blob: 1b8df914dee9adba2fb571bcde6ce0df778f3bb7 [file] [log] [blame]
// Copyright 2014 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 "remoting/host/cast_extension_session.h"
#include "base/bind.h"
#include "base/json/json_reader.h"
#include "base/json/json_writer.h"
#include "base/logging.h"
#include "base/synchronization/waitable_event.h"
#include "net/url_request/url_request_context_getter.h"
#include "remoting/host/client_session.h"
#include "remoting/proto/control.pb.h"
#include "remoting/protocol/chromium_port_allocator_factory.h"
#include "remoting/protocol/client_stub.h"
#include "remoting/protocol/webrtc_video_capturer_adapter.h"
#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
#include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h"
#include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h"
namespace remoting {
// Used as the type attribute of all Cast protocol::ExtensionMessages.
const char kExtensionMessageType[] = "cast_message";
// Top-level keys used in all extension messages between host and client.
// Must keep synced with webapp.
const char kTopLevelData[] = "chromoting_data";
const char kTopLevelSubject[] = "subject";
// Keys used to describe the subject of a cast extension message. WebRTC-related
// message subjects are prepended with "webrtc_".
// Must keep synced with webapp.
const char kSubjectReady[] = "ready";
const char kSubjectTest[] = "test";
const char kSubjectNewCandidate[] = "webrtc_candidate";
const char kSubjectOffer[] = "webrtc_offer";
const char kSubjectAnswer[] = "webrtc_answer";
// WebRTC headers used inside messages with subject = "webrtc_*".
const char kWebRtcCandidate[] = "candidate";
const char kWebRtcSessionDescType[] = "type";
const char kWebRtcSessionDescSDP[] = "sdp";
const char kWebRtcSDPMid[] = "sdpMid";
const char kWebRtcSDPMLineIndex[] = "sdpMLineIndex";
// Media labels used over the PeerConnection.
const char kVideoLabel[] = "cast_video_label";
const char kStreamLabel[] = "stream_label";
// Default STUN server used to construct
// webrtc::PeerConnectionInterface::RTCConfiguration for the PeerConnection.
const char kDefaultStunURI[] = "stun:stun.l.google.com:19302";
const char kWorkerThreadName[] = "CastExtensionSessionWorkerThread";
// Interval between each call to PollPeerConnectionStats().
const int kStatsLogIntervalSec = 10;
// Minimum frame rate for video streaming over the PeerConnection in frames per
// second, added as a media constraint when constructing the video source for
// the Peer Connection.
const int kMinFramesPerSecond = 5;
// A webrtc::SetSessionDescriptionObserver implementation used to receive the
// results of setting local and remote descriptions of the PeerConnection.
class CastSetSessionDescriptionObserver
: public webrtc::SetSessionDescriptionObserver {
public:
static CastSetSessionDescriptionObserver* Create() {
return new rtc::RefCountedObject<CastSetSessionDescriptionObserver>();
}
void OnSuccess() override {
VLOG(1) << "Setting session description succeeded.";
}
void OnFailure(const std::string& error) override {
LOG(ERROR) << "Setting session description failed: " << error;
}
protected:
CastSetSessionDescriptionObserver() {}
~CastSetSessionDescriptionObserver() override {}
DISALLOW_COPY_AND_ASSIGN(CastSetSessionDescriptionObserver);
};
// A webrtc::CreateSessionDescriptionObserver implementation used to receive the
// results of creating descriptions for this end of the PeerConnection.
class CastCreateSessionDescriptionObserver
: public webrtc::CreateSessionDescriptionObserver {
public:
static CastCreateSessionDescriptionObserver* Create(
CastExtensionSession* session) {
return new rtc::RefCountedObject<CastCreateSessionDescriptionObserver>(
session);
}
void OnSuccess(webrtc::SessionDescriptionInterface* desc) override {
if (cast_extension_session_ == nullptr) {
LOG(ERROR)
<< "No CastExtensionSession. Creating session description succeeded.";
return;
}
cast_extension_session_->OnCreateSessionDescription(desc);
}
void OnFailure(const std::string& error) override {
if (cast_extension_session_ == nullptr) {
LOG(ERROR)
<< "No CastExtensionSession. Creating session description failed.";
return;
}
cast_extension_session_->OnCreateSessionDescriptionFailure(error);
}
void SetCastExtensionSession(CastExtensionSession* cast_extension_session) {
cast_extension_session_ = cast_extension_session;
}
protected:
explicit CastCreateSessionDescriptionObserver(CastExtensionSession* session)
: cast_extension_session_(session) {}
~CastCreateSessionDescriptionObserver() override {}
private:
CastExtensionSession* cast_extension_session_;
DISALLOW_COPY_AND_ASSIGN(CastCreateSessionDescriptionObserver);
};
// A webrtc::StatsObserver implementation used to receive statistics about the
// current PeerConnection.
class CastStatsObserver : public webrtc::StatsObserver {
public:
static CastStatsObserver* Create() {
return new rtc::RefCountedObject<CastStatsObserver>();
}
void OnComplete(const webrtc::StatsReports& reports) override {
VLOG(1) << "Received " << reports.size() << " new StatsReports.";
int index = 0;
for (const auto* report : reports) {
VLOG(1) << "Report " << index++ << ":";
for (const auto& v : report->values()) {
VLOG(1) << "Stat: " << v.second->display_name() << "="
<< v.second->ToString() << ".";
}
}
}
protected:
CastStatsObserver() {}
~CastStatsObserver() override {}
DISALLOW_COPY_AND_ASSIGN(CastStatsObserver);
};
// TODO(aiguha): Fix PeerConnnection-related tear down crash caused by premature
// destruction of cricket::CaptureManager (which occurs on releasing
// |peer_conn_factory_|). See crbug.com/403840.
CastExtensionSession::~CastExtensionSession() {
DCHECK(caller_task_runner_->BelongsToCurrentThread());
// Explicitly clear |create_session_desc_observer_|'s pointer to |this|,
// since the CastExtensionSession is destructing. Otherwise,
// |create_session_desc_observer_| would be left with a dangling pointer.
create_session_desc_observer_->SetCastExtensionSession(nullptr);
CleanupPeerConnection();
}
// static
scoped_ptr<CastExtensionSession> CastExtensionSession::Create(
scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner,
scoped_refptr<net::URLRequestContextGetter> url_request_context_getter,
const protocol::NetworkSettings& network_settings,
ClientSessionControl* client_session_control,
protocol::ClientStub* client_stub) {
scoped_ptr<CastExtensionSession> cast_extension_session(
new CastExtensionSession(caller_task_runner,
url_request_context_getter,
network_settings,
client_session_control,
client_stub));
if (!cast_extension_session->WrapTasksAndSave() ||
!cast_extension_session->InitializePeerConnection()) {
return nullptr;
}
return cast_extension_session.Pass();
}
void CastExtensionSession::OnCreateSessionDescription(
webrtc::SessionDescriptionInterface* desc) {
if (!caller_task_runner_->BelongsToCurrentThread()) {
caller_task_runner_->PostTask(
FROM_HERE,
base::Bind(&CastExtensionSession::OnCreateSessionDescription,
base::Unretained(this),
desc));
return;
}
peer_connection_->SetLocalDescription(
CastSetSessionDescriptionObserver::Create(), desc);
base::DictionaryValue json;
json.SetString(kWebRtcSessionDescType, desc->type());
std::string subject =
(desc->type() == "offer") ? kSubjectOffer : kSubjectAnswer;
std::string desc_str;
desc->ToString(&desc_str);
json.SetString(kWebRtcSessionDescSDP, desc_str);
std::string json_str;
if (!base::JSONWriter::Write(json, &json_str)) {
LOG(ERROR) << "Failed to serialize sdp message.";
return;
}
SendMessageToClient(subject.c_str(), json_str);
}
void CastExtensionSession::OnCreateSessionDescriptionFailure(
const std::string& error) {
VLOG(1) << "Creating Session Description failed: " << error;
}
// TODO(aiguha): Support the case(s) where we've grabbed the capturer already,
// but another extension reset the video pipeline. We should remove the
// stream from the peer connection here, and then attempt to re-setup the
// peer connection in the OnRenegotiationNeeded() callback.
// See crbug.com/403843.
void CastExtensionSession::OnCreateVideoCapturer(
scoped_ptr<webrtc::DesktopCapturer>* capturer) {
if (has_grabbed_capturer_) {
LOG(ERROR) << "The video pipeline was reset unexpectedly.";
has_grabbed_capturer_ = false;
peer_connection_->RemoveStream(stream_.release());
return;
}
if (received_offer_) {
has_grabbed_capturer_ = true;
if (SetupVideoStream(capturer->Pass())) {
peer_connection_->CreateAnswer(create_session_desc_observer_, nullptr);
} else {
has_grabbed_capturer_ = false;
// Ignore the received offer, since we failed to setup a video stream.
received_offer_ = false;
}
return;
}
}
bool CastExtensionSession::ModifiesVideoPipeline() const {
return true;
}
// Returns true if the |message| is a Cast ExtensionMessage, even if
// it was badly formed or a resulting action failed. This is done so that
// the host does not continue to attempt to pass |message| to other
// HostExtensionSessions.
bool CastExtensionSession::OnExtensionMessage(
ClientSessionControl* client_session_control,
protocol::ClientStub* client_stub,
const protocol::ExtensionMessage& message) {
if (message.type() != kExtensionMessageType) {
return false;
}
scoped_ptr<base::Value> value = base::JSONReader::Read(message.data());
base::DictionaryValue* client_message;
if (!(value && value->GetAsDictionary(&client_message))) {
LOG(ERROR) << "Could not read cast extension message.";
return true;
}
std::string subject;
if (!client_message->GetString(kTopLevelSubject, &subject)) {
LOG(ERROR) << "Invalid Cast Extension Message (missing subject header).";
return true;
}
if (subject == kSubjectOffer && !received_offer_) {
// Reset the video pipeline so we can grab the screen capturer and setup
// a video stream.
if (ParseAndSetRemoteDescription(client_message)) {
received_offer_ = true;
LOG(INFO) << "About to ResetVideoPipeline.";
client_session_control_->ResetVideoPipeline();
}
} else if (subject == kSubjectAnswer) {
ParseAndSetRemoteDescription(client_message);
} else if (subject == kSubjectNewCandidate) {
ParseAndAddICECandidate(client_message);
} else {
VLOG(1) << "Unexpected CastExtension Message: " << message.data();
}
return true;
}
// Private methods ------------------------------------------------------------
CastExtensionSession::CastExtensionSession(
scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner,
scoped_refptr<net::URLRequestContextGetter> url_request_context_getter,
const protocol::NetworkSettings& network_settings,
ClientSessionControl* client_session_control,
protocol::ClientStub* client_stub)
: caller_task_runner_(caller_task_runner),
url_request_context_getter_(url_request_context_getter),
network_settings_(network_settings),
client_session_control_(client_session_control),
client_stub_(client_stub),
stats_observer_(CastStatsObserver::Create()),
received_offer_(false),
has_grabbed_capturer_(false),
signaling_thread_wrapper_(nullptr),
worker_thread_wrapper_(nullptr),
worker_thread_(kWorkerThreadName) {
DCHECK(caller_task_runner_->BelongsToCurrentThread());
DCHECK(url_request_context_getter_.get());
DCHECK(client_session_control_);
DCHECK(client_stub_);
// The worker thread is created with base::MessageLoop::TYPE_IO because
// the PeerConnection performs some port allocation operations on this thread
// that require it. See crbug.com/404013.
base::Thread::Options options(base::MessageLoop::TYPE_IO, 0);
worker_thread_.StartWithOptions(options);
worker_task_runner_ = worker_thread_.task_runner();
}
bool CastExtensionSession::ParseAndSetRemoteDescription(
base::DictionaryValue* message) {
DCHECK(peer_connection_.get() != nullptr);
base::DictionaryValue* message_data;
if (!message->GetDictionary(kTopLevelData, &message_data)) {
LOG(ERROR) << "Invalid Cast Extension Message (missing data).";
return false;
}
std::string webrtc_type;
if (!message_data->GetString(kWebRtcSessionDescType, &webrtc_type)) {
LOG(ERROR)
<< "Invalid Cast Extension Message (missing webrtc type header).";
return false;
}
std::string sdp;
if (!message_data->GetString(kWebRtcSessionDescSDP, &sdp)) {
LOG(ERROR) << "Invalid Cast Extension Message (missing webrtc sdp header).";
return false;
}
webrtc::SdpParseError error;
webrtc::SessionDescriptionInterface* session_description(
webrtc::CreateSessionDescription(webrtc_type, sdp, &error));
if (!session_description) {
LOG(ERROR) << "Invalid Cast Extension Message (could not parse sdp).";
VLOG(1) << "SdpParseError was: " << error.description;
return false;
}
peer_connection_->SetRemoteDescription(
CastSetSessionDescriptionObserver::Create(), session_description);
return true;
}
bool CastExtensionSession::ParseAndAddICECandidate(
base::DictionaryValue* message) {
DCHECK(peer_connection_.get() != nullptr);
base::DictionaryValue* message_data;
if (!message->GetDictionary(kTopLevelData, &message_data)) {
LOG(ERROR) << "Invalid Cast Extension Message (missing data).";
return false;
}
std::string candidate_str;
std::string sdp_mid;
int sdp_mlineindex = 0;
if (!message_data->GetString(kWebRtcSDPMid, &sdp_mid) ||
!message_data->GetInteger(kWebRtcSDPMLineIndex, &sdp_mlineindex) ||
!message_data->GetString(kWebRtcCandidate, &candidate_str)) {
LOG(ERROR) << "Invalid Cast Extension Message (could not parse).";
return false;
}
rtc::scoped_ptr<webrtc::IceCandidateInterface> candidate(
webrtc::CreateIceCandidate(sdp_mid, sdp_mlineindex, candidate_str,
nullptr));
if (!candidate.get()) {
LOG(ERROR)
<< "Invalid Cast Extension Message (could not create candidate).";
return false;
}
if (!peer_connection_->AddIceCandidate(candidate.get())) {
LOG(ERROR) << "Failed to apply received ICE Candidate to PeerConnection.";
return false;
}
VLOG(1) << "Received and Added ICE Candidate: " << candidate_str;
return true;
}
bool CastExtensionSession::SendMessageToClient(const std::string& subject,
const std::string& data) {
DCHECK(caller_task_runner_->BelongsToCurrentThread());
if (client_stub_ == nullptr) {
LOG(ERROR) << "No Client Stub. Cannot send message to client.";
return false;
}
base::DictionaryValue message_dict;
message_dict.SetString(kTopLevelSubject, subject);
message_dict.SetString(kTopLevelData, data);
std::string message_json;
if (!base::JSONWriter::Write(message_dict, &message_json)) {
LOG(ERROR) << "Failed to serialize JSON message.";
return false;
}
protocol::ExtensionMessage message;
message.set_type(kExtensionMessageType);
message.set_data(message_json);
client_stub_->DeliverHostMessage(message);
return true;
}
void CastExtensionSession::EnsureTaskAndSetSend(rtc::Thread** ptr,
base::WaitableEvent* event) {
jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
*ptr = jingle_glue::JingleThreadWrapper::current();
if (event != nullptr) {
event->Signal();
}
}
bool CastExtensionSession::WrapTasksAndSave() {
DCHECK(caller_task_runner_->BelongsToCurrentThread());
EnsureTaskAndSetSend(&signaling_thread_wrapper_);
if (signaling_thread_wrapper_ == nullptr)
return false;
base::WaitableEvent wrap_worker_thread_event(true, false);
worker_task_runner_->PostTask(
FROM_HERE,
base::Bind(&CastExtensionSession::EnsureTaskAndSetSend,
base::Unretained(this),
&worker_thread_wrapper_,
&wrap_worker_thread_event));
wrap_worker_thread_event.Wait();
return (worker_thread_wrapper_ != nullptr);
}
bool CastExtensionSession::InitializePeerConnection() {
DCHECK(caller_task_runner_->BelongsToCurrentThread());
DCHECK(!peer_conn_factory_);
DCHECK(!peer_connection_);
DCHECK(worker_thread_wrapper_ != nullptr);
DCHECK(signaling_thread_wrapper_ != nullptr);
peer_conn_factory_ = webrtc::CreatePeerConnectionFactory(
worker_thread_wrapper_, signaling_thread_wrapper_, nullptr, nullptr,
nullptr);
if (!peer_conn_factory_.get()) {
CleanupPeerConnection();
return false;
}
VLOG(1) << "Created PeerConnectionFactory successfully.";
webrtc::PeerConnectionInterface::IceServers servers;
webrtc::PeerConnectionInterface::IceServer server;
server.uri = kDefaultStunURI;
servers.push_back(server);
webrtc::PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.servers = servers;
// DTLS-SRTP is the preferred encryption method. If set to kValueFalse, the
// peer connection uses SDES. Disabling SDES as well will cause the peer
// connection to fail to connect.
// Note: For protection and unprotection of SRTP packets, the libjingle
// ENABLE_EXTERNAL_AUTH flag must not be set.
webrtc::FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
webrtc::MediaConstraintsInterface::kValueTrue);
rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
port_allocator_factory = protocol::ChromiumPortAllocatorFactory::Create(
network_settings_, url_request_context_getter_);
peer_connection_ = peer_conn_factory_->CreatePeerConnection(
rtc_config, &constraints, port_allocator_factory, nullptr, this);
if (!peer_connection_.get()) {
CleanupPeerConnection();
return false;
}
VLOG(1) << "Created PeerConnection successfully.";
create_session_desc_observer_ =
CastCreateSessionDescriptionObserver::Create(this);
// Send a test message to the client. Then, notify the client to start
// webrtc offer/answer negotiation.
if (!SendMessageToClient(kSubjectTest, "Hello, client.") ||
!SendMessageToClient(kSubjectReady, "Host ready to receive offers.")) {
LOG(ERROR) << "Failed to send messages to client.";
return false;
}
return true;
}
bool CastExtensionSession::SetupVideoStream(
scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) {
DCHECK(caller_task_runner_->BelongsToCurrentThread());
DCHECK(desktop_capturer);
if (stream_) {
VLOG(1) << "Already added MediaStream. Aborting Setup.";
return false;
}
scoped_ptr<WebrtcVideoCapturerAdapter> video_capturer_adapter(
new WebrtcVideoCapturerAdapter(desktop_capturer.Pass()));
// Set video stream constraints.
webrtc::FakeConstraints video_constraints;
video_constraints.AddMandatory(
webrtc::MediaConstraintsInterface::kMinFrameRate, kMinFramesPerSecond);
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track =
peer_conn_factory_->CreateVideoTrack(
kVideoLabel,
peer_conn_factory_->CreateVideoSource(
video_capturer_adapter.release(), &video_constraints));
stream_ = peer_conn_factory_->CreateLocalMediaStream(kStreamLabel);
if (!stream_->AddTrack(video_track) ||
!peer_connection_->AddStream(stream_)) {
return false;
}
VLOG(1) << "Setup video stream successfully.";
return true;
}
void CastExtensionSession::PollPeerConnectionStats() {
if (!connection_active()) {
VLOG(1) << "Cannot poll stats while PeerConnection is inactive.";
}
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> video_track =
stream_->FindVideoTrack(kVideoLabel);
peer_connection_->GetStats(
stats_observer_, video_track.release(),
webrtc::PeerConnectionInterface::kStatsOutputLevelStandard);
}
void CastExtensionSession::CleanupPeerConnection() {
peer_connection_->Close();
peer_connection_ = nullptr;
stream_ = nullptr;
peer_conn_factory_ = nullptr;
worker_thread_.Stop();
}
bool CastExtensionSession::connection_active() const {
return peer_connection_.get() != nullptr;
}
// webrtc::PeerConnectionObserver implementation -------------------------------
void CastExtensionSession::OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) {
VLOG(1) << "PeerConnectionObserver: SignalingState changed to:" << new_state;
}
void CastExtensionSession::OnStateChange(
webrtc::PeerConnectionObserver::StateType state_changed) {
VLOG(1) << "PeerConnectionObserver: StateType changed to: " << state_changed;
}
void CastExtensionSession::OnAddStream(webrtc::MediaStreamInterface* stream) {
VLOG(1) << "PeerConnectionObserver: stream added: " << stream->label();
}
void CastExtensionSession::OnRemoveStream(
webrtc::MediaStreamInterface* stream) {
VLOG(1) << "PeerConnectionObserver: stream removed: " << stream->label();
}
void CastExtensionSession::OnDataChannel(
webrtc::DataChannelInterface* data_channel) {
VLOG(1) << "PeerConnectionObserver: data channel: " << data_channel->label();
}
void CastExtensionSession::OnRenegotiationNeeded() {
VLOG(1) << "PeerConnectionObserver: renegotiation needed.";
}
void CastExtensionSession::OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) {
VLOG(1) << "PeerConnectionObserver: IceConnectionState changed to: "
<< new_state;
// TODO(aiguha): Maybe start timer only if enabled by command-line flag or
// at a particular verbosity level.
if (!stats_polling_timer_.IsRunning() &&
new_state == webrtc::PeerConnectionInterface::kIceConnectionConnected) {
stats_polling_timer_.Start(
FROM_HERE,
base::TimeDelta::FromSeconds(kStatsLogIntervalSec),
this,
&CastExtensionSession::PollPeerConnectionStats);
}
}
void CastExtensionSession::OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) {
VLOG(1) << "PeerConnectionObserver: IceGatheringState changed to: "
<< new_state;
}
void CastExtensionSession::OnIceComplete() {
VLOG(1) << "PeerConnectionObserver: all ICE candidates found.";
}
void CastExtensionSession::OnIceCandidate(
const webrtc::IceCandidateInterface* candidate) {
std::string candidate_str;
if (!candidate->ToString(&candidate_str)) {
LOG(ERROR) << "PeerConnectionObserver: failed to serialize candidate.";
return;
}
base::DictionaryValue json;
json.SetString(kWebRtcSDPMid, candidate->sdp_mid());
json.SetInteger(kWebRtcSDPMLineIndex, candidate->sdp_mline_index());
json.SetString(kWebRtcCandidate, candidate_str);
std::string json_str;
if (!base::JSONWriter::Write(json, &json_str)) {
LOG(ERROR) << "Failed to serialize candidate message.";
return;
}
SendMessageToClient(kSubjectNewCandidate, json_str);
}
} // namespace remoting