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Side by Side Diff: webrtc/call/rtc_event_log_unittest.cc

Issue 1571283002: Fixes a bug which incorrectly logs incoming RTCP as outgoing. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added RTC_DISALLOW_IMPLICT_CONSTRUCTORS Created 4 years, 11 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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
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306 306
307 RTPSender rtp_sender(false, // bool audio 307 RTPSender rtp_sender(false, // bool audio
308 clock, // Clock* clock 308 clock, // Clock* clock
309 nullptr, // Transport* 309 nullptr, // Transport*
310 nullptr, // RtpAudioFeedback* 310 nullptr, // RtpAudioFeedback*
311 nullptr, // PacedSender* 311 nullptr, // PacedSender*
312 nullptr, // PacketRouter* 312 nullptr, // PacketRouter*
313 nullptr, // SendTimeObserver* 313 nullptr, // SendTimeObserver*
314 nullptr, // BitrateStatisticsObserver* 314 nullptr, // BitrateStatisticsObserver*
315 nullptr, // FrameCountObserver* 315 nullptr, // FrameCountObserver*
316 nullptr); // SendSideDelayObserver* 316 nullptr, // SendSideDelayObserver*
317 nullptr); // RtcEventLog*
317 318
318 std::vector<uint32_t> csrcs; 319 std::vector<uint32_t> csrcs;
319 for (unsigned i = 0; i < csrcs_count; i++) { 320 for (unsigned i = 0; i < csrcs_count; i++) {
320 csrcs.push_back(prng->Rand<uint32_t>()); 321 csrcs.push_back(prng->Rand<uint32_t>());
321 } 322 }
322 rtp_sender.SetCsrcs(csrcs); 323 rtp_sender.SetCsrcs(csrcs);
323 rtp_sender.SetSSRC(prng->Rand<uint32_t>()); 324 rtp_sender.SetSSRC(prng->Rand<uint32_t>());
324 rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true); 325 rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true);
325 rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>()); 326 rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>());
326 327
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681 // Enable all header extensions 682 // Enable all header extensions
682 uint32_t extensions = (1u << kNumExtensions) - 1; 683 uint32_t extensions = (1u << kNumExtensions) - 1;
683 uint32_t csrcs_count = 2; 684 uint32_t csrcs_count = 2;
684 DropOldEvents(extensions, csrcs_count, 141421356); 685 DropOldEvents(extensions, csrcs_count, 141421356);
685 DropOldEvents(extensions, csrcs_count, 173205080); 686 DropOldEvents(extensions, csrcs_count, 173205080);
686 } 687 }
687 688
688 } // namespace webrtc 689 } // namespace webrtc
689 690
690 #endif // ENABLE_RTC_EVENT_LOG 691 #endif // ENABLE_RTC_EVENT_LOG
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