mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 23:01:21 +01:00
92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
58 lines
1.9 KiB
C++
58 lines
1.9 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
|
|
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
|
|
|
|
#include <array>
|
|
#include <memory>
|
|
|
|
#include "api/optional.h"
|
|
#include "modules/audio_processing/aec3/aec3_common.h"
|
|
#include "modules/audio_processing/aec3/render_buffer.h"
|
|
#include "rtc_base/constructormagic.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Provides functionality for analyzing the properties of the render signal.
|
|
class RenderSignalAnalyzer {
|
|
public:
|
|
RenderSignalAnalyzer();
|
|
~RenderSignalAnalyzer();
|
|
|
|
// Updates the render signal analysis with the most recent render signal.
|
|
void Update(const RenderBuffer& render_buffer,
|
|
const rtc::Optional<size_t>& delay_partitions);
|
|
|
|
// Returns true if the render signal is poorly exciting.
|
|
bool PoorSignalExcitation() const {
|
|
RTC_DCHECK_LT(2, narrow_band_counters_.size());
|
|
return std::any_of(narrow_band_counters_.begin(),
|
|
narrow_band_counters_.end(),
|
|
[](size_t a) { return a > 10; });
|
|
}
|
|
|
|
// Zeros the array around regions with narrow bands signal characteristics.
|
|
void MaskRegionsAroundNarrowBands(
|
|
std::array<float, kFftLengthBy2Plus1>* v) const;
|
|
|
|
rtc::Optional<int> NarrowPeakBand() const { return narrow_peak_band_; }
|
|
|
|
private:
|
|
std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_;
|
|
rtc::Optional<int> narrow_peak_band_;
|
|
size_t narrow_peak_counter_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(RenderSignalAnalyzer);
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_SIGNAL_ANALYZER_H_
|