webrtc_m130/talk/app/webrtc/webrtcsession.h
Fredrik Solenberg 709ed67c38 Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).

BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1269863005 .

Cr-Commit-Position: refs/heads/master@{#9939}
2015-09-15 10:26:45 +00:00

422 lines
17 KiB
C++

/*
* libjingle
* Copyright 2012 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_
#define TALK_APP_WEBRTC_WEBRTCSESSION_H_
#include <string>
#include "talk/app/webrtc/datachannel.h"
#include "talk/app/webrtc/dtmfsender.h"
#include "talk/app/webrtc/mediacontroller.h"
#include "talk/app/webrtc/mediastreamprovider.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/statstypes.h"
#include "talk/media/base/mediachannel.h"
#include "webrtc/p2p/base/session.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/sslidentity.h"
#include "webrtc/base/thread.h"
namespace cricket {
class BaseChannel;
class ChannelManager;
class DataChannel;
class StatsReport;
class Transport;
class VideoCapturer;
class VideoChannel;
class VoiceChannel;
} // namespace cricket
namespace webrtc {
class IceRestartAnswerLatch;
class JsepIceCandidate;
class MediaStreamSignaling;
class WebRtcSessionDescriptionFactory;
extern const char kBundleWithoutRtcpMux[];
extern const char kCreateChannelFailed[];
extern const char kInvalidCandidates[];
extern const char kInvalidSdp[];
extern const char kMlineMismatch[];
extern const char kPushDownTDFailed[];
extern const char kSdpWithoutDtlsFingerprint[];
extern const char kSdpWithoutSdesCrypto[];
extern const char kSdpWithoutIceUfragPwd[];
extern const char kSdpWithoutSdesAndDtlsDisabled[];
extern const char kSessionError[];
extern const char kSessionErrorDesc[];
extern const char kDtlsSetupFailureRtp[];
extern const char kDtlsSetupFailureRtcp[];
// Maximum number of received video streams that will be processed by webrtc
// even if they are not signalled beforehand.
extern const int kMaxUnsignalledRecvStreams;
// ICE state callback interface.
class IceObserver {
public:
IceObserver() {}
// Called any time the IceConnectionState changes
// TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to
// conform to the w3c standard.
virtual void OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {}
// Called any time the IceGatheringState changes
virtual void OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {}
// New Ice candidate have been found.
virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
// All Ice candidates have been found.
// TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
// (via PeerConnectionObserver)
virtual void OnIceComplete() {}
// Called whenever the state changes between receiving and not receiving.
virtual void OnIceConnectionReceivingChange(bool receiving) {}
protected:
~IceObserver() {}
private:
DISALLOW_COPY_AND_ASSIGN(IceObserver);
};
class WebRtcSession : public cricket::BaseSession,
public AudioProviderInterface,
public DataChannelFactory,
public VideoProviderInterface,
public DtmfProviderInterface,
public DataChannelProviderInterface {
public:
WebRtcSession(cricket::ChannelManager* channel_manager,
rtc::Thread* signaling_thread,
rtc::Thread* worker_thread,
cricket::PortAllocator* port_allocator,
MediaStreamSignaling* mediastream_signaling);
virtual ~WebRtcSession();
bool Initialize(
const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints,
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
// Deletes the voice, video and data channel and changes the session state
// to STATE_RECEIVEDTERMINATE.
void Terminate();
void RegisterIceObserver(IceObserver* observer) {
ice_observer_ = observer;
}
virtual cricket::VoiceChannel* voice_channel() {
return voice_channel_.get();
}
virtual cricket::VideoChannel* video_channel() {
return video_channel_.get();
}
virtual cricket::DataChannel* data_channel() {
return data_channel_.get();
}
virtual const MediaStreamSignaling* mediastream_signaling() const {
return mediastream_signaling_;
}
void SetSdesPolicy(cricket::SecurePolicy secure_policy);
cricket::SecurePolicy SdesPolicy() const;
// Get current ssl role from transport.
bool GetSslRole(rtc::SSLRole* role);
// Generic error message callback from WebRtcSession.
// TODO - It may be necessary to supply error code as well.
sigslot::signal0<> SignalError;
void CreateOffer(
CreateSessionDescriptionObserver* observer,
const PeerConnectionInterface::RTCOfferAnswerOptions& options);
void CreateAnswer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints);
// The ownership of |desc| will be transferred after this call.
bool SetLocalDescription(SessionDescriptionInterface* desc,
std::string* err_desc);
// The ownership of |desc| will be transferred after this call.
bool SetRemoteDescription(SessionDescriptionInterface* desc,
std::string* err_desc);
bool ProcessIceMessage(const IceCandidateInterface* ice_candidate);
bool SetIceTransports(PeerConnectionInterface::IceTransportsType type);
const SessionDescriptionInterface* local_description() const {
return local_desc_.get();
}
const SessionDescriptionInterface* remote_description() const {
return remote_desc_.get();
}
// TODO(pthatcher): Cleanup the distinction between
// SessionDescription and SessionDescriptionInterface and remove
// these if possible.
const cricket::SessionDescription* base_local_description() const {
return BaseSession::local_description();
}
const cricket::SessionDescription* base_remote_description() const {
return BaseSession::remote_description();
}
// Get the id used as a media stream track's "id" field from ssrc.
virtual bool GetLocalTrackIdBySsrc(uint32 ssrc, std::string* track_id);
virtual bool GetRemoteTrackIdBySsrc(uint32 ssrc, std::string* track_id);
// AudioMediaProviderInterface implementation.
void SetAudioPlayout(uint32 ssrc,
bool enable,
cricket::AudioRenderer* renderer) override;
void SetAudioSend(uint32 ssrc,
bool enable,
const cricket::AudioOptions& options,
cricket::AudioRenderer* renderer) override;
void SetAudioPlayoutVolume(uint32 ssrc, double volume) override;
// Implements VideoMediaProviderInterface.
bool SetCaptureDevice(uint32 ssrc, cricket::VideoCapturer* camera) override;
void SetVideoPlayout(uint32 ssrc,
bool enable,
cricket::VideoRenderer* renderer) override;
void SetVideoSend(uint32 ssrc,
bool enable,
const cricket::VideoOptions* options) override;
// Implements DtmfProviderInterface.
virtual bool CanInsertDtmf(const std::string& track_id);
virtual bool InsertDtmf(const std::string& track_id,
int code, int duration);
virtual sigslot::signal0<>* GetOnDestroyedSignal();
// Implements DataChannelProviderInterface.
bool SendData(const cricket::SendDataParams& params,
const rtc::Buffer& payload,
cricket::SendDataResult* result) override;
bool ConnectDataChannel(DataChannel* webrtc_data_channel) override;
void DisconnectDataChannel(DataChannel* webrtc_data_channel) override;
void AddSctpDataStream(int sid) override;
void RemoveSctpDataStream(int sid) override;
bool ReadyToSendData() const override;
// Returns stats for all channels of all transports.
// This avoids exposing the internal structures used to track them.
virtual bool GetTransportStats(cricket::SessionStats* stats);
// Implements DataChannelFactory.
rtc::scoped_refptr<DataChannel> CreateDataChannel(
const std::string& label,
const InternalDataChannelInit* config) override;
cricket::DataChannelType data_channel_type() const;
bool IceRestartPending() const;
void ResetIceRestartLatch();
// Called when an RTCCertificate is generated or retrieved by
// WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
void OnCertificateReady(
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
void OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp);
// For unit test.
bool waiting_for_certificate_for_testing() const;
void set_metrics_observer(
webrtc::MetricsObserverInterface* metrics_observer) {
metrics_observer_ = metrics_observer;
}
private:
// Indicates the type of SessionDescription in a call to SetLocalDescription
// and SetRemoteDescription.
enum Action {
kOffer,
kPrAnswer,
kAnswer,
};
// Invokes ConnectChannels() on transport proxies, which initiates ice
// candidates allocation.
bool StartCandidatesAllocation();
bool UpdateSessionState(Action action, cricket::ContentSource source,
std::string* err_desc);
static Action GetAction(const std::string& type);
// Push the media parts of the local or remote session description
// down to all of the channels.
bool PushdownMediaDescription(cricket::ContentAction action,
cricket::ContentSource source,
std::string* error_desc);
// Transport related callbacks, override from cricket::BaseSession.
virtual void OnTransportRequestSignaling(cricket::Transport* transport);
virtual void OnTransportConnecting(cricket::Transport* transport);
virtual void OnTransportWritable(cricket::Transport* transport);
virtual void OnTransportCompleted(cricket::Transport* transport);
virtual void OnTransportFailed(cricket::Transport* transport);
virtual void OnTransportProxyCandidatesReady(
cricket::TransportProxy* proxy,
const cricket::Candidates& candidates);
virtual void OnCandidatesAllocationDone();
void OnTransportReceiving(cricket::Transport* transport) override;
// Enables media channels to allow sending of media.
void EnableChannels();
// Creates a JsepIceCandidate and adds it to the local session description
// and notify observers. Called when a new local candidate have been found.
void ProcessNewLocalCandidate(const std::string& content_name,
const cricket::Candidates& candidates);
// Returns the media index for a local ice candidate given the content name.
// Returns false if the local session description does not have a media
// content called |content_name|.
bool GetLocalCandidateMediaIndex(const std::string& content_name,
int* sdp_mline_index);
// Uses all remote candidates in |remote_desc| in this session.
bool UseCandidatesInSessionDescription(
const SessionDescriptionInterface* remote_desc);
// Uses |candidate| in this session.
bool UseCandidate(const IceCandidateInterface* candidate);
// Deletes the corresponding channel of contents that don't exist in |desc|.
// |desc| can be null. This means that all channels are deleted.
void RemoveUnusedChannelsAndTransports(
const cricket::SessionDescription* desc);
// Allocates media channels based on the |desc|. If |desc| doesn't have
// the BUNDLE option, this method will disable BUNDLE in PortAllocator.
// This method will also delete any existing media channels before creating.
bool CreateChannels(const cricket::SessionDescription* desc);
// Helper methods to create media channels.
bool CreateVoiceChannel(const cricket::ContentInfo* content);
bool CreateVideoChannel(const cricket::ContentInfo* content);
bool CreateDataChannel(const cricket::ContentInfo* content);
// Copy the candidates from |saved_candidates_| to |dest_desc|.
// The |saved_candidates_| will be cleared after this function call.
void CopySavedCandidates(SessionDescriptionInterface* dest_desc);
// Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
// messages.
void OnDataChannelMessageReceived(cricket::DataChannel* channel,
const cricket::ReceiveDataParams& params,
const rtc::Buffer& payload);
std::string BadStateErrMsg(State state);
void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
void SetIceConnectionReceiving(bool receiving);
bool ValidateBundleSettings(const cricket::SessionDescription* desc);
bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
// Below methods are helper methods which verifies SDP.
bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
cricket::ContentSource source,
std::string* err_desc);
// Check if a call to SetLocalDescription is acceptable with |action|.
bool ExpectSetLocalDescription(Action action);
// Check if a call to SetRemoteDescription is acceptable with |action|.
bool ExpectSetRemoteDescription(Action action);
// Verifies a=setup attribute as per RFC 5763.
bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
Action action);
// Returns true if we are ready to push down the remote candidate.
// |remote_desc| is the new remote description, or NULL if the current remote
// description should be used. Output |valid| is true if the candidate media
// index is valid.
bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate,
const SessionDescriptionInterface* remote_desc,
bool* valid);
std::string GetSessionErrorMsg();
// Invoked when OnTransportCompleted is signaled to gather the usage
// of IPv4/IPv6 as best connection.
void ReportBestConnectionState(const cricket::TransportStats& stats);
void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
rtc::scoped_ptr<MediaControllerInterface> media_controller_;
rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
rtc::scoped_ptr<cricket::DataChannel> data_channel_;
cricket::ChannelManager* channel_manager_;
MediaStreamSignaling* mediastream_signaling_;
IceObserver* ice_observer_;
PeerConnectionInterface::IceConnectionState ice_connection_state_;
bool ice_connection_receiving_;
rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
// Candidates that arrived before the remote description was set.
std::vector<IceCandidateInterface*> saved_candidates_;
// If the remote peer is using a older version of implementation.
bool older_version_remote_peer_;
bool dtls_enabled_;
// Specifies which kind of data channel is allowed. This is controlled
// by the chrome command-line flag and constraints:
// 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled,
// constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is
// not set or false, SCTP is allowed (DCT_SCTP);
// 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP);
// 3. If both 1&2 are false, data channel is not allowed (DCT_NONE).
cricket::DataChannelType data_channel_type_;
rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
webrtc_session_desc_factory_;
sigslot::signal0<> SignalVoiceChannelDestroyed;
sigslot::signal0<> SignalVideoChannelDestroyed;
sigslot::signal0<> SignalDataChannelDestroyed;
// Member variables for caching global options.
cricket::AudioOptions audio_options_;
cricket::VideoOptions video_options_;
MetricsObserverInterface* metrics_observer_;
// Declares the bundle policy for the WebRTCSession.
PeerConnectionInterface::BundlePolicy bundle_policy_;
// Declares the RTCP mux policy for the WebRTCSession.
PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_;
DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
};
} // namespace webrtc
#endif // TALK_APP_WEBRTC_WEBRTCSESSION_H_