Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(816)

Side by Side Diff: webrtc/modules/audio_processing/aec3/aec_state.h

Issue 2784023002: Major AEC3 render pipeline changes (Closed)
Patch Set: Disabled one more DEATH test that caused issues due to bug on bots Created 3 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
13 13
14 #include <algorithm> 14 #include <algorithm>
15 #include <memory> 15 #include <memory>
16 #include <vector> 16 #include <vector>
17 17
18 #include "webrtc/base/array_view.h" 18 #include "webrtc/base/array_view.h"
19 #include "webrtc/base/constructormagic.h" 19 #include "webrtc/base/constructormagic.h"
20 #include "webrtc/base/optional.h" 20 #include "webrtc/base/optional.h"
21 #include "webrtc/modules/audio_processing/aec3/aec3_common.h" 21 #include "webrtc/modules/audio_processing/aec3/aec3_common.h"
22 #include "webrtc/modules/audio_processing/aec3/echo_path_variability.h" 22 #include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
23 #include "webrtc/modules/audio_processing/aec3/erl_estimator.h"
23 #include "webrtc/modules/audio_processing/aec3/erle_estimator.h" 24 #include "webrtc/modules/audio_processing/aec3/erle_estimator.h"
24 #include "webrtc/modules/audio_processing/aec3/erl_estimator.h" 25 #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
25 #include "webrtc/modules/audio_processing/aec3/fft_buffer.h"
26 26
27 namespace webrtc { 27 namespace webrtc {
28 28
29 class ApmDataDumper; 29 class ApmDataDumper;
30 30
31 // Handles the state and the conditions for the echo removal functionality. 31 // Handles the state and the conditions for the echo removal functionality.
32 class AecState { 32 class AecState {
33 public: 33 public:
34 AecState(); 34 AecState();
35 ~AecState(); 35 ~AecState();
(...skipping 54 matching lines...) Expand 10 before | Expand all | Expand 10 after
90 capture_signal_saturation_ = capture_signal_saturation; 90 capture_signal_saturation_ = capture_signal_saturation;
91 } 91 }
92 92
93 // Returns whether a probable headset setup has been detected. 93 // Returns whether a probable headset setup has been detected.
94 bool HeadsetDetected() const { return headset_detected_; } 94 bool HeadsetDetected() const { return headset_detected_; }
95 95
96 // Updates the aec state. 96 // Updates the aec state.
97 void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>& 97 void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
98 filter_frequency_response, 98 filter_frequency_response,
99 const rtc::Optional<size_t>& external_delay_samples, 99 const rtc::Optional<size_t>& external_delay_samples,
100 const FftBuffer& X_buffer, 100 const RenderBuffer& X_buffer,
101 const std::array<float, kFftLengthBy2Plus1>& E2_main, 101 const std::array<float, kFftLengthBy2Plus1>& E2_main,
102 const std::array<float, kFftLengthBy2Plus1>& E2_shadow, 102 const std::array<float, kFftLengthBy2Plus1>& E2_shadow,
103 const std::array<float, kFftLengthBy2Plus1>& Y2, 103 const std::array<float, kFftLengthBy2Plus1>& Y2,
104 rtc::ArrayView<const float> x, 104 rtc::ArrayView<const float> x,
105 const EchoPathVariability& echo_path_variability, 105 const EchoPathVariability& echo_path_variability,
106 bool echo_leakage_detected); 106 bool echo_leakage_detected);
107 107
108 private: 108 private:
109 static int instance_count_; 109 static int instance_count_;
110 std::unique_ptr<ApmDataDumper> data_dumper_; 110 std::unique_ptr<ApmDataDumper> data_dumper_;
(...skipping 12 matching lines...) Expand all
123 std::array<bool, kFftLengthBy2Plus1> bands_with_reliable_filter_; 123 std::array<bool, kFftLengthBy2Plus1> bands_with_reliable_filter_;
124 std::array<float, kFftLengthBy2Plus1> filter_estimate_strength_; 124 std::array<float, kFftLengthBy2Plus1> filter_estimate_strength_;
125 size_t filter_length_; 125 size_t filter_length_;
126 126
127 RTC_DISALLOW_COPY_AND_ASSIGN(AecState); 127 RTC_DISALLOW_COPY_AND_ASSIGN(AecState);
128 }; 128 };
129 129
130 } // namespace webrtc 130 } // namespace webrtc
131 131
132 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_ 132 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
OLDNEW
« no previous file with comments | « webrtc/modules/audio_processing/aec3/aec3_common.h ('k') | webrtc/modules/audio_processing/aec3/aec_state.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698