AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M131 release

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Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
This commit is contained in:
Arun Raghavan
2024-12-24 19:32:07 -05:00
parent 8bdb53d91c
commit b5c48b97f6
263 changed files with 4628 additions and 20416 deletions
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14
webrtc/modules/audio_processing/agc2/vad_wrapper.h
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@ -14,10 +14,9 @@
#include <memory>
#include <vector>
#include "api/array_view.h"
#include "api/audio/audio_view.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "modules/audio_processing/agc2/cpu_features.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
@ -37,7 +36,7 @@ class VoiceActivityDetectorWrapper {
// Resets the internal state.
virtual void Reset() = 0;
// Analyzes an audio frame and returns the speech probability.
virtual float Analyze(rtc::ArrayView<const float> frame) = 0;
virtual float Analyze(MonoView<const float> frame) = 0;
};
// Ctor. Uses `cpu_features` to instantiate the default VAD.
@ -60,21 +59,18 @@ class VoiceActivityDetectorWrapper {
delete;
~VoiceActivityDetectorWrapper();
// Initializes the VAD wrapper.
void Initialize(int sample_rate_hz);
// Analyzes the first channel of `frame` and returns the speech probability.
// `frame` must be a 10 ms frame with the sample rate specified in the last
// `Initialize()` call.
float Analyze(AudioFrameView<const float> frame);
float Analyze(DeinterleavedView<const float> frame);
private:
const int vad_reset_period_frames_;
int frame_size_;
const int frame_size_;
int time_to_vad_reset_;
PushResampler<float> resampler_;
std::unique_ptr<MonoVad> vad_;
std::vector<float> resampled_buffer_;
PushResampler<float> resampler_;
};
} // namespace webrtc