Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(102)

Side by Side Diff: webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

Issue 1702983002: Replace scoped_ptr with unique_ptr in webrtc/voice_engine/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
(...skipping 90 matching lines...) Expand 10 before | Expand all | Expand 10 after
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_
OLDNEW
« no previous file with comments | « webrtc/voice_engine/shared_data.cc ('k') | webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698