webrtc/modules/audio_processing/vad/vad_with_level.h
Alex Loiko 2bac896d5e Adaptive Digital gain control structure.
This CL defines the control flow of the adaptive AGC. It also defines
method and class stubs.

Contents:
1. Divide the 'agc2' build target into 'fixed_digital' and
'adaptive_digital'.
1. Update the dependencies of everything that depended on 'agc2'.
2. Define the sub-modules of the adaptive digital AGC 2. They are:
   1. Level Estimator - it gets the energy and a speech probability
      and updates a speech level estimate.
   2. Noise Estimator - it gets an immutable view of the speech frame
      and updates the noise level estimate
   3. Gain applier - it gets the speech frame, the current speech and
      noise estimates, and the speech probability. It finds a gain to
      apply and applies it.
   4. AdaptiveAgc - sets up and controls the sub-modules described
      above.

Bug: webrtc:7494
Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1
Reviewed-on: https://webrtc-review.googlesource.com/64440
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 14:12:50 +00:00

40 lines
1.3 KiB
C++

/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_WITH_LEVEL_H_
#define MODULES_AUDIO_PROCESSING_VAD_VAD_WITH_LEVEL_H_
#include "api/array_view.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
class VadWithLevel {
public:
struct LevelAndProbability {
constexpr LevelAndProbability(float prob, float rms, float peak)
: speech_probability(prob),
speech_rms_dbfs(rms),
speech_peak_dbfs(peak) {}
LevelAndProbability() = default;
float speech_probability = 0;
float speech_rms_dbfs = 0; // Root mean square in decibels to full-scale.
float speech_peak_dbfs = 0;
};
// TODO(webrtc:7494): This is a stub. Add implementation.
rtc::ArrayView<const LevelAndProbability> AnalyzeFrame(
AudioFrameView<const float> frame) {
return {nullptr, 0};
}
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_VAD_VAD_WITH_LEVEL_H_