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1 /* | 1 /* |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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130 bool DispatchPackets(); | 130 bool DispatchPackets(); |
131 | 131 |
132 rtc::CriticalSection pq_crit_; | 132 rtc::CriticalSection pq_crit_; |
133 rtc::CriticalSection stream_crit_; | 133 rtc::CriticalSection stream_crit_; |
134 const std::unique_ptr<webrtc::EventWrapper> packet_event_; | 134 const std::unique_ptr<webrtc::EventWrapper> packet_event_; |
135 rtc::PlatformThread thread_; | 135 rtc::PlatformThread thread_; |
136 | 136 |
137 unsigned int rtt_ms_; | 137 unsigned int rtt_ms_; |
138 unsigned int stream_count_; | 138 unsigned int stream_count_; |
139 | 139 |
140 std::map<unsigned int, std::pair<int, int>> streams_ GUARDED_BY(stream_crit_); | 140 std::map<unsigned int, std::pair<int, int>> streams_ |
141 std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_); | 141 RTC_GUARDED_BY(stream_crit_); |
| 142 std::deque<Packet> packet_queue_ RTC_GUARDED_BY(pq_crit_); |
142 | 143 |
143 int local_sender_; // Channel Id of local sender | 144 int local_sender_; // Channel Id of local sender |
144 int reflector_; | 145 int reflector_; |
145 | 146 |
146 webrtc::VoiceEngine* local_voe_; | 147 webrtc::VoiceEngine* local_voe_; |
147 webrtc::VoEBase* local_base_; | 148 webrtc::VoEBase* local_base_; |
148 webrtc::VoERTP_RTCP* local_rtp_rtcp_; | 149 webrtc::VoERTP_RTCP* local_rtp_rtcp_; |
149 webrtc::VoENetwork* local_network_; | 150 webrtc::VoENetwork* local_network_; |
150 rtc::scoped_refptr<webrtc::AudioProcessing> local_apm_; | 151 rtc::scoped_refptr<webrtc::AudioProcessing> local_apm_; |
151 | 152 |
152 webrtc::VoiceEngine* remote_voe_; | 153 webrtc::VoiceEngine* remote_voe_; |
153 webrtc::VoEBase* remote_base_; | 154 webrtc::VoEBase* remote_base_; |
154 webrtc::VoECodec* remote_codec_; | 155 webrtc::VoECodec* remote_codec_; |
155 webrtc::VoERTP_RTCP* remote_rtp_rtcp_; | 156 webrtc::VoERTP_RTCP* remote_rtp_rtcp_; |
156 webrtc::VoENetwork* remote_network_; | 157 webrtc::VoENetwork* remote_network_; |
157 webrtc::VoEFile* remote_file_; | 158 webrtc::VoEFile* remote_file_; |
158 rtc::scoped_refptr<webrtc::AudioProcessing> remote_apm_; | 159 rtc::scoped_refptr<webrtc::AudioProcessing> remote_apm_; |
159 LoudestFilter loudest_filter_; | 160 LoudestFilter loudest_filter_; |
160 | 161 |
161 const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; | 162 const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; |
162 }; | 163 }; |
163 | 164 |
164 } // namespace voetest | 165 } // namespace voetest |
165 } // namespace webrtc | 166 } // namespace webrtc |
166 | 167 |
167 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | 168 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ |
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