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| 1 /* | 1 /* |
| 2 * Copyright 2016 The WebRTC Project Authors. All rights reserved. | 2 * Copyright 2016 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/pc/rtcstatscollector.h" | 11 #include "webrtc/pc/rtcstatscollector.h" |
| 12 | 12 |
| 13 #include <memory> | 13 #include <memory> |
| 14 #include <sstream> | 14 #include <sstream> |
| 15 #include <utility> | 15 #include <utility> |
| 16 #include <vector> | 16 #include <vector> |
| 17 | 17 |
| 18 #include "webrtc/api/mediastreaminterface.h" | 18 #include "webrtc/api/mediastreaminterface.h" |
| 19 #include "webrtc/api/peerconnectioninterface.h" | 19 #include "webrtc/api/peerconnectioninterface.h" |
| 20 #include "webrtc/media/base/mediachannel.h" | 20 #include "webrtc/media/base/mediachannel.h" |
| 21 #include "webrtc/p2p/base/candidate.h" | 21 #include "webrtc/p2p/base/candidate.h" |
| 22 #include "webrtc/p2p/base/p2pconstants.h" | 22 #include "webrtc/p2p/base/p2pconstants.h" |
| 23 #include "webrtc/p2p/base/port.h" | 23 #include "webrtc/p2p/base/port.h" |
| 24 #include "webrtc/pc/peerconnection.h" | 24 #include "webrtc/pc/peerconnection.h" |
| 25 #include "webrtc/pc/webrtcsession.h" | 25 #include "webrtc/pc/webrtcsession.h" |
| 26 #include "webrtc/rtc_base/checks.h" | 26 #include "webrtc/rtc_base/checks.h" |
| 27 #include "webrtc/rtc_base/stringutils.h" | |
| 27 #include "webrtc/rtc_base/timeutils.h" | 28 #include "webrtc/rtc_base/timeutils.h" |
| 29 #include "webrtc/rtc_base/trace_event.h" | |
| 28 | 30 |
| 29 namespace webrtc { | 31 namespace webrtc { |
| 30 | 32 |
| 31 namespace { | 33 namespace { |
| 32 | 34 |
| 35 std::string TraceNameFromStatsTypeAndMemberName(const char* stats_type, | |
| 36 const char* member_name) { | |
| 37 char buffer[80]; | |
| 38 rtc::sprintfn(&buffer[0], sizeof(buffer), "%s.%s", stats_type, member_name); | |
|
tommi
2017/07/12 16:53:03
You might want to add dchecks for the length of th
| |
| 39 return buffer; | |
| 40 } | |
| 41 | |
| 33 std::string RTCCertificateIDFromFingerprint(const std::string& fingerprint) { | 42 std::string RTCCertificateIDFromFingerprint(const std::string& fingerprint) { |
| 34 return "RTCCertificate_" + fingerprint; | 43 return "RTCCertificate_" + fingerprint; |
| 35 } | 44 } |
| 36 | 45 |
| 37 std::string RTCCodecStatsIDFromDirectionMediaAndPayload( | 46 std::string RTCCodecStatsIDFromDirectionMediaAndPayload( |
| 38 bool inbound, bool audio, uint32_t payload_type) { | 47 bool inbound, bool audio, uint32_t payload_type) { |
| 39 // TODO(hbos): The present codec ID assignment is not sufficient to support | 48 // TODO(hbos): The present codec ID assignment is not sufficient to support |
| 40 // Unified Plan or unbundled connections in all cases. When we are able to | 49 // Unified Plan or unbundled connections in all cases. When we are able to |
| 41 // handle multiple m= lines of the same media type (and multiple BaseChannels | 50 // handle multiple m= lines of the same media type (and multiple BaseChannels |
| 42 // for the same type is possible?) this needs to be updated to differentiate | 51 // for the same type is possible?) this needs to be updated to differentiate |
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| 744 else | 753 else |
| 745 partial_report_->TakeMembersFrom(partial_report); | 754 partial_report_->TakeMembersFrom(partial_report); |
| 746 --num_pending_partial_reports_; | 755 --num_pending_partial_reports_; |
| 747 if (!num_pending_partial_reports_) { | 756 if (!num_pending_partial_reports_) { |
| 748 cache_timestamp_us_ = partial_report_timestamp_us_; | 757 cache_timestamp_us_ = partial_report_timestamp_us_; |
| 749 cached_report_ = partial_report_; | 758 cached_report_ = partial_report_; |
| 750 partial_report_ = nullptr; | 759 partial_report_ = nullptr; |
| 751 channel_name_pairs_.reset(); | 760 channel_name_pairs_.reset(); |
| 752 track_media_info_map_.reset(); | 761 track_media_info_map_.reset(); |
| 753 track_to_id_.clear(); | 762 track_to_id_.clear(); |
| 763 // Trace WebRTC Stats when getStats is called on Javascript. | |
| 764 // This allows access to WebRTC stats from trace logs. To enable them, | |
| 765 // select the "webrtc_stats" category when recording traces. | |
| 766 for (const RTCStats& stats : *cached_report_) { | |
| 767 for (const RTCStatsMemberInterface* member : stats.Members()) { | |
| 768 if (member->is_defined()) { | |
| 769 TRACE_EVENT_INSTANT2("webrtc_stats", | |
| 770 TraceNameFromStatsTypeAndMemberName( | |
| 771 stats.type(), member->name()).c_str(), | |
|
tommi
2017/07/12 16:53:03
Have you checked if this is a concern?
https://cs.
ehmaldonado_webrtc
2017/07/12 18:57:10
Right, I missed that. :(
I think the easiest way
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| 772 "value", member->ValueToString(), | |
| 773 "id", stats.id()); | |
| 774 } | |
| 775 } | |
| 776 } | |
| 754 DeliverCachedReport(); | 777 DeliverCachedReport(); |
| 755 } | 778 } |
| 756 } | 779 } |
| 757 | 780 |
| 758 void RTCStatsCollector::DeliverCachedReport() { | 781 void RTCStatsCollector::DeliverCachedReport() { |
| 759 RTC_DCHECK(signaling_thread_->IsCurrent()); | 782 RTC_DCHECK(signaling_thread_->IsCurrent()); |
| 760 RTC_DCHECK(!callbacks_.empty()); | 783 RTC_DCHECK(!callbacks_.empty()); |
| 761 RTC_DCHECK(cached_report_); | 784 RTC_DCHECK(cached_report_); |
| 762 for (const rtc::scoped_refptr<RTCStatsCollectorCallback>& callback : | 785 for (const rtc::scoped_refptr<RTCStatsCollectorCallback>& callback : |
| 763 callbacks_) { | 786 callbacks_) { |
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| 1248 const std::string& type) { | 1271 const std::string& type) { |
| 1249 return CandidateTypeToRTCIceCandidateType(type); | 1272 return CandidateTypeToRTCIceCandidateType(type); |
| 1250 } | 1273 } |
| 1251 | 1274 |
| 1252 const char* DataStateToRTCDataChannelStateForTesting( | 1275 const char* DataStateToRTCDataChannelStateForTesting( |
| 1253 DataChannelInterface::DataState state) { | 1276 DataChannelInterface::DataState state) { |
| 1254 return DataStateToRTCDataChannelState(state); | 1277 return DataStateToRTCDataChannelState(state); |
| 1255 } | 1278 } |
| 1256 | 1279 |
| 1257 } // namespace webrtc | 1280 } // namespace webrtc |
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