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// 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.
#ifndef REMOTING_HOST_CAST_EXTENSION_SESSION_H_
#define REMOTING_HOST_CAST_EXTENSION_SESSION_H_
#include <string>
#include "base/macros.h"
#include "base/memory/ref_counted.h"
#include "base/memory/scoped_ptr.h"
#include "base/threading/thread.h"
#include "base/timer/timer.h"
#include "base/values.h"
#include "jingle/glue/thread_wrapper.h"
#include "remoting/host/host_extension_session.h"
#include "third_party/webrtc/api/peerconnectioninterface.h"
#include "third_party/webrtc/base/scoped_ref_ptr.h"
#include "third_party/webrtc/modules/desktop_capture/desktop_capturer.h"
namespace base {
class SingleThreadTaskRunner;
class WaitableEvent;
} // namespace base
namespace webrtc {
class MediaStreamInterface;
} // namespace webrtc
namespace remoting {
class CastCreateSessionDescriptionObserver;
namespace protocol {
class TransportContext;
} // namespace protocol
// A HostExtensionSession implementation that enables WebRTC support using
// the PeerConnection native API.
class CastExtensionSession : public HostExtensionSession,
public webrtc::PeerConnectionObserver {
public:
~CastExtensionSession() override;
// Creates and returns a CastExtensionSession object, after performing
// initialization steps on it. The caller must take ownership of the returned
// object.
static scoped_ptr<CastExtensionSession> Create(
scoped_refptr<protocol::TransportContext> transport_context,
ClientSessionControl* client_session_control,
protocol::ClientStub* client_stub);
// Called by webrtc::CreateSessionDescriptionObserver implementation.
void OnCreateSessionDescription(webrtc::SessionDescriptionInterface* desc);
void OnCreateSessionDescriptionFailure(const std::string& error);
// HostExtensionSession interface.
void OnCreateVideoCapturer(
scoped_ptr<webrtc::DesktopCapturer>* capturer) override;
bool ModifiesVideoPipeline() const override;
bool OnExtensionMessage(ClientSessionControl* client_session_control,
protocol::ClientStub* client_stub,
const protocol::ExtensionMessage& message) override;
// webrtc::PeerConnectionObserver interface.
void OnSignalingChange(
webrtc::PeerConnectionInterface::SignalingState new_state) override;
void OnAddStream(webrtc::MediaStreamInterface* stream) override;
void OnRemoveStream(webrtc::MediaStreamInterface* stream) override;
void OnDataChannel(webrtc::DataChannelInterface* data_channel) override;
void OnRenegotiationNeeded() override;
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
void OnIceGatheringChange(
webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
private:
CastExtensionSession(
scoped_refptr<protocol::TransportContext> transport_context,
ClientSessionControl* client_session_control,
protocol::ClientStub* client_stub);
// Parses |message| for a Session Description and sets the remote
// description, returning true if successful.
bool ParseAndSetRemoteDescription(base::DictionaryValue* message);
// Parses |message| for a PeerConnection ICE candidate and adds it to the
// Peer Connection, returning true if successful.
bool ParseAndAddICECandidate(base::DictionaryValue* message);
// Sends a message to the client through |client_stub_|. This method must be
// called on the network thread.
//
// A protocol::ExtensionMessage consists of two string fields: type and data.
//
// The type field must be |kExtensionMessageType|.
// The data field must be a JSON formatted string with two compulsory
// top level keys: |kTopLevelSubject| and |kTopLevelData|.
//
// The |subject| of a message describes the message to the receiving peer,
// effectively identifying the command the receiving peer should perform.
// The |subject| MUST be one of constants formatted as kSubject* defined in
// the .cc file. This set of subjects is identical between host and client,
// thus standardizing how they communicate.
// The |data| of a message depends on the |subject| of the message.
//
// Examples of what ExtensionMessage.data() could look like:
//
// Host Ready Message:
// Notifies the remote peer that we are ready to receive an offer.
//
// {
// "subject": "ready",
// "chromoting_data": "Host Ready to receive offers"
// }
//
// WebRTC Offer Message:
// Represents the offer received from the remote peer. The local
// peer would then respond with a webrtc_answer message.
// {
// "subject": "webrtc_offer",
// "chromoting_data": {
// "sdp" : "...",
// "type" : "offer"
// }
// }
//
// WebRTC Candidate Message:
// Represents an ICE candidate received from the remote peer. Each peer
// shares its local ICE candidates in this way, until a connection is
// established.
//
// {
// "subject": "webrtc_candidate",
// "chromoting_data": {
// "candidate" : "...",
// "sdpMid" : "...",
// "sdpMLineIndex" : "..."
// }
// }
//
bool SendMessageToClient(const std::string& subject, const std::string& data);
// Creates the jingle wrapper for the current thread, sets send to allowed,
// and saves a pointer to the relevant thread pointer in ptr. If |event|
// is not nullptr, signals the event on completion.
void EnsureTaskAndSetSend(rtc::Thread** ptr,
base::WaitableEvent* event = nullptr);
// Wraps each task runner in JingleThreadWrapper using EnsureTaskAndSetSend(),
// returning true if successful. Wrapping the task runners allows them to be
// shared with and used by the (about to be created) PeerConnectionFactory.
bool WrapTasksAndSave();
// Initializes PeerConnectionFactory and PeerConnection and sends a "ready"
// message to client. Returns true if these steps are performed successfully.
bool InitializePeerConnection();
// Constructs a CastVideoCapturerAdapter, a VideoSource, a VideoTrack and a
// MediaStream |stream_|, which it adds to the |peer_connection_|. Returns
// true if these steps are performed successfully. This method is called only
// when a PeerConnection offer is received from the client.
bool SetupVideoStream(scoped_ptr<webrtc::DesktopCapturer> desktop_capturer);
// Polls a single stats report from the PeerConnection immediately. Called
// periodically using |stats_polling_timer_| after a PeerConnection has been
// established.
void PollPeerConnectionStats();
// Closes |peer_connection_|, releases |peer_connection_|, |stream_| and
// |peer_conn_factory_| and stops the worker thread.
void CleanupPeerConnection();
// Check if the connection is active.
bool connection_active() const;
// TaskRunners that will be used to setup the PeerConnectionFactory's
// signalling thread and worker thread respectively.
scoped_refptr<base::SingleThreadTaskRunner> caller_task_runner_;
scoped_refptr<base::SingleThreadTaskRunner> worker_task_runner_;
// Objects related to the WebRTC PeerConnection.
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> peer_conn_factory_;
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_;
rtc::scoped_refptr<CastCreateSessionDescriptionObserver>
create_session_desc_observer_;
// TransportContext for P2P transport.
scoped_refptr<protocol::TransportContext> transport_context_;
// Interface to interact with ClientSession.
ClientSessionControl* client_session_control_;
// Interface through which messages can be sent to the client.
protocol::ClientStub* client_stub_;
// Used to track webrtc connection statistics.
rtc::scoped_refptr<webrtc::StatsObserver> stats_observer_;
// Used to repeatedly poll stats from the |peer_connection_|.
base::RepeatingTimer stats_polling_timer_;
// True if a PeerConnection offer from the client has been received. This
// necessarily means that the host is not the caller in this attempted
// peer connection.
bool received_offer_;
// True if the webrtc::ScreenCapturer has been grabbed through the
// OnCreateVideoCapturer() callback.
bool has_grabbed_capturer_;
// PeerConnection signaling and worker threads created from
// JingleThreadWrappers. Each is created by calling
// jingle_glue::EnsureForCurrentMessageLoop() and thus deletes itself
// automatically when the associated MessageLoop is destroyed.
rtc::Thread* signaling_thread_wrapper_;
rtc::Thread* worker_thread_wrapper_;
// Worker thread that is wrapped to create |worker_thread_wrapper_|.
base::Thread worker_thread_;
DISALLOW_COPY_AND_ASSIGN(CastExtensionSession);
};
} // namespace remoting
#endif // REMOTING_HOST_CAST_EXTENSION_SESSION_H_
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