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}
This commit is contained in:
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a94a4cc197
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@ -43,6 +43,7 @@
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#include "rtc_base/task_utils/to_queued_task.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/time_utils.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/field_trial.h"
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#include "system_wrappers/include/metrics.h"
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@ -193,6 +194,7 @@ P2PTransportChannel::P2PTransportChannel(
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true /* presume_writable_when_fully_relayed */,
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REGATHER_ON_FAILED_NETWORKS_INTERVAL,
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RECEIVING_SWITCHING_DELAY) {
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TRACE_EVENT0("webrtc", "P2PTransportChannel::P2PTransportChannel");
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RTC_DCHECK(allocator_ != nullptr);
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weak_ping_interval_ = GetWeakPingIntervalInFieldTrial();
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// Validate IceConfig even for mostly built-in constant default values in case
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@ -247,6 +249,7 @@ P2PTransportChannel::P2PTransportChannel(
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ice_controller_factory) {}
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P2PTransportChannel::~P2PTransportChannel() {
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TRACE_EVENT0("webrtc", "P2PTransportChannel::~P2PTransportChannel");
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RTC_DCHECK_RUN_ON(network_thread_);
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std::vector<Connection*> copy(connections().begin(), connections().end());
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for (Connection* con : copy) {
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@ -33,6 +33,7 @@
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#include "rtc_base/string_utils.h"
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#include "rtc_base/strings/string_builder.h"
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#include "rtc_base/third_party/base64/base64.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/field_trial.h"
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namespace {
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@ -836,6 +837,7 @@ void Port::Prune() {
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// Call to stop any currently pending operations from running.
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void Port::CancelPendingTasks() {
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TRACE_EVENT0("webrtc", "Port::CancelPendingTasks");
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RTC_DCHECK_RUN_ON(thread_);
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thread_->Clear(this);
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}
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@ -27,6 +27,7 @@
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#include "rtc_base/checks.h"
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#include "rtc_base/helpers.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/field_trial.h"
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#include "system_wrappers/include/metrics.h"
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@ -268,12 +269,16 @@ BasicPortAllocatorSession::BasicPortAllocatorSession(
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network_manager_started_(false),
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allocation_sequences_created_(false),
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turn_port_prune_policy_(allocator->turn_port_prune_policy()) {
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TRACE_EVENT0("webrtc",
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"BasicPortAllocatorSession::BasicPortAllocatorSession");
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allocator_->network_manager()->SignalNetworksChanged.connect(
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this, &BasicPortAllocatorSession::OnNetworksChanged);
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allocator_->network_manager()->StartUpdating();
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}
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BasicPortAllocatorSession::~BasicPortAllocatorSession() {
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TRACE_EVENT0("webrtc",
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"BasicPortAllocatorSession::~BasicPortAllocatorSession");
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RTC_DCHECK_RUN_ON(network_thread_);
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allocator_->network_manager()->StopUpdating();
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if (network_thread_ != NULL)
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@ -1248,6 +1253,7 @@ void AllocationSequence::Init() {
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}
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void AllocationSequence::Clear() {
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TRACE_EVENT0("webrtc", "AllocationSequence::Clear");
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udp_port_ = NULL;
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relay_ports_.clear();
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}
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@ -220,6 +220,7 @@ void BaseChannel::Deinit() {
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}
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bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
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TRACE_EVENT0("webrtc", "BaseChannel::SetRtpTransport");
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RTC_DCHECK_RUN_ON(network_thread());
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if (rtp_transport == rtp_transport_) {
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return true;
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@ -524,6 +525,7 @@ void BaseChannel::DisableMedia_w() {
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}
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void BaseChannel::UpdateWritableState_n() {
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TRACE_EVENT0("webrtc", "BaseChannel::UpdateWritableState_n");
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if (rtp_transport_->IsWritable(/*rtcp=*/true) &&
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rtp_transport_->IsWritable(/*rtcp=*/false)) {
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ChannelWritable_n();
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@ -533,6 +535,7 @@ void BaseChannel::UpdateWritableState_n() {
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}
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void BaseChannel::ChannelWritable_n() {
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TRACE_EVENT0("webrtc", "BaseChannel::ChannelWritable_n");
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if (writable_) {
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return;
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}
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@ -552,6 +555,7 @@ void BaseChannel::ChannelWritable_n() {
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}
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void BaseChannel::ChannelNotWritable_n() {
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TRACE_EVENT0("webrtc", "BaseChannel::ChannelNotWritable_n");
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if (!writable_) {
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return;
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}
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@ -26,6 +26,7 @@
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#include "rtc_base/logging.h"
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#include "rtc_base/ref_counted_object.h"
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#include "rtc_base/strings/string_builder.h"
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#include "rtc_base/trace_event.h"
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using webrtc::SdpType;
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@ -104,6 +105,7 @@ JsepTransport::JsepTransport(
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? rtc::make_ref_counted<webrtc::SctpTransport>(
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std::move(sctp_transport))
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: nullptr) {
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TRACE_EVENT0("webrtc", "JsepTransport::JsepTransport");
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RTC_DCHECK(ice_transport_);
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RTC_DCHECK(rtp_dtls_transport_);
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// |rtcp_ice_transport_| must be present iff |rtcp_dtls_transport_| is
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@ -129,6 +131,7 @@ JsepTransport::JsepTransport(
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}
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JsepTransport::~JsepTransport() {
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TRACE_EVENT0("webrtc", "JsepTransport::~JsepTransport");
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if (sctp_transport_) {
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sctp_transport_->Clear();
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}
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@ -147,7 +150,7 @@ webrtc::RTCError JsepTransport::SetLocalJsepTransportDescription(
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const JsepTransportDescription& jsep_description,
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SdpType type) {
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webrtc::RTCError error;
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TRACE_EVENT0("webrtc", "JsepTransport::SetLocalJsepTransportDescription");
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RTC_DCHECK_RUN_ON(network_thread_);
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IceParameters ice_parameters =
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@ -233,6 +236,7 @@ webrtc::RTCError JsepTransport::SetLocalJsepTransportDescription(
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webrtc::RTCError JsepTransport::SetRemoteJsepTransportDescription(
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const JsepTransportDescription& jsep_description,
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webrtc::SdpType type) {
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TRACE_EVENT0("webrtc", "JsepTransport::SetLocalJsepTransportDescription");
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webrtc::RTCError error;
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RTC_DCHECK_RUN_ON(network_thread_);
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@ -344,6 +348,7 @@ absl::optional<rtc::SSLRole> JsepTransport::GetDtlsRole() const {
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}
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bool JsepTransport::GetStats(TransportStats* stats) {
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TRACE_EVENT0("webrtc", "JsepTransport::GetStats");
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RTC_DCHECK_RUN_ON(network_thread_);
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stats->transport_name = mid();
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stats->channel_stats.clear();
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@ -362,6 +367,7 @@ bool JsepTransport::GetStats(TransportStats* stats) {
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webrtc::RTCError JsepTransport::VerifyCertificateFingerprint(
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const rtc::RTCCertificate* certificate,
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const rtc::SSLFingerprint* fingerprint) const {
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TRACE_EVENT0("webrtc", "JsepTransport::VerifyCertificateFingerprint");
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RTC_DCHECK_RUN_ON(network_thread_);
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if (!fingerprint) {
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return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
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@ -400,6 +406,7 @@ void JsepTransport::SetActiveResetSrtpParams(bool active_reset_srtp_params) {
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void JsepTransport::SetRemoteIceParameters(
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const IceParameters& ice_parameters,
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IceTransportInternal* ice_transport) {
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TRACE_EVENT0("webrtc", "JsepTransport::SetRemoteIceParameters");
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RTC_DCHECK_RUN_ON(network_thread_);
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RTC_DCHECK(ice_transport);
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RTC_DCHECK(remote_description_);
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@ -32,6 +32,7 @@
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#include "rtc_base/net_helper.h"
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#include "rtc_base/socket_address.h"
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#include "rtc_base/thread.h"
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#include "rtc_base/trace_event.h"
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using webrtc::SdpType;
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@ -77,6 +78,7 @@ JsepTransportController::~JsepTransportController() {
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RTCError JsepTransportController::SetLocalDescription(
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SdpType type,
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const cricket::SessionDescription* description) {
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TRACE_EVENT0("webrtc", "JsepTransportController::SetLocalDescription");
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if (!network_thread_->IsCurrent()) {
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return network_thread_->Invoke<RTCError>(
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RTC_FROM_HERE, [=] { return SetLocalDescription(type, description); });
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@ -97,6 +99,7 @@ RTCError JsepTransportController::SetLocalDescription(
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RTCError JsepTransportController::SetRemoteDescription(
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SdpType type,
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const cricket::SessionDescription* description) {
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TRACE_EVENT0("webrtc", "JsepTransportController::SetRemoteDescription");
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if (!network_thread_->IsCurrent()) {
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return network_thread_->Invoke<RTCError>(
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RTC_FROM_HERE, [=] { return SetRemoteDescription(type, description); });
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@ -539,6 +542,7 @@ RTCError JsepTransportController::ApplyDescription_n(
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bool local,
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SdpType type,
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const cricket::SessionDescription* description) {
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TRACE_EVENT0("webrtc", "JsepTransportController::ApplyDescription_n");
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RTC_DCHECK(description);
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if (local) {
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@ -866,6 +870,7 @@ void JsepTransportController::HandleRejectedContent(
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bool JsepTransportController::HandleBundledContent(
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const cricket::ContentInfo& content_info,
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const cricket::ContentGroup& bundle_group) {
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TRACE_EVENT0("webrtc", "JsepTransportController::HandleBundledContent");
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RTC_DCHECK(bundle_group.FirstContentName());
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auto jsep_transport =
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GetJsepTransportByName(*bundle_group.FirstContentName());
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@ -887,6 +892,7 @@ bool JsepTransportController::HandleBundledContent(
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bool JsepTransportController::SetTransportForMid(
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const std::string& mid,
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cricket::JsepTransport* jsep_transport) {
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TRACE_EVENT0("webrtc", "JsepTransportController::SetTransportForMid");
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RTC_DCHECK_RUN_ON(network_thread_);
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RTC_DCHECK(jsep_transport);
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@ -924,6 +930,8 @@ JsepTransportController::CreateJsepTransportDescription(
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const cricket::TransportInfo& transport_info,
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const std::vector<int>& encrypted_extension_ids,
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int rtp_abs_sendtime_extn_id) {
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TRACE_EVENT0("webrtc",
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"JsepTransportController::CreateJsepTransportDescription");
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const cricket::MediaContentDescription* content_desc =
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content_info.media_description();
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RTC_DCHECK(content_desc);
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@ -1127,6 +1135,7 @@ RTCError JsepTransportController::MaybeCreateJsepTransport(
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void JsepTransportController::MaybeDestroyJsepTransport(
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const std::string& mid) {
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TRACE_EVENT0("webrtc", "JsepTransportController::MaybeDestroyJsepTransport");
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auto jsep_transport = GetJsepTransportByName(mid);
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if (!jsep_transport) {
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return;
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@ -1280,6 +1289,7 @@ void JsepTransportController::OnTransportStateChanged_n(
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}
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void JsepTransportController::UpdateAggregateStates_n() {
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TRACE_EVENT0("webrtc", "JsepTransportController::UpdateAggregateStates_n");
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auto dtls_transports = GetDtlsTransports();
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cricket::IceConnectionState new_connection_state =
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cricket::kIceConnectionConnecting;
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@ -2188,6 +2188,7 @@ cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
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std::map<std::string, cricket::TransportStats>
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PeerConnection::GetTransportStatsByNames(
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const std::set<std::string>& transport_names) {
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TRACE_EVENT0("webrtc", "PeerConnection::GetTransportStatsByNames");
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RTC_DCHECK_RUN_ON(network_thread());
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if (!network_thread_safety_->alive())
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return {};
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@ -2636,6 +2637,7 @@ void PeerConnection::OnTransportControllerGatheringState(
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// Runs on network_thread().
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void PeerConnection::ReportTransportStats() {
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TRACE_EVENT0("webrtc", "PeerConnection::ReportTransportStats");
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rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
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std::map<std::string, std::set<cricket::MediaType>>
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media_types_by_transport_name;
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@ -1273,6 +1273,8 @@ void RTCStatsCollector::ProducePartialResultsOnSignalingThreadImpl(
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void RTCStatsCollector::ProducePartialResultsOnNetworkThread(
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int64_t timestamp_us,
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absl::optional<std::string> sctp_transport_name) {
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TRACE_EVENT0("webrtc",
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"RTCStatsCollector::ProducePartialResultsOnNetworkThread");
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RTC_DCHECK_RUN_ON(network_thread_);
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rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
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@ -1251,6 +1251,7 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
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std::unique_ptr<SessionDescriptionInterface> desc,
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const std::map<std::string, const cricket::ContentGroup*>&
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bundle_groups_by_mid) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ApplyLocalDescription");
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RTC_DCHECK_RUN_ON(signaling_thread());
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RTC_DCHECK(desc);
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@ -1540,6 +1541,7 @@ RTCError SdpOfferAnswerHandler::ApplyRemoteDescription(
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std::unique_ptr<SessionDescriptionInterface> desc,
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const std::map<std::string, const cricket::ContentGroup*>&
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bundle_groups_by_mid) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ApplyRemoteDescription");
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RTC_DCHECK_RUN_ON(signaling_thread());
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RTC_DCHECK(desc);
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@ -2025,6 +2027,7 @@ void SdpOfferAnswerHandler::DoCreateOffer(
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void SdpOfferAnswerHandler::CreateAnswer(
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CreateSessionDescriptionObserver* observer,
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const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateAnswer");
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RTC_DCHECK_RUN_ON(signaling_thread());
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// Chain this operation. If asynchronous operations are pending on the chain,
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// this operation will be queued to be invoked, otherwise the contents of the
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@ -2325,6 +2328,7 @@ AddIceCandidateResult SdpOfferAnswerHandler::AddIceCandidateInternal(
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void SdpOfferAnswerHandler::AddIceCandidate(
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std::unique_ptr<IceCandidateInterface> candidate,
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std::function<void(RTCError)> callback) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AddIceCandidate");
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RTC_DCHECK_RUN_ON(signaling_thread());
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// Chain this operation. If asynchronous operations are pending on the chain,
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// this operation will be queued to be invoked, otherwise the contents of the
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@ -2456,6 +2460,7 @@ PeerConnectionInterface::SignalingState SdpOfferAnswerHandler::signaling_state()
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void SdpOfferAnswerHandler::ChangeSignalingState(
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PeerConnectionInterface::SignalingState signaling_state) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ChangeSignalingState");
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RTC_DCHECK_RUN_ON(signaling_thread());
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if (signaling_state_ == signaling_state) {
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return;
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@ -2660,6 +2665,7 @@ void SdpOfferAnswerHandler::OnVideoTrackRemoved(VideoTrackInterface* track,
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}
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RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::Rollback");
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auto state = signaling_state();
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if (state != PeerConnectionInterface::kHaveLocalOffer &&
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state != PeerConnectionInterface::kHaveRemoteOffer) {
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@ -3119,6 +3125,8 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels(
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const SessionDescriptionInterface* old_remote_description,
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const std::map<std::string, const cricket::ContentGroup*>&
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bundle_groups_by_mid) {
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TRACE_EVENT0("webrtc",
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"SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels");
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RTC_DCHECK_RUN_ON(signaling_thread());
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RTC_DCHECK(IsUnifiedPlan());
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@ -3205,6 +3213,7 @@ SdpOfferAnswerHandler::AssociateTransceiver(
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const ContentInfo& content,
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const ContentInfo* old_local_content,
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const ContentInfo* old_remote_content) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AssociateTransceiver");
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RTC_DCHECK(IsUnifiedPlan());
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#if RTC_DCHECK_IS_ON
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// If this is an offer then the m= section might be recycled. If the m=
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@ -3338,6 +3347,7 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiverChannel(
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transceiver,
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const cricket::ContentInfo& content,
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const cricket::ContentGroup* bundle_group) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateTransceiverChannel");
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RTC_DCHECK(IsUnifiedPlan());
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RTC_DCHECK(transceiver);
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cricket::ChannelInterface* channel = transceiver->internal()->channel();
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@ -4022,6 +4032,7 @@ void SdpOfferAnswerHandler::RemoveSenders(cricket::MediaType media_type) {
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void SdpOfferAnswerHandler::UpdateLocalSenders(
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const std::vector<cricket::StreamParams>& streams,
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cricket::MediaType media_type) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateLocalSenders");
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RTC_DCHECK_RUN_ON(signaling_thread());
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std::vector<RtpSenderInfo>* current_senders =
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rtp_manager()->GetLocalSenderInfos(media_type);
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@ -4064,6 +4075,7 @@ void SdpOfferAnswerHandler::UpdateRemoteSendersList(
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bool default_sender_needed,
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cricket::MediaType media_type,
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StreamCollection* new_streams) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateRemoteSendersList");
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RTC_DCHECK_RUN_ON(signaling_thread());
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RTC_DCHECK(!IsUnifiedPlan());
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@ -4163,6 +4175,7 @@ void SdpOfferAnswerHandler::UpdateRemoteSendersList(
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}
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void SdpOfferAnswerHandler::EnableSending() {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::EnableSending");
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RTC_DCHECK_RUN_ON(signaling_thread());
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for (const auto& transceiver : transceivers()->ListInternal()) {
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cricket::ChannelInterface* channel = transceiver->channel();
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@ -4177,6 +4190,7 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
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cricket::ContentSource source,
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const std::map<std::string, const cricket::ContentGroup*>&
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bundle_groups_by_mid) {
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TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::PushdownMediaDescription");
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const SessionDescriptionInterface* sdesc =
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(source == cricket::CS_LOCAL ? local_description()
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: remote_description());
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@ -4270,6 +4284,7 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
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RTCError SdpOfferAnswerHandler::PushdownTransportDescription(
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cricket::ContentSource source,
|
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SdpType type) {
|
||||
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::PushdownTransportDescription");
|
||||
RTC_DCHECK_RUN_ON(signaling_thread());
|
||||
|
||||
if (source == cricket::CS_LOCAL) {
|
||||
@ -4286,6 +4301,7 @@ RTCError SdpOfferAnswerHandler::PushdownTransportDescription(
|
||||
}
|
||||
|
||||
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 @@ RTCErrorOr<const cricket::ContentInfo*> SdpOfferAnswerHandler::FindContentInfo(
|
||||
}
|
||||
|
||||
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 @@ RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
|
||||
// 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 @@ cricket::VoiceChannel* SdpOfferAnswerHandler::CreateVoiceChannel(
|
||||
// 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 @@ bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
|
||||
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::DestroyDataChannelTransport() {
|
||||
|
||||
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 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
|
||||
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
|
||||
|
||||
@ -201,12 +201,12 @@ bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
|
||||
|
||||
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::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
|
||||
|
||||
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)) {
|
||||
|
||||
@ -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 @@ StatsReport* StatsCollector::AddCandidateReport(
|
||||
}
|
||||
|
||||
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 @@ StatsCollector::SessionStats StatsCollector::ExtractSessionInfo_n(
|
||||
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;
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user