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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 |
11 #ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | 11 #ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ |
12 #define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | 12 #define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ |
13 | 13 |
14 #include <deque> | 14 #include <deque> |
15 #include <map> | 15 #include <map> |
| 16 #include <memory> |
16 #include <utility> | 17 #include <utility> |
17 | 18 |
18 #include "testing/gtest/include/gtest/gtest.h" | 19 #include "testing/gtest/include/gtest/gtest.h" |
19 #include "webrtc/base/basictypes.h" | 20 #include "webrtc/base/basictypes.h" |
20 #include "webrtc/base/criticalsection.h" | 21 #include "webrtc/base/criticalsection.h" |
21 #include "webrtc/base/platform_thread.h" | 22 #include "webrtc/base/platform_thread.h" |
22 #include "webrtc/base/scoped_ptr.h" | |
23 #include "webrtc/common_types.h" | 23 #include "webrtc/common_types.h" |
24 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 24 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
25 #include "webrtc/system_wrappers/include/event_wrapper.h" | 25 #include "webrtc/system_wrappers/include/event_wrapper.h" |
26 #include "webrtc/voice_engine/include/voe_base.h" | 26 #include "webrtc/voice_engine/include/voe_base.h" |
27 #include "webrtc/voice_engine/include/voe_codec.h" | 27 #include "webrtc/voice_engine/include/voe_codec.h" |
28 #include "webrtc/voice_engine/include/voe_file.h" | 28 #include "webrtc/voice_engine/include/voe_file.h" |
29 #include "webrtc/voice_engine/include/voe_network.h" | 29 #include "webrtc/voice_engine/include/voe_network.h" |
30 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | 30 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
31 #include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h" | 31 #include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h" |
32 | 32 |
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123 return static_cast<ConferenceTransport*>(transport)->DispatchPackets(); | 123 return static_cast<ConferenceTransport*>(transport)->DispatchPackets(); |
124 } | 124 } |
125 | 125 |
126 int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const; | 126 int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const; |
127 void StorePacket(Packet::Type type, const void* data, size_t len); | 127 void StorePacket(Packet::Type type, const void* data, size_t len); |
128 void SendPacket(const Packet& packet); | 128 void SendPacket(const Packet& packet); |
129 bool DispatchPackets(); | 129 bool DispatchPackets(); |
130 | 130 |
131 rtc::CriticalSection pq_crit_; | 131 rtc::CriticalSection pq_crit_; |
132 rtc::CriticalSection stream_crit_; | 132 rtc::CriticalSection stream_crit_; |
133 const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_; | 133 const std::unique_ptr<webrtc::EventWrapper> packet_event_; |
134 rtc::PlatformThread thread_; | 134 rtc::PlatformThread thread_; |
135 | 135 |
136 unsigned int rtt_ms_; | 136 unsigned int rtt_ms_; |
137 unsigned int stream_count_; | 137 unsigned int stream_count_; |
138 | 138 |
139 std::map<unsigned int, std::pair<int, int>> streams_ GUARDED_BY(stream_crit_); | 139 std::map<unsigned int, std::pair<int, int>> streams_ GUARDED_BY(stream_crit_); |
140 std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_); | 140 std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_); |
141 | 141 |
142 int local_sender_; // Channel Id of local sender | 142 int local_sender_; // Channel Id of local sender |
143 int reflector_; | 143 int reflector_; |
144 | 144 |
145 webrtc::VoiceEngine* local_voe_; | 145 webrtc::VoiceEngine* local_voe_; |
146 webrtc::VoEBase* local_base_; | 146 webrtc::VoEBase* local_base_; |
147 webrtc::VoERTP_RTCP* local_rtp_rtcp_; | 147 webrtc::VoERTP_RTCP* local_rtp_rtcp_; |
148 webrtc::VoENetwork* local_network_; | 148 webrtc::VoENetwork* local_network_; |
149 | 149 |
150 webrtc::VoiceEngine* remote_voe_; | 150 webrtc::VoiceEngine* remote_voe_; |
151 webrtc::VoEBase* remote_base_; | 151 webrtc::VoEBase* remote_base_; |
152 webrtc::VoECodec* remote_codec_; | 152 webrtc::VoECodec* remote_codec_; |
153 webrtc::VoERTP_RTCP* remote_rtp_rtcp_; | 153 webrtc::VoERTP_RTCP* remote_rtp_rtcp_; |
154 webrtc::VoENetwork* remote_network_; | 154 webrtc::VoENetwork* remote_network_; |
155 webrtc::VoEFile* remote_file_; | 155 webrtc::VoEFile* remote_file_; |
156 | 156 |
157 LoudestFilter loudest_filter_; | 157 LoudestFilter loudest_filter_; |
158 | 158 |
159 const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; | 159 const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; |
160 }; | 160 }; |
161 } // namespace voetest | 161 } // namespace voetest |
162 | 162 |
163 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ | 163 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ |
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