Add more trace events to interesting places.
Bug: webrtc:12840
Change-Id: I57e5373ae33060bd3743cea8ada21c845cbbd944
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221365
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34237}
diff --git a/p2p/base/p2p_transport_channel.cc b/p2p/base/p2p_transport_channel.cc
index eff79ab..836721c 100644
--- a/p2p/base/p2p_transport_channel.cc
+++ b/p2p/base/p2p_transport_channel.cc
@@ -43,6 +43,7 @@
#include "rtc_base/task_utils/to_queued_task.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
@@ -193,6 +194,7 @@
true /* presume_writable_when_fully_relayed */,
REGATHER_ON_FAILED_NETWORKS_INTERVAL,
RECEIVING_SWITCHING_DELAY) {
+ TRACE_EVENT0("webrtc", "P2PTransportChannel::P2PTransportChannel");
RTC_DCHECK(allocator_ != nullptr);
weak_ping_interval_ = GetWeakPingIntervalInFieldTrial();
// Validate IceConfig even for mostly built-in constant default values in case
@@ -247,6 +249,7 @@
ice_controller_factory) {}
P2PTransportChannel::~P2PTransportChannel() {
+ TRACE_EVENT0("webrtc", "P2PTransportChannel::~P2PTransportChannel");
RTC_DCHECK_RUN_ON(network_thread_);
std::vector<Connection*> copy(connections().begin(), connections().end());
for (Connection* con : copy) {
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index d24d40f..0a123b4 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -33,6 +33,7 @@
#include "rtc_base/string_utils.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/third_party/base64/base64.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
namespace {
@@ -836,6 +837,7 @@
// Call to stop any currently pending operations from running.
void Port::CancelPendingTasks() {
+ TRACE_EVENT0("webrtc", "Port::CancelPendingTasks");
RTC_DCHECK_RUN_ON(thread_);
thread_->Clear(this);
}
diff --git a/p2p/client/basic_port_allocator.cc b/p2p/client/basic_port_allocator.cc
index 7e1f970..a190fb7 100644
--- a/p2p/client/basic_port_allocator.cc
+++ b/p2p/client/basic_port_allocator.cc
@@ -27,6 +27,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
@@ -268,12 +269,16 @@
network_manager_started_(false),
allocation_sequences_created_(false),
turn_port_prune_policy_(allocator->turn_port_prune_policy()) {
+ TRACE_EVENT0("webrtc",
+ "BasicPortAllocatorSession::BasicPortAllocatorSession");
allocator_->network_manager()->SignalNetworksChanged.connect(
this, &BasicPortAllocatorSession::OnNetworksChanged);
allocator_->network_manager()->StartUpdating();
}
BasicPortAllocatorSession::~BasicPortAllocatorSession() {
+ TRACE_EVENT0("webrtc",
+ "BasicPortAllocatorSession::~BasicPortAllocatorSession");
RTC_DCHECK_RUN_ON(network_thread_);
allocator_->network_manager()->StopUpdating();
if (network_thread_ != NULL)
@@ -1248,6 +1253,7 @@
}
void AllocationSequence::Clear() {
+ TRACE_EVENT0("webrtc", "AllocationSequence::Clear");
udp_port_ = NULL;
relay_ports_.clear();
}
diff --git a/pc/channel.cc b/pc/channel.cc
index a8ecd4b..8630703 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -220,6 +220,7 @@
}
bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
+ TRACE_EVENT0("webrtc", "BaseChannel::SetRtpTransport");
RTC_DCHECK_RUN_ON(network_thread());
if (rtp_transport == rtp_transport_) {
return true;
@@ -524,6 +525,7 @@
}
void BaseChannel::UpdateWritableState_n() {
+ TRACE_EVENT0("webrtc", "BaseChannel::UpdateWritableState_n");
if (rtp_transport_->IsWritable(/*rtcp=*/true) &&
rtp_transport_->IsWritable(/*rtcp=*/false)) {
ChannelWritable_n();
@@ -533,6 +535,7 @@
}
void BaseChannel::ChannelWritable_n() {
+ TRACE_EVENT0("webrtc", "BaseChannel::ChannelWritable_n");
if (writable_) {
return;
}
@@ -552,6 +555,7 @@
}
void BaseChannel::ChannelNotWritable_n() {
+ TRACE_EVENT0("webrtc", "BaseChannel::ChannelNotWritable_n");
if (!writable_) {
return;
}
diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc
index dc4649b..e720888 100644
--- a/pc/jsep_transport.cc
+++ b/pc/jsep_transport.cc
@@ -26,6 +26,7 @@
#include "rtc_base/logging.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/trace_event.h"
using webrtc::SdpType;
@@ -104,6 +105,7 @@
? rtc::make_ref_counted<webrtc::SctpTransport>(
std::move(sctp_transport))
: nullptr) {
+ TRACE_EVENT0("webrtc", "JsepTransport::JsepTransport");
RTC_DCHECK(ice_transport_);
RTC_DCHECK(rtp_dtls_transport_);
// |rtcp_ice_transport_| must be present iff |rtcp_dtls_transport_| is
@@ -129,6 +131,7 @@
}
JsepTransport::~JsepTransport() {
+ TRACE_EVENT0("webrtc", "JsepTransport::~JsepTransport");
if (sctp_transport_) {
sctp_transport_->Clear();
}
@@ -147,7 +150,7 @@
const JsepTransportDescription& jsep_description,
SdpType type) {
webrtc::RTCError error;
-
+ TRACE_EVENT0("webrtc", "JsepTransport::SetLocalJsepTransportDescription");
RTC_DCHECK_RUN_ON(network_thread_);
IceParameters ice_parameters =
@@ -233,6 +236,7 @@
webrtc::RTCError JsepTransport::SetRemoteJsepTransportDescription(
const JsepTransportDescription& jsep_description,
webrtc::SdpType type) {
+ TRACE_EVENT0("webrtc", "JsepTransport::SetLocalJsepTransportDescription");
webrtc::RTCError error;
RTC_DCHECK_RUN_ON(network_thread_);
@@ -344,6 +348,7 @@
}
bool JsepTransport::GetStats(TransportStats* stats) {
+ TRACE_EVENT0("webrtc", "JsepTransport::GetStats");
RTC_DCHECK_RUN_ON(network_thread_);
stats->transport_name = mid();
stats->channel_stats.clear();
@@ -362,6 +367,7 @@
webrtc::RTCError JsepTransport::VerifyCertificateFingerprint(
const rtc::RTCCertificate* certificate,
const rtc::SSLFingerprint* fingerprint) const {
+ TRACE_EVENT0("webrtc", "JsepTransport::VerifyCertificateFingerprint");
RTC_DCHECK_RUN_ON(network_thread_);
if (!fingerprint) {
return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
@@ -400,6 +406,7 @@
void JsepTransport::SetRemoteIceParameters(
const IceParameters& ice_parameters,
IceTransportInternal* ice_transport) {
+ TRACE_EVENT0("webrtc", "JsepTransport::SetRemoteIceParameters");
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK(ice_transport);
RTC_DCHECK(remote_description_);
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 78aa032..a124389 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -32,6 +32,7 @@
#include "rtc_base/net_helper.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/thread.h"
+#include "rtc_base/trace_event.h"
using webrtc::SdpType;
@@ -77,6 +78,7 @@
RTCError JsepTransportController::SetLocalDescription(
SdpType type,
const cricket::SessionDescription* description) {
+ TRACE_EVENT0("webrtc", "JsepTransportController::SetLocalDescription");
if (!network_thread_->IsCurrent()) {
return network_thread_->Invoke<RTCError>(
RTC_FROM_HERE, [=] { return SetLocalDescription(type, description); });
@@ -97,6 +99,7 @@
RTCError JsepTransportController::SetRemoteDescription(
SdpType type,
const cricket::SessionDescription* description) {
+ TRACE_EVENT0("webrtc", "JsepTransportController::SetRemoteDescription");
if (!network_thread_->IsCurrent()) {
return network_thread_->Invoke<RTCError>(
RTC_FROM_HERE, [=] { return SetRemoteDescription(type, description); });
@@ -539,6 +542,7 @@
bool local,
SdpType type,
const cricket::SessionDescription* description) {
+ TRACE_EVENT0("webrtc", "JsepTransportController::ApplyDescription_n");
RTC_DCHECK(description);
if (local) {
@@ -866,6 +870,7 @@
bool JsepTransportController::HandleBundledContent(
const cricket::ContentInfo& content_info,
const cricket::ContentGroup& bundle_group) {
+ TRACE_EVENT0("webrtc", "JsepTransportController::HandleBundledContent");
RTC_DCHECK(bundle_group.FirstContentName());
auto jsep_transport =
GetJsepTransportByName(*bundle_group.FirstContentName());
@@ -887,6 +892,7 @@
bool JsepTransportController::SetTransportForMid(
const std::string& mid,
cricket::JsepTransport* jsep_transport) {
+ TRACE_EVENT0("webrtc", "JsepTransportController::SetTransportForMid");
RTC_DCHECK_RUN_ON(network_thread_);
RTC_DCHECK(jsep_transport);
@@ -924,6 +930,8 @@
const cricket::TransportInfo& transport_info,
const std::vector<int>& encrypted_extension_ids,
int rtp_abs_sendtime_extn_id) {
+ TRACE_EVENT0("webrtc",
+ "JsepTransportController::CreateJsepTransportDescription");
const cricket::MediaContentDescription* content_desc =
content_info.media_description();
RTC_DCHECK(content_desc);
@@ -1127,6 +1135,7 @@
void JsepTransportController::MaybeDestroyJsepTransport(
const std::string& mid) {
+ TRACE_EVENT0("webrtc", "JsepTransportController::MaybeDestroyJsepTransport");
auto jsep_transport = GetJsepTransportByName(mid);
if (!jsep_transport) {
return;
@@ -1280,6 +1289,7 @@
}
void JsepTransportController::UpdateAggregateStates_n() {
+ TRACE_EVENT0("webrtc", "JsepTransportController::UpdateAggregateStates_n");
auto dtls_transports = GetDtlsTransports();
cricket::IceConnectionState new_connection_state =
cricket::kIceConnectionConnecting;
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 2ef00de..54b4934 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -2188,6 +2188,7 @@
std::map<std::string, cricket::TransportStats>
PeerConnection::GetTransportStatsByNames(
const std::set<std::string>& transport_names) {
+ TRACE_EVENT0("webrtc", "PeerConnection::GetTransportStatsByNames");
RTC_DCHECK_RUN_ON(network_thread());
if (!network_thread_safety_->alive())
return {};
@@ -2636,6 +2637,7 @@
// Runs on network_thread().
void PeerConnection::ReportTransportStats() {
+ TRACE_EVENT0("webrtc", "PeerConnection::ReportTransportStats");
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::map<std::string, std::set<cricket::MediaType>>
media_types_by_transport_name;
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index d7b7ba7..4b38abc 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -1273,6 +1273,8 @@
void RTCStatsCollector::ProducePartialResultsOnNetworkThread(
int64_t timestamp_us,
absl::optional<std::string> sctp_transport_name) {
+ TRACE_EVENT0("webrtc",
+ "RTCStatsCollector::ProducePartialResultsOnNetworkThread");
RTC_DCHECK_RUN_ON(network_thread_);
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 761ae7c..8aa1947 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -1251,6 +1251,7 @@
std::unique_ptr<SessionDescriptionInterface> desc,
const std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ApplyLocalDescription");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc);
@@ -1540,6 +1541,7 @@
std::unique_ptr<SessionDescriptionInterface> desc,
const std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ApplyRemoteDescription");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc);
@@ -2025,6 +2027,7 @@
void SdpOfferAnswerHandler::CreateAnswer(
CreateSessionDescriptionObserver* observer,
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateAnswer");
RTC_DCHECK_RUN_ON(signaling_thread());
// Chain this operation. If asynchronous operations are pending on the chain,
// this operation will be queued to be invoked, otherwise the contents of the
@@ -2325,6 +2328,7 @@
void SdpOfferAnswerHandler::AddIceCandidate(
std::unique_ptr<IceCandidateInterface> candidate,
std::function<void(RTCError)> callback) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AddIceCandidate");
RTC_DCHECK_RUN_ON(signaling_thread());
// Chain this operation. If asynchronous operations are pending on the chain,
// this operation will be queued to be invoked, otherwise the contents of the
@@ -2456,6 +2460,7 @@
void SdpOfferAnswerHandler::ChangeSignalingState(
PeerConnectionInterface::SignalingState signaling_state) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ChangeSignalingState");
RTC_DCHECK_RUN_ON(signaling_thread());
if (signaling_state_ == signaling_state) {
return;
@@ -2660,6 +2665,7 @@
}
RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::Rollback");
auto state = signaling_state();
if (state != PeerConnectionInterface::kHaveLocalOffer &&
state != PeerConnectionInterface::kHaveRemoteOffer) {
@@ -3119,6 +3125,8 @@
const SessionDescriptionInterface* old_remote_description,
const std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid) {
+ TRACE_EVENT0("webrtc",
+ "SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
@@ -3205,6 +3213,7 @@
const ContentInfo& content,
const ContentInfo* old_local_content,
const ContentInfo* old_remote_content) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AssociateTransceiver");
RTC_DCHECK(IsUnifiedPlan());
#if RTC_DCHECK_IS_ON
// If this is an offer then the m= section might be recycled. If the m=
@@ -3338,6 +3347,7 @@
transceiver,
const cricket::ContentInfo& content,
const cricket::ContentGroup* bundle_group) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateTransceiverChannel");
RTC_DCHECK(IsUnifiedPlan());
RTC_DCHECK(transceiver);
cricket::ChannelInterface* channel = transceiver->internal()->channel();
@@ -4022,6 +4032,7 @@
void SdpOfferAnswerHandler::UpdateLocalSenders(
const std::vector<cricket::StreamParams>& streams,
cricket::MediaType media_type) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateLocalSenders");
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<RtpSenderInfo>* current_senders =
rtp_manager()->GetLocalSenderInfos(media_type);
@@ -4064,6 +4075,7 @@
bool default_sender_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateRemoteSendersList");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsUnifiedPlan());
@@ -4163,6 +4175,7 @@
}
void SdpOfferAnswerHandler::EnableSending() {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::EnableSending");
RTC_DCHECK_RUN_ON(signaling_thread());
for (const auto& transceiver : transceivers()->ListInternal()) {
cricket::ChannelInterface* channel = transceiver->channel();
@@ -4177,6 +4190,7 @@
cricket::ContentSource source,
const std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::PushdownMediaDescription");
const SessionDescriptionInterface* sdesc =
(source == cricket::CS_LOCAL ? local_description()
: remote_description());
@@ -4270,6 +4284,7 @@
RTCError SdpOfferAnswerHandler::PushdownTransportDescription(
cricket::ContentSource source,
SdpType type) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::PushdownTransportDescription");
RTC_DCHECK_RUN_ON(signaling_thread());
if (source == cricket::CS_LOCAL) {
@@ -4286,6 +4301,7 @@
}
void SdpOfferAnswerHandler::RemoveStoppedTransceivers() {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::RemoveStoppedTransceivers");
RTC_DCHECK_RUN_ON(signaling_thread());
// 3.2.10.1: For each transceiver in the connection's set of transceivers
// run the following steps:
@@ -4505,6 +4521,7 @@
}
RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateChannels");
// Creating the media channels. Transports should already have been created
// at this point.
RTC_DCHECK_RUN_ON(signaling_thread());
@@ -4545,6 +4562,7 @@
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VoiceChannel* SdpOfferAnswerHandler::CreateVoiceChannel(
const std::string& mid) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateVoiceChannel");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!channel_manager()->media_engine())
return nullptr;
@@ -4563,6 +4581,7 @@
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VideoChannel* SdpOfferAnswerHandler::CreateVideoChannel(
const std::string& mid) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateVideoChannel");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!channel_manager()->media_engine())
return nullptr;
@@ -4600,6 +4619,7 @@
void SdpOfferAnswerHandler::DestroyTransceiverChannel(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DestroyTransceiverChannel");
RTC_DCHECK(transceiver);
RTC_LOG_THREAD_BLOCK_COUNT();
@@ -4648,6 +4668,7 @@
void SdpOfferAnswerHandler::DestroyChannelInterface(
cricket::ChannelInterface* channel) {
+ TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DestroyChannelInterface");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(channel_manager()->media_engine());
RTC_DCHECK(channel);
@@ -4802,6 +4823,8 @@
cricket::ContentSource source,
const std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid) {
+ TRACE_EVENT0("webrtc",
+ "SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState");
RTC_DCHECK_RUN_ON(signaling_thread());
// We may need to delete any created default streams and disable creation of
// new ones on the basis of payload type. This is needed to avoid SSRC
diff --git a/pc/srtp_transport.cc b/pc/srtp_transport.cc
index ee07349..c90b3fa 100644
--- a/pc/srtp_transport.cc
+++ b/pc/srtp_transport.cc
@@ -201,12 +201,12 @@
void SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
+ TRACE_EVENT0("webrtc", "SrtpTransport::OnRtpPacketReceived");
if (!IsSrtpActive()) {
RTC_LOG(LS_WARNING)
<< "Inactive SRTP transport received an RTP packet. Drop it.";
return;
}
- TRACE_EVENT0("webrtc", "SRTP Decode");
char* data = packet.MutableData<char>();
int len = rtc::checked_cast<int>(packet.size());
if (!UnprotectRtp(data, len, &len)) {
@@ -233,12 +233,12 @@
void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
+ TRACE_EVENT0("webrtc", "SrtpTransport::OnRtcpPacketReceived");
if (!IsSrtpActive()) {
RTC_LOG(LS_WARNING)
<< "Inactive SRTP transport received an RTCP packet. Drop it.";
return;
}
- TRACE_EVENT0("webrtc", "SRTP Decode");
char* data = packet.MutableData<char>();
int len = rtc::checked_cast<int>(packet.size());
if (!UnprotectRtcp(data, len, &len)) {
diff --git a/pc/stats_collector.cc b/pc/stats_collector.cc
index 6d4c224..1728cb4 100644
--- a/pc/stats_collector.cc
+++ b/pc/stats_collector.cc
@@ -50,6 +50,7 @@
#include "rtc_base/string_encode.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
@@ -849,6 +850,7 @@
}
std::map<std::string, std::string> StatsCollector::ExtractSessionInfo() {
+ TRACE_EVENT0("webrtc", "StatsCollector::ExtractSessionInfo");
RTC_DCHECK_RUN_ON(pc_->signaling_thread());
SessionStats stats;
@@ -870,6 +872,7 @@
RtpTransceiverProxyWithInternal<RtpTransceiver>>>& transceivers,
absl::optional<std::string> sctp_transport_name,
absl::optional<std::string> sctp_mid) {
+ TRACE_EVENT0("webrtc", "StatsCollector::ExtractSessionInfo_n");
RTC_DCHECK_RUN_ON(pc_->network_thread());
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
SessionStats stats;
