Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(300)

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

Issue 2997393002: Move rtp dump writer from quality test to test transport (Closed)
Patch Set: Deps Created 3 years, 3 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
« no previous file with comments | « webrtc/test/layer_filtering_transport.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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_TEST_RTP_RTCP_OBSERVER_H_ 10 #ifndef WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
(...skipping 83 matching lines...) Expand 10 before | Expand all | Expand 10 after
94 94
95 PacketTransport(SingleThreadedTaskQueueForTesting* task_queue, 95 PacketTransport(SingleThreadedTaskQueueForTesting* task_queue,
96 Call* send_call, 96 Call* send_call,
97 RtpRtcpObserver* observer, 97 RtpRtcpObserver* observer,
98 TransportType transport_type, 98 TransportType transport_type,
99 const std::map<uint8_t, MediaType>& payload_type_map, 99 const std::map<uint8_t, MediaType>& payload_type_map,
100 const FakeNetworkPipe::Config& configuration) 100 const FakeNetworkPipe::Config& configuration)
101 : test::DirectTransport(task_queue, 101 : test::DirectTransport(task_queue,
102 configuration, 102 configuration,
103 send_call, 103 send_call,
104 payload_type_map), 104 payload_type_map,
105 std::unique_ptr<test::RtpFileWriter>()),
105 observer_(observer), 106 observer_(observer),
106 transport_type_(transport_type) {} 107 transport_type_(transport_type) {}
107 108
108 private: 109 private:
109 bool SendRtp(const uint8_t* packet, 110 bool SendRtp(const uint8_t* packet,
110 size_t length, 111 size_t length,
111 const PacketOptions& options) override { 112 const PacketOptions& options) override {
112 EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length)); 113 EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, length));
113 RtpRtcpObserver::Action action; 114 RtpRtcpObserver::Action action;
114 { 115 {
(...skipping 33 matching lines...) Expand 10 before | Expand all | Expand 10 after
148 return true; // Will never happen, makes compiler happy. 149 return true; // Will never happen, makes compiler happy.
149 } 150 }
150 151
151 RtpRtcpObserver* const observer_; 152 RtpRtcpObserver* const observer_;
152 TransportType transport_type_; 153 TransportType transport_type_;
153 }; 154 };
154 } // namespace test 155 } // namespace test
155 } // namespace webrtc 156 } // namespace webrtc
156 157
157 #endif // WEBRTC_TEST_RTP_RTCP_OBSERVER_H_ 158 #endif // WEBRTC_TEST_RTP_RTCP_OBSERVER_H_
OLDNEW
« no previous file with comments | « webrtc/test/layer_filtering_transport.cc ('k') | webrtc/video/end_to_end_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698