Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(139)

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

Issue 1290813008: Add RemoteEstimatorProxy for capturing receive times (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed borked Rebase Created 5 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
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 "testing/gmock/include/gmock/gmock.h" 14 #include "testing/gmock/include/gmock/gmock.h"
15 15
16 #include "webrtc/modules/interface/module.h" 16 #include "webrtc/modules/interface/module.h"
17 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 17 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
19 20
20 namespace webrtc { 21 namespace webrtc {
21 22
22 class MockRtpData : public RtpData { 23 class MockRtpData : public RtpData {
23 public: 24 public:
24 MOCK_METHOD3(OnReceivedPayloadData, 25 MOCK_METHOD3(OnReceivedPayloadData,
25 int32_t(const uint8_t* payloadData, 26 int32_t(const uint8_t* payloadData,
26 const size_t payloadSize, 27 const size_t payloadSize,
27 const WebRtcRTPHeader* rtpHeader)); 28 const WebRtcRTPHeader* rtpHeader));
28 29
(...skipping 169 matching lines...) Expand 10 before | Expand all | Expand 10 after
198 int()); 199 int());
199 MOCK_METHOD1(SetSelectiveRetransmissions, 200 MOCK_METHOD1(SetSelectiveRetransmissions,
200 int(uint8_t settings)); 201 int(uint8_t settings));
201 MOCK_METHOD2(SendNACK, 202 MOCK_METHOD2(SendNACK,
202 int32_t(const uint16_t* nackList, const uint16_t size)); 203 int32_t(const uint16_t* nackList, const uint16_t size));
203 MOCK_METHOD2(SetStorePacketsStatus, 204 MOCK_METHOD2(SetStorePacketsStatus,
204 void(const bool enable, const uint16_t numberToStore)); 205 void(const bool enable, const uint16_t numberToStore));
205 MOCK_CONST_METHOD0(StorePackets, bool()); 206 MOCK_CONST_METHOD0(StorePackets, bool());
206 MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*)); 207 MOCK_METHOD1(RegisterRtcpStatisticsCallback, void(RtcpStatisticsCallback*));
207 MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*()); 208 MOCK_METHOD0(GetRtcpStatisticsCallback, RtcpStatisticsCallback*());
209 MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
208 MOCK_METHOD1(RegisterAudioCallback, 210 MOCK_METHOD1(RegisterAudioCallback,
209 int32_t(RtpAudioFeedback* messagesCallback)); 211 int32_t(RtpAudioFeedback* messagesCallback));
210 MOCK_METHOD1(SetAudioPacketSize, 212 MOCK_METHOD1(SetAudioPacketSize,
211 int32_t(const uint16_t packetSizeSamples)); 213 int32_t(const uint16_t packetSizeSamples));
212 MOCK_METHOD3(SendTelephoneEventOutband, 214 MOCK_METHOD3(SendTelephoneEventOutband,
213 int32_t(const uint8_t key, const uint16_t time_ms, const uint8_t level)); 215 int32_t(const uint8_t key, const uint16_t time_ms, const uint8_t level));
214 MOCK_METHOD1(SetSendREDPayloadType, 216 MOCK_METHOD1(SetSendREDPayloadType,
215 int32_t(const int8_t payloadType)); 217 int32_t(const int8_t payloadType));
216 MOCK_CONST_METHOD1(SendREDPayloadType, 218 MOCK_CONST_METHOD1(SendREDPayloadType,
217 int32_t(int8_t& payloadType)); 219 int32_t(int8_t& payloadType));
(...skipping 34 matching lines...) Expand 10 before | Expand all | Expand 10 after
252 void(StreamDataCountersCallback*)); 254 void(StreamDataCountersCallback*));
253 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, 255 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
254 StreamDataCountersCallback*(void)); 256 StreamDataCountersCallback*(void));
255 // Members. 257 // Members.
256 unsigned int remote_ssrc_; 258 unsigned int remote_ssrc_;
257 }; 259 };
258 260
259 } // namespace webrtc 261 } // namespace webrtc
260 262
261 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ 263 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h ('k') | webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698