mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-04 02:27:46 +01:00
5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
63 lines
2.1 KiB
C++
63 lines
2.1 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 <algorithm>
|
|
#include <array>
|
|
#include <cstddef>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "api/audio/echo_canceller3_config.h"
|
|
#include "modules/audio_processing/aec3/aec3_common.h"
|
|
#include "modules/audio_processing/aec3/render_buffer.h"
|
|
#include "rtc_base/checks.h"
|
|
#include "rtc_base/constructormagic.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// Provides functionality for analyzing the properties of the render signal.
|
|
class RenderSignalAnalyzer {
|
|
public:
|
|
explicit RenderSignalAnalyzer(const EchoCanceller3Config& config);
|
|
~RenderSignalAnalyzer();
|
|
|
|
// Updates the render signal analysis with the most recent render signal.
|
|
void Update(const RenderBuffer& render_buffer,
|
|
const absl::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;
|
|
|
|
absl::optional<int> NarrowPeakBand() const { return narrow_peak_band_; }
|
|
|
|
private:
|
|
const int strong_peak_freeze_duration_;
|
|
std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_;
|
|
absl::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_
|