blob: 5818a106bacba2b32f53e027f50fa4568a943da4 [file] [log] [blame]
* Copyright (c) 2017 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.
#include <memory>
#include <vector>
#include "webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h"
#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
#include "webrtc/modules/audio_processing/aec3/render_buffer.h"
#include "webrtc/modules/audio_processing/aec3/render_signal_analyzer.h"
#include "webrtc/modules/audio_processing/aec3/subtractor_output.h"
#include "webrtc/rtc_base/constructormagic.h"
namespace webrtc {
class ApmDataDumper;
// Provides functionality for computing the adaptive gain for the main filter.
class MainFilterUpdateGain {
// Takes action in the case of a known echo path change.
void HandleEchoPathChange();
// Computes the gain.
void Compute(const RenderBuffer& render_buffer,
const RenderSignalAnalyzer& render_signal_analyzer,
const SubtractorOutput& subtractor_output,
const AdaptiveFirFilter& filter,
bool saturated_capture_signal,
FftData* gain_fft);
static int instance_count_;
std::unique_ptr<ApmDataDumper> data_dumper_;
std::array<float, kFftLengthBy2Plus1> H_error_;
size_t poor_excitation_counter_;
size_t call_counter_ = 0;
} // namespace webrtc