Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(63)

Side by Side Diff: webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc

Issue 1268383002: Use RtcpPacket to send SLI in RtcpSender (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 4 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/modules/rtp_rtcp/source/rtcp_sender.cc ('k') | no next file » | 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) 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 #include <algorithm> 11 #include <algorithm>
12 #include <vector> 12 #include <vector>
13 13
14 #include "testing/gmock/include/gmock/gmock.h" 14 #include "testing/gmock/include/gmock/gmock.h"
15 #include "testing/gtest/include/gtest/gtest.h" 15 #include "testing/gtest/include/gtest/gtest.h"
16 #include "webrtc/common_types.h" 16 #include "webrtc/common_types.h"
17 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" 17 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
19 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 19 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
20 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h" 20 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
21 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h" 21 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
22 22
23 using namespace webrtc; 23 using namespace webrtc;
24 24
25 const uint64_t kTestPictureId = 12345678; 25 const uint64_t kTestPictureId = 12345678;
26 const uint8_t kSliPictureId = 156;
26 27
27 class RtcpCallback : public RtcpIntraFrameObserver { 28 class RtcpCallback : public RtcpIntraFrameObserver {
28 public: 29 public:
29 void SetModule(RtpRtcp* module) { 30 void SetModule(RtpRtcp* module) {
30 _rtpRtcpModule = module; 31 _rtpRtcpModule = module;
31 }; 32 };
32 virtual void OnRTCPPacketTimeout(const int32_t id) { 33 virtual void OnRTCPPacketTimeout(const int32_t id) {
33 } 34 }
34 virtual void OnLipSyncUpdate(const int32_t id, 35 virtual void OnLipSyncUpdate(const int32_t id,
35 const int32_t audioVideoOffset) { 36 const int32_t audioVideoOffset) {
36 }; 37 };
37 virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) { 38 virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {
38 }; 39 };
39 virtual void OnReceivedSLI(uint32_t ssrc, 40 virtual void OnReceivedSLI(uint32_t ssrc,
40 uint8_t pictureId) { 41 uint8_t pictureId) {
41 EXPECT_EQ(28, pictureId); 42 EXPECT_EQ(kSliPictureId & 0x3f, pictureId);
42 }; 43 };
43 virtual void OnReceivedRPSI(uint32_t ssrc, 44 virtual void OnReceivedRPSI(uint32_t ssrc,
44 uint64_t pictureId) { 45 uint64_t pictureId) {
45 EXPECT_EQ(kTestPictureId, pictureId); 46 EXPECT_EQ(kTestPictureId, pictureId);
46 }; 47 };
47 virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {}; 48 virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) {};
48 private: 49 private:
49 RtpRtcp* _rtpRtcpModule; 50 RtpRtcp* _rtpRtcpModule;
50 }; 51 };
51 52
(...skipping 145 matching lines...) Expand 10 before | Expand all | Expand 10 after
197 198
198 uint32_t test_ssrc; 199 uint32_t test_ssrc;
199 uint32_t test_timestamp; 200 uint32_t test_timestamp;
200 uint16_t test_sequence_number; 201 uint16_t test_sequence_number;
201 std::vector<uint32_t> test_csrcs; 202 std::vector<uint32_t> test_csrcs;
202 SimulatedClock fake_clock; 203 SimulatedClock fake_clock;
203 }; 204 };
204 205
205 TEST_F(RtpRtcpRtcpTest, RTCP_PLI_RPSI) { 206 TEST_F(RtpRtcpRtcpTest, RTCP_PLI_RPSI) {
206 EXPECT_EQ(0, module1->SendRTCPReferencePictureSelection(kTestPictureId)); 207 EXPECT_EQ(0, module1->SendRTCPReferencePictureSelection(kTestPictureId));
207 EXPECT_EQ(0, module1->SendRTCPSliceLossIndication(156)); 208 EXPECT_EQ(0, module1->SendRTCPSliceLossIndication(kSliPictureId));
208 } 209 }
209 210
210 TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) { 211 TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
211 uint32_t testOfCSRC[webrtc::kRtpCsrcSize]; 212 uint32_t testOfCSRC[webrtc::kRtpCsrcSize];
212 EXPECT_EQ(2, rtp_receiver2_->CSRCs(testOfCSRC)); 213 EXPECT_EQ(2, rtp_receiver2_->CSRCs(testOfCSRC));
213 EXPECT_EQ(test_csrcs[0], testOfCSRC[0]); 214 EXPECT_EQ(test_csrcs[0], testOfCSRC[0]);
214 EXPECT_EQ(test_csrcs[1], testOfCSRC[1]); 215 EXPECT_EQ(test_csrcs[1], testOfCSRC[1]);
215 216
216 // Set cname of mixed. 217 // Set cname of mixed.
217 EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[0], "john@192.168.0.1")); 218 EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[0], "john@192.168.0.1"));
(...skipping 45 matching lines...) Expand 10 before | Expand all | Expand 10 after
263 264
264 // |test_ssrc+1| is the SSRC of module2 that send the report. 265 // |test_ssrc+1| is the SSRC of module2 that send the report.
265 EXPECT_EQ(test_ssrc+1, report_blocks[0].remoteSSRC); 266 EXPECT_EQ(test_ssrc+1, report_blocks[0].remoteSSRC);
266 EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC); 267 EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC);
267 268
268 EXPECT_EQ(0u, report_blocks[0].cumulativeLost); 269 EXPECT_EQ(0u, report_blocks[0].cumulativeLost);
269 EXPECT_LT(0u, report_blocks[0].delaySinceLastSR); 270 EXPECT_LT(0u, report_blocks[0].delaySinceLastSR);
270 EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum); 271 EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
271 EXPECT_EQ(0u, report_blocks[0].fractionLost); 272 EXPECT_EQ(0u, report_blocks[0].fractionLost);
272 } 273 }
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_sender.cc ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698