OLD | NEW |
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 |
(...skipping 84 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
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 |
(...skipping 168 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
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_ |
OLD | NEW |