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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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| 95 | 95 |
| 96 const webrtc::AudioProcessing::Config& GetApmConfigForTest() const { | 96 const webrtc::AudioProcessing::Config& GetApmConfigForTest() const { |
| 97 return apm_config_; | 97 return apm_config_; |
| 98 } | 98 } |
| 99 | 99 |
| 100 private: | 100 private: |
| 101 // Every option that is "set" will be applied. Every option not "set" will be | 101 // Every option that is "set" will be applied. Every option not "set" will be |
| 102 // ignored. This allows us to selectively turn on and off different options | 102 // ignored. This allows us to selectively turn on and off different options |
| 103 // easily at any time. | 103 // easily at any time. |
| 104 bool ApplyOptions(const AudioOptions& options); | 104 bool ApplyOptions(const AudioOptions& options); |
| 105 void SetDefaultDevices(); | |
| 106 | 105 |
| 107 // webrtc::TraceCallback: | 106 // webrtc::TraceCallback: |
| 108 void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 107 void Print(webrtc::TraceLevel level, const char* trace, int length) override; |
| 109 | 108 |
| 110 void StartAecDump(const std::string& filename); | 109 void StartAecDump(const std::string& filename); |
| 111 int CreateVoEChannel(); | 110 int CreateVoEChannel(); |
| 112 webrtc::AudioDeviceModule* adm(); | 111 webrtc::AudioDeviceModule* adm(); |
| 113 webrtc::AudioProcessing* apm(); | 112 webrtc::AudioProcessing* apm(); |
| 114 webrtc::voe::TransmitMixer* transmit_mixer(); | 113 webrtc::voe::TransmitMixer* transmit_mixer(); |
| 115 | 114 |
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| 284 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 283 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
| 285 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 284 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
| 286 | 285 |
| 287 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; | 286 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; |
| 288 | 287 |
| 289 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 288 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
| 290 }; | 289 }; |
| 291 } // namespace cricket | 290 } // namespace cricket |
| 292 | 291 |
| 293 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 292 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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