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

Issue 2967603002: Added the ability to adjust the AEC3 performance for large rooms (Closed)
Patch Set: Changes in response to reviewer commments Created 3 years, 5 months ago
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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_ECHO_REMOVER_H_ 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
13 13
14 #include <vector> 14 #include <vector>
15 15
16 #include "webrtc/base/optional.h" 16 #include "webrtc/base/optional.h"
17 #include "webrtc/modules/audio_processing/aec3/echo_path_variability.h" 17 #include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
18 #include "webrtc/modules/audio_processing/aec3/render_buffer.h" 18 #include "webrtc/modules/audio_processing/aec3/render_buffer.h"
19 #include "webrtc/modules/audio_processing/include/audio_processing.h"
19 20
20 namespace webrtc { 21 namespace webrtc {
21 22
22 // Class for removing the echo from the capture signal. 23 // Class for removing the echo from the capture signal.
23 class EchoRemover { 24 class EchoRemover {
24 public: 25 public:
25 static EchoRemover* Create(int sample_rate_hz); 26 static EchoRemover* Create(
27 const AudioProcessing::Config::EchoCanceller3& config,
28 int sample_rate_hz);
26 virtual ~EchoRemover() = default; 29 virtual ~EchoRemover() = default;
27 30
28 // Removes the echo from a block of samples from the capture signal. The 31 // Removes the echo from a block of samples from the capture signal. The
29 // supplied render signal is assumed to be pre-aligned with the capture 32 // supplied render signal is assumed to be pre-aligned with the capture
30 // signal. 33 // signal.
31 virtual void ProcessCapture( 34 virtual void ProcessCapture(
32 const rtc::Optional<size_t>& echo_path_delay_samples, 35 const rtc::Optional<size_t>& echo_path_delay_samples,
33 const EchoPathVariability& echo_path_variability, 36 const EchoPathVariability& echo_path_variability,
34 bool capture_signal_saturation, 37 bool capture_signal_saturation,
35 const RenderBuffer& render_buffer, 38 const RenderBuffer& render_buffer,
36 std::vector<std::vector<float>>* capture) = 0; 39 std::vector<std::vector<float>>* capture) = 0;
37 40
38 // Updates the status on whether echo leakage is detected in the output of the 41 // Updates the status on whether echo leakage is detected in the output of the
39 // echo remover. 42 // echo remover.
40 virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0; 43 virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0;
41 }; 44 };
42 45
43 } // namespace webrtc 46 } // namespace webrtc
44 47
45 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_ 48 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_H_
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