webrtc/talk/session/media/call.h

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/*
* libjingle
* Copyright 2004 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_SESSION_MEDIA_CALL_H_
#define TALK_SESSION_MEDIA_CALL_H_
#include <deque>
#include <map>
#include <string>
#include <vector>
#include "talk/media/base/mediachannel.h"
#include "talk/media/base/screencastid.h"
#include "talk/media/base/streamparams.h"
#include "talk/media/base/videocommon.h"
#include "talk/p2p/base/session.h"
#include "talk/p2p/client/socketmonitor.h"
#include "talk/session/media/audiomonitor.h"
#include "talk/session/media/currentspeakermonitor.h"
#include "talk/session/media/mediamessages.h"
#include "talk/session/media/mediasession.h"
#include "talk/xmpp/jid.h"
#include "webrtc/base/messagequeue.h"
namespace cricket {
struct AudioInfo;
class Call;
class MediaSessionClient;
class BaseChannel;
class VoiceChannel;
class VideoChannel;
class DataChannel;
// Can't typedef this easily since it's forward declared as struct elsewhere.
struct CallOptions : public MediaSessionOptions {
};
// CurrentSpeakerMonitor used to have a dependency on Call. To remove this
// dependency, we create AudioSourceContext. CurrentSpeakerMonitor depends on
// AudioSourceContext.
// AudioSourceProxy acts as a proxy so that when SignalAudioMonitor
// in Call is triggered, SignalAudioMonitor in AudioSourceContext is triggered.
// Likewise, when OnMediaStreamsUpdate in Call is triggered,
// OnMediaStreamsUpdate in AudioSourceContext is triggered.
class AudioSourceProxy: public AudioSourceContext, public sigslot::has_slots<> {
public:
explicit AudioSourceProxy(Call* call);
private:
void OnAudioMonitor(Call* call, const AudioInfo& info);
void OnMediaStreamsUpdate(Call* call, cricket::Session*,
const cricket::MediaStreams&, const cricket::MediaStreams&);
AudioSourceContext* audio_source_context_;
Call* call_;
};
class Call : public rtc::MessageHandler, public sigslot::has_slots<> {
public:
explicit Call(MediaSessionClient* session_client);
~Call();
// |initiator| can be empty.
Session* InitiateSession(const buzz::Jid& to, const buzz::Jid& initiator,
const CallOptions& options);
Session* InitiateSession(const std::string& id, const buzz::Jid& to,
const CallOptions& options);
void AcceptSession(Session* session, const CallOptions& options);
void RejectSession(Session* session);
void TerminateSession(Session* session);
void Terminate();
bool SendViewRequest(Session* session,
const ViewRequest& view_request);
void SetLocalRenderer(VideoRenderer* renderer);
void SetVideoRenderer(Session* session, uint32 ssrc,
VideoRenderer* renderer);
void StartConnectionMonitor(Session* session, int cms);
void StopConnectionMonitor(Session* session);
void StartAudioMonitor(Session* session, int cms);
void StopAudioMonitor(Session* session);
bool IsAudioMonitorRunning(Session* session);
void StartSpeakerMonitor(Session* session);
void StopSpeakerMonitor(Session* session);
void Mute(bool mute);
void MuteVideo(bool mute);
bool SendData(Session* session,
const SendDataParams& params,
const rtc::Buffer& payload,
SendDataResult* result);
void PressDTMF(int event);
bool StartScreencast(Session* session,
const std::string& stream_name, uint32 ssrc,
const ScreencastId& screenid, int fps);
bool StopScreencast(Session* session,
const std::string& stream_name, uint32 ssrc);
std::vector<Session*> sessions();
uint32 id();
bool has_video() const { return has_video_; }
bool has_data() const { return has_data_; }
bool muted() const { return muted_; }
bool video() const { return has_video_; }
bool secure() const;
bool video_muted() const { return video_muted_; }
const std::vector<StreamParams>* GetDataRecvStreams(Session* session) const {
MediaStreams* recv_streams = GetMediaStreams(session);
return recv_streams ? &recv_streams->data() : NULL;
}
const std::vector<StreamParams>* GetVideoRecvStreams(Session* session) const {
MediaStreams* recv_streams = GetMediaStreams(session);
return recv_streams ? &recv_streams->video() : NULL;
}
const std::vector<StreamParams>* GetAudioRecvStreams(Session* session) const {
MediaStreams* recv_streams = GetMediaStreams(session);
return recv_streams ? &recv_streams->audio() : NULL;
}
VoiceChannel* GetVoiceChannel(Session* session) const;
VideoChannel* GetVideoChannel(Session* session) const;
DataChannel* GetDataChannel(Session* session) const;
// Public just for unit tests
VideoContentDescription* CreateVideoStreamUpdate(const StreamParams& stream);
// Takes ownership of video.
void SendVideoStreamUpdate(Session* session, VideoContentDescription* video);
// Setting this to false will cause the call to have a longer timeout and
// for the SignalSetupToCallVoicemail to never fire.
void set_send_to_voicemail(bool send_to_voicemail) {
send_to_voicemail_ = send_to_voicemail;
}
bool send_to_voicemail() { return send_to_voicemail_; }
const VoiceMediaInfo& last_voice_media_info() const {
return last_voice_media_info_;
}
// Sets a flag on the chatapp that will redirect the call to voicemail once
// the call has been terminated
sigslot::signal0<> SignalSetupToCallVoicemail;
sigslot::signal2<Call*, Session*> SignalAddSession;
sigslot::signal2<Call*, Session*> SignalRemoveSession;
sigslot::signal3<Call*, Session*, Session::State>
SignalSessionState;
sigslot::signal3<Call*, Session*, Session::Error>
SignalSessionError;
sigslot::signal3<Call*, Session*, const std::string &>
SignalReceivedTerminateReason;
sigslot::signal2<Call*, const std::vector<ConnectionInfo> &>
SignalConnectionMonitor;
sigslot::signal2<Call*, const VoiceMediaInfo&> SignalMediaMonitor;
sigslot::signal2<Call*, const AudioInfo&> SignalAudioMonitor;
// Empty nick on StreamParams means "unknown".
// No ssrcs in StreamParams means "no current speaker".
sigslot::signal3<Call*,
Session*,
const StreamParams&> SignalSpeakerMonitor;
sigslot::signal2<Call*, const std::vector<ConnectionInfo> &>
SignalVideoConnectionMonitor;
sigslot::signal2<Call*, const VideoMediaInfo&> SignalVideoMediaMonitor;
// Gives added streams and removed streams, in that order.
sigslot::signal4<Call*,
Session*,
const MediaStreams&,
const MediaStreams&> SignalMediaStreamsUpdate;
sigslot::signal3<Call*,
const ReceiveDataParams&,
const rtc::Buffer&> SignalDataReceived;
AudioSourceProxy* GetAudioSourceProxy();
private:
void OnMessage(rtc::Message* message);
void OnSessionState(BaseSession* base_session, BaseSession::State state);
void OnSessionError(BaseSession* base_session, Session::Error error);
void OnSessionInfoMessage(
Session* session, const buzz::XmlElement* action_elem);
void OnViewRequest(
Session* session, const ViewRequest& view_request);
void OnRemoteDescriptionUpdate(
BaseSession* base_session, const ContentInfos& updated_contents);
void OnReceivedTerminateReason(Session* session, const std::string &reason);
void IncomingSession(Session* session, const SessionDescription* offer);
// Returns true on success.
bool AddSession(Session* session, const SessionDescription* offer);
void RemoveSession(Session* session);
void EnableChannels(bool enable);
void EnableSessionChannels(Session* session, bool enable);
void Join(Call* call, bool enable);
void OnConnectionMonitor(VoiceChannel* channel,
const std::vector<ConnectionInfo> &infos);
void OnMediaMonitor(VoiceChannel* channel, const VoiceMediaInfo& info);
void OnAudioMonitor(VoiceChannel* channel, const AudioInfo& info);
void OnSpeakerMonitor(CurrentSpeakerMonitor* monitor, uint32 ssrc);
void OnConnectionMonitor(VideoChannel* channel,
const std::vector<ConnectionInfo> &infos);
void OnMediaMonitor(VideoChannel* channel, const VideoMediaInfo& info);
void OnDataReceived(DataChannel* channel,
const ReceiveDataParams& params,
const rtc::Buffer& payload);
MediaStreams* GetMediaStreams(Session* session) const;
void UpdateRemoteMediaStreams(Session* session,
const ContentInfos& updated_contents,
bool update_channels);
bool UpdateVoiceChannelRemoteContent(Session* session,
const AudioContentDescription* audio);
bool UpdateVideoChannelRemoteContent(Session* session,
const VideoContentDescription* video);
bool UpdateDataChannelRemoteContent(Session* session,
const DataContentDescription* data);
void UpdateRecvStreams(const std::vector<StreamParams>& update_streams,
BaseChannel* channel,
std::vector<StreamParams>* recv_streams,
std::vector<StreamParams>* added_streams,
std::vector<StreamParams>* removed_streams);
void AddRecvStreams(const std::vector<StreamParams>& added_streams,
BaseChannel* channel,
std::vector<StreamParams>* recv_streams);
void AddRecvStream(const StreamParams& stream,
BaseChannel* channel,
std::vector<StreamParams>* recv_streams);
void RemoveRecvStreams(const std::vector<StreamParams>& removed_streams,
BaseChannel* channel,
std::vector<StreamParams>* recv_streams);
void RemoveRecvStream(const StreamParams& stream,
BaseChannel* channel,
std::vector<StreamParams>* recv_streams);
void ContinuePlayDTMF();
bool StopScreencastWithoutSendingUpdate(Session* session, uint32 ssrc);
bool StopAllScreencastsWithoutSendingUpdate(Session* session);
bool SessionDescriptionContainsCrypto(const SessionDescription* sdesc) const;
Session* InternalInitiateSession(const std::string& id,
const buzz::Jid& to,
const std::string& initiator_name,
const CallOptions& options);
uint32 id_;
MediaSessionClient* session_client_;
struct StartedCapture {
StartedCapture(cricket::VideoCapturer* capturer,
const cricket::VideoFormat& format) :
capturer(capturer),
format(format) {
}
cricket::VideoCapturer* capturer;
cricket::VideoFormat format;
};
typedef std::map<uint32, StartedCapture> StartedScreencastMap;
struct MediaSession {
Session* session;
VoiceChannel* voice_channel;
VideoChannel* video_channel;
DataChannel* data_channel;
MediaStreams* recv_streams;
StartedScreencastMap started_screencasts;
};
// Create a map of media sessions, keyed off session->id().
typedef std::map<std::string, MediaSession> MediaSessionMap;
MediaSessionMap media_session_map_;
std::map<std::string, CurrentSpeakerMonitor*> speaker_monitor_map_;
VideoRenderer* local_renderer_;
bool has_video_;
bool has_data_;
bool muted_;
bool video_muted_;
bool send_to_voicemail_;
// DTMF tones have to be queued up so that we don't flood the call. We
// keep a deque (doubely ended queue) of them around. While one is playing we
// set the playing_dtmf_ bit and schedule a message in XX msec to clear that
// bit or start the next tone playing.
std::deque<int> queued_dtmf_;
bool playing_dtmf_;
VoiceMediaInfo last_voice_media_info_;
rtc::scoped_ptr<AudioSourceProxy> audio_source_proxy_;
friend class MediaSessionClient;
};
} // namespace cricket
#endif // TALK_SESSION_MEDIA_CALL_H_