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Side by Side Diff: webrtc/test/rtp_rtcp_observer.h

Issue 1376673004: Add a PacketOptions struct to webrtc::Transport. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Comment added Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 #ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_ 10 #ifndef WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
(...skipping 108 matching lines...) Expand 10 before | Expand all | Expand 10 after
119 PacketTransportAction on_rtp, 119 PacketTransportAction on_rtp,
120 PacketTransportAction on_rtcp, 120 PacketTransportAction on_rtcp,
121 const FakeNetworkPipe::Config& configuration) 121 const FakeNetworkPipe::Config& configuration)
122 : test::DirectTransport(configuration), 122 : test::DirectTransport(configuration),
123 crit_(lock), 123 crit_(lock),
124 observer_(observer), 124 observer_(observer),
125 on_rtp_(on_rtp), 125 on_rtp_(on_rtp),
126 on_rtcp_(on_rtcp) {} 126 on_rtcp_(on_rtcp) {}
127 127
128 private: 128 private:
129 bool SendRtp(const uint8_t* packet, size_t length) override { 129 bool SendRtp(const uint8_t* packet,
130 size_t length,
131 const PacketOptions& options) override {
130 EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length)); 132 EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
131 Action action; 133 Action action;
132 { 134 {
133 rtc::CritScope lock(crit_); 135 rtc::CritScope lock(crit_);
134 action = (observer_->*on_rtp_)(packet, length); 136 action = (observer_->*on_rtp_)(packet, length);
135 } 137 }
136 switch (action) { 138 switch (action) {
137 case DROP_PACKET: 139 case DROP_PACKET:
138 // Drop packet silently. 140 // Drop packet silently.
139 return true; 141 return true;
140 case SEND_PACKET: 142 case SEND_PACKET:
141 return test::DirectTransport::SendRtp(packet, length); 143 return test::DirectTransport::SendRtp(packet, length, options);
142 } 144 }
143 return true; // Will never happen, makes compiler happy. 145 return true; // Will never happen, makes compiler happy.
144 } 146 }
145 147
146 bool SendRtcp(const uint8_t* packet, size_t length) override { 148 bool SendRtcp(const uint8_t* packet, size_t length) override {
147 EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length)); 149 EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, length));
148 Action action; 150 Action action;
149 { 151 {
150 rtc::CritScope lock(crit_); 152 rtc::CritScope lock(crit_);
151 action = (observer_->*on_rtcp_)(packet, length); 153 action = (observer_->*on_rtcp_)(packet, length);
(...skipping 21 matching lines...) Expand all
173 const rtc::scoped_ptr<RtpHeaderParser> parser_; 175 const rtc::scoped_ptr<RtpHeaderParser> parser_;
174 PacketTransport send_transport_, receive_transport_; 176 PacketTransport send_transport_, receive_transport_;
175 177
176 private: 178 private:
177 unsigned int timeout_ms_; 179 unsigned int timeout_ms_;
178 }; 180 };
179 } // namespace test 181 } // namespace test
180 } // namespace webrtc 182 } // namespace webrtc
181 183
182 #endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_ 184 #endif // WEBRTC_VIDEO_ENGINE_TEST_COMMON_RTP_RTCP_OBSERVER_H_
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