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

Issue 1748403002: Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Undid unneccessary changes to rtp_rtcp module. Created 4 years, 9 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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308 RTPSender rtp_sender(false, // bool audio 308 RTPSender rtp_sender(false, // bool audio
309 clock, // Clock* clock 309 clock, // Clock* clock
310 nullptr, // Transport* 310 nullptr, // Transport*
311 nullptr, // RtpAudioFeedback* 311 nullptr, // RtpAudioFeedback*
312 nullptr, // PacedSender* 312 nullptr, // PacedSender*
313 nullptr, // PacketRouter* 313 nullptr, // PacketRouter*
314 nullptr, // SendTimeObserver* 314 nullptr, // SendTimeObserver*
315 nullptr, // BitrateStatisticsObserver* 315 nullptr, // BitrateStatisticsObserver*
316 nullptr, // FrameCountObserver* 316 nullptr, // FrameCountObserver*
317 nullptr, // SendSideDelayObserver* 317 nullptr, // SendSideDelayObserver*
318 nullptr); // RtcEventLog* 318 nullptr); // RtcEventLogProxy*
the sun 2016/03/21 13:03:08 RtcEventLog
ivoc 2016/03/22 13:44:54 Oops, missed that one, thanks.
319 319
320 std::vector<uint32_t> csrcs; 320 std::vector<uint32_t> csrcs;
321 for (unsigned i = 0; i < csrcs_count; i++) { 321 for (unsigned i = 0; i < csrcs_count; i++) {
322 csrcs.push_back(prng->Rand<uint32_t>()); 322 csrcs.push_back(prng->Rand<uint32_t>());
323 } 323 }
324 rtp_sender.SetCsrcs(csrcs); 324 rtp_sender.SetCsrcs(csrcs);
325 rtp_sender.SetSSRC(prng->Rand<uint32_t>()); 325 rtp_sender.SetSSRC(prng->Rand<uint32_t>());
326 rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true); 326 rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true);
327 rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>()); 327 rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>());
328 328
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685 // Enable all header extensions 685 // Enable all header extensions
686 uint32_t extensions = (1u << kNumExtensions) - 1; 686 uint32_t extensions = (1u << kNumExtensions) - 1;
687 uint32_t csrcs_count = 2; 687 uint32_t csrcs_count = 2;
688 DropOldEvents(extensions, csrcs_count, 141421356); 688 DropOldEvents(extensions, csrcs_count, 141421356);
689 DropOldEvents(extensions, csrcs_count, 173205080); 689 DropOldEvents(extensions, csrcs_count, 173205080);
690 } 690 }
691 691
692 } // namespace webrtc 692 } // namespace webrtc
693 693
694 #endif // ENABLE_RTC_EVENT_LOG 694 #endif // ENABLE_RTC_EVENT_LOG
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