forked from xiaozhi/xiaozhi-esp32
Fix frame samples for server AEC
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@@ -12,8 +12,7 @@ AfeAudioProcessor::AfeAudioProcessor()
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void AfeAudioProcessor::Initialize(AudioCodec* codec, int frame_duration_ms) {
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codec_ = codec;
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frame_duration_ms_ = frame_duration_ms;
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frame_samples_ = frame_duration_ms_ * codec_->input_sample_rate() / 1000;
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frame_samples_ = frame_duration_ms * 16000 / 1000;
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// Pre-allocate output buffer capacity
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output_buffer_.reserve(frame_samples_);
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@@ -35,7 +35,6 @@ private:
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std::function<void(std::vector<int16_t>&& data)> output_callback_;
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std::function<void(bool speaking)> vad_state_change_callback_;
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AudioCodec* codec_ = nullptr;
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int frame_duration_ms_ = 0;
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int frame_samples_ = 0;
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bool is_speaking_ = false;
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std::vector<int16_t> output_buffer_;
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@@ -3,10 +3,9 @@
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#define TAG "NoAudioProcessor"
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void NoAudioProcessor::Initialize(AudioCodec* codec, int frame_duration_ms) :
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codec_(codec),
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frame_duration_ms_(frame_duration_ms) {
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frame_samples_ = frame_duration_ms_ * codec_->input_sample_rate() / 1000;
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void NoAudioProcessor::Initialize(AudioCodec* codec, int frame_duration_ms) {
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codec_ = codec;
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frame_samples_ = frame_duration_ms * 16000 / 1000;
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}
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void NoAudioProcessor::Feed(std::vector<int16_t>&& data) {
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@@ -24,7 +24,6 @@ public:
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private:
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AudioCodec* codec_ = nullptr;
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int frame_duration_ms_ = 0;
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int frame_samples_ = 0;
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std::function<void(std::vector<int16_t>&& data)> output_callback_;
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std::function<void(bool speaking)> vad_state_change_callback_;
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