}
timestamp_queue_.pop_front();
}

3
main/audio/processors/afe_audio_processor.cc
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@@ -12,8 +12,7 @@ AfeAudioProcessor::AfeAudioProcessor()
void AfeAudioProcessor::Initialize(AudioCodec* codec, int frame_duration_ms) {
codec_ = codec;
frame_duration_ms_ = frame_duration_ms;
frame_samples_ = frame_duration_ms_ * codec_->input_sample_rate() / 1000;
frame_samples_ = frame_duration_ms * 16000 / 1000;
// Pre-allocate output buffer capacity
output_buffer_.reserve(frame_samples_);

1
main/audio/processors/afe_audio_processor.h
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@@ -35,7 +35,6 @@ private:
std::function<void(std::vector<int16_t>&& data)> output_callback_;
std::function<void(bool speaking)> vad_state_change_callback_;
AudioCodec* codec_ = nullptr;
int frame_duration_ms_ = 0;
int frame_samples_ = 0;
bool is_speaking_ = false;
std::vector<int16_t> output_buffer_;

7
main/audio/processors/no_audio_processor.cc
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@@ -3,10 +3,9 @@
#define TAG "NoAudioProcessor"
void NoAudioProcessor::Initialize(AudioCodec* codec, int frame_duration_ms) :
codec_(codec),
frame_duration_ms_(frame_duration_ms) {
frame_samples_ = frame_duration_ms_ * codec_->input_sample_rate() / 1000;
void NoAudioProcessor::Initialize(AudioCodec* codec, int frame_duration_ms) {
codec_ = codec;
frame_samples_ = frame_duration_ms * 16000 / 1000;
}
void NoAudioProcessor::Feed(std::vector<int16_t>&& data) {

1
main/audio/processors/no_audio_processor.h
View File

@@ -24,7 +24,6 @@ public:
private:
AudioCodec* codec_ = nullptr;
int frame_duration_ms_ = 0;
int frame_samples_ = 0;
std::function<void(std::vector<int16_t>&& data)> output_callback_;
std::function<void(bool speaking)> vad_state_change_callback_;