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chromium / chromium / src / 29a25683789611e2b345bccff42746764500ebda / . / content / renderer / media / webrtc / rtc_peer_connection_handler.cc
blob: a9bc65e0ac865a48fb5511f83628bfe7eb7acc20 [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/strings/string_util.h"
#include "base/strings/utf_string_conversions.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_certificate.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_media_stream_adapter.h"
#include "content/renderer/media/webrtc/webrtc_set_remote_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/WebKit/public/platform/WebMediaConstraints.h"
#include "third_party/WebKit/public/platform/WebRTCAnswerOptions.h"
#include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
#include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
#include "third_party/WebKit/public/platform/WebRTCError.h"
#include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
#include "third_party/WebKit/public/platform/WebRTCLegacyStats.h"
#include "third_party/WebKit/public/platform/WebRTCOfferOptions.h"
#include "third_party/WebKit/public/platform/WebRTCRtpSender.h"
#include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
#include "third_party/WebKit/public/platform/WebRTCSessionDescriptionRequest.h"
#include "third_party/WebKit/public/platform/WebRTCVoidRequest.h"
#include "third_party/WebKit/public/platform/WebURL.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
};
// Converter functions from libjingle types to WebKit types.
blink::WebRTCPeerConnectionHandlerClient::ICEGatheringState
GetWebKitIceGatheringState(
webrtc::PeerConnectionInterface::IceGatheringState state) {
using blink::WebRTCPeerConnectionHandlerClient;
switch (state) {
case webrtc::PeerConnectionInterface::kIceGatheringNew:
return WebRTCPeerConnectionHandlerClient::kICEGatheringStateNew;
case webrtc::PeerConnectionInterface::kIceGatheringGathering:
return WebRTCPeerConnectionHandlerClient::kICEGatheringStateGathering;
case webrtc::PeerConnectionInterface::kIceGatheringComplete:
return WebRTCPeerConnectionHandlerClient::kICEGatheringStateComplete;
default:
NOTREACHED();
return WebRTCPeerConnectionHandlerClient::kICEGatheringStateNew;
}
}
blink::WebRTCPeerConnectionHandlerClient::ICEConnectionState
GetWebKitIceConnectionState(
webrtc::PeerConnectionInterface::IceConnectionState ice_state) {
using blink::WebRTCPeerConnectionHandlerClient;
switch (ice_state) {
case webrtc::PeerConnectionInterface::kIceConnectionNew:
return WebRTCPeerConnectionHandlerClient::kICEConnectionStateStarting;
case webrtc::PeerConnectionInterface::kIceConnectionChecking:
return WebRTCPeerConnectionHandlerClient::kICEConnectionStateChecking;
case webrtc::PeerConnectionInterface::kIceConnectionConnected:
return WebRTCPeerConnectionHandlerClient::kICEConnectionStateConnected;
case webrtc::PeerConnectionInterface::kIceConnectionCompleted:
return WebRTCPeerConnectionHandlerClient::kICEConnectionStateCompleted;
case webrtc::PeerConnectionInterface::kIceConnectionFailed:
return WebRTCPeerConnectionHandlerClient::kICEConnectionStateFailed;
case webrtc::PeerConnectionInterface::kIceConnectionDisconnected:
return WebRTCPeerConnectionHandlerClient::kICEConnectionStateDisconnected;
case webrtc::PeerConnectionInterface::kIceConnectionClosed:
return WebRTCPeerConnectionHandlerClient::kICEConnectionStateClosed;
default:
NOTREACHED();
return WebRTCPeerConnectionHandlerClient::kICEConnectionStateClosed;
}
}
blink::WebRTCPeerConnectionHandlerClient::SignalingState
GetWebKitSignalingState(webrtc::PeerConnectionInterface::SignalingState state) {
using blink::WebRTCPeerConnectionHandlerClient;
switch (state) {
case webrtc::PeerConnectionInterface::kStable:
return WebRTCPeerConnectionHandlerClient::kSignalingStateStable;
case webrtc::PeerConnectionInterface::kHaveLocalOffer:
return WebRTCPeerConnectionHandlerClient::kSignalingStateHaveLocalOffer;
case webrtc::PeerConnectionInterface::kHaveLocalPrAnswer:
return WebRTCPeerConnectionHandlerClient::
kSignalingStateHaveLocalPrAnswer;
case webrtc::PeerConnectionInterface::kHaveRemoteOffer:
return WebRTCPeerConnectionHandlerClient::kSignalingStateHaveRemoteOffer;
case webrtc::PeerConnectionInterface::kHaveRemotePrAnswer:
return WebRTCPeerConnectionHandlerClient::
kSignalingStateHaveRemotePrAnswer;
case webrtc::PeerConnectionInterface::kClosed:
return WebRTCPeerConnectionHandlerClient::kSignalingStateClosed;
default:
NOTREACHED();
return WebRTCPeerConnectionHandlerClient::kSignalingStateClosed;
}
}
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 ConvertToWebKitRTCError(const webrtc::RTCError& webrtc_error,
blink::WebRTCError* blink_error) {
switch (webrtc_error.type()) {
case webrtc::RTCErrorType::NONE:
blink_error->SetType(blink::WebRTCErrorType::kNone);
break;
case webrtc::RTCErrorType::UNSUPPORTED_PARAMETER:
blink_error->SetType(blink::WebRTCErrorType::kUnsupportedParameter);
break;
case webrtc::RTCErrorType::INVALID_PARAMETER:
blink_error->SetType(blink::WebRTCErrorType::kInvalidParameter);
break;
case webrtc::RTCErrorType::INVALID_RANGE:
blink_error->SetType(blink::WebRTCErrorType::kInvalidRange);
break;
case webrtc::RTCErrorType::SYNTAX_ERROR:
blink_error->SetType(blink::WebRTCErrorType::kSyntaxError);
break;
case webrtc::RTCErrorType::INVALID_STATE:
blink_error->SetType(blink::WebRTCErrorType::kInvalidState);
break;
case webrtc::RTCErrorType::INVALID_MODIFICATION:
blink_error->SetType(blink::WebRTCErrorType::kInvalidModification);
break;
case webrtc::RTCErrorType::NETWORK_ERROR:
blink_error->SetType(blink::WebRTCErrorType::kNetworkError);
break;
case webrtc::RTCErrorType::INTERNAL_ERROR:
blink_error->SetType(blink::WebRTCErrorType::kInternalError);
break;
default:
// If adding a new error type, need 3 CLs: One to add the enum to webrtc,
// one to update this mapping code, and one to start using the enum in
// webrtc.
NOTREACHED() << "webrtc::RTCErrorType " << webrtc_error.type()
<< " not covered by switch statement.";
break;
}
}
void RunClosureWithTrace(const base::Closure& closure,
const char* trace_event_name) {
TRACE_EVENT0("webrtc", trace_event_name);
closure.Run();
}
void RunSynchronousClosure(const base::Closure& closure,
const char* trace_event_name,
base::WaitableEvent* event) {
{
TRACE_EVENT0("webrtc", trace_event_name);
closure.Run();
}
event->Signal();
}
void GetSdpAndTypeFromSessionDescription(
const base::Callback<const webrtc::SessionDescriptionInterface*()>&
description_callback,
std::string* sdp, std::string* type) {
const webrtc::SessionDescriptionInterface* description =
description_callback.Run();
if (description) {
description->ToString(sdp);
*type = description->type();
}
}
// Converter functions from Blink types to WebRTC types.
// This function doesn't assume |webrtc_config| is empty. Any fields in
// |blink_config| replace the corresponding fields in |webrtc_config|, but
// fields that only exist in |webrtc_config| are left alone.
void GetNativeRtcConfiguration(
const blink::WebRTCConfiguration& blink_config,
webrtc::PeerConnectionInterface::RTCConfiguration* webrtc_config) {
DCHECK(webrtc_config);
webrtc_config->servers.clear();
for (const blink::WebRTCIceServer& blink_server : blink_config.ice_servers) {
webrtc::PeerConnectionInterface::IceServer server;
server.username = blink_server.username.Utf8();
server.password = blink_server.credential.Utf8();
server.uri = blink_server.url.GetString().Utf8();
webrtc_config->servers.push_back(server);
}
switch (blink_config.ice_transport_policy) {
case blink::WebRTCIceTransportPolicy::kRelay:
webrtc_config->type = webrtc::PeerConnectionInterface::kRelay;
break;
case blink::WebRTCIceTransportPolicy::kAll:
webrtc_config->type = webrtc::PeerConnectionInterface::kAll;
break;
default:
NOTREACHED();
}
switch (blink_config.bundle_policy) {
case blink::WebRTCBundlePolicy::kBalanced:
webrtc_config->bundle_policy =
webrtc::PeerConnectionInterface::kBundlePolicyBalanced;
break;
case blink::WebRTCBundlePolicy::kMaxBundle:
webrtc_config->bundle_policy =
webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
break;
case blink::WebRTCBundlePolicy::kMaxCompat:
webrtc_config->bundle_policy =
webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat;
break;
default:
NOTREACHED();
}
switch (blink_config.rtcp_mux_policy) {
case blink::WebRTCRtcpMuxPolicy::kNegotiate:
webrtc_config->rtcp_mux_policy =
webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
break;
case blink::WebRTCRtcpMuxPolicy::kRequire:
webrtc_config->rtcp_mux_policy =
webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire;
break;
default:
NOTREACHED();
}
switch (blink_config.sdp_semantics) {
case blink::WebRTCSdpSemantics::kDefault:
webrtc_config->sdp_semantics = webrtc::SdpSemantics::kDefault;
break;
case blink::WebRTCSdpSemantics::kPlanB:
webrtc_config->sdp_semantics = webrtc::SdpSemantics::kPlanB;
break;
case blink::WebRTCSdpSemantics::kUnifiedPlan:
webrtc_config->sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
break;
default:
NOTREACHED();
}
webrtc_config->certificates.clear();
for (const std::unique_ptr<blink::WebRTCCertificate>& blink_certificate :
blink_config.certificates) {
webrtc_config->certificates.push_back(
static_cast<RTCCertificate*>(blink_certificate.get())
->rtcCertificate());
}
webrtc_config->ice_candidate_pool_size = blink_config.ice_candidate_pool_size;
}
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 = rtc::Optional<int>(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 SessionDescriptionRequestTracker {
public:
SessionDescriptionRequestTracker(
const base::WeakPtr<RTCPeerConnectionHandler>& handler,
const base::WeakPtr<PeerConnectionTracker>& tracker,
PeerConnectionTracker::Action action)
: handler_(handler), tracker_(tracker), action_(action) {}
void TrackOnSuccess(const webrtc::SessionDescriptionInterface* desc) {
DCHECK(thread_checker_.CalledOnValidThread());
if (tracker_ && handler_) {
std::string value;
if (desc) {
desc->ToString(&value);
value = "type: " + desc->type() + ", sdp: " + value;
}
tracker_->TrackSessionDescriptionCallback(
handler_.get(), action_, "OnSuccess", value);
}
}
void TrackOnFailure(const std::string& error) {
DCHECK(thread_checker_.CalledOnValidThread());
if (handler_ && tracker_) {
tracker_->TrackSessionDescriptionCallback(
handler_.get(), action_, "OnFailure", error);
}
}
private:
const base::WeakPtr<RTCPeerConnectionHandler> handler_;
const base::WeakPtr<PeerConnectionTracker> tracker_;
PeerConnectionTracker::Action action_;
base::ThreadChecker thread_checker_;
};
// 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),
tracker_(handler, tracker, action) {
}
void OnSuccess(webrtc::SessionDescriptionInterface* desc) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE, base::BindOnce(&CreateSessionDescriptionRequest::OnSuccess,
this, desc));
return;
}
tracker_.TrackOnSuccess(desc);
webkit_request_.RequestSucceeded(CreateWebKitSessionDescription(desc));
webkit_request_.Reset();
delete desc;
}
void OnFailure(const std::string& error) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE, base::BindOnce(&CreateSessionDescriptionRequest::OnFailure,
this, error));
return;
}
tracker_.TrackOnFailure(error);
webkit_request_.RequestFailed(blink::WebString::FromUTF8(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_;
SessionDescriptionRequestTracker tracker_;
};
// Class mapping responses from calls to libjingle SetLocalDescription and a
// blink::WebRTCVoidRequest.
class SetLocalDescriptionRequest
: public webrtc::SetSessionDescriptionObserver {
public:
explicit SetLocalDescriptionRequest(
const scoped_refptr<base::SingleThreadTaskRunner>& main_thread,
const blink::WebRTCVoidRequest& request,
const base::WeakPtr<RTCPeerConnectionHandler>& handler,
const base::WeakPtr<PeerConnectionTracker>& tracker,
PeerConnectionTracker::Action action)
: main_thread_(main_thread),
webkit_request_(request),
tracker_(handler, tracker, action) {}
void OnSuccess() override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(&SetLocalDescriptionRequest::OnSuccess, this));
return;
}
tracker_.TrackOnSuccess(nullptr);
webkit_request_.RequestSucceeded();
webkit_request_.Reset();
}
void OnFailure(const std::string& error) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(&SetLocalDescriptionRequest::OnFailure, this, error));
return;
}
tracker_.TrackOnFailure(error);
webkit_request_.RequestFailed(blink::WebString::FromUTF8(error));
webkit_request_.Reset();
}
protected:
~SetLocalDescriptionRequest() 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())
<< "SetLocalDescriptionRequest not completed. Shutting down?";
}
private:
const scoped_refptr<base::SingleThreadTaskRunner> main_thread_;
blink::WebRTCVoidRequest webkit_request_;
SessionDescriptionRequestTracker tracker_;
};
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);
signaling_thread_checker_.DetachFromThread();
}
void OnComplete(const StatsReports& reports) override {
DCHECK(signaling_thread_checker_.CalledOnValidThread());
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, 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)
: thread_checker_(),
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(thread_checker_.CalledOnValidThread());
}
// 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:
const base::ThreadChecker 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_;
base::ThreadChecker 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());
}
}
// A stats collector callback.
// It is invoked on the WebRTC signaling thread and will post a task to invoke
// |callback| on the thread given in the |main_thread| argument.
// The argument to the callback will be a |blink::WebRTCStatsReport|.
class GetRTCStatsCallback : public webrtc::RTCStatsCollectorCallback {
public:
static rtc::scoped_refptr<GetRTCStatsCallback> Create(
const scoped_refptr<base::SingleThreadTaskRunner>& main_thread,
std::unique_ptr<blink::WebRTCStatsReportCallback> callback) {
return rtc::scoped_refptr<GetRTCStatsCallback>(
new rtc::RefCountedObject<GetRTCStatsCallback>(
main_thread, callback.release()));
}
void OnStatsDelivered(
const rtc::scoped_refptr<const webrtc::RTCStatsReport>& report) override {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(&GetRTCStatsCallback::OnStatsDeliveredOnMainThread, this,
report));
}
protected:
GetRTCStatsCallback(
const scoped_refptr<base::SingleThreadTaskRunner>& main_thread,
blink::WebRTCStatsReportCallback* callback)
: main_thread_(main_thread),
callback_(callback) {
}
~GetRTCStatsCallback() override { DCHECK(!callback_); }
void OnStatsDeliveredOnMainThread(
const rtc::scoped_refptr<const webrtc::RTCStatsReport>& report) {
DCHECK(main_thread_->BelongsToCurrentThread());
DCHECK(report);
DCHECK(callback_);
callback_->OnStatsDelivered(std::unique_ptr<blink::WebRTCStatsReport>(
new RTCStatsReport(base::WrapRefCounted(report.get()))));
// Make sure the callback is destroyed in the main thread as well.
callback_.reset();
}
const scoped_refptr<base::SingleThreadTaskRunner> main_thread_;
std::unique_ptr<blink::WebRTCStatsReportCallback> callback_;
};
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(
GetRTCStatsCallback::Create(main_thread, std::move(callback)));
}
class PeerConnectionUMAObserver : public webrtc::UMAObserver {
public:
PeerConnectionUMAObserver() {}
~PeerConnectionUMAObserver() override {}
void IncrementEnumCounter(webrtc::PeerConnectionEnumCounterType counter_type,
int counter,
int counter_max) override {
switch (counter_type) {
case webrtc::kEnumCounterAddressFamily:
UMA_HISTOGRAM_EXACT_LINEAR("WebRTC.PeerConnection.IPMetrics", counter,
counter_max);
break;
case webrtc::kEnumCounterIceCandidatePairTypeUdp:
UMA_HISTOGRAM_EXACT_LINEAR(
"WebRTC.PeerConnection.CandidatePairType_UDP", counter,
counter_max);
break;
case webrtc::kEnumCounterIceCandidatePairTypeTcp:
UMA_HISTOGRAM_EXACT_LINEAR(
"WebRTC.PeerConnection.CandidatePairType_TCP", counter,
counter_max);
break;
case webrtc::kEnumCounterDtlsHandshakeError:
UMA_HISTOGRAM_EXACT_LINEAR("WebRTC.PeerConnection.DtlsHandshakeError",
counter, counter_max);
break;
case webrtc::kEnumCounterIceRestart:
UMA_HISTOGRAM_EXACT_LINEAR("WebRTC.PeerConnection.IceRestartState",
counter, counter_max);
break;
case webrtc::kEnumCounterIceRegathering:
UMA_HISTOGRAM_EXACT_LINEAR("WebRTC.PeerConnection.IceRegatheringReason",
counter, counter_max);
break;
case webrtc::kEnumCounterKeyProtocol:
UMA_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.KeyProtocol",
static_cast<webrtc::KeyExchangeProtocolType>(counter),
webrtc::kEnumCounterKeyProtocolMax);
break;
case webrtc::kEnumCounterSdpSemanticRequested:
UMA_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpSemanticRequested",
static_cast<webrtc::SdpSemanticRequested>(counter),
webrtc::kSdpSemanticRequestMax);
break;
case webrtc::kEnumCounterSdpSemanticNegotiated:
UMA_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpSemanticNegotiated",
static_cast<webrtc::SdpSemanticNegotiated>(counter),
webrtc::kSdpSemanticNegotiatedMax);
break;
case webrtc::kEnumCounterKeyProtocolMediaType:
UMA_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.KeyProtocolByMedia",
static_cast<webrtc::KeyExchangeProtocolMedia>(counter),
webrtc::kEnumCounterKeyProtocolMediaTypeMax);
break;
default:
// The default clause is expected to be reached when new enum types are
// added.
break;
}
}
void IncrementSparseEnumCounter(
webrtc::PeerConnectionEnumCounterType counter_type,
int counter) override {
switch (counter_type) {
case webrtc::kEnumCounterAudioSrtpCipher:
base::UmaHistogramSparse("WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
counter);
break;
case webrtc::kEnumCounterAudioSslCipher:
base::UmaHistogramSparse("WebRTC.PeerConnection.SslCipherSuite.Audio",
counter);
break;
case webrtc::kEnumCounterVideoSrtpCipher:
base::UmaHistogramSparse("WebRTC.PeerConnection.SrtpCryptoSuite.Video",
counter);
break;
case webrtc::kEnumCounterVideoSslCipher:
base::UmaHistogramSparse("WebRTC.PeerConnection.SslCipherSuite.Video",
counter);
break;
case webrtc::kEnumCounterDataSrtpCipher:
base::UmaHistogramSparse("WebRTC.PeerConnection.SrtpCryptoSuite.Data",
counter);
break;
case webrtc::kEnumCounterDataSslCipher:
base::UmaHistogramSparse("WebRTC.PeerConnection.SslCipherSuite.Data",
counter);
break;
default:
// The default clause is expected to reach when new enum types are
// added.
break;
}
}
void AddHistogramSample(webrtc::PeerConnectionUMAMetricsName type,
int value) override {
// Runs on libjingle's signaling thread.
switch (type) {
case webrtc::kTimeToConnect:
UMA_HISTOGRAM_MEDIUM_TIMES(
"WebRTC.PeerConnection.TimeToConnect",
base::TimeDelta::FromMilliseconds(value));
break;
case webrtc::kNetworkInterfaces_IPv4:
UMA_HISTOGRAM_COUNTS_100("WebRTC.PeerConnection.IPv4Interfaces",
value);
break;
case webrtc::kNetworkInterfaces_IPv6:
UMA_HISTOGRAM_COUNTS_100("WebRTC.PeerConnection.IPv6Interfaces",
value);
break;
default:
// The default clause is expected to reach when new enum types are
// added.
break;
}
}
};
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 |web_stream| as an associated stream.
size_t GetLocalStreamUsageCount(
const std::vector<std::unique_ptr<RTCRtpSender>>& rtp_senders,
const blink::WebMediaStream& web_stream) {
size_t usage_count = 0;
for (const auto& sender : rtp_senders) {
for (const auto& stream_ref : sender->stream_refs()) {
if (stream_ref->adapter().web_stream().UniqueId() ==
web_stream.UniqueId()) {
++usage_count;
break;
}
}
}
return usage_count;
}
// Counts the number of receivers that have |webrtc_stream| as an associated
// stream.
size_t GetRemoteStreamUsageCount(
const std::map<uintptr_t, std::unique_ptr<RTCRtpReceiver>>& rtp_receivers,
const webrtc::MediaStreamInterface* webrtc_stream) {
size_t usage_count = 0;
for (const auto& receiver_entry : rtp_receivers) {
if (receiver_entry.second->HasStream(webrtc_stream))
++usage_count;
}
return usage_count;
}
} // 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 SetRemoteDescription call.
class RTCPeerConnectionHandler::WebRtcSetRemoteDescriptionObserverImpl
: public WebRtcSetRemoteDescriptionObserver {
public:
WebRtcSetRemoteDescriptionObserverImpl(
base::WeakPtr<RTCPeerConnectionHandler> handler,
blink::WebRTCVoidRequest web_request,
base::WeakPtr<PeerConnectionTracker> tracker,
scoped_refptr<base::SingleThreadTaskRunner> task_runner)
: handler_(handler),
main_thread_(task_runner),
web_request_(web_request),
tracker_(tracker) {}
void OnSetRemoteDescriptionComplete(
webrtc::RTCErrorOr<WebRtcSetRemoteDescriptionObserver::States>
states_or_error) override {
if (!states_or_error.ok()) {
auto& error = states_or_error.error();
if (tracker_ && handler_) {
tracker_->TrackSessionDescriptionCallback(
handler_.get(),
PeerConnectionTracker::ACTION_SET_REMOTE_DESCRIPTION, "OnFailure",
error.message());
}
// TODO(hbos): Use |error.type()| to reject the promise with the
// appropriate DOMException.
web_request_.RequestFailed(blink::WebString::FromUTF8(error.message()));
web_request_.Reset();
return;
}
auto& states = states_or_error.value();
if (handler_) {
// 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->second, states.receiver_states))
removed_receivers.push_back(it->second.get());
}
// Update stream states (which tracks belong to which streams).
for (auto& stream_state : GetStreamStates(states, removed_receivers)) {
stream_state.stream_ref->adapter().SetTracks(
std::move(stream_state.track_refs));
}
// Process the addition of remote receivers/tracks.
for (auto& receiver_state : states.receiver_states) {
if (ReceiverWasAdded(receiver_state)) {
handler_->OnAddRemoteTrack(receiver_state.receiver,
std::move(receiver_state.track_ref),
std::move(receiver_state.stream_refs));
}
}
// Process the removal of remote receivers/tracks.
for (auto* removed_receiver : removed_receivers)
handler_->OnRemoveRemoteTrack(removed_receiver->webrtc_receiver());
if (tracker_) {
tracker_->TrackSessionDescriptionCallback(
handler_.get(),
PeerConnectionTracker::ACTION_SET_REMOTE_DESCRIPTION, "OnSuccess",
"");
}
}
// 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::WebRtcSetRemoteDescriptionObserverImpl::
ResolvePromise,
this));
}
private:
~WebRtcSetRemoteDescriptionObserverImpl() override {}
// Describes which tracks belong to a stream in terms of AdapterRefs.
struct StreamState {
StreamState(
std::unique_ptr<WebRtcMediaStreamAdapterMap::AdapterRef> stream_ref)
: stream_ref(std::move(stream_ref)) {}
std::unique_ptr<WebRtcMediaStreamAdapterMap::AdapterRef> stream_ref;
std::vector<std::unique_ptr<WebRtcMediaStreamTrackAdapterMap::AdapterRef>>
track_refs;
};
void ResolvePromise() {
web_request_.RequestSucceeded();
web_request_.Reset();
}
bool ReceiverWasAdded(const WebRtcReceiverState& receiver_state) {
return handler_->rtp_receivers_.find(
RTCRtpReceiver::getId(receiver_state.receiver.get())) ==
handler_->rtp_receivers_.end();
}
bool ReceiverWasRemoved(
const RTCRtpReceiver& receiver,
const std::vector<WebRtcReceiverState>& receiver_states) {
for (const auto& receiver_state : receiver_states) {
if (receiver_state.receiver == receiver.webrtc_receiver())
return false;
}
return true;
}
// Determines the stream states from the current state of receivers and the
// receivers that are about to be removed. Produces a stable order of streams.
std::vector<StreamState> GetStreamStates(
const WebRtcSetRemoteDescriptionObserver::States& states,
const std::vector<RTCRtpReceiver*>& removed_receivers) {
states.CheckInvariants();
std::vector<StreamState> stream_states;
// The receiver's track belongs to all of its streams. A stream may be
// associated with multiple tracks (multiple receivers).
for (auto& receiver_state : states.receiver_states) {
for (auto& stream_ref : receiver_state.stream_refs) {
CHECK(stream_ref);
CHECK(stream_ref->adapter().is_initialized());
CHECK(!stream_ref->adapter().web_stream().IsNull());
CHECK(stream_ref->adapter().webrtc_stream());
auto* stream_state =
GetOrAddStreamStateForStream(*stream_ref, &stream_states);
auto track_ref = receiver_state.track_ref->Copy();
CHECK(!track_ref->web_track().IsNull());
CHECK(track_ref->webrtc_track());
stream_state->track_refs.push_back(std::move(track_ref));
}
}
// The track of removed receivers do not belong to any stream. Make sure we
// have a stream state for any streams belonging to receivers about to be
// removed in case it was the last receiver referencing that stream.
for (auto* removed_receiver : removed_receivers) {
for (auto& stream_ref : removed_receiver->StreamAdapterRefs()) {
CHECK(!stream_ref->adapter().web_stream().IsNull());
CHECK(stream_ref->adapter().webrtc_stream());
GetOrAddStreamStateForStream(*stream_ref, &stream_states);
}
}
states.CheckInvariants();
return stream_states;
}
StreamState* GetOrAddStreamStateForStream(
const WebRtcMediaStreamAdapterMap::AdapterRef& stream_ref,
std::vector<StreamState>* stream_states) {
auto* webrtc_stream = stream_ref.adapter().webrtc_stream().get();
for (auto& stream_state : *stream_states) {
if (stream_state.stream_ref->adapter().webrtc_stream().get() ==
webrtc_stream) {
return &stream_state;
}
}
stream_states->push_back(StreamState(stream_ref.Copy()));
return &stream_states->back();
}
base::WeakPtr<RTCPeerConnectionHandler> handler_;
scoped_refptr<base::SequencedTaskRunner> main_thread_;
blink::WebRTCVoidRequest web_request_;
base::WeakPtr<PeerConnectionTracker> tracker_;
};
// 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;
void OnSignalingChange(
PeerConnectionInterface::SignalingState new_state) override {
if (!main_thread_->BelongsToCurrentThread()) {
main_thread_->PostTask(
FROM_HERE,
base::BindOnce(&RTCPeerConnectionHandler::Observer::OnSignalingChange,
this, new_state));
} else if (handler_) {
handler_->OnSignalingChange(new_state);
}
}
// TODO(hbos): Remove once no longer mandatory to implement.
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 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);
}
}
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)),
stream_adapter_map_(new WebRtcMediaStreamAdapterMap(dependency_factory_,
task_runner,
track_adapter_map_)),
task_runner_(std::move(task_runner)),
weak_factory_(this) {
CHECK(client_);
GetPeerConnectionHandlers()->insert(this);
}
RTCPeerConnectionHandler::~RTCPeerConnectionHandler() {
DCHECK(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
DCHECK(frame);
frame_ = frame;
}
bool RTCPeerConnectionHandler::Initialize(
const blink::WebRTCConfiguration& server_configuration,
const blink::WebMediaConstraints& options) {
DCHECK(thread_checker_.CalledOnValidThread());
DCHECK(frame_);
CHECK(!initialize_called_);
initialize_called_ = true;
peer_connection_tracker_ =
RenderThreadImpl::current()->peer_connection_tracker()->AsWeakPtr();
GetNativeRtcConfiguration(server_configuration, &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));
// 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_);
}
uma_observer_ = new rtc::RefCountedObject<PeerConnectionUMAObserver>();
native_peer_connection_->RegisterUMAObserver(uma_observer_.get());
return true;
}
bool RTCPeerConnectionHandler::InitializeForTest(
const blink::WebRTCConfiguration& server_configuration,
const blink::WebMediaConstraints& options,
const base::WeakPtr<PeerConnectionTracker>& peer_connection_tracker) {
DCHECK(thread_checker_.CalledOnValidThread());
CHECK(!initialize_called_);
initialize_called_ = true;
GetNativeRtcConfiguration(server_configuration, &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(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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(blink::WebString::FromUTF8(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<SetLocalDescriptionRequest> set_request(
new rtc::RefCountedObject<SetLocalDescriptionRequest>(
task_runner_, request, weak_factory_.GetWeakPtr(),
peer_connection_tracker_,
PeerConnectionTracker::ACTION_SET_LOCAL_DESCRIPTION));
signaling_thread()->PostTask(
FROM_HERE,
base::BindOnce(
&RunClosureWithTrace,
base::Bind(&webrtc::PeerConnectionInterface::SetLocalDescription,
native_peer_connection_, base::RetainedRef(set_request),
base::Unretained(native_desc)),
"SetLocalDescription"));
}
void RTCPeerConnectionHandler::SetRemoteDescription(
const blink::WebRTCVoidRequest& request,
const blink::WebRTCSessionDescription& description) {
DCHECK(thread_checker_.CalledOnValidThread());
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(blink::WebString::FromUTF8(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<WebRtcSetRemoteDescriptionObserverImpl> content_observer(
new WebRtcSetRemoteDescriptionObserverImpl(
weak_factory_.GetWeakPtr(), request, peer_connection_tracker_,
task_runner_));
rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>
webrtc_observer(WebRtcSetRemoteDescriptionObserverHandler::Create(
task_runner_, native_peer_connection_,
stream_adapter_map_, content_observer)
.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(thread_checker_.CalledOnValidThread());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::localDescription");
// Since local_description returns 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.
std::string sdp, type;
base::Callback<const webrtc::SessionDescriptionInterface*()> description_cb =
base::Bind(&webrtc::PeerConnectionInterface::local_description,
native_peer_connection_);
RunSynchronousClosureOnSignalingThread(
base::Bind(&GetSdpAndTypeFromSessionDescription, description_cb,
base::Unretained(&sdp), base::Unretained(&type)),
"localDescription");
return CreateWebKitSessionDescription(sdp, type);
}
blink::WebRTCSessionDescription RTCPeerConnectionHandler::RemoteDescription() {
DCHECK(thread_checker_.CalledOnValidThread());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::remoteDescription");
// Since local_description returns 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.
std::string sdp, type;
base::Callback<const webrtc::SessionDescriptionInterface*()> description_cb =
base::Bind(&webrtc::PeerConnectionInterface::remote_description,
native_peer_connection_);
RunSynchronousClosureOnSignalingThread(
base::Bind(&GetSdpAndTypeFromSessionDescription, description_cb,
base::Unretained(&sdp), base::Unretained(&type)),
"remoteDescription");
return CreateWebKitSessionDescription(sdp, type);
}
blink::WebRTCErrorType RTCPeerConnectionHandler::SetConfiguration(
const blink::WebRTCConfiguration& blink_config) {
DCHECK(thread_checker_.CalledOnValidThread());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::setConfiguration");
GetNativeRtcConfiguration(blink_config, &configuration_);
if (peer_connection_tracker_)
peer_connection_tracker_->TrackSetConfiguration(this, configuration_);
webrtc::RTCError webrtc_error;
blink::WebRTCError blink_error;
bool ret =
native_peer_connection_->SetConfiguration(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);
ConvertToWebKitRTCError(webrtc_error, &blink_error);
return blink_error.GetType();
}
bool RTCPeerConnectionHandler::AddICECandidate(
const blink::WebRTCVoidRequest& request,
scoped_refptr<blink::WebRTCICECandidate> candidate) {
DCHECK(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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(
blink::WebString::FromUTF8("Error processing ICE candidate"));
}
return webkit_request.RequestSucceeded();
}
void RTCPeerConnectionHandler::GetStats(
const blink::WebRTCStatsRequest& request) {
DCHECK(thread_checker_.CalledOnValidThread());
scoped_refptr<LocalRTCStatsRequest> inner_request(
new rtc::RefCountedObject<LocalRTCStatsRequest>(request));
getStats(inner_request);
}
void RTCPeerConnectionHandler::getStats(
const scoped_refptr<LocalRTCStatsRequest>& request) {
DCHECK(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
signaling_thread()->PostTask(
FROM_HERE, base::BindOnce(&GetRTCStatsOnSignalingThread, task_runner_,
native_peer_connection_, std::move(callback)));
}
std::unique_ptr<blink::WebRTCRtpSender> RTCPeerConnectionHandler::AddTrack(
const blink::WebMediaStreamTrack& track,
const blink::WebVector<blink::WebMediaStream>& streams) {
DCHECK(thread_checker_.CalledOnValidThread());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::AddTrack");
// Get or create the associated track and stream adapters.
std::unique_ptr<WebRtcMediaStreamTrackAdapterMap::AdapterRef> track_adapter =
track_adapter_map_->GetOrCreateLocalTrackAdapter(track);
std::vector<std::unique_ptr<WebRtcMediaStreamAdapterMap::AdapterRef>>
stream_adapters(streams.size());
std::vector<webrtc::MediaStreamInterface*> webrtc_streams(streams.size());
for (size_t i = 0; i < streams.size(); ++i) {
stream_adapters[i] =
stream_adapter_map_->GetOrCreateLocalStreamAdapter(streams[i]);
webrtc_streams[i] = stream_adapters[i]->adapter().webrtc_stream().get();
}
rtc::scoped_refptr<webrtc::RtpSenderInterface> webrtc_sender =
native_peer_connection_->AddTrack(track_adapter->webrtc_track(),
webrtc_streams);
if (!webrtc_sender)
return nullptr;
DCHECK(FindSender(RTCRtpSender::getId(webrtc_sender)) == rtp_senders_.end());
rtp_senders_.push_back(std::make_unique<RTCRtpSender>(
task_runner_, signaling_thread(), stream_adapter_map_,
std::move(webrtc_sender), std::move(track_adapter),
std::move(stream_adapters)));
for (const auto& stream_ref : rtp_senders_.back()->stream_refs()) {
if (GetLocalStreamUsageCount(rtp_senders_,
stream_ref->adapter().web_stream()) == 1u) {
// This is the first occurrence of this stream.
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackAddStream(
this, stream_ref->adapter().web_stream(),
PeerConnectionTracker::SOURCE_LOCAL);
}
PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
track_metrics_.AddStream(MediaStreamTrackMetrics::SENT_STREAM,
stream_ref->adapter().webrtc_stream().get());
}
}
return rtp_senders_.back()->ShallowCopy();
}
bool RTCPeerConnectionHandler::RemoveTrack(blink::WebRTCRtpSender* web_sender) {
DCHECK(thread_checker_.CalledOnValidThread());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::RemoveTrack");
auto it = FindSender(web_sender->Id());
if (it == rtp_senders_.end())
return false;
if (!(*it)->RemoveFromPeerConnection(native_peer_connection_.get()))
return false;
auto stream_refs = (*it)->stream_refs();
// TODO(hbos): In Unified Plan, senders are never removed. The lower layer
// needs to tell us what to do with the sender: Update its states and/or
// remove it. https://crbug.com/799030
rtp_senders_.erase(it);
for (const auto& stream_ref : stream_refs) {
if (GetLocalStreamUsageCount(rtp_senders_,
stream_ref->adapter().web_stream()) == 0u) {
// This was the last occurrence of this stream.
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackRemoveStream(
this, stream_ref->adapter().web_stream(),
PeerConnectionTracker::SOURCE_LOCAL);
}
PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
track_metrics_.RemoveStream(MediaStreamTrackMetrics::SENT_STREAM,
stream_ref->adapter().webrtc_stream().get());
}
}
return true;
}
void RTCPeerConnectionHandler::CloseClientPeerConnection() {
DCHECK(thread_checker_.CalledOnValidThread());
if (!is_closed_)
client_->ClosePeerConnection();
}
void RTCPeerConnectionHandler::StartEventLog(IPC::PlatformFileForTransit file,
int64_t max_file_size_bytes) {
DCHECK(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
// 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(thread_checker_.CalledOnValidThread());
native_peer_connection_->StopRtcEventLog();
}
void RTCPeerConnectionHandler::OnWebRtcEventLogWrite(
const std::string& output) {
DCHECK(thread_checker_.CalledOnValidThread());
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackRtcEventLogWrite(this, output);
}
}
blink::WebRTCDataChannelHandler* RTCPeerConnectionHandler::CreateDataChannel(
const blink::WebString& label,
const blink::WebRTCDataChannelInit& init) {
DCHECK(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnSignalingChange");
blink::WebRTCPeerConnectionHandlerClient::SignalingState state =
GetWebKitSignalingState(new_state);
if (peer_connection_tracker_)
peer_connection_tracker_->TrackSignalingStateChange(this, state);
if (!is_closed_)
client_->DidChangeSignalingState(state);
}
// Called any time the IceConnectionState changes
void RTCPeerConnectionHandler::OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) {
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnIceConnectionChange");
DCHECK(thread_checker_.CalledOnValidThread());
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);
blink::WebRTCPeerConnectionHandlerClient::ICEConnectionState state =
GetWebKitIceConnectionState(new_state);
if (peer_connection_tracker_)
peer_connection_tracker_->TrackIceConnectionStateChange(this, state);
if (!is_closed_)
client_->DidChangeICEConnectionState(state);
}
// Called any time the IceGatheringState changes
void RTCPeerConnectionHandler::OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) {
DCHECK(thread_checker_.CalledOnValidThread());
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();
}
blink::WebRTCPeerConnectionHandlerClient::ICEGatheringState state =
GetWebKitIceGatheringState(new_state);
if (peer_connection_tracker_)
peer_connection_tracker_->TrackIceGatheringStateChange(this, state);
if (!is_closed_)
client_->DidChangeICEGatheringState(state);
}
void RTCPeerConnectionHandler::OnRenegotiationNeeded() {
DCHECK(thread_checker_.CalledOnValidThread());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnRenegotiationNeeded");
if (peer_connection_tracker_)
peer_connection_tracker_->TrackOnRenegotiationNeeded(this);
if (!is_closed_)
client_->NegotiationNeeded();
}
void RTCPeerConnectionHandler::OnAddRemoteTrack(
scoped_refptr<webrtc::RtpReceiverInterface> webrtc_receiver,
std::unique_ptr<WebRtcMediaStreamTrackAdapterMap::AdapterRef>
remote_track_adapter_ref,
std::vector<std::unique_ptr<WebRtcMediaStreamAdapterMap::AdapterRef>>
remote_stream_adapter_refs) {
DCHECK(thread_checker_.CalledOnValidThread());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnAddRemoteTrack");
for (const auto& remote_stream_adapter_ref : remote_stream_adapter_refs) {
// New remote stream?
if (GetRemoteStreamUsageCount(
rtp_receivers_,
remote_stream_adapter_ref->adapter().webrtc_stream().get()) == 0) {
// Update metrics.
// TODO(hbos): Update metrics to correspond to track added/removed events,
// not streams. https://crbug.com/765170
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackAddStream(
this, remote_stream_adapter_ref->adapter().web_stream(),
PeerConnectionTracker::SOURCE_REMOTE);
}
PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
track_metrics_.AddStream(
MediaStreamTrackMetrics::RECEIVED_STREAM,
remote_stream_adapter_ref->adapter().webrtc_stream().get());
}
}
uintptr_t receiver_id = RTCRtpReceiver::getId(webrtc_receiver.get());
DCHECK(rtp_receivers_.find(receiver_id) == rtp_receivers_.end());
const std::unique_ptr<RTCRtpReceiver>& rtp_receiver =
rtp_receivers_
.insert(std::make_pair(
receiver_id,
std::make_unique<RTCRtpReceiver>(
webrtc_receiver.get(), std::move(remote_track_adapter_ref),
std::move(remote_stream_adapter_refs))))
.first->second;
if (!is_closed_)
client_->DidAddRemoteTrack(rtp_receiver->ShallowCopy());
}
void RTCPeerConnectionHandler::OnRemoveRemoteTrack(
scoped_refptr<webrtc::RtpReceiverInterface> webrtc_receiver) {
DCHECK(thread_checker_.CalledOnValidThread());
TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::OnRemoveRemoteTrack");
std::vector<std::unique_ptr<WebRtcMediaStreamAdapterMap::AdapterRef>>
remote_stream_adapter_refs;
{
uintptr_t receiver_id = RTCRtpReceiver::getId(webrtc_receiver.get());
auto it = rtp_receivers_.find(receiver_id);
DCHECK(it != rtp_receivers_.end());
remote_stream_adapter_refs = it->second->StreamAdapterRefs();
if (!is_closed_)
client_->DidRemoveRemoteTrack(it->second->ShallowCopy());
rtp_receivers_.erase(it);
}
for (const auto& remote_stream_adapter_ref : remote_stream_adapter_refs) {
// Was this the last usage of the remote stream?
if (GetRemoteStreamUsageCount(
rtp_receivers_,
remote_stream_adapter_ref->adapter().webrtc_stream().get()) == 0) {
// Update metrics.
track_metrics_.RemoveStream(
MediaStreamTrackMetrics::RECEIVED_STREAM,
remote_stream_adapter_ref->adapter().webrtc_stream().get());
PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
if (peer_connection_tracker_) {
peer_connection_tracker_->TrackRemoveStream(
this, remote_stream_adapter_ref->adapter().web_stream(),
PeerConnectionTracker::SOURCE_REMOTE);
}
}
}
}
void RTCPeerConnectionHandler::OnDataChannel(
std::unique_ptr<RtcDataChannelHandler> handler) {
DCHECK(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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));
}
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();
}
scoped_refptr<base::SingleThreadTaskRunner>
RTCPeerConnectionHandler::signaling_thread() const {
DCHECK(thread_checker_.CalledOnValidThread());
return dependency_factory_->GetWebRtcSignalingThread();
}
void RTCPeerConnectionHandler::RunSynchronousClosureOnSignalingThread(
const base::Closure& closure,
const char* trace_event_name) {
DCHECK(thread_checker_.CalledOnValidThread());
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();
}
}
void RTCPeerConnectionHandler::ReportICEState(
webrtc::PeerConnectionInterface::IceConnectionState new_state) {
DCHECK(thread_checker_.CalledOnValidThread());
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(thread_checker_.CalledOnValidThread());
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