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

Issue 1362503003: Use suffixed {uint,int}{8,16,32,64}_t types. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: google::int32 Created 5 years, 3 months ago
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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
(...skipping 88 matching lines...) Expand 10 before | Expand all | Expand 10 after
99 99
100 // Inherit from class webrtc::Transport. 100 // Inherit from class webrtc::Transport.
101 int SendPacket(const void *data, size_t len) override; 101 int SendPacket(const void *data, size_t len) override;
102 int SendRTCPPacket(const void *data, size_t len) override; 102 int SendRTCPPacket(const void *data, size_t len) override;
103 103
104 private: 104 private:
105 struct Packet { 105 struct Packet {
106 enum Type { Rtp, Rtcp, } type_; 106 enum Type { Rtp, Rtcp, } type_;
107 107
108 Packet() : len_(0) {} 108 Packet() : len_(0) {}
109 Packet(Type type, const void* data, size_t len, uint32 time_ms) 109 Packet(Type type, const void* data, size_t len, uint32_t time_ms)
110 : type_(type), len_(len), send_time_ms_(time_ms) { 110 : type_(type), len_(len), send_time_ms_(time_ms) {
111 EXPECT_LE(len_, kMaxPacketSizeByte); 111 EXPECT_LE(len_, kMaxPacketSizeByte);
112 memcpy(data_, data, len_); 112 memcpy(data_, data, len_);
113 } 113 }
114 114
115 uint8_t data_[kMaxPacketSizeByte]; 115 uint8_t data_[kMaxPacketSizeByte];
116 size_t len_; 116 size_t len_;
117 uint32 send_time_ms_; 117 uint32_t send_time_ms_;
118 }; 118 };
119 119
120 static bool Run(void* transport) { 120 static bool Run(void* transport) {
121 return static_cast<ConferenceTransport*>(transport)->DispatchPackets(); 121 return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
122 } 122 }
123 123
124 int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const; 124 int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
125 void StorePacket(Packet::Type type, const void* data, size_t len); 125 void StorePacket(Packet::Type type, const void* data, size_t len);
126 void SendPacket(const Packet& packet); 126 void SendPacket(const Packet& packet);
127 bool DispatchPackets(); 127 bool DispatchPackets();
(...skipping 25 matching lines...) Expand all
153 webrtc::VoENetwork* remote_network_; 153 webrtc::VoENetwork* remote_network_;
154 webrtc::VoEFile* remote_file_; 154 webrtc::VoEFile* remote_file_;
155 155
156 LoudestFilter loudest_filter_; 156 LoudestFilter loudest_filter_;
157 157
158 const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; 158 const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
159 }; 159 };
160 } // namespace voetest 160 } // namespace voetest
161 161
162 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ 162 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
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