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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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| 124 bool ApplyOptions(const AudioOptions& options); | 124 bool ApplyOptions(const AudioOptions& options); |
| 125 void SetDefaultDevices(); | 125 void SetDefaultDevices(); |
| 126 | 126 |
| 127 // webrtc::TraceCallback: | 127 // webrtc::TraceCallback: |
| 128 void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 128 void Print(webrtc::TraceLevel level, const char* trace, int length) override; |
| 129 | 129 |
| 130 void StartAecDump(const std::string& filename); | 130 void StartAecDump(const std::string& filename); |
| 131 int CreateVoEChannel(); | 131 int CreateVoEChannel(); |
| 132 webrtc::AudioDeviceModule* adm(); | 132 webrtc::AudioDeviceModule* adm(); |
| 133 | 133 |
| 134 AudioCodecs CollectRecvCodecs() const; | |
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ivoc
2016/07/05 14:43:51
I would prefer not to abbreviate "Recv" in the met
ossu
2016/07/05 15:31:57
It's done eeeeverywhere, except where it isn't. :)
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| 135 | |
| 134 rtc::ThreadChecker signal_thread_checker_; | 136 rtc::ThreadChecker signal_thread_checker_; |
| 135 rtc::ThreadChecker worker_thread_checker_; | 137 rtc::ThreadChecker worker_thread_checker_; |
| 136 | 138 |
| 137 // The audio device manager. | 139 // The audio device manager. |
| 138 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; | 140 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; |
| 139 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; | 141 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; |
| 140 // The primary instance of WebRtc VoiceEngine. | 142 // The primary instance of WebRtc VoiceEngine. |
| 141 std::unique_ptr<VoEWrapper> voe_wrapper_; | 143 std::unique_ptr<VoEWrapper> voe_wrapper_; |
| 142 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 144 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
| 143 std::vector<AudioCodec> codecs_; | 145 std::vector<AudioCodec> send_codecs_; |
| 146 std::vector<AudioCodec> recv_codecs_; | |
| 144 std::vector<WebRtcVoiceMediaChannel*> channels_; | 147 std::vector<WebRtcVoiceMediaChannel*> channels_; |
| 145 webrtc::Config voe_config_; | 148 webrtc::Config voe_config_; |
| 146 bool is_dumping_aec_ = false; | 149 bool is_dumping_aec_ = false; |
| 147 | 150 |
| 148 webrtc::AgcConfig default_agc_config_; | 151 webrtc::AgcConfig default_agc_config_; |
| 149 // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns and | 152 // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns and |
| 150 // intelligibility_enhancer values, and apply them in case they are missing | 153 // intelligibility_enhancer values, and apply them in case they are missing |
| 151 // in the audio options. We need to do this because SetExtraOptions() will | 154 // in the audio options. We need to do this because SetExtraOptions() will |
| 152 // revert to defaults for options which are not provided. | 155 // revert to defaults for options which are not provided. |
| 153 rtc::Optional<bool> extended_filter_aec_; | 156 rtc::Optional<bool> extended_filter_aec_; |
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| 302 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 305 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
| 303 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 306 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
| 304 | 307 |
| 305 SendCodecSpec send_codec_spec_; | 308 SendCodecSpec send_codec_spec_; |
| 306 | 309 |
| 307 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 310 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
| 308 }; | 311 }; |
| 309 } // namespace cricket | 312 } // namespace cricket |
| 310 | 313 |
| 311 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 314 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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