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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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| 116   bool ApplyOptions(const AudioOptions& options); | 116   bool ApplyOptions(const AudioOptions& options); | 
| 117   void SetDefaultDevices(); | 117   void SetDefaultDevices(); | 
| 118 | 118 | 
| 119   // webrtc::TraceCallback: | 119   // webrtc::TraceCallback: | 
| 120   void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 120   void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 
| 121 | 121 | 
| 122   void StartAecDump(const std::string& filename); | 122   void StartAecDump(const std::string& filename); | 
| 123   int CreateVoEChannel(); | 123   int CreateVoEChannel(); | 
| 124   webrtc::AudioDeviceModule* adm(); | 124   webrtc::AudioDeviceModule* adm(); | 
| 125 | 125 | 
| 126   AudioCodecs CollectRecvCodecs() const; |  | 
| 127 |  | 
| 128   rtc::ThreadChecker signal_thread_checker_; | 126   rtc::ThreadChecker signal_thread_checker_; | 
| 129   rtc::ThreadChecker worker_thread_checker_; | 127   rtc::ThreadChecker worker_thread_checker_; | 
| 130 | 128 | 
| 131   // The audio device manager. | 129   // The audio device manager. | 
| 132   rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; | 130   rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; | 
| 133   rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; | 131   rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; | 
| 134   // The primary instance of WebRtc VoiceEngine. | 132   // The primary instance of WebRtc VoiceEngine. | 
| 135   std::unique_ptr<VoEWrapper> voe_wrapper_; | 133   std::unique_ptr<VoEWrapper> voe_wrapper_; | 
| 136   rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 134   rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 
| 137   std::vector<AudioCodec> send_codecs_; | 135   std::vector<AudioCodec> codecs_; | 
| 138   std::vector<AudioCodec> recv_codecs_; |  | 
| 139   std::vector<WebRtcVoiceMediaChannel*> channels_; | 136   std::vector<WebRtcVoiceMediaChannel*> channels_; | 
| 140   webrtc::Config voe_config_; | 137   webrtc::Config voe_config_; | 
| 141   bool is_dumping_aec_ = false; | 138   bool is_dumping_aec_ = false; | 
| 142 | 139 | 
| 143   webrtc::AgcConfig default_agc_config_; | 140   webrtc::AgcConfig default_agc_config_; | 
| 144   // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns | 141   // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns | 
| 145   // level controller, and intelligibility_enhancer values, and apply them | 142   // level controller, and intelligibility_enhancer values, and apply them | 
| 146   // in case they are missing in the audio options. We need to do this because | 143   // in case they are missing in the audio options. We need to do this because | 
| 147   // SetExtraOptions() will revert to defaults for options which are not | 144   // SetExtraOptions() will revert to defaults for options which are not | 
| 148   // provided. | 145   // provided. | 
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| 299   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 296   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 
| 300   std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 297   std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 
| 301 | 298 | 
| 302   SendCodecSpec send_codec_spec_; | 299   SendCodecSpec send_codec_spec_; | 
| 303 | 300 | 
| 304   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 301   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 
| 305 }; | 302 }; | 
| 306 }  // namespace cricket | 303 }  // namespace cricket | 
| 307 | 304 | 
| 308 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 305 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 
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