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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: rebase + revert basictypes.h (to be landed separately just in case of a revert due to unexpected us… Created 5 years, 2 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
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101 bool SendRtp(const uint8_t* data, 101 bool SendRtp(const uint8_t* data,
102 size_t len, 102 size_t len,
103 const webrtc::PacketOptions& options) override; 103 const webrtc::PacketOptions& options) override;
104 bool SendRtcp(const uint8_t *data, size_t len) override; 104 bool SendRtcp(const uint8_t *data, size_t len) override;
105 105
106 private: 106 private:
107 struct Packet { 107 struct Packet {
108 enum Type { Rtp, Rtcp, } type_; 108 enum Type { Rtp, Rtcp, } type_;
109 109
110 Packet() : len_(0) {} 110 Packet() : len_(0) {}
111 Packet(Type type, const void* data, size_t len, uint32 time_ms) 111 Packet(Type type, const void* data, size_t len, uint32_t time_ms)
112 : type_(type), len_(len), send_time_ms_(time_ms) { 112 : type_(type), len_(len), send_time_ms_(time_ms) {
113 EXPECT_LE(len_, kMaxPacketSizeByte); 113 EXPECT_LE(len_, kMaxPacketSizeByte);
114 memcpy(data_, data, len_); 114 memcpy(data_, data, len_);
115 } 115 }
116 116
117 uint8_t data_[kMaxPacketSizeByte]; 117 uint8_t data_[kMaxPacketSizeByte];
118 size_t len_; 118 size_t len_;
119 uint32 send_time_ms_; 119 uint32_t send_time_ms_;
120 }; 120 };
121 121
122 static bool Run(void* transport) { 122 static bool Run(void* transport) {
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();
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155 webrtc::VoENetwork* remote_network_; 155 webrtc::VoENetwork* remote_network_;
156 webrtc::VoEFile* remote_file_; 156 webrtc::VoEFile* remote_file_;
157 157
158 LoudestFilter loudest_filter_; 158 LoudestFilter loudest_filter_;
159 159
160 const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; 160 const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
161 }; 161 };
162 } // namespace voetest 162 } // namespace voetest
163 163
164 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ 164 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
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