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