Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions

View File

@ -1,5 +1,5 @@
/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
* Copyright (c) 2020 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
@ -8,30 +8,24 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
#ifndef MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
#define MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
#include <deque>
#include <set>
#include "webrtc/base/scoped_ptr.h"
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
#include "webrtc/test/testsupport/gtest_prod_util.h"
#endif
#include "webrtc/typedefs.h"
#include <stddef.h>
#include <stdint.h>
#include <memory>
namespace webrtc {
class TransientDetector;
// Detects transients in an audio stream and suppress them using a simple
// restoration algorithm that attenuates unexpected spikes in the spectrum.
class TransientSuppressor {
public:
TransientSuppressor();
~TransientSuppressor();
virtual ~TransientSuppressor() {}
int Initialize(int sample_rate_hz, int detector_rate_hz, int num_channels);
virtual int Initialize(int sample_rate_hz,
int detector_rate_hz,
int num_channels) = 0;
// Processes a |data| chunk, and returns it with keystrokes suppressed from
// it. The float format is assumed to be int16 ranged. If there are more than
@ -50,75 +44,17 @@ class TransientSuppressor {
// always be set to 1.
// |key_pressed| determines if a key was pressed on this audio chunk.
// Returns 0 on success and -1 otherwise.
int Suppress(float* data,
size_t data_length,
int num_channels,
const float* detection_data,
size_t detection_length,
const float* reference_data,
size_t reference_length,
float voice_probability,
bool key_pressed);
private:
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
FRIEND_TEST_ALL_PREFIXES(TransientSuppressorTest,
TypingDetectionLogicWorksAsExpectedForMono);
#endif
void Suppress(float* in_ptr, float* spectral_mean, float* out_ptr);
void UpdateKeypress(bool key_pressed);
void UpdateRestoration(float voice_probability);
void UpdateBuffers(float* data);
void HardRestoration(float* spectral_mean);
void SoftRestoration(float* spectral_mean);
rtc::scoped_ptr<TransientDetector> detector_;
size_t data_length_;
size_t detection_length_;
size_t analysis_length_;
size_t buffer_delay_;
size_t complex_analysis_length_;
int num_channels_;
// Input buffer where the original samples are stored.
rtc::scoped_ptr<float[]> in_buffer_;
rtc::scoped_ptr<float[]> detection_buffer_;
// Output buffer where the restored samples are stored.
rtc::scoped_ptr<float[]> out_buffer_;
// Arrays for fft.
rtc::scoped_ptr<size_t[]> ip_;
rtc::scoped_ptr<float[]> wfft_;
rtc::scoped_ptr<float[]> spectral_mean_;
// Stores the data for the fft.
rtc::scoped_ptr<float[]> fft_buffer_;
rtc::scoped_ptr<float[]> magnitudes_;
const float* window_;
rtc::scoped_ptr<float[]> mean_factor_;
float detector_smoothed_;
int keypress_counter_;
int chunks_since_keypress_;
bool detection_enabled_;
bool suppression_enabled_;
bool use_hard_restoration_;
int chunks_since_voice_change_;
uint32_t seed_;
bool using_reference_;
virtual int Suppress(float* data,
size_t data_length,
int num_channels,
const float* detection_data,
size_t detection_length,
const float* reference_data,
size_t reference_length,
float voice_probability,
bool key_pressed) = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_
#endif // MODULES_AUDIO_PROCESSING_TRANSIENT_TRANSIENT_SUPPRESSOR_H_