Google Git
Sign in
chromium / chromium / src / 93094c4962f94a897a6601a75724e4b2db329fae / . / content / renderer / media / webrtc / rtc_peer_connection_handler.cc
blob: 7fc3477c33a35430d73f6659a98f175301a7940b [file] [log] [blame]
// Copyright (c) 2012 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "content/renderer/media/webrtc/rtc_peer_connection_handler.h"
#include <ctype.h>
#include <string.h>
#include <algorithm>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "base/command_line.h"
#include "base/location.h"
#include "base/logging.h"
#include "base/metrics/histogram_functions.h"
#include "base/metrics/histogram_macros.h"
#include "base/stl_util.h"
#include "base/strings/string_util.h"
#include "base/strings/utf_string_conversions.h"
#include "base/threading/thread_checker.h"
#include "base/threading/thread_task_runner_handle.h"
#include "base/trace_event/trace_event.h"
#include "base/unguessable_token.h"
#include "content/public/common/content_features.h"
#include "content/public/common/content_switches.h"
#include "content/renderer/media/stream/media_stream_constraints_util.h"
#include "content/renderer/media/stream/media_stream_track.h"
#include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
#include "content/renderer/media/webrtc/peer_connection_tracker.h"
#include "content/renderer/media/webrtc/rtc_data_channel_handler.h"
#include "content/renderer/media/webrtc/rtc_dtmf_sender_handler.h"
#include "content/renderer/media/webrtc/rtc_event_log_output_sink.h"
#include "content/renderer/media/webrtc/rtc_event_log_output_sink_proxy.h"
#include "content/renderer/media/webrtc/rtc_stats.h"
#include "content/renderer/media/webrtc/webrtc_audio_device_impl.h"
#include "content/renderer/media/webrtc/webrtc_set_description_observer.h"
#include "content/renderer/media/webrtc/webrtc_uma_histograms.h"
#include "content/renderer/render_thread_impl.h"
#include "media/base/media_switches.h"
#include "third_party/blink/public/platform/web_media_constraints.h"
#include "third_party/blink/public/platform/web_rtc_answer_options.h"
#include "third_party/blink/public/platform/web_rtc_data_channel_init.h"
#include "third_party/blink/public/platform/web_rtc_ice_candidate.h"
#include "third_party/blink/public/platform/web_rtc_legacy_stats.h"
#include "third_party/blink/public/platform/web_rtc_offer_options.h"
#include "third_party/blink/public/platform/web_rtc_rtp_sender.h"
#include "third_party/blink/public/platform/web_rtc_rtp_transceiver.h"
#include "third_party/blink/public/platform/web_rtc_session_description.h"
#include "third_party/blink/public/platform/web_rtc_session_description_request.h"
#include "third_party/blink/public/platform/web_rtc_void_request.h"
#include "third_party/blink/public/platform/web_url.h"
#include "third_party/webrtc/api/rtceventlogoutput.h"
#include "third_party/webrtc/pc/mediasession.h"
using webrtc::DataChannelInterface;
using webrtc::IceCandidateInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
using webrtc::StatsReport;
using webrtc::StatsReports;
namespace content {
namespace {
// Used to back histogram value of "WebRTC.PeerConnection.RtcpMux",
// so treat as append-only.
enum RtcpMux {
RTCP_MUX_DISABLED,
RTCP_MUX_ENABLED,
RTCP_MUX_NO_MEDIA,
RTCP_MUX_MAX
};
blink::WebRTCSessionDescription CreateWebKitSessionDescription(
const std::string& sdp, const std::string& type) {
blink::WebRTCSessionDescription description;
description.Initialize(blink::WebString::FromUTF8(type),
blink::WebString::FromUTF8(sdp));
return description;
}
blink::WebRTCSessionDescription
CreateWebKitSessionDescription(
const webrtc::SessionDescriptionInterface* native_desc) {
if (!native_desc) {
LOG(ERROR) << "Native session description is null.";
return blink::WebRTCSessionDescription();
}
std::string sdp;
if (!native_desc->ToString(&sdp)) {
LOG(ERROR) << "Failed to get SDP string of native session description.";
return blink::WebRTCSessionDescription();
}
return CreateWebKitSessionDescription(sdp, native_desc->type());
}
void RunClosureWithTrace(const base::Closure& closure,
const char* trace_event_name) {
TRACE_EVENT0("webrtc", trace_event_name);
closure.Run();
}
void RunSynchronousOnceClosure(base::OnceClosure closure,
const char* trace_event_name,
base::WaitableEvent* event) {
{
TRACE_EVENT0("webrtc", trace_event_name);
std::move(closure).Run();
}
event->Signal();
}
void RunSynchronousClosure(const base::Closure& closure,
const char* trace_event_name,
base::WaitableEvent* event) {
{
TRACE_EVENT0("webrtc", trace_event_name);
closure.Run();
}
event->Signal();
}
// Initializes |web_description| if |description_callback| returns non-null,
// otherwise does nothing.
void GetWebRTCSessionDescriptionFromSessionDescriptionCallback(
base::OnceCallback<const webrtc::SessionDescriptionInterface*()>
description_callback,
blink::WebRTCSessionDescription* web_description) {
const webrtc::SessionDescriptionInterface* description =
std::move(description_callback).Run();
if (description) {
std::string sdp;
description->ToString(&sdp);
web_description->Initialize(blink::WebString::FromUTF8(description->type()),
blink::WebString::FromUTF8(sdp));
}
}
// Converter functions from Blink types to WebRTC types.
absl::optional<bool> ConstraintToOptional(
const blink::WebMediaConstraints& constraints,
const blink::BooleanConstraint blink::WebMediaTrackConstraintSet::*picker) {
bool value;
if (GetConstraintValueAsBoolean(constraints, picker, &value)) {
return absl::optional<bool>(value);
}
return absl::nullopt;
}
void CopyConstraintsIntoRtcConfiguration(
const blink::WebMediaConstraints constraints,
webrtc::PeerConnectionInterface::RTCConfiguration* configuration) {
// Copy info from constraints into configuration, if present.
if (constraints.IsEmpty()) {
return;
}
bool the_value;
if (GetConstraintValueAsBoolean(
constraints, &blink::WebMediaTrackConstraintSet::enable_i_pv6,
&the_value)) {
configuration->disable_ipv6 = !the_value;
} else {
// Note: IPv6 WebRTC value is "disable" while Blink is "enable".
configuration->disable_ipv6 = false;
}
if (GetConstraintValueAsBoolean(
constraints, &blink::WebMediaTrackConstraintSet::enable_dscp,
&the_value)) {
configuration->set_dscp(the_value);
}
if (GetConstraintValueAsBoolean(
constraints,
&blink::WebMediaTrackConstraintSet::goog_cpu_overuse_detection,
&the_value)) {
configuration->set_cpu_adaptation(the_value);
}
if (GetConstraintValueAsBoolean(
constraints,
&blink::WebMediaTrackConstraintSet::
goog_enable_video_suspend_below_min_bitrate,
&the_value)) {
configuration->set_suspend_below_min_bitrate(the_value);
}
if (!GetConstraintValueAsBoolean(
constraints,
&blink::WebMediaTrackConstraintSet::enable_rtp_data_channels,
&configuration->enable_rtp_data_channel)) {
configuration->enable_rtp_data_channel = false;
}
int rate;
if (GetConstraintValueAsInteger(
constraints,
&blink::WebMediaTrackConstraintSet::goog_screencast_min_bitrate,
&rate)) {
configuration->screencast_min_bitrate = rate;
}
configuration->combined_audio_video_bwe = ConstraintToOptional(
constraints,
&blink::WebMediaTrackConstraintSet::goog_combined_audio_video_bwe);
configuration->enable_dtls_srtp = ConstraintToOptional(
constraints, &blink::WebMediaTrackConstraintSet::enable_dtls_srtp);
}
// Class mapping responses from calls to libjingle CreateOffer/Answer and
// the blink::WebRTCSessionDescriptionRequest.
class CreateSessionDescriptionRequest
: public webrtc::CreateSessionDescriptionObserver {
public:
explicit CreateSessionDescriptionRequest(
const scoped_refptr<base::SingleThreadTaskRunner>& main_thread,
const blink::WebRTCSessionDescriptionRequest& request,
const base::WeakPtr<RTCPeerConnectionHandler>& handler,
const base::WeakPtr<PeerConnectionTracker>& tracker,
PeerConnectionTracker::Action action)
: main_thread_(main_thread),
webkit_request_(request),
handler_(handler),
tracker_(tracker),
action_(action) {}
void OnSuccess(webrtc::SessionDescriptionInterface* desc) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(
&CreateSessionDescriptionRequest::OnSuccess,
rtc::scoped_refptr<CreateSessionDescriptionRequest>(this), desc));
return;
}
if (tracker_ && handler_) {
std::string value;
if (desc) {
desc->ToString(&value);
value = "type: " + desc->type() + ", sdp: " + value;
}
tracker_->TrackSessionDescriptionCallback(handler_.get(), action_,
"OnSuccess", value);
}
webkit_request_.RequestSucceeded(CreateWebKitSessionDescription(desc));
webkit_request_.Reset();
delete desc;
}
void OnFailure(webrtc::RTCError error) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(
&CreateSessionDescriptionRequest::OnFailure,
rtc::scoped_refptr<CreateSessionDescriptionRequest>(this),
std::move(error)));
return;
}
if (handler_ && tracker_) {
tracker_->TrackSessionDescriptionCallback(handler_.get(), action_,
"OnFailure", error.message());
}
// TODO(hta): Convert CreateSessionDescriptionRequest.OnFailure
webkit_request_.RequestFailed(error);
webkit_request_.Reset();
}
protected:
~CreateSessionDescriptionRequest() override {
// This object is reference counted and its callback methods |OnSuccess| and
// |OnFailure| will be invoked on libjingle's signaling thread and posted to
// the main thread. Since the main thread may complete before the signaling
// thread has deferenced this object there is no guarantee that this object
// is destructed on the main thread.
DLOG_IF(ERROR, !webkit_request_.IsNull())
<< "CreateSessionDescriptionRequest not completed. Shutting down?";
}
const scoped_refptr<base::SingleThreadTaskRunner> main_thread_;
blink::WebRTCSessionDescriptionRequest webkit_request_;
const base::WeakPtr<RTCPeerConnectionHandler> handler_;
const base::WeakPtr<PeerConnectionTracker> tracker_;
PeerConnectionTracker::Action action_;
};
blink::WebRTCLegacyStatsMemberType
WebRTCLegacyStatsMemberTypeFromStatsValueType(
webrtc::StatsReport::Value::Type type) {
switch (type) {
case StatsReport::Value::kInt:
return blink::kWebRTCLegacyStatsMemberTypeInt;
case StatsReport::Value::kInt64:
return blink::kWebRTCLegacyStatsMemberTypeInt64;
case StatsReport::Value::kFloat:
return blink::kWebRTCLegacyStatsMemberTypeFloat;
case StatsReport::Value::kString:
case StatsReport::Value::kStaticString:
return blink::kWebRTCLegacyStatsMemberTypeString;
case StatsReport::Value::kBool:
return blink::kWebRTCLegacyStatsMemberTypeBool;
case StatsReport::Value::kId:
return blink::kWebRTCLegacyStatsMemberTypeId;
}
NOTREACHED();
return blink::kWebRTCLegacyStatsMemberTypeInt;
}
// Class mapping responses from calls to libjingle
// GetStats into a blink::WebRTCStatsCallback.
class StatsResponse : public webrtc::StatsObserver {
public:
StatsResponse(const scoped_refptr<LocalRTCStatsRequest>& request,
scoped_refptr<base::SingleThreadTaskRunner> task_runner)
: request_(request.get()), main_thread_(task_runner) {
// Measure the overall time it takes to satisfy a getStats request.
TRACE_EVENT_ASYNC_BEGIN0("webrtc", "getStats_Native", this);
DETACH_FROM_THREAD(signaling_thread_checker_);
}
void OnComplete(const StatsReports& reports) override {
DCHECK_CALLED_ON_VALID_THREAD(signaling_thread_checker_);
TRACE_EVENT0("webrtc", "StatsResponse::OnComplete");
// We can't use webkit objects directly since they use a single threaded
// heap allocator.
std::vector<Report*>* report_copies = new std::vector<Report*>();
report_copies->reserve(reports.size());
for (auto* r : reports)
report_copies->push_back(new Report(r));
main_thread_->PostTaskAndReply(
FROM_HERE,
base::BindOnce(&StatsResponse::DeliverCallback,
rtc::scoped_refptr<StatsResponse>(this),
base::Unretained(report_copies)),
base::BindOnce(&StatsResponse::DeleteReports,
base::Unretained(report_copies)));
}
private:
class Report : public blink::WebRTCLegacyStats {
public:
class MemberIterator : public blink::WebRTCLegacyStatsMemberIterator {
public:
MemberIterator(const StatsReport::Values::const_iterator& it,
const StatsReport::Values::const_iterator& end)
: it_(it), end_(end) {}
// blink::WebRTCLegacyStatsMemberIterator
bool IsEnd() const override { return it_ == end_; }
void Next() override { ++it_; }
blink::WebString GetName() const override {
return blink::WebString::FromUTF8(it_->second->display_name());
}
blink::WebRTCLegacyStatsMemberType GetType() const override {
return WebRTCLegacyStatsMemberTypeFromStatsValueType(
it_->second->type());
}
int ValueInt() const override { return it_->second->int_val(); }
int64_t ValueInt64() const override { return it_->second->int64_val(); }
float ValueFloat() const override { return it_->second->float_val(); }
blink::WebString ValueString() const override {
const StatsReport::ValuePtr& value = it_->second;
if (value->type() == StatsReport::Value::kString)
return blink::WebString::FromUTF8(value->string_val());
DCHECK_EQ(value->type(), StatsReport::Value::kStaticString);
return blink::WebString::FromUTF8(value->static_string_val());
}
bool ValueBool() const override { return it_->second->bool_val(); }
blink::WebString ValueToString() const override {
const StatsReport::ValuePtr& value = it_->second;
if (value->type() == StatsReport::Value::kString)
return blink::WebString::FromUTF8(value->string_val());
if (value->type() == StatsReport::Value::kStaticString)
return blink::WebString::FromUTF8(value->static_string_val());
return blink::WebString::FromUTF8(value->ToString());
}
private:
StatsReport::Values::const_iterator it_;
StatsReport::Values::const_iterator end_;
};
Report(const StatsReport* report)
: id_(report->id()->ToString()),
type_(report->type()),
type_name_(report->TypeToString()),
timestamp_(report->timestamp()),
values_(report->values()) {}
~Report() override {
// Since the values vector holds pointers to const objects that are bound
// to the signaling thread, they must be released on the same thread.
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
}
// blink::WebRTCLegacyStats
blink::WebString Id() const override {
return blink::WebString::FromUTF8(id_);
}
blink::WebString GetType() const override {
return blink::WebString::FromUTF8(type_name_);
}
double Timestamp() const override { return timestamp_; }
blink::WebRTCLegacyStatsMemberIterator* Iterator() const override {
return new MemberIterator(values_.cbegin(), values_.cend());
}
bool HasValues() const {
return values_.size() > 0;
}
private:
THREAD_CHECKER(thread_checker_);
const std::string id_;
const StatsReport::StatsType type_;
const std::string type_name_;
const double timestamp_;
const StatsReport::Values values_;
};
static void DeleteReports(std::vector<Report*>* reports) {
TRACE_EVENT0("webrtc", "StatsResponse::DeleteReports");
for (auto* p : *reports)
delete p;
delete reports;
}
void DeliverCallback(const std::vector<Report*>* reports) {
DCHECK(main_thread_->BelongsToCurrentThread());
TRACE_EVENT0("webrtc", "StatsResponse::DeliverCallback");
rtc::scoped_refptr<LocalRTCStatsResponse> response(
request_->createResponse().get());
for (const auto* report : *reports) {
if (report->HasValues())
AddReport(response.get(), *report);
}
// Record the getStats operation as done before calling into Blink so that
// we don't skew the perf measurements of the native code with whatever the
// callback might be doing.
TRACE_EVENT_ASYNC_END0("webrtc", "getStats_Native", this);
request_->requestSucceeded(response);
request_ = nullptr; // must be freed on the main thread.
}
void AddReport(LocalRTCStatsResponse* response, const Report& report) {
response->addStats(report);
}
rtc::scoped_refptr<LocalRTCStatsRequest> request_;
const scoped_refptr<base::SingleThreadTaskRunner> main_thread_;
THREAD_CHECKER(signaling_thread_checker_);
};
void GetStatsOnSignalingThread(
const scoped_refptr<webrtc::PeerConnectionInterface>& pc,
webrtc::PeerConnectionInterface::StatsOutputLevel level,
const scoped_refptr<webrtc::StatsObserver>& observer,
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> selector) {
TRACE_EVENT0("webrtc", "GetStatsOnSignalingThread");
if (selector) {
bool belongs_to_pc = false;
for (const auto& sender : pc->GetSenders()) {
if (sender->track() == selector) {
belongs_to_pc = true;
break;
}
}
if (!belongs_to_pc) {
for (const auto& receiver : pc->GetReceivers()) {
if (receiver->track() == selector) {
belongs_to_pc = true;
break;
}
}
}
if (!belongs_to_pc) {
DVLOG(1) << "GetStats: Track not found.";
observer->OnComplete(StatsReports());
return;
}
}
if (!pc->GetStats(observer.get(), selector.get(), level)) {
DVLOG(1) << "GetStats failed.";
observer->OnComplete(StatsReports());
}
}
void GetRTCStatsOnSignalingThread(
const scoped_refptr<base::SingleThreadTaskRunner>& main_thread,
scoped_refptr<webrtc::PeerConnectionInterface> native_peer_connection,
std::unique_ptr<blink::WebRTCStatsReportCallback> callback) {
TRACE_EVENT0("webrtc", "GetRTCStatsOnSignalingThread");
native_peer_connection->GetStats(
RTCStatsCollectorCallbackImpl::Create(main_thread, std::move(callback)));
}
void ConvertOfferOptionsToWebrtcOfferOptions(
const blink::WebRTCOfferOptions& options,
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions* output) {
output->offer_to_receive_audio = options.OfferToReceiveAudio();
output->offer_to_receive_video = options.OfferToReceiveVideo();
output->voice_activity_detection = options.VoiceActivityDetection();
output->ice_restart = options.IceRestart();
}
void ConvertAnswerOptionsToWebrtcAnswerOptions(
const blink::WebRTCAnswerOptions& options,
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions* output) {
output->voice_activity_detection = options.VoiceActivityDetection();
}
void ConvertConstraintsToWebrtcOfferOptions(
const blink::WebMediaConstraints& constraints,
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions* output) {
if (constraints.IsEmpty()) {
return;
}
std::string failing_name;
if (constraints.Basic().HasMandatoryOutsideSet(
{constraints.Basic().offer_to_receive_audio.GetName(),
constraints.Basic().offer_to_receive_video.GetName(),
constraints.Basic().voice_activity_detection.GetName(),
constraints.Basic().ice_restart.GetName()},
failing_name)) {
// TODO(hta): Reject the calling operation with "constraint error"
// https://crbug.com/594894
DLOG(ERROR) << "Invalid mandatory constraint to CreateOffer/Answer: "
<< failing_name;
}
GetConstraintValueAsInteger(
constraints, &blink::WebMediaTrackConstraintSet::offer_to_receive_audio,
&output->offer_to_receive_audio);
GetConstraintValueAsInteger(
constraints, &blink::WebMediaTrackConstraintSet::offer_to_receive_video,
&output->offer_to_receive_video);
GetConstraintValueAsBoolean(
constraints, &blink::WebMediaTrackConstraintSet::voice_activity_detection,
&output->voice_activity_detection);
GetConstraintValueAsBoolean(constraints,
&blink::WebMediaTrackConstraintSet::ice_restart,
&output->ice_restart);
}
std::set<RTCPeerConnectionHandler*>* GetPeerConnectionHandlers() {
static std::set<RTCPeerConnectionHandler*>* handlers =
new std::set<RTCPeerConnectionHandler*>();
return handlers;
}
// Counts the number of senders that have |stream_id| as an associated stream.
size_t GetLocalStreamUsageCount(
const std::vector<std::unique_ptr<RTCRtpSender>>& rtp_senders,
const std::string stream_id) {
size_t usage_count = 0;
for (const auto& sender : rtp_senders) {
for (const auto& sender_stream_id : sender->state().stream_ids()) {
if (sender_stream_id == stream_id) {
++usage_count;
break;
}
}
}
return usage_count;
}
bool IsRemoteStream(
const std::vector<std::unique_ptr<RTCRtpReceiver>>& rtp_receivers,
const std::string& stream_id) {
for (const auto& receiver : rtp_receivers) {
for (const auto& receiver_stream_id : receiver->state().stream_ids()) {
if (stream_id == receiver_stream_id)
return true;
}
}
return false;
}
MediaStreamTrackMetrics::Kind MediaStreamTrackMetricsKind(
const blink::WebMediaStreamTrack& track) {
return track.Source().GetType() == blink::WebMediaStreamSource::kTypeAudio
? MediaStreamTrackMetrics::Kind::kAudio
: MediaStreamTrackMetrics::Kind::kVideo;
}
} // namespace
// Implementation of LocalRTCStatsRequest.
LocalRTCStatsRequest::LocalRTCStatsRequest(blink::WebRTCStatsRequest impl)
: impl_(impl) {
}
LocalRTCStatsRequest::LocalRTCStatsRequest() {}
LocalRTCStatsRequest::~LocalRTCStatsRequest() {}
bool LocalRTCStatsRequest::hasSelector() const {
return impl_.HasSelector();
}
blink::WebMediaStreamTrack LocalRTCStatsRequest::component() const {
return impl_.Component();
}
scoped_refptr<LocalRTCStatsResponse> LocalRTCStatsRequest::createResponse() {
return scoped_refptr<LocalRTCStatsResponse>(
new rtc::RefCountedObject<LocalRTCStatsResponse>(impl_.CreateResponse()));
}
void LocalRTCStatsRequest::requestSucceeded(
const LocalRTCStatsResponse* response) {
impl_.RequestSucceeded(response->webKitStatsResponse());
}
// Implementation of LocalRTCStatsResponse.
blink::WebRTCStatsResponse LocalRTCStatsResponse::webKitStatsResponse() const {
return impl_;
}
void LocalRTCStatsResponse::addStats(const blink::WebRTCLegacyStats& stats) {
impl_.AddStats(stats);
}
// Processes the resulting state changes of a SetLocalDescription() or
// SetRemoteDescription() call.
class RTCPeerConnectionHandler::WebRtcSetDescriptionObserverImpl
: public WebRtcSetDescriptionObserver {
public:
WebRtcSetDescriptionObserverImpl(
base::WeakPtr<RTCPeerConnectionHandler> handler,
blink::WebRTCVoidRequest web_request,
base::WeakPtr<PeerConnectionTracker> tracker,
scoped_refptr<base::SingleThreadTaskRunner> task_runner,
PeerConnectionTracker::Action action,
webrtc::SdpSemantics sdp_semantics)
: handler_(handler),
main_thread_(task_runner),
web_request_(web_request),
tracker_(tracker),
action_(action),
sdp_semantics_(sdp_semantics) {}
void OnSetDescriptionComplete(
webrtc::RTCError error,
WebRtcSetDescriptionObserver::States states) override {
if (!error.ok()) {
if (tracker_ && handler_) {
tracker_->TrackSessionDescriptionCallback(handler_.get(), action_,
"OnFailure", error.message());
}
web_request_.RequestFailed(error);
web_request_.Reset();
return;
}
if (handler_) {
handler_->OnSignalingChange(states.signaling_state);
// Process the rest of the state changes differently depending on SDP
// semantics.
if (sdp_semantics_ == webrtc::SdpSemantics::kPlanB) {
ProcessStateChangesPlanB(std::move(states));
} else {
DCHECK_EQ(sdp_semantics_, webrtc::SdpSemantics::kUnifiedPlan);
ProcessStateChangesUnifiedPlan(std::move(states));
}
if (tracker_) {
tracker_->TrackSessionDescriptionCallback(handler_.get(), action_,
"OnSuccess", "");
}
}
if (action_ == PeerConnectionTracker::ACTION_SET_REMOTE_DESCRIPTION) {
// Resolve the promise in a post to ensure any events scheduled for
// dispatching will have fired by the time the promise is resolved.
// TODO(hbos): Don't schedule/post to fire events/resolve the promise.
// Instead, do it all synchronously. This must happen as the last step
// before returning so that all effects of SRD have occurred when the
// event executes. https://crbug.com/788558
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(&RTCPeerConnectionHandler::
WebRtcSetDescriptionObserverImpl::ResolvePromise,
this));
} else {
// Resolve promise immediately if we can. https://crbug.com/788558 still
// needs to be addressed for "setLocalDescription(answer)" rejecting a
// transceiver in Unified Plan, but this is a minor edge-case.
ResolvePromise();
}
}
private:
~WebRtcSetDescriptionObserverImpl() override {}
void ResolvePromise() {
web_request_.RequestSucceeded();
web_request_.Reset();
}
void ProcessStateChangesPlanB(WebRtcSetDescriptionObserver::States states) {
DCHECK_EQ(sdp_semantics_, webrtc::SdpSemantics::kPlanB);
// Determine which receivers have been removed before processing the
// removal as to not invalidate the iterator.
std::vector<RTCRtpReceiver*> removed_receivers;
for (auto it = handler_->rtp_receivers_.begin();
it != handler_->rtp_receivers_.end(); ++it) {
if (ReceiverWasRemoved(*(*it), states.transceiver_states))
removed_receivers.push_back(it->get());
}
// Process the addition of remote receivers/tracks.
for (auto& transceiver_state : states.transceiver_states) {
if (ReceiverWasAdded(transceiver_state)) {
handler_->OnAddReceiverPlanB(transceiver_state.MoveReceiverState());
}
}
// Process the removal of remote receivers/tracks.
for (auto* removed_receiver : removed_receivers) {
handler_->OnRemoveReceiverPlanB(RTCRtpReceiver::getId(
removed_receiver->state().webrtc_receiver().get()));
}
}
bool ReceiverWasAdded(const RtpTransceiverState& transceiver_state) {
uintptr_t receiver_id = RTCRtpReceiver::getId(
transceiver_state.receiver_state()->webrtc_receiver().get());
for (const auto& receiver : handler_->rtp_receivers_) {
if (receiver->Id() == receiver_id)
return false;
}
return true;
}
bool ReceiverWasRemoved(
const RTCRtpReceiver& receiver,
const std::vector<RtpTransceiverState>& transceiver_states) {
for (const auto& transceiver_state : transceiver_states) {
if (transceiver_state.receiver_state()->webrtc_receiver() ==
receiver.state().webrtc_receiver()) {
return false;
}
}
return true;
}
void ProcessStateChangesUnifiedPlan(
WebRtcSetDescriptionObserver::States states) {
DCHECK_EQ(sdp_semantics_, webrtc::SdpSemantics::kUnifiedPlan);
handler_->OnModifyTransceivers(
std::move(states.transceiver_states),
action_ == PeerConnectionTracker::ACTION_SET_REMOTE_DESCRIPTION);
}
base::WeakPtr<RTCPeerConnectionHandler> handler_;
scoped_refptr<base::SequencedTaskRunner> main_thread_;
blink::WebRTCVoidRequest web_request_;
base::WeakPtr<PeerConnectionTracker> tracker_;
PeerConnectionTracker::Action action_;
webrtc::SdpSemantics sdp_semantics_;
};
// Receives notifications from a PeerConnection object about state changes. The
// callbacks we receive here come on the webrtc signaling thread, so this class
// takes care of delivering them to an RTCPeerConnectionHandler instance on the
// main thread. In order to do safe PostTask-ing, the class is reference counted
// and checks for the existence of the RTCPeerConnectionHandler instance before
// delivering callbacks on the main thread.
class RTCPeerConnectionHandler::Observer
: public base::RefCountedThreadSafe<RTCPeerConnectionHandler::Observer>,
public PeerConnectionObserver,
public RtcEventLogOutputSink {
public:
Observer(const base::WeakPtr<RTCPeerConnectionHandler>& handler,
scoped_refptr<base::SingleThreadTaskRunner> task_runner)
: handler_(handler), main_thread_(task_runner) {}
// When an RTC event log is sent back from PeerConnection, it arrives here.
void OnWebRtcEventLogWrite(const std::string& output) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(
&RTCPeerConnectionHandler::Observer::OnWebRtcEventLogWrite, this,
output));
} else if (handler_) {
handler_->OnWebRtcEventLogWrite(output);
}
}
protected:
friend class base::RefCountedThreadSafe<RTCPeerConnectionHandler::Observer>;
~Observer() override = default;
// TODO(hbos): Remove once no longer mandatory to implement.
void OnSignalingChange(PeerConnectionInterface::SignalingState) override {}
void OnAddStream(rtc::scoped_refptr<MediaStreamInterface>) override {}
void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface>) override {}
void OnDataChannel(
rtc::scoped_refptr<DataChannelInterface> data_channel) override {
std::unique_ptr<RtcDataChannelHandler> handler(
new RtcDataChannelHandler(main_thread_, data_channel));
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(&RTCPeerConnectionHandler::Observer::OnDataChannelImpl,
this, std::move(handler)));
}
void OnRenegotiationNeeded() override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(
&RTCPeerConnectionHandler::Observer::OnRenegotiationNeeded,
this));
} else if (handler_) {
handler_->OnRenegotiationNeeded();
}
}
void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(
&RTCPeerConnectionHandler::Observer::OnIceConnectionChange, this,
new_state));
} else if (handler_) {
handler_->OnIceConnectionChange(new_state);
}
}
void OnConnectionChange(
PeerConnectionInterface::PeerConnectionState new_state) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(
&RTCPeerConnectionHandler::Observer::OnConnectionChange, this,
new_state));
} else if (handler_) {
handler_->OnConnectionChange(new_state);
}
}
void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(
&RTCPeerConnectionHandler::Observer::OnIceGatheringChange, this,
new_state));
} else if (handler_) {
handler_->OnIceGatheringChange(new_state);
}
}
void OnIceCandidate(const IceCandidateInterface* candidate) override {
std::string sdp;
if (!candidate->ToString(&sdp)) {
NOTREACHED() << "OnIceCandidate: Could not get SDP string.";
return;
}
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(&RTCPeerConnectionHandler::Observer::OnIceCandidateImpl,
this, sdp, candidate->sdp_mid(),
candidate->sdp_mline_index(),
candidate->candidate().component(),
candidate->candidate().address().family()));
}
void OnDataChannelImpl(std::unique_ptr<RtcDataChannelHandler> handler) {
DCHECK(main_thread_->BelongsToCurrentThread());
if (handler_)
handler_->OnDataChannel(std::move(handler));
}
void OnIceCandidateImpl(const std::string& sdp, const std::string& sdp_mid,
int sdp_mline_index, int component, int address_family) {
DCHECK(main_thread_->BelongsToCurrentThread());
if (handler_) {
handler_->OnIceCandidate(sdp, sdp_mid, sdp_mline_index, component,
address_family);
}
}
void OnInterestingUsage(int usage_pattern) override {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(
&RTCPeerConnectionHandler::Observer::OnInterestingUsageImpl, this,
usage_pattern));
}
void OnInterestingUsageImpl(int usage_pattern) {
DCHECK(main_thread_->BelongsToCurrentThread());
if (handler_) {
handler_->OnInterestingUsage(usage_pattern);
}
}
private:
const base::WeakPtr<RTCPeerConnectionHandler> handler_;
const scoped_refptr<base::SingleThreadTaskRunner> main_thread_;
};
RTCPeerConnectionHandler::RTCPeerConnectionHandler(
blink::WebRTCPeerConnectionHandlerClient* client,
PeerConnectionDependencyFactory* dependency_factory,
scoped_refptr<base::SingleThreadTaskRunner> task_runner)
: id_(base::ToUpperASCII(base::UnguessableToken::Create().ToString())),
initialize_called_(false),
client_(client),
is_closed_(false),
dependency_factory_(dependency_factory),
track_adapter_map_(
new WebRtcMediaStreamTrackAdapterMap(dependency_factory_,
task_runner)),
task_runner_(std::move(task_runner)),
weak_factory_(this) {
CHECK(client_);
GetPeerConnectionHandlers()->insert(this);
}
RTCPeerConnectionHandler::~RTCPeerConnectionHandler() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
Stop();
GetPeerConnectionHandlers()->erase(this);
if (peer_connection_tracker_)
peer_connection_tracker_->UnregisterPeerConnection(this);
UMA_HISTOGRAM_COUNTS_10000(
"WebRTC.NumDataChannelsPerPeerConnection", num_data_channels_created_);
}
void RTCPeerConnectionHandler::associateWithFrame(blink::WebLocalFrame* frame) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
DCHECK(frame);
frame_ = frame;
}
bool RTCPeerConnectionHandler::Initialize(
const webrtc::PeerConnectionInterface::RTCConfiguration&
server_configuration,
const blink::WebMediaConstraints& options) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
DCHECK(frame_);
CHECK(!initialize_called_);
initialize_called_ = true;
peer_connection_tracker_ =
RenderThreadImpl::current()->peer_connection_tracker()->AsWeakPtr();
configuration_ = server_configuration;
// Choose between RTC smoothness algorithm and prerenderer smoothing.
// Prerenderer smoothing is turned on if RTC smoothness is turned off.
configuration_.set_prerenderer_smoothing(
base::CommandLine::ForCurrentProcess()->HasSwitch(
switches::kDisableRTCSmoothnessAlgorithm));
configuration_.set_experiment_cpu_load_estimator(
base::FeatureList::IsEnabled(media::kNewEncodeCpuLoadEstimator));
// Configure optional SRTP configurations enabled via the command line.
configuration_.crypto_options = webrtc::CryptoOptions{};
configuration_.crypto_options->srtp.enable_gcm_crypto_suites =
base::CommandLine::ForCurrentProcess()->HasSwitch(
switches::kEnableWebRtcSrtpAesGcm);
configuration_.crypto_options->srtp.enable_encrypted_rtp_header_extensions =
base::CommandLine::ForCurrentProcess()->HasSwitch(
switches::kEnableWebRtcSrtpEncryptedHeaders);
// Copy all the relevant constraints into |config|.
CopyConstraintsIntoRtcConfiguration(options, &configuration_);
peer_connection_observer_ =
new Observer(weak_factory_.GetWeakPtr(), task_runner_);
native_peer_connection_ = dependency_factory_->CreatePeerConnection(
configuration_, frame_, peer_connection_observer_.get());
if (!native_peer_connection_.get()) {
LOG(ERROR) << "Failed to initialize native PeerConnection.";
return false;
}
if (peer_connection_tracker_) {
peer_connection_tracker_->RegisterPeerConnection(this, configuration_,
options, frame_);
}
return true;
}
bool RTCPeerConnectionHandler::InitializeForTest(
const webrtc::PeerConnectionInterface::RTCConfiguration&
server_configuration,
const blink::WebMediaConstraints& options,
const base::WeakPtr<PeerConnectionTracker>& peer_connection_tracker) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
CHECK(!initialize_called_);
initialize_called_ = true;
configuration_ = server_configuration;
peer_connection_observer_ =
new Observer(weak_factory_.GetWeakPtr(), task_runner_);
CopyConstraintsIntoRtcConfiguration(options, &configuration_);
native_peer_connection_ = dependency_factory_->CreatePeerConnection(
configuration_, nullptr, peer_connection_observer_.get());
if (!native_peer_connection_.get()) {
LOG(ERROR) << "Failed to initialize native PeerConnection.";
return false;
}
peer_connection_tracker_ = peer_connection_tracker;
return true;
}
void RTCPeerConnectionHandler::CreateOffer(
const blink::WebRTCSessionDescriptionRequest& request,
const blink::WebMediaConstraints& options) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createOffer");
scoped_refptr<CreateSessionDescriptionRequest> description_request(
new rtc::RefCountedObject<CreateSessionDescriptionRequest>(
task_runner_, request, weak_factory_.GetWeakPtr(),
peer_connection_tracker_,
PeerConnectionTracker::ACTION_CREATE_OFFER));
// TODO(tommi): Do this asynchronously via e.g. PostTaskAndReply.
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions webrtc_options;
ConvertConstraintsToWebrtcOfferOptions(options, &webrtc_options);
native_peer_connection_->CreateOffer(description_request.get(),
webrtc_options);
if (peer_connection_tracker_)
peer_connection_tracker_->TrackCreateOffer(this, options);
}
void RTCPeerConnectionHandler::CreateOffer(
const blink::WebRTCSessionDescriptionRequest& request,
const blink::WebRTCOfferOptions& options) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createOffer");
scoped_refptr<CreateSessionDescriptionRequest> description_request(
new rtc::RefCountedObject<CreateSessionDescriptionRequest>(
task_runner_, request, weak_factory_.GetWeakPtr(),
peer_connection_tracker_,
PeerConnectionTracker::ACTION_CREATE_OFFER));
// TODO(tommi): Do this asynchronously via e.g. PostTaskAndReply.
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions webrtc_options;
ConvertOfferOptionsToWebrtcOfferOptions(options, &webrtc_options);
native_peer_connection_->CreateOffer(description_request.get(),
webrtc_options);
if (peer_connection_tracker_)
peer_connection_tracker_->TrackCreateOffer(this, options);
}
void RTCPeerConnectionHandler::CreateAnswer(
const blink::WebRTCSessionDescriptionRequest& request,
const blink::WebMediaConstraints& options) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createAnswer");
scoped_refptr<CreateSessionDescriptionRequest> description_request(
new rtc::RefCountedObject<CreateSessionDescriptionRequest>(
task_runner_, request, weak_factory_.GetWeakPtr(),
peer_connection_tracker_,
PeerConnectionTracker::ACTION_CREATE_ANSWER));
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions webrtc_options;
ConvertConstraintsToWebrtcOfferOptions(options, &webrtc_options);
// TODO(tommi): Do this asynchronously via e.g. PostTaskAndReply.
native_peer_connection_->CreateAnswer(description_request.get(),
webrtc_options);
if (peer_connection_tracker_)
peer_connection_tracker_->TrackCreateAnswer(this, options);
}
void RTCPeerConnectionHandler::CreateAnswer(
const blink::WebRTCSessionDescriptionRequest& request,
const blink::WebRTCAnswerOptions& options) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createAnswer");
scoped_refptr<CreateSessionDescriptionRequest> description_request(
new rtc::RefCountedObject<CreateSessionDescriptionRequest>(
task_runner_, request, weak_factory_.GetWeakPtr(),
peer_connection_tracker_,
PeerConnectionTracker::ACTION_CREATE_ANSWER));
// TODO(tommi): Do this asynchronously via e.g. PostTaskAndReply.
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions webrtc_options;
ConvertAnswerOptionsToWebrtcAnswerOptions(options, &webrtc_options);
native_peer_connection_->CreateAnswer(description_request.get(),
webrtc_options);
if (peer_connection_tracker_)
peer_connection_tracker_->TrackCreateAnswer(this, options);
}
bool IsOfferOrAnswer(const webrtc::SessionDescriptionInterface* native_desc) {
DCHECK(native_desc);
return native_desc->type() == "offer" || native_desc->type() == "answer";
}
void RTCPeerConnectionHandler::SetLocalDescription(
const blink::WebRTCVoidRequest& request,
const blink::WebRTCSessionDescription& description) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::setLocalDescription");
std::string sdp = description.Sdp().Utf8();
std::string type = description.GetType().Utf8();
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackSetSessionDescription(
this, sdp, type, PeerConnectionTracker::SOURCE_LOCAL);
}
webrtc::SdpParseError error;
// Since CreateNativeSessionDescription uses the dependency factory, we need
// to make this call on the current thread to be safe.
webrtc::SessionDescriptionInterface* native_desc =
CreateNativeSessionDescription(sdp, type, &error);
if (!native_desc) {
std::string reason_str = "Failed to parse SessionDescription. ";
reason_str.append(error.line);
reason_str.append(" ");
reason_str.append(error.description);
LOG(ERROR) << reason_str;
request.RequestFailed(webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR,
std::move(reason_str)));
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackSessionDescriptionCallback(
this, PeerConnectionTracker::ACTION_SET_LOCAL_DESCRIPTION,
"OnFailure", reason_str);
}
return;
}
if (!first_local_description_ && IsOfferOrAnswer(native_desc)) {
first_local_description_.reset(new FirstSessionDescription(native_desc));
if (first_remote_description_) {
ReportFirstSessionDescriptions(
*first_local_description_,
*first_remote_description_);
}
}
scoped_refptr<WebRtcSetDescriptionObserverImpl> content_observer(
new WebRtcSetDescriptionObserverImpl(
weak_factory_.GetWeakPtr(), request, peer_connection_tracker_,
task_runner_, PeerConnectionTracker::ACTION_SET_LOCAL_DESCRIPTION,
configuration_.sdp_semantics));
bool surface_receivers_only =
(configuration_.sdp_semantics == webrtc::SdpSemantics::kPlanB);
scoped_refptr<webrtc::SetSessionDescriptionObserver> webrtc_observer(
WebRtcSetLocalDescriptionObserverHandler::Create(
task_runner_, signaling_thread(), native_peer_connection_,
track_adapter_map_, content_observer, surface_receivers_only)
.get());
signaling_thread()->PostTask(
FROM_HERE,
base::BindOnce(
&RunClosureWithTrace,
base::Bind(&webrtc::PeerConnectionInterface::SetLocalDescription,
native_peer_connection_,
base::RetainedRef(webrtc_observer),
base::Unretained(native_desc)),
"SetLocalDescription"));
}
void RTCPeerConnectionHandler::SetRemoteDescription(
const blink::WebRTCVoidRequest& request,
const blink::WebRTCSessionDescription& description) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::setRemoteDescription");
std::string sdp = description.Sdp().Utf8();
std::string type = description.GetType().Utf8();
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackSetSessionDescription(
this, sdp, type, PeerConnectionTracker::SOURCE_REMOTE);
}
webrtc::SdpParseError error;
// Since CreateNativeSessionDescription uses the dependency factory, we need
// to make this call on the current thread to be safe.
std::unique_ptr<webrtc::SessionDescriptionInterface> native_desc(
CreateNativeSessionDescription(sdp, type, &error));
if (!native_desc) {
std::string reason_str = "Failed to parse SessionDescription. ";
reason_str.append(error.line);
reason_str.append(" ");
reason_str.append(error.description);
LOG(ERROR) << reason_str;
request.RequestFailed(webrtc::RTCError(
webrtc::RTCErrorType::UNSUPPORTED_OPERATION, std::move(reason_str)));
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackSessionDescriptionCallback(
this, PeerConnectionTracker::ACTION_SET_REMOTE_DESCRIPTION,
"OnFailure", reason_str);
}
return;
}
if (!first_remote_description_ && IsOfferOrAnswer(native_desc.get())) {
first_remote_description_.reset(
new FirstSessionDescription(native_desc.get()));
if (first_local_description_) {
ReportFirstSessionDescriptions(
*first_local_description_,
*first_remote_description_);
}
}
scoped_refptr<WebRtcSetDescriptionObserverImpl> content_observer(
new WebRtcSetDescriptionObserverImpl(
weak_factory_.GetWeakPtr(), request, peer_connection_tracker_,
task_runner_, PeerConnectionTracker::ACTION_SET_REMOTE_DESCRIPTION,
configuration_.sdp_semantics));
bool surface_receivers_only =
(configuration_.sdp_semantics == webrtc::SdpSemantics::kPlanB);
rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>
webrtc_observer(WebRtcSetRemoteDescriptionObserverHandler::Create(
task_runner_, signaling_thread(),
native_peer_connection_, track_adapter_map_,
content_observer, surface_receivers_only)
.get());
signaling_thread()->PostTask(
FROM_HERE,
base::BindOnce(
&RunClosureWithTrace,
base::Bind(
static_cast<void (webrtc::PeerConnectionInterface::*)(
std::unique_ptr<webrtc::SessionDescriptionInterface>,
rtc::scoped_refptr<
webrtc::SetRemoteDescriptionObserverInterface>)>(
&webrtc::PeerConnectionInterface::SetRemoteDescription),
native_peer_connection_, base::Passed(&native_desc),
webrtc_observer),
"SetRemoteDescription"));
}
blink::WebRTCSessionDescription RTCPeerConnectionHandler::LocalDescription() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::localDescription");
base::OnceCallback<const webrtc::SessionDescriptionInterface*()>
description_cb =
base::BindOnce(&webrtc::PeerConnectionInterface::local_description,
native_peer_connection_);
return GetWebRTCSessionDescriptionOnSignalingThread(std::move(description_cb),
"localDescription");
}
blink::WebRTCSessionDescription RTCPeerConnectionHandler::RemoteDescription() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::remoteDescription");
base::OnceCallback<const webrtc::SessionDescriptionInterface*()>
description_cb =
base::BindOnce(&webrtc::PeerConnectionInterface::remote_description,
native_peer_connection_);
return GetWebRTCSessionDescriptionOnSignalingThread(std::move(description_cb),
"remoteDescription");
}
blink::WebRTCSessionDescription
RTCPeerConnectionHandler::CurrentLocalDescription() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::currentLocalDescription");
base::OnceCallback<const webrtc::SessionDescriptionInterface*()>
description_cb = base::BindOnce(
&webrtc::PeerConnectionInterface::current_local_description,
native_peer_connection_);
return GetWebRTCSessionDescriptionOnSignalingThread(
std::move(description_cb), "currentLocalDescription");
}
blink::WebRTCSessionDescription
RTCPeerConnectionHandler::CurrentRemoteDescription() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::currentRemoteDescription");
base::OnceCallback<const webrtc::SessionDescriptionInterface*()>
description_cb = base::BindOnce(
&webrtc::PeerConnectionInterface::current_remote_description,
native_peer_connection_);
return GetWebRTCSessionDescriptionOnSignalingThread(
std::move(description_cb), "currentRemoteDescription");
}
blink::WebRTCSessionDescription
RTCPeerConnectionHandler::PendingLocalDescription() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::pendingLocalDescription");
base::OnceCallback<const webrtc::SessionDescriptionInterface*()>
description_cb = base::BindOnce(
&webrtc::PeerConnectionInterface::pending_local_description,
native_peer_connection_);
return GetWebRTCSessionDescriptionOnSignalingThread(
std::move(description_cb), "pendingLocalDescription");
}
blink::WebRTCSessionDescription
RTCPeerConnectionHandler::PendingRemoteDescription() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::pendingRemoteDescription");
base::OnceCallback<const webrtc::SessionDescriptionInterface*()>
description_cb = base::BindOnce(
&webrtc::PeerConnectionInterface::pending_remote_description,
native_peer_connection_);
return GetWebRTCSessionDescriptionOnSignalingThread(
std::move(description_cb), "pendingRemoteDescription");
}
const webrtc::PeerConnectionInterface::RTCConfiguration&
RTCPeerConnectionHandler::GetConfiguration() const {
return configuration_;
}
webrtc::RTCErrorType RTCPeerConnectionHandler::SetConfiguration(
const webrtc::PeerConnectionInterface::RTCConfiguration& blink_config) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::setConfiguration");
// Update the configuration with the potentially modified fields
webrtc::PeerConnectionInterface::RTCConfiguration new_configuration =
configuration_;
new_configuration.servers = blink_config.servers;
new_configuration.type = blink_config.type;
new_configuration.bundle_policy = blink_config.bundle_policy;
new_configuration.rtcp_mux_policy = blink_config.rtcp_mux_policy;
new_configuration.sdp_semantics = blink_config.sdp_semantics;
new_configuration.certificates = blink_config.certificates;
new_configuration.ice_candidate_pool_size =
blink_config.ice_candidate_pool_size;
if (peer_connection_tracker_)
peer_connection_tracker_->TrackSetConfiguration(this, new_configuration);
webrtc::RTCError webrtc_error;
bool ret = native_peer_connection_->SetConfiguration(new_configuration,
&webrtc_error);
// The boolean return value is made redundant by the error output param; just
// DCHECK that they're consistent.
DCHECK_EQ(ret, webrtc_error.type() == webrtc::RTCErrorType::NONE);
if (webrtc_error.ok()) {
configuration_ = new_configuration;
}
return webrtc_error.type();
}
bool RTCPeerConnectionHandler::AddICECandidate(
const blink::WebRTCVoidRequest& request,
scoped_refptr<blink::WebRTCICECandidate> candidate) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::addICECandidate");
// Libjingle currently does not accept callbacks for addICECandidate.
// For that reason we are going to call callbacks from here.
// TODO(tommi): Instead of calling addICECandidate here, we can do a
// PostTaskAndReply kind of a thing.
bool result = AddICECandidate(std::move(candidate));
task_runner_->PostTask(
FROM_HERE,
base::BindOnce(&RTCPeerConnectionHandler::OnaddICECandidateResult,
weak_factory_.GetWeakPtr(), request, result));
// On failure callback will be triggered.
return true;
}
bool RTCPeerConnectionHandler::AddICECandidate(
scoped_refptr<blink::WebRTCICECandidate> candidate) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::addICECandidate");
std::unique_ptr<webrtc::IceCandidateInterface> native_candidate(
dependency_factory_->CreateIceCandidate(candidate->SdpMid().Utf8(),
candidate->SdpMLineIndex(),
candidate->Candidate().Utf8()));
bool return_value = false;
if (native_candidate) {
return_value =
native_peer_connection_->AddIceCandidate(native_candidate.get());
LOG_IF(ERROR, !return_value) << "Error processing ICE candidate.";
} else {
LOG(ERROR) << "Could not create native ICE candidate.";
}
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackAddIceCandidate(
this, std::move(candidate), PeerConnectionTracker::SOURCE_REMOTE,
return_value);
}
return return_value;
}
void RTCPeerConnectionHandler::OnaddICECandidateResult(
const blink::WebRTCVoidRequest& webkit_request, bool result) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnaddICECandidateResult");
if (!result) {
// We don't have the actual error code from the libjingle, so for now
// using a generic error string.
return webkit_request.RequestFailed(
webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
std::move("Error processing ICE candidate")));
}
return webkit_request.RequestSucceeded();
}
void RTCPeerConnectionHandler::GetStats(
const blink::WebRTCStatsRequest& request) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
scoped_refptr<LocalRTCStatsRequest> inner_request(
new rtc::RefCountedObject<LocalRTCStatsRequest>(request));
getStats(inner_request);
}
void RTCPeerConnectionHandler::getStats(
const scoped_refptr<LocalRTCStatsRequest>& request) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::getStats");
rtc::scoped_refptr<webrtc::StatsObserver> observer(
new rtc::RefCountedObject<StatsResponse>(request, task_runner_));
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> selector;
if (request->hasSelector()) {
auto track_adapter_ref =
track_adapter_map_->GetLocalTrackAdapter(request->component());
if (!track_adapter_ref) {
track_adapter_ref =
track_adapter_map_->GetRemoteTrackAdapter(request->component());
}
if (track_adapter_ref)
selector = track_adapter_ref->webrtc_track();
}
GetStats(observer, webrtc::PeerConnectionInterface::kStatsOutputLevelStandard,
std::move(selector));
}
// TODO(tommi,hbos): It's weird to have three {g|G}etStats methods for the
// legacy stats collector API and even more for the new stats API. Clean it up.
// TODO(hbos): Rename old |getStats| and related functions to "getLegacyStats",
// rename new |getStats|'s helper functions from "GetRTCStats*" to "GetStats*".
void RTCPeerConnectionHandler::GetStats(
webrtc::StatsObserver* observer,
webrtc::PeerConnectionInterface::StatsOutputLevel level,
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> selector) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
signaling_thread()->PostTask(
FROM_HERE,
base::BindOnce(&GetStatsOnSignalingThread, native_peer_connection_, level,
base::WrapRefCounted(observer), std::move(selector)));
}
void RTCPeerConnectionHandler::GetStats(
std::unique_ptr<blink::WebRTCStatsReportCallback> callback) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
signaling_thread()->PostTask(
FROM_HERE, base::BindOnce(&GetRTCStatsOnSignalingThread, task_runner_,
native_peer_connection_, std::move(callback)));
}
webrtc::RTCErrorOr<std::unique_ptr<blink::WebRTCRtpTransceiver>>
RTCPeerConnectionHandler::AddTransceiverWithTrack(
const blink::WebMediaStreamTrack& web_track,
const webrtc::RtpTransceiverInit& init) {
DCHECK_EQ(configuration_.sdp_semantics, webrtc::SdpSemantics::kUnifiedPlan);
DCHECK(task_runner_->RunsTasksInCurrentSequence());
std::unique_ptr<WebRtcMediaStreamTrackAdapterMap::AdapterRef> track_ref =
track_adapter_map_->GetOrCreateLocalTrackAdapter(web_track);
TransceiverStateSurfacer transceiver_state_surfacer(task_runner_,
signaling_thread());
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>>
error_or_transceiver;
RunSynchronousClosureOnSignalingThread(
base::BindRepeating(
&RTCPeerConnectionHandler::AddTransceiverWithTrackOnSignalingThread,
base::Unretained(this), base::RetainedRef(track_ref->webrtc_track()),
base::ConstRef(init), base::Unretained(&transceiver_state_surfacer),
base::Unretained(&error_or_transceiver)),
"AddTransceiverWithTrackOnSignalingThread");
if (!error_or_transceiver.ok()) {
// Don't leave the surfacer in a pending state.
transceiver_state_surfacer.ObtainStates();
return error_or_transceiver.MoveError();
}
auto transceiver_states = transceiver_state_surfacer.ObtainStates();
auto transceiver =
CreateOrUpdateTransceiver(std::move(transceiver_states[0]));
std::unique_ptr<blink::WebRTCRtpTransceiver> web_transceiver =
std::move(transceiver);
return std::move(web_transceiver);
}
void RTCPeerConnectionHandler::AddTransceiverWithTrackOnSignalingThread(
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> webrtc_track,
webrtc::RtpTransceiverInit init,
TransceiverStateSurfacer* transceiver_state_surfacer,
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>>*
error_or_transceiver) {
*error_or_transceiver =
native_peer_connection_->AddTransceiver(webrtc_track, init);
std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> transceivers;
if (error_or_transceiver->ok())
transceivers.push_back(error_or_transceiver->value());
transceiver_state_surfacer->Initialize(track_adapter_map_, transceivers);
}
webrtc::RTCErrorOr<std::unique_ptr<blink::WebRTCRtpTransceiver>>
RTCPeerConnectionHandler::AddTransceiverWithKind(
std::string kind,
const webrtc::RtpTransceiverInit& init) {
DCHECK_EQ(configuration_.sdp_semantics, webrtc::SdpSemantics::kUnifiedPlan);
DCHECK(task_runner_->RunsTasksInCurrentSequence());
cricket::MediaType media_type;
if (kind == webrtc::MediaStreamTrackInterface::kAudioKind) {
media_type = cricket::MEDIA_TYPE_AUDIO;
} else {
DCHECK_EQ(kind, webrtc::MediaStreamTrackInterface::kVideoKind);
media_type = cricket::MEDIA_TYPE_VIDEO;
}
TransceiverStateSurfacer transceiver_state_surfacer(task_runner_,
signaling_thread());
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>>
error_or_transceiver;
RunSynchronousClosureOnSignalingThread(
base::BindRepeating(&RTCPeerConnectionHandler::
AddTransceiverWithMediaTypeOnSignalingThread,
base::Unretained(this), base::ConstRef(media_type),
base::ConstRef(init),
base::Unretained(&transceiver_state_surfacer),
base::Unretained(&error_or_transceiver)),
"AddTransceiverWithMediaTypeOnSignalingThread");
if (!error_or_transceiver.ok()) {
// Don't leave the surfacer in a pending state.
transceiver_state_surfacer.ObtainStates();
return error_or_transceiver.MoveError();
}
auto transceiver_states = transceiver_state_surfacer.ObtainStates();
auto transceiver =
CreateOrUpdateTransceiver(std::move(transceiver_states[0]));
std::unique_ptr<blink::WebRTCRtpTransceiver> web_transceiver =
std::move(transceiver);
return std::move(web_transceiver);
}
void RTCPeerConnectionHandler::AddTransceiverWithMediaTypeOnSignalingThread(
cricket::MediaType media_type,
webrtc::RtpTransceiverInit init,
TransceiverStateSurfacer* transceiver_state_surfacer,
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>>*
error_or_transceiver) {
*error_or_transceiver =
native_peer_connection_->AddTransceiver(media_type, init);
std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> transceivers;
if (error_or_transceiver->ok())
transceivers.push_back(error_or_transceiver->value());
transceiver_state_surfacer->Initialize(track_adapter_map_, transceivers);
}
webrtc::RTCErrorOr<std::unique_ptr<blink::WebRTCRtpTransceiver>>
RTCPeerConnectionHandler::AddTrack(
const blink::WebMediaStreamTrack& track,
const blink::WebVector<blink::WebMediaStream>& streams) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::AddTrack");
std::unique_ptr<WebRtcMediaStreamTrackAdapterMap::AdapterRef> track_ref =
track_adapter_map_->GetOrCreateLocalTrackAdapter(track);
std::vector<std::string> stream_ids(streams.size());
for (size_t i = 0; i < streams.size(); ++i)
stream_ids[i] = streams[i].Id().Utf8();
// Invoke native AddTrack() on the signaling thread and surface the resulting
// transceiver (Plan B: sender only).
// TODO(hbos): Implement and surface full transceiver support under Unified
// Plan. https://crbug.com/777617
TransceiverStateSurfacer transceiver_state_surfacer(task_runner_,
signaling_thread());
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>>
error_or_sender;
RunSynchronousClosureOnSignalingThread(
base::BindRepeating(
&RTCPeerConnectionHandler::AddTrackOnSignalingThread,
base::Unretained(this), base::RetainedRef(track_ref->webrtc_track()),
std::move(stream_ids), base::Unretained(&transceiver_state_surfacer),
base::Unretained(&error_or_sender)),
"AddTrackOnSignalingThread");
DCHECK(transceiver_state_surfacer.is_initialized());
if (!error_or_sender.ok()) {
// Don't leave the surfacer in a pending state.
transceiver_state_surfacer.ObtainStates();
return error_or_sender.MoveError();
}
track_metrics_.AddTrack(MediaStreamTrackMetrics::Direction::kSend,
MediaStreamTrackMetricsKind(track),
track.Id().Utf8());
auto transceiver_states = transceiver_state_surfacer.ObtainStates();
DCHECK_EQ(transceiver_states.size(), 1u);
auto transceiver_state = std::move(transceiver_states[0]);
std::unique_ptr<blink::WebRTCRtpTransceiver> web_transceiver;
if (configuration_.sdp_semantics == webrtc::SdpSemantics::kPlanB) {
// Plan B: Create sender only.
DCHECK(transceiver_state.sender_state());
auto webrtc_sender = transceiver_state.sender_state()->webrtc_sender();
DCHECK(FindSender(RTCRtpSender::getId(webrtc_sender.get())) ==
rtp_senders_.end());
RtpSenderState sender_state = transceiver_state.MoveSenderState();
DCHECK(sender_state.is_initialized());
rtp_senders_.push_back(std::make_unique<RTCRtpSender>(
native_peer_connection_, track_adapter_map_, std::move(sender_state)));
web_transceiver = std::make_unique<RTCRtpSenderOnlyTransceiver>(
std::make_unique<RTCRtpSender>(*rtp_senders_.back().get()));
} else {
DCHECK_EQ(configuration_.sdp_semantics, webrtc::SdpSemantics::kUnifiedPlan);
// Unified Plan: Create or recycle a transceiver.
auto transceiver = CreateOrUpdateTransceiver(std::move(transceiver_state));
web_transceiver = std::move(transceiver);
}
if (peer_connection_tracker_) {
size_t transceiver_index = GetTransceiverIndex(*web_transceiver.get());
peer_connection_tracker_->TrackAddTransceiver(
this, PeerConnectionTracker::TransceiverUpdatedReason::kAddTrack,
*web_transceiver.get(), transceiver_index);
}
for (const auto& stream_id : rtp_senders_.back()->state().stream_ids()) {
if (GetLocalStreamUsageCount(rtp_senders_, stream_id) == 1u) {
// This is the first occurrence of this stream.
PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
}
}
return web_transceiver;
}
void RTCPeerConnectionHandler::AddTrackOnSignalingThread(
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track,
std::vector<std::string> stream_ids,
TransceiverStateSurfacer* transceiver_state_surfacer,
webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>>*
error_or_sender) {
*error_or_sender = native_peer_connection_->AddTrack(track, stream_ids);
std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> transceivers;
if (error_or_sender->ok()) {
auto sender = error_or_sender->value();
if (configuration_.sdp_semantics == webrtc::SdpSemantics::kPlanB) {
transceivers = {new SurfaceSenderStateOnly(sender)};
} else {
DCHECK_EQ(configuration_.sdp_semantics,
webrtc::SdpSemantics::kUnifiedPlan);
rtc::scoped_refptr<webrtc::RtpTransceiverInterface>
transceiver_for_sender = nullptr;
for (const auto& transceiver :
native_peer_connection_->GetTransceivers()) {
if (transceiver->sender() == sender) {
transceiver_for_sender = transceiver;
break;
}
}
DCHECK(transceiver_for_sender);
transceivers = {transceiver_for_sender};
}
}
transceiver_state_surfacer->Initialize(track_adapter_map_,
std::move(transceivers));
}
webrtc::RTCErrorOr<std::unique_ptr<blink::WebRTCRtpTransceiver>>
RTCPeerConnectionHandler::RemoveTrack(blink::WebRTCRtpSender* web_sender) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::RemoveTrack");
if (configuration_.sdp_semantics == webrtc::SdpSemantics::kPlanB) {
if (RemoveTrackPlanB(web_sender)) {
// In Plan B, null indicates success.
std::unique_ptr<blink::WebRTCRtpTransceiver> web_transceiver = nullptr;
return std::move(web_transceiver);
}
// TODO(hbos): Surface RTCError from third_party/webrtc when
// peerconnectioninterface.h is updated. https://crbug.com/webrtc/9534
return webrtc::RTCError(webrtc::RTCErrorType::INVALID_STATE);
}
DCHECK_EQ(configuration_.sdp_semantics, webrtc::SdpSemantics::kUnifiedPlan);
return RemoveTrackUnifiedPlan(web_sender);
}
bool RTCPeerConnectionHandler::RemoveTrackPlanB(
blink::WebRTCRtpSender* web_sender) {
DCHECK_EQ(configuration_.sdp_semantics, webrtc::SdpSemantics::kPlanB);
auto web_track = web_sender->Track();
auto it = FindSender(web_sender->Id());
if (it == rtp_senders_.end())
return false;
if (!(*it)->RemoveFromPeerConnection(native_peer_connection_.get()))
return false;
track_metrics_.RemoveTrack(MediaStreamTrackMetrics::Direction::kSend,
MediaStreamTrackMetricsKind(web_track),
web_track.Id().Utf8());
if (peer_connection_tracker_) {
auto sender_only_transceiver =
std::make_unique<RTCRtpSenderOnlyTransceiver>(
std::make_unique<RTCRtpSender>(*it->get()));
size_t sender_index = GetTransceiverIndex(*sender_only_transceiver);
peer_connection_tracker_->TrackRemoveTransceiver(
this, PeerConnectionTracker::TransceiverUpdatedReason::kRemoveTrack,
*sender_only_transceiver.get(), sender_index);
}
std::vector<std::string> stream_ids = (*it)->state().stream_ids();
rtp_senders_.erase(it);
for (const auto& stream_id : stream_ids) {
if (GetLocalStreamUsageCount(rtp_senders_, stream_id) == 0u) {
// This was the last occurrence of this stream.
PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
}
}
return true;
}
webrtc::RTCErrorOr<std::unique_ptr<blink::WebRTCRtpTransceiver>>
RTCPeerConnectionHandler::RemoveTrackUnifiedPlan(
blink::WebRTCRtpSender* web_sender) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
DCHECK_EQ(configuration_.sdp_semantics, webrtc::SdpSemantics::kUnifiedPlan);
auto it = FindSender(web_sender->Id());
if (it == rtp_senders_.end())
return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER);
const auto& sender = *it;
auto webrtc_sender = sender->state().webrtc_sender();
TransceiverStateSurfacer transceiver_state_surfacer(task_runner_,
signaling_thread());
bool result;
RunSynchronousClosureOnSignalingThread(
base::BindRepeating(
&RTCPeerConnectionHandler::RemoveTrackUnifiedPlanOnSignalingThread,
base::Unretained(this), base::RetainedRef(webrtc_sender),
base::Unretained(&transceiver_state_surfacer),
base::Unretained(&result)),
"RemoveTrackUnifiedPlanOnSignalingThread");
DCHECK(transceiver_state_surfacer.is_initialized());
if (!result) {
// Don't leave the surfacer in a pending state.
transceiver_state_surfacer.ObtainStates();
// TODO(hbos): Surface RTCError from third_party/webrtc when
// peerconnectioninterface.h is updated. https://crbug.com/webrtc/9534
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
auto transceiver_states = transceiver_state_surfacer.ObtainStates();
DCHECK_EQ(transceiver_states.size(), 1u);
auto transceiver_state = std::move(transceiver_states[0]);
// Update the transceiver.
auto transceiver = CreateOrUpdateTransceiver(std::move(transceiver_state));
if (peer_connection_tracker_) {
size_t transceiver_index = GetTransceiverIndex(*transceiver);
peer_connection_tracker_->TrackModifyTransceiver(
this, PeerConnectionTracker::TransceiverUpdatedReason::kRemoveTrack,
*transceiver.get(), transceiver_index);
}
std::unique_ptr<blink::WebRTCRtpTransceiver> web_transceiver =
std::move(transceiver);
return web_transceiver;
}
void RTCPeerConnectionHandler::RemoveTrackUnifiedPlanOnSignalingThread(
rtc::scoped_refptr<webrtc::RtpSenderInterface> sender,
TransceiverStateSurfacer* transceiver_state_surfacer,
bool* result) {
*result = native_peer_connection_->RemoveTrack(sender);
std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> transceivers;
if (*result) {
rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver_for_sender =
nullptr;
for (const auto& transceiver : native_peer_connection_->GetTransceivers()) {
if (transceiver->sender() == sender) {
transceiver_for_sender = transceiver;
break;
}
}
DCHECK(transceiver_for_sender);
transceivers = {transceiver_for_sender};
}
transceiver_state_surfacer->Initialize(track_adapter_map_,
std::move(transceivers));
}
void RTCPeerConnectionHandler::CloseClientPeerConnection() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
if (!is_closed_)
client_->ClosePeerConnection();
}
void RTCPeerConnectionHandler::StartEventLog(IPC::PlatformFileForTransit file,
int64_t max_file_size_bytes) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
DCHECK(file != IPC::InvalidPlatformFileForTransit());
// TODO(eladalon): StartRtcEventLog() return value is not useful; remove it
// or find a way to be able to use it.
// https://crbug.com/775415
native_peer_connection_->StartRtcEventLog(
IPC::PlatformFileForTransitToPlatformFile(file), max_file_size_bytes);
}
void RTCPeerConnectionHandler::StartEventLog() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
// TODO(eladalon): StartRtcEventLog() return value is not useful; remove it
// or find a way to be able to use it.
// https://crbug.com/775415
native_peer_connection_->StartRtcEventLog(
std::make_unique<RtcEventLogOutputSinkProxy>(
peer_connection_observer_.get()),
webrtc::RtcEventLog::kImmediateOutput);
}
void RTCPeerConnectionHandler::StopEventLog() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
native_peer_connection_->StopRtcEventLog();
}
void RTCPeerConnectionHandler::OnWebRtcEventLogWrite(
const std::string& output) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackRtcEventLogWrite(this, output);
}
}
blink::WebRTCDataChannelHandler* RTCPeerConnectionHandler::CreateDataChannel(
const blink::WebString& label,
const blink::WebRTCDataChannelInit& init) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDataChannel");
DVLOG(1) << "createDataChannel label " << label.Utf8();
webrtc::DataChannelInit config;
// TODO(jiayl): remove the deprecated reliable field once Libjingle is updated
// to handle that.
config.reliable = false;
config.id = init.id;
config.ordered = init.ordered;
config.negotiated = init.negotiated;
config.maxRetransmits = init.max_retransmits;
config.maxRetransmitTime = init.max_retransmit_time;
config.protocol = init.protocol.Utf8();
rtc::scoped_refptr<webrtc::DataChannelInterface> webrtc_channel(
native_peer_connection_->CreateDataChannel(label.Utf8(), &config));
if (!webrtc_channel) {
DLOG(ERROR) << "Could not create native data channel.";
return nullptr;
}
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackCreateDataChannel(
this, webrtc_channel.get(), PeerConnectionTracker::SOURCE_LOCAL);
}
++num_data_channels_created_;
return new RtcDataChannelHandler(task_runner_, webrtc_channel);
}
void RTCPeerConnectionHandler::Stop() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
DVLOG(1) << "RTCPeerConnectionHandler::stop";
if (is_closed_ || !native_peer_connection_.get())
return; // Already stopped.
if (peer_connection_tracker_)
peer_connection_tracker_->TrackStop(this);
native_peer_connection_->Close();
// This object may no longer forward call backs to blink.
is_closed_ = true;
}
blink::WebString RTCPeerConnectionHandler::Id() const {
return blink::WebString::FromASCII(id_);
}
void RTCPeerConnectionHandler::OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnSignalingChange");
if (peer_connection_tracker_)
peer_connection_tracker_->TrackSignalingStateChange(this, new_state);
if (!is_closed_)
client_->DidChangeSignalingState(new_state);
}
// Called any time the IceConnectionState changes
void RTCPeerConnectionHandler::OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) {
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnIceConnectionChange");
DCHECK(task_runner_->RunsTasksInCurrentSequence());
ReportICEState(new_state);
if (new_state == webrtc::PeerConnectionInterface::kIceConnectionChecking) {
ice_connection_checking_start_ = base::TimeTicks::Now();
} else if (new_state ==
webrtc::PeerConnectionInterface::kIceConnectionConnected) {
// If the state becomes connected, send the time needed for PC to become
// connected from checking to UMA. UMA data will help to know how much
// time needed for PC to connect with remote peer.
if (ice_connection_checking_start_.is_null()) {
// From UMA, we have observed a large number of calls falling into the
// overflow buckets. One possibility is that the Checking is not signaled
// before Connected. This is to guard against that situation to make the
// metric more robust.
UMA_HISTOGRAM_MEDIUM_TIMES("WebRTC.PeerConnection.TimeToConnect",
base::TimeDelta());
} else {
UMA_HISTOGRAM_MEDIUM_TIMES(
"WebRTC.PeerConnection.TimeToConnect",
base::TimeTicks::Now() - ice_connection_checking_start_);
}
}
track_metrics_.IceConnectionChange(new_state);
if (peer_connection_tracker_)
peer_connection_tracker_->TrackIceConnectionStateChange(this, new_state);
if (!is_closed_)
client_->DidChangeIceConnectionState(new_state);
}
// Called any time the combined peerconnection state changes
void RTCPeerConnectionHandler::OnConnectionChange(
webrtc::PeerConnectionInterface::PeerConnectionState new_state) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
if (!is_closed_)
client_->DidChangePeerConnectionState(new_state);
}
// Called any time the IceGatheringState changes
void RTCPeerConnectionHandler::OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnIceGatheringChange");
if (new_state == webrtc::PeerConnectionInterface::kIceGatheringComplete) {
UMA_HISTOGRAM_COUNTS_100("WebRTC.PeerConnection.IPv4LocalCandidates",
num_local_candidates_ipv4_);
UMA_HISTOGRAM_COUNTS_100("WebRTC.PeerConnection.IPv6LocalCandidates",
num_local_candidates_ipv6_);
} else if (new_state ==
webrtc::PeerConnectionInterface::kIceGatheringGathering) {
// ICE restarts will change gathering state back to "gathering",
// reset the counter.
ResetUMAStats();
}
if (peer_connection_tracker_)
peer_connection_tracker_->TrackIceGatheringStateChange(this, new_state);
if (!is_closed_)
client_->DidChangeIceGatheringState(new_state);
}
void RTCPeerConnectionHandler::OnRenegotiationNeeded() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnRenegotiationNeeded");
if (peer_connection_tracker_)
peer_connection_tracker_->TrackOnRenegotiationNeeded(this);
if (!is_closed_)
client_->NegotiationNeeded();
}
void RTCPeerConnectionHandler::OnAddReceiverPlanB(
RtpReceiverState receiver_state) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
DCHECK(receiver_state.is_initialized());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnAddReceiverPlanB");
auto web_track = receiver_state.track_ref()->web_track();
// Update metrics.
track_metrics_.AddTrack(MediaStreamTrackMetrics::Direction::kReceive,
MediaStreamTrackMetricsKind(web_track),
web_track.Id().Utf8());
for (const auto& stream_id : receiver_state.stream_ids()) {
// New remote stream?
if (!IsRemoteStream(rtp_receivers_, stream_id))
PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
}
uintptr_t receiver_id =
RTCRtpReceiver::getId(receiver_state.webrtc_receiver().get());
DCHECK(FindReceiver(receiver_id) == rtp_receivers_.end());
auto rtp_receiver = std::make_unique<RTCRtpReceiver>(
native_peer_connection_, std::move(receiver_state));
rtp_receivers_.push_back(std::make_unique<RTCRtpReceiver>(*rtp_receiver));
if (peer_connection_tracker_) {
auto receiver_only_transceiver =
std::make_unique<RTCRtpReceiverOnlyTransceiver>(
std::make_unique<RTCRtpReceiver>(*rtp_receiver));
size_t receiver_index = GetTransceiverIndex(*receiver_only_transceiver);
peer_connection_tracker_->TrackAddTransceiver(
this,
PeerConnectionTracker::TransceiverUpdatedReason::kSetRemoteDescription,
*receiver_only_transceiver.get(), receiver_index);
}
if (!is_closed_)
client_->DidAddReceiverPlanB(rtp_receiver->ShallowCopy());
}
void RTCPeerConnectionHandler::OnRemoveReceiverPlanB(uintptr_t receiver_id) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnRemoveReceiverPlanB");
auto it = FindReceiver(receiver_id);
DCHECK(it != rtp_receivers_.end());
auto receiver = std::make_unique<RTCRtpReceiver>(*(*it));
// Update metrics.
track_metrics_.RemoveTrack(MediaStreamTrackMetrics::Direction::kReceive,
MediaStreamTrackMetricsKind(receiver->Track()),
receiver->Track().Id().Utf8());
if (peer_connection_tracker_) {
auto receiver_only_transceiver =
std::make_unique<RTCRtpReceiverOnlyTransceiver>(
std::make_unique<RTCRtpReceiver>(*receiver));
size_t receiver_index = GetTransceiverIndex(*receiver_only_transceiver);
peer_connection_tracker_->TrackRemoveTransceiver(
this,
PeerConnectionTracker::TransceiverUpdatedReason::kSetRemoteDescription,
*receiver_only_transceiver.get(), receiver_index);
}
rtp_receivers_.erase(it);
for (const auto& stream_id : receiver->state().stream_ids()) {
// This was the last occurence of the stream?
if (!IsRemoteStream(rtp_receivers_, stream_id))
PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
}
if (!is_closed_)
client_->DidRemoveReceiverPlanB(std::move(receiver));
}
void RTCPeerConnectionHandler::OnModifyTransceivers(
std::vector<RtpTransceiverState> transceiver_states,
bool is_remote_description) {
DCHECK_EQ(configuration_.sdp_semantics, webrtc::SdpSemantics::kUnifiedPlan);
std::vector<std::unique_ptr<blink::WebRTCRtpTransceiver>> web_transceivers(
transceiver_states.size());
PeerConnectionTracker::TransceiverUpdatedReason update_reason =
!is_remote_description ? PeerConnectionTracker::TransceiverUpdatedReason::
kSetLocalDescription
: PeerConnectionTracker::TransceiverUpdatedReason::
kSetRemoteDescription;
for (size_t i = 0; i < transceiver_states.size(); ++i) {
// Figure out if this transceiver is new or if setting the state modified
// the transceiver such that it should be logged by the
// |peer_connection_tracker_|.
uintptr_t transceiver_id = RTCRtpTransceiver::GetId(
transceiver_states[i].webrtc_transceiver().get());
auto it = FindTransceiver(transceiver_id);
bool transceiver_is_new = (it == rtp_transceivers_.end());
bool transceiver_was_modified = false;
if (!transceiver_is_new) {
const auto& previous_state = (*it)->state();
transceiver_was_modified =
previous_state.mid() != transceiver_states[i].mid() ||
previous_state.stopped() != transceiver_states[i].stopped() ||
previous_state.direction() != transceiver_states[i].direction() ||
previous_state.current_direction() !=
transceiver_states[i].current_direction();
}
// Update the transceiver.
web_transceivers[i] =
CreateOrUpdateTransceiver(std::move(transceiver_states[i]));
// Log a "transcieverAdded" or "transceiverModified" event in
// chrome://webrtc-internals if new or modified.
if (peer_connection_tracker_ &&
(transceiver_is_new || transceiver_was_modified)) {
size_t transceiver_index = GetTransceiverIndex(*web_transceivers[i]);
if (transceiver_is_new) {
peer_connection_tracker_->TrackAddTransceiver(
this, update_reason, *web_transceivers[i].get(), transceiver_index);
} else if (transceiver_was_modified) {
peer_connection_tracker_->TrackModifyTransceiver(
this, update_reason, *web_transceivers[i].get(), transceiver_index);
}
}
}
if (!is_closed_) {
client_->DidModifyTransceivers(std::move(web_transceivers),
is_remote_description);
}
}
void RTCPeerConnectionHandler::OnDataChannel(
std::unique_ptr<RtcDataChannelHandler> handler) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnDataChannelImpl");
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackCreateDataChannel(
this, handler->channel().get(), PeerConnectionTracker::SOURCE_REMOTE);
}
if (!is_closed_)
client_->DidAddRemoteDataChannel(handler.release());
}
void RTCPeerConnectionHandler::OnIceCandidate(
const std::string& sdp, const std::string& sdp_mid, int sdp_mline_index,
int component, int address_family) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnIceCandidateImpl");
scoped_refptr<blink::WebRTCICECandidate> web_candidate =
blink::WebRTCICECandidate::Create(blink::WebString::FromUTF8(sdp),
blink::WebString::FromUTF8(sdp_mid),
sdp_mline_index);
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackAddIceCandidate(
this, web_candidate, PeerConnectionTracker::SOURCE_LOCAL, true);
}
// Only the first m line's first component is tracked to avoid
// miscounting when doing BUNDLE or rtcp mux.
if (sdp_mline_index == 0 && component == 1) {
if (address_family == AF_INET) {
++num_local_candidates_ipv4_;
} else if (address_family == AF_INET6) {
++num_local_candidates_ipv6_;
} else {
NOTREACHED();
}
}
if (!is_closed_)
client_->DidGenerateICECandidate(std::move(web_candidate));
}
void RTCPeerConnectionHandler::OnInterestingUsage(int usage_pattern) {
client_->DidNoteInterestingUsage(usage_pattern);
}
webrtc::SessionDescriptionInterface*
RTCPeerConnectionHandler::CreateNativeSessionDescription(
const std::string& sdp, const std::string& type,
webrtc::SdpParseError* error) {
webrtc::SessionDescriptionInterface* native_desc =
dependency_factory_->CreateSessionDescription(type, sdp, error);
LOG_IF(ERROR, !native_desc) << "Failed to create native session description."
<< " Type: " << type << " SDP: " << sdp;
return native_desc;
}
RTCPeerConnectionHandler::FirstSessionDescription::FirstSessionDescription(
const webrtc::SessionDescriptionInterface* sdesc) {
DCHECK(sdesc);
for (const auto& content : sdesc->description()->contents()) {
if (content.type == cricket::NS_JINGLE_RTP) {
const auto* mdesc =
static_cast<cricket::MediaContentDescription*>(content.description);
audio = audio || (mdesc->type() == cricket::MEDIA_TYPE_AUDIO);
video = video || (mdesc->type() == cricket::MEDIA_TYPE_VIDEO);
rtcp_mux = rtcp_mux || mdesc->rtcp_mux();
}
}
}
void RTCPeerConnectionHandler::ReportFirstSessionDescriptions(
const FirstSessionDescription& local,
const FirstSessionDescription& remote) {
RtcpMux rtcp_mux = RTCP_MUX_ENABLED;
if ((!local.audio && !local.video) || (!remote.audio && !remote.video)) {
rtcp_mux = RTCP_MUX_NO_MEDIA;
} else if (!local.rtcp_mux || !remote.rtcp_mux) {
rtcp_mux = RTCP_MUX_DISABLED;
}
UMA_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.RtcpMux", rtcp_mux, RTCP_MUX_MAX);
// TODO(pthatcher): Reports stats about whether we have audio and
// video or not.
}
std::vector<std::unique_ptr<RTCRtpSender>>::iterator
RTCPeerConnectionHandler::FindSender(uintptr_t id) {
for (auto it = rtp_senders_.begin(); it != rtp_senders_.end(); ++it) {
if ((*it)->Id() == id)
return it;
}
return rtp_senders_.end();
}
std::vector<std::unique_ptr<RTCRtpReceiver>>::iterator
RTCPeerConnectionHandler::FindReceiver(uintptr_t id) {
for (auto it = rtp_receivers_.begin(); it != rtp_receivers_.end(); ++it) {
if ((*it)->Id() == id)
return it;
}
return rtp_receivers_.end();
}
std::vector<std::unique_ptr<RTCRtpTransceiver>>::iterator
RTCPeerConnectionHandler::FindTransceiver(uintptr_t id) {
for (auto it = rtp_transceivers_.begin(); it != rtp_transceivers_.end();
++it) {
if ((*it)->Id() == id)
return it;
}
return rtp_transceivers_.end();
}
size_t RTCPeerConnectionHandler::GetTransceiverIndex(
const blink::WebRTCRtpTransceiver& web_transceiver) {
if (web_transceiver.ImplementationType() ==
blink::WebRTCRtpTransceiverImplementationType::kFullTransceiver) {
for (size_t i = 0; i < rtp_transceivers_.size(); ++i) {
if (web_transceiver.Id() == rtp_transceivers_[i]->Id())
return i;
}
} else if (web_transceiver.ImplementationType() ==
blink::WebRTCRtpTransceiverImplementationType::kPlanBSenderOnly) {
const auto web_sender = web_transceiver.Sender();
for (size_t i = 0; i < rtp_senders_.size(); ++i) {
if (web_sender->Id() == rtp_senders_[i]->Id())
return i;
}
} else {
RTC_DCHECK(
web_transceiver.ImplementationType() ==
blink::WebRTCRtpTransceiverImplementationType::kPlanBReceiverOnly);
const auto web_receiver = web_transceiver.Receiver();
for (size_t i = 0; i < rtp_receivers_.size(); ++i) {
if (web_receiver->Id() == rtp_receivers_[i]->Id())
return i;
}
}
NOTREACHED();
return 0u;
}
std::unique_ptr<RTCRtpTransceiver>
RTCPeerConnectionHandler::CreateOrUpdateTransceiver(
RtpTransceiverState transceiver_state) {
DCHECK_EQ(configuration_.sdp_semantics, webrtc::SdpSemantics::kUnifiedPlan);
DCHECK(transceiver_state.is_initialized());
DCHECK(transceiver_state.sender_state());
DCHECK(transceiver_state.receiver_state());
auto webrtc_transceiver = transceiver_state.webrtc_transceiver();
auto webrtc_sender = transceiver_state.sender_state()->webrtc_sender();
auto webrtc_receiver = transceiver_state.receiver_state()->webrtc_receiver();
std::unique_ptr<RTCRtpTransceiver> transceiver;
auto it = FindTransceiver(RTCRtpTransceiver::GetId(webrtc_transceiver.get()));
if (it == rtp_transceivers_.end()) {
// Create a new transceiver, including a sender and a receiver.
transceiver = std::make_unique<RTCRtpTransceiver>(
native_peer_connection_, track_adapter_map_,
std::move(transceiver_state));
rtp_transceivers_.push_back(transceiver->ShallowCopy());
DCHECK(FindSender(RTCRtpSender::getId(webrtc_sender.get())) ==
rtp_senders_.end());
rtp_senders_.push_back(
std::make_unique<RTCRtpSender>(*transceiver->content_sender()));
DCHECK(FindReceiver(RTCRtpReceiver::getId(webrtc_receiver.get())) ==
rtp_receivers_.end());
rtp_receivers_.push_back(
std::make_unique<RTCRtpReceiver>(*transceiver->content_receiver()));
} else {
// Update the transceiver. This also updates the sender and receiver.
transceiver = (*it)->ShallowCopy();
transceiver->set_state(std::move(transceiver_state));
DCHECK(FindSender(RTCRtpSender::getId(webrtc_sender.get())) !=
rtp_senders_.end());
DCHECK(FindReceiver(RTCRtpReceiver::getId(webrtc_receiver.get())) !=
rtp_receivers_.end());
}
return transceiver;
}
scoped_refptr<base::SingleThreadTaskRunner>
RTCPeerConnectionHandler::signaling_thread() const {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
return dependency_factory_->GetWebRtcSignalingThread();
}
void RTCPeerConnectionHandler::RunSynchronousOnceClosureOnSignalingThread(
base::OnceClosure closure,
const char* trace_event_name) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
scoped_refptr<base::SingleThreadTaskRunner> thread(signaling_thread());
if (!thread.get() || thread->BelongsToCurrentThread()) {
TRACE_EVENT0("webrtc", trace_event_name);
std::move(closure).Run();
} else {
base::WaitableEvent event(base::WaitableEvent::ResetPolicy::AUTOMATIC,
base::WaitableEvent::InitialState::NOT_SIGNALED);
thread->PostTask(
FROM_HERE,
base::BindOnce(&RunSynchronousOnceClosure, std::move(closure),
base::Unretained(trace_event_name),
base::Unretained(&event)));
event.Wait();
}
}
// Deprecated version - uses a RepeatingCosure (aka old-style Closure)
void RTCPeerConnectionHandler::RunSynchronousClosureOnSignalingThread(
const base::RepeatingClosure& closure,
const char* trace_event_name) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
scoped_refptr<base::SingleThreadTaskRunner> thread(signaling_thread());
if (!thread.get() || thread->BelongsToCurrentThread()) {
TRACE_EVENT0("webrtc", trace_event_name);
closure.Run();
} else {
base::WaitableEvent event(base::WaitableEvent::ResetPolicy::AUTOMATIC,
base::WaitableEvent::InitialState::NOT_SIGNALED);
thread->PostTask(FROM_HERE,
base::BindOnce(&RunSynchronousClosure, closure,
base::Unretained(trace_event_name),
base::Unretained(&event)));
event.Wait();
}
}
blink::WebRTCSessionDescription
RTCPeerConnectionHandler::GetWebRTCSessionDescriptionOnSignalingThread(
base::OnceCallback<const webrtc::SessionDescriptionInterface*()>
description_cb,
const char* log_text) {
// Since the webrtc::PeerConnectionInterface::*_description() functions
// return a pointer to a non-reference-counted object that lives on the
// signaling thread, we cannot fetch a pointer to it and use it directly
// here.
// Instead, we access the object completely on the signaling thread.
// Initializing |description| on the signaling thread is safe because we
// own it and wait for it to be initialized here.
blink::WebRTCSessionDescription description; // IsNull() by default.
RunSynchronousOnceClosureOnSignalingThread(
base::BindOnce(&GetWebRTCSessionDescriptionFromSessionDescriptionCallback,
std::move(description_cb), base::Unretained(&description)),
log_text);
return description;
}
void RTCPeerConnectionHandler::ReportICEState(
webrtc::PeerConnectionInterface::IceConnectionState new_state) {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
if (ice_state_seen_[new_state])
return;
ice_state_seen_[new_state] = true;
UMA_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.ConnectionState", new_state,
webrtc::PeerConnectionInterface::kIceConnectionMax);
}
void RTCPeerConnectionHandler::ResetUMAStats() {
DCHECK(task_runner_->RunsTasksInCurrentSequence());
num_local_candidates_ipv6_ = 0;
num_local_candidates_ipv4_ = 0;
ice_connection_checking_start_ = base::TimeTicks();
memset(ice_state_seen_, 0, sizeof(ice_state_seen_));
}
} // namespace content