mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-19 00:27:51 +01:00
93e5750a92
Hypothetical scenario: short weak speech at start of call, then high noise. The digital adaptive AGC2 would pick a high gain, and then continue to apply it on the noise. Unless the noise is detected by the noise estimator, the gain would never be reduced. This CL addresses the issue by sending limiter gain info to the adaptive digital AGC2. Bug: webrtc:7494 Change-Id: Idf5c2686af0f5e5bad981d39a95b8efc9ffb9d64 Reviewed-on: https://webrtc-review.googlesource.com/102641 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24922}
43 lines
1.4 KiB
C++
43 lines
1.4 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_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
|
|
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
|
|
|
|
#include "modules/audio_processing/agc2/saturation_protector.h"
|
|
#include "modules/audio_processing/agc2/vad_with_level.h"
|
|
|
|
namespace webrtc {
|
|
class ApmDataDumper;
|
|
|
|
class AdaptiveModeLevelEstimator {
|
|
public:
|
|
explicit AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper);
|
|
void UpdateEstimation(const VadWithLevel::LevelAndProbability& vad_data);
|
|
float LatestLevelEstimate() const;
|
|
void Reset();
|
|
bool LevelEstimationIsConfident() const {
|
|
return buffer_size_ms_ >= kFullBufferSizeMs;
|
|
}
|
|
|
|
private:
|
|
void DebugDumpEstimate();
|
|
|
|
size_t buffer_size_ms_ = 0;
|
|
float last_estimate_with_offset_dbfs_ = kInitialSpeechLevelEstimateDbfs;
|
|
float estimate_numerator_ = 0.f;
|
|
float estimate_denominator_ = 0.f;
|
|
SaturationProtector saturation_protector_;
|
|
ApmDataDumper* const apm_data_dumper_;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
|