Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(134)

Side by Side Diff: webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

Issue 1743543003: RtpRtcp allows to register header extension by name (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 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
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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
11 #ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
12 #define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
13 13
14 #include <set> 14 #include <set>
15 #include <string>
15 #include <utility> 16 #include <utility>
16 #include <vector> 17 #include <vector>
17 18
18 #include "testing/gmock/include/gmock/gmock.h" 19 #include "testing/gmock/include/gmock/gmock.h"
19 20
20 #include "webrtc/modules/include/module.h" 21 #include "webrtc/modules/include/module.h"
21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 23 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
23 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" 24 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
24 25
(...skipping 46 matching lines...) Expand 10 before | Expand all | Expand 10 after
71 MOCK_CONST_METHOD0(MaxDataPayloadLength, 72 MOCK_CONST_METHOD0(MaxDataPayloadLength,
72 uint16_t()); 73 uint16_t());
73 MOCK_METHOD1(RegisterSendPayload, 74 MOCK_METHOD1(RegisterSendPayload,
74 int32_t(const CodecInst& voiceCodec)); 75 int32_t(const CodecInst& voiceCodec));
75 MOCK_METHOD1(RegisterSendPayload, 76 MOCK_METHOD1(RegisterSendPayload,
76 int32_t(const VideoCodec& videoCodec)); 77 int32_t(const VideoCodec& videoCodec));
77 MOCK_METHOD1(DeRegisterSendPayload, 78 MOCK_METHOD1(DeRegisterSendPayload,
78 int32_t(const int8_t payloadType)); 79 int32_t(const int8_t payloadType));
79 MOCK_METHOD2(RegisterSendRtpHeaderExtension, 80 MOCK_METHOD2(RegisterSendRtpHeaderExtension,
80 int32_t(const RTPExtensionType type, const uint8_t id)); 81 int32_t(const RTPExtensionType type, const uint8_t id));
82 MOCK_METHOD2(RegisterRtpHeaderExtension,
83 bool(const std::string& type, uint8_t id));
81 MOCK_METHOD1(DeregisterSendRtpHeaderExtension, 84 MOCK_METHOD1(DeregisterSendRtpHeaderExtension,
82 int32_t(const RTPExtensionType type)); 85 int32_t(const RTPExtensionType type));
83 MOCK_CONST_METHOD0(StartTimestamp, 86 MOCK_CONST_METHOD0(StartTimestamp,
84 uint32_t()); 87 uint32_t());
85 MOCK_METHOD1(SetStartTimestamp, void(const uint32_t timestamp)); 88 MOCK_METHOD1(SetStartTimestamp, void(const uint32_t timestamp));
86 MOCK_CONST_METHOD0(SequenceNumber, 89 MOCK_CONST_METHOD0(SequenceNumber,
87 uint16_t()); 90 uint16_t());
88 MOCK_METHOD1(SetSequenceNumber, void(const uint16_t seq)); 91 MOCK_METHOD1(SetSequenceNumber, void(const uint16_t seq));
89 MOCK_METHOD2(SetRtpStateForSsrc, 92 MOCK_METHOD2(SetRtpStateForSsrc,
90 bool(uint32_t ssrc, const RtpState& rtp_state)); 93 bool(uint32_t ssrc, const RtpState& rtp_state));
(...skipping 167 matching lines...) Expand 10 before | Expand all | Expand 10 after
258 void(StreamDataCountersCallback*)); 261 void(StreamDataCountersCallback*));
259 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, 262 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
260 StreamDataCountersCallback*(void)); 263 StreamDataCountersCallback*(void));
261 // Members. 264 // Members.
262 unsigned int remote_ssrc_; 265 unsigned int remote_ssrc_;
263 }; 266 };
264 267
265 } // namespace webrtc 268 } // namespace webrtc
266 269
267 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ 270 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/include/rtp_rtcp.h ('k') | webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698