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Side by Side Diff: webrtc/media/engine/webrtcvoiceengine.h

Issue 2415403002: Introduced the new parameter setting scheme for activating the high-pass filter in APM (Closed)
Patch Set: Changes in response to reviewer comments Created 4 years ago
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1 /* 1 /*
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2004 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
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82 82
83 // Starts AEC dump using an existing file. A maximum file size in bytes can be 83 // Starts AEC dump using an existing file. A maximum file size in bytes can be
84 // specified. When the maximum file size is reached, logging is stopped and 84 // specified. When the maximum file size is reached, logging is stopped and
85 // the file is closed. If max_size_bytes is set to <= 0, no limit will be 85 // the file is closed. If max_size_bytes is set to <= 0, no limit will be
86 // used. 86 // used.
87 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes); 87 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
88 88
89 // Stops AEC dump. 89 // Stops AEC dump.
90 void StopAecDump(); 90 void StopAecDump();
91 91
92 const webrtc::AudioProcessing::Config& GetApmConfigForTest() const {
93 return apm_config_;
94 }
95
92 private: 96 private:
93 // Every option that is "set" will be applied. Every option not "set" will be 97 // Every option that is "set" will be applied. Every option not "set" will be
94 // ignored. This allows us to selectively turn on and off different options 98 // ignored. This allows us to selectively turn on and off different options
95 // easily at any time. 99 // easily at any time.
96 bool ApplyOptions(const AudioOptions& options); 100 bool ApplyOptions(const AudioOptions& options);
97 void SetDefaultDevices(); 101 void SetDefaultDevices();
98 102
99 // webrtc::TraceCallback: 103 // webrtc::TraceCallback:
100 void Print(webrtc::TraceLevel level, const char* trace, int length) override; 104 void Print(webrtc::TraceLevel level, const char* trace, int length) override;
101 105
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275 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; 279 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
276 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 280 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
277 281
278 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; 282 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;
279 283
280 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); 284 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
281 }; 285 };
282 } // namespace cricket 286 } // namespace cricket
283 287
284 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ 288 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
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