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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_ |
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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); |
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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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