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1 /* | 1 /* |
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 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 "webrtc/api/statscollector.h" | 11 #include "webrtc/api/statscollector.h" |
12 | 12 |
13 #include <memory> | 13 #include <memory> |
14 #include <utility> | 14 #include <utility> |
15 #include <vector> | 15 #include <vector> |
16 | 16 |
17 #include "webrtc/api/peerconnection.h" | 17 #include "webrtc/api/peerconnection.h" |
18 #include "webrtc/base/base64.h" | 18 #include "webrtc/base/base64.h" |
19 #include "webrtc/base/checks.h" | 19 #include "webrtc/base/checks.h" |
20 #include "webrtc/base/timing.h" | |
21 #include "webrtc/pc/channel.h" | 20 #include "webrtc/pc/channel.h" |
22 | 21 |
23 namespace webrtc { | 22 namespace webrtc { |
24 namespace { | 23 namespace { |
25 | 24 |
26 // The following is the enum RTCStatsIceCandidateType from | 25 // The following is the enum RTCStatsIceCandidateType from |
27 // http://w3c.github.io/webrtc-stats/#rtcstatsicecandidatetype-enum such that | 26 // http://w3c.github.io/webrtc-stats/#rtcstatsicecandidatetype-enum such that |
28 // our stats report for ice candidate type could conform to that. | 27 // our stats report for ice candidate type could conform to that. |
29 const char STATSREPORT_LOCAL_PORT_TYPE[] = "host"; | 28 const char STATSREPORT_LOCAL_PORT_TYPE[] = "host"; |
30 const char STATSREPORT_STUN_PORT_TYPE[] = "serverreflexive"; | 29 const char STATSREPORT_STUN_PORT_TYPE[] = "serverreflexive"; |
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368 | 367 |
369 StatsCollector::StatsCollector(PeerConnection* pc) | 368 StatsCollector::StatsCollector(PeerConnection* pc) |
370 : pc_(pc), stats_gathering_started_(0) { | 369 : pc_(pc), stats_gathering_started_(0) { |
371 RTC_DCHECK(pc_); | 370 RTC_DCHECK(pc_); |
372 } | 371 } |
373 | 372 |
374 StatsCollector::~StatsCollector() { | 373 StatsCollector::~StatsCollector() { |
375 RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent()); | 374 RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent()); |
376 } | 375 } |
377 | 376 |
377 // Wallclock time in ms. | |
378 double StatsCollector::GetTimeNow() { | 378 double StatsCollector::GetTimeNow() { |
379 return rtc::Timing::WallTimeNow() * rtc::kNumMillisecsPerSec; | 379 return rtc::TimeUTCMicros() / |
380 static_cast<double>(rtc::kNumMicrosecsPerMillisec); | |
380 } | 381 } |
381 | 382 |
382 // Adds a MediaStream with tracks that can be used as a |selector| in a call | 383 // Adds a MediaStream with tracks that can be used as a |selector| in a call |
383 // to GetStats. | 384 // to GetStats. |
384 void StatsCollector::AddStream(MediaStreamInterface* stream) { | 385 void StatsCollector::AddStream(MediaStreamInterface* stream) { |
385 RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent()); | 386 RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent()); |
386 RTC_DCHECK(stream != NULL); | 387 RTC_DCHECK(stream != NULL); |
387 | 388 |
388 CreateTrackReports<AudioTrackVector>(stream->GetAudioTracks(), | 389 CreateTrackReports<AudioTrackVector>(stream->GetAudioTracks(), |
389 &reports_, track_ids_); | 390 &reports_, track_ids_); |
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460 r->FindValue(StatsReport::kStatsValueNameTrackId); | 461 r->FindValue(StatsReport::kStatsValueNameTrackId); |
461 if (v && v->string_val() == track->id()) | 462 if (v && v->string_val() == track->id()) |
462 reports->push_back(r); | 463 reports->push_back(r); |
463 } | 464 } |
464 } | 465 } |
465 | 466 |
466 void | 467 void |
467 StatsCollector::UpdateStats(PeerConnectionInterface::StatsOutputLevel level) { | 468 StatsCollector::UpdateStats(PeerConnectionInterface::StatsOutputLevel level) { |
468 RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent()); | 469 RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent()); |
469 double time_now = GetTimeNow(); | 470 double time_now = GetTimeNow(); |
471 | |
472 LOG(LS_INFO) << "UpdateStats: time_now: " << time_now; | |
Taylor Brandstetter
2016/09/07 18:39:17
nit: The two colons seem odd, could change it to "
nisse-webrtc
2016/09/08 07:38:09
Done.
| |
473 | |
470 // Calls to UpdateStats() that occur less than kMinGatherStatsPeriod number of | 474 // Calls to UpdateStats() that occur less than kMinGatherStatsPeriod number of |
471 // ms apart will be ignored. | 475 // ms apart will be ignored. |
472 const double kMinGatherStatsPeriod = 50; | 476 const double kMinGatherStatsPeriod = 50; |
473 if (stats_gathering_started_ != 0 && | 477 if (stats_gathering_started_ != 0 && |
474 stats_gathering_started_ + kMinGatherStatsPeriod > time_now) { | 478 stats_gathering_started_ + kMinGatherStatsPeriod > time_now) { |
475 return; | 479 return; |
476 } | 480 } |
477 stats_gathering_started_ = time_now; | 481 stats_gathering_started_ = time_now; |
478 | 482 |
479 // TODO(pthatcher): Merge PeerConnection and WebRtcSession so there is no | 483 // TODO(pthatcher): Merge PeerConnection and WebRtcSession so there is no |
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961 StatsReport* report = entry.second; | 965 StatsReport* report = entry.second; |
962 report->set_timestamp(stats_gathering_started_); | 966 report->set_timestamp(stats_gathering_started_); |
963 } | 967 } |
964 } | 968 } |
965 | 969 |
966 void StatsCollector::ClearUpdateStatsCacheForTest() { | 970 void StatsCollector::ClearUpdateStatsCacheForTest() { |
967 stats_gathering_started_ = 0; | 971 stats_gathering_started_ = 0; |
968 } | 972 } |
969 | 973 |
970 } // namespace webrtc | 974 } // namespace webrtc |
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