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-rw-r--r--pc/peer_connection.cc72
1 files changed, 40 insertions, 32 deletions
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 95abb108b9..276af1787d 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -465,10 +465,9 @@ RTCErrorOr<rtc::scoped_refptr<PeerConnection>> PeerConnection::Create(
}
// The PeerConnection constructor consumes some, but not all, dependencies.
- rtc::scoped_refptr<PeerConnection> pc(
- new rtc::RefCountedObject<PeerConnection>(
- context, options, is_unified_plan, std::move(event_log),
- std::move(call), dependencies, dtls_enabled));
+ auto pc = rtc::make_ref_counted<PeerConnection>(
+ context, options, is_unified_plan, std::move(event_log), std::move(call),
+ dependencies, dtls_enabled);
RTCError init_error = pc->Initialize(configuration, std::move(dependencies));
if (!init_error.ok()) {
RTC_LOG(LS_ERROR) << "PeerConnection initialization failed";
@@ -632,10 +631,12 @@ RTCError PeerConnection::Initialize(
if (!IsUnifiedPlan()) {
rtp_manager()->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
- signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO)));
+ signaling_thread(),
+ new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO, channel_manager())));
rtp_manager()->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
- signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
+ signaling_thread(),
+ new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO, channel_manager())));
}
int delay_ms = configuration.report_usage_pattern_delay_ms
@@ -996,9 +997,11 @@ PeerConnection::AddTransceiver(
parameters.encodings = init.send_encodings;
// Encodings are dropped from the tail if too many are provided.
- if (parameters.encodings.size() > kMaxSimulcastStreams) {
+ size_t max_simulcast_streams =
+ media_type == cricket::MEDIA_TYPE_VIDEO ? kMaxSimulcastStreams : 1u;
+ if (parameters.encodings.size() > max_simulcast_streams) {
parameters.encodings.erase(
- parameters.encodings.begin() + kMaxSimulcastStreams,
+ parameters.encodings.begin() + max_simulcast_streams,
parameters.encodings.end());
}
@@ -1276,9 +1279,9 @@ absl::optional<bool> PeerConnection::can_trickle_ice_candidates() {
"trickle");
}
-rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel(
- const std::string& label,
- const DataChannelInit* config) {
+RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>>
+PeerConnection::CreateDataChannelOrError(const std::string& label,
+ const DataChannelInit* config) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
@@ -1288,14 +1291,16 @@ rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel(
if (config) {
internal_config.reset(new InternalDataChannelInit(*config));
}
+ // TODO(bugs.webrtc.org/12796): Return a more specific error.
rtc::scoped_refptr<DataChannelInterface> channel(
data_channel_controller_.InternalCreateDataChannelWithProxy(
label, internal_config.get()));
if (!channel.get()) {
- return nullptr;
+ return RTCError(RTCErrorType::INTERNAL_ERROR,
+ "Data channel creation failed");
}
- // Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
+ // Trigger the onRenegotiationNeeded event for
// the first SCTP DataChannel.
if (first_datachannel) {
sdp_handler_->UpdateNegotiationNeeded();
@@ -1878,6 +1883,16 @@ void PeerConnection::SetConnectionState(
configuration_.ice_candidate_pool_size, 0, 255, 256);
break;
}
+
+ // Record whether there was a local or remote provisional answer.
+ ProvisionalAnswerUsage pranswer = kProvisionalAnswerNotUsed;
+ if (local_description()->GetType() == SdpType::kPrAnswer) {
+ pranswer = kProvisionalAnswerLocal;
+ } else if (remote_description()->GetType() == SdpType::kPrAnswer) {
+ pranswer = kProvisionalAnswerRemote;
+ }
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.ProvisionalAnswer",
+ pranswer, kProvisionalAnswerMax);
}
}
@@ -2183,6 +2198,7 @@ cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
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 {};
@@ -2413,21 +2429,20 @@ void PeerConnection::TeardownDataChannelTransport_n() {
}
// Returns false if bundle is enabled and rtcp_mux is disabled.
-bool PeerConnection::ValidateBundleSettings(const SessionDescription* desc) {
- bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
- if (!bundle_enabled)
+bool PeerConnection::ValidateBundleSettings(
+ const SessionDescription* desc,
+ const std::map<std::string, const cricket::ContentGroup*>&
+ bundle_groups_by_mid) {
+ if (bundle_groups_by_mid.empty())
return true;
- const cricket::ContentGroup* bundle_group =
- desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
- RTC_DCHECK(bundle_group != NULL);
-
const cricket::ContentInfos& contents = desc->contents();
for (cricket::ContentInfos::const_iterator citer = contents.begin();
citer != contents.end(); ++citer) {
const cricket::ContentInfo* content = (&*citer);
RTC_DCHECK(content != NULL);
- if (bundle_group->HasContentName(content->name) && !content->rejected &&
+ auto it = bundle_groups_by_mid.find(content->name);
+ if (it != bundle_groups_by_mid.end() && !content->rejected &&
content->type == MediaProtocolType::kRtp) {
if (!HasRtcpMuxEnabled(content))
return false;
@@ -2632,6 +2647,7 @@ void PeerConnection::OnTransportControllerGatheringState(
// 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;
@@ -2865,18 +2881,10 @@ std::function<void(const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us)>
PeerConnection::InitializeRtcpCallback() {
RTC_DCHECK_RUN_ON(network_thread());
- return [this, flag = worker_thread_safety_](
- const rtc::CopyOnWriteBuffer& packet, int64_t packet_time_us) {
+ return [this](const rtc::CopyOnWriteBuffer& packet, int64_t packet_time_us) {
RTC_DCHECK_RUN_ON(network_thread());
- // TODO(bugs.webrtc.org/11993): We should actually be delivering this call
- // directly to the Call class somehow directly on the network thread and not
- // incur this hop here. The DeliverPacket() method will eventually just have
- // to hop back over to the network thread.
- worker_thread()->PostTask(ToQueuedTask(flag, [this, packet,
- packet_time_us] {
- RTC_DCHECK_RUN_ON(worker_thread());
- call_->Receiver()->DeliverPacket(MediaType::ANY, packet, packet_time_us);
- }));
+ call_ptr_->Receiver()->DeliverPacket(MediaType::ANY, packet,
+ packet_time_us);
};
}
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-27 12:48:19 +0000