Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(595)

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

Issue 2089773002: Add EncodedImageCallback::OnEncodedImage(). (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 4 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
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
(...skipping 147 matching lines...) Expand 10 before | Expand all | Expand 10 after
158 voice_codec.plname, 158 voice_codec.plname,
159 voice_codec.pltype, 159 voice_codec.pltype,
160 voice_codec.plfreq, 160 voice_codec.plfreq,
161 voice_codec.channels, 161 voice_codec.channels,
162 (voice_codec.rate < 0) ? 0 : voice_codec.rate)); 162 (voice_codec.rate < 0) ? 0 : voice_codec.rate));
163 163
164 // We need to send one RTP packet to get the RTCP packet to be accepted by 164 // We need to send one RTP packet to get the RTCP packet to be accepted by
165 // the receiving module. 165 // the receiving module.
166 // send RTP packet with the data "testtest" 166 // send RTP packet with the data "testtest"
167 const uint8_t test[9] = "testtest"; 167 const uint8_t test[9] = "testtest";
168 EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 168 EXPECT_EQ(true,
169 0, -1, test, 8)); 169 module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
170 test, 8, nullptr, nullptr, nullptr));
170 } 171 }
171 172
172 virtual void TearDown() { 173 virtual void TearDown() {
173 delete module1; 174 delete module1;
174 delete module2; 175 delete module2;
175 delete myRTCPFeedback1; 176 delete myRTCPFeedback1;
176 delete myRTCPFeedback2; 177 delete myRTCPFeedback2;
177 delete transport1; 178 delete transport1;
178 delete transport2; 179 delete transport2;
179 delete receiver; 180 delete receiver;
(...skipping 87 matching lines...) Expand 10 before | Expand all | Expand 10 after
267 EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC); 268 EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC);
268 269
269 EXPECT_EQ(0u, report_blocks[0].cumulativeLost); 270 EXPECT_EQ(0u, report_blocks[0].cumulativeLost);
270 EXPECT_LT(0u, report_blocks[0].delaySinceLastSR); 271 EXPECT_LT(0u, report_blocks[0].delaySinceLastSR);
271 EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum); 272 EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
272 EXPECT_EQ(0u, report_blocks[0].fractionLost); 273 EXPECT_EQ(0u, report_blocks[0].fractionLost);
273 } 274 }
274 275
275 } // namespace 276 } // namespace
276 } // namespace webrtc 277 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc ('k') | webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698