webrtc/talk/session/media/currentspeakermonitor.h
henrike@webrtc.org 28e2075280 Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-10 00:45:36 +00:00

101 lines
3.7 KiB
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/*
* libjingle
* Copyright 2011 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.
*/
// CurrentSpeakerMonitor monitors the audio levels for a session and determines
// which participant is currently speaking.
#ifndef TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
#define TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
#include <map>
#include "talk/base/basictypes.h"
#include "talk/base/sigslot.h"
namespace cricket {
class BaseSession;
class Call;
class Session;
struct AudioInfo;
struct MediaStreams;
// Note that the call's audio monitor must be started before this is started.
// It's recommended that the audio monitor be started with a 100 ms period.
class CurrentSpeakerMonitor : public sigslot::has_slots<> {
public:
CurrentSpeakerMonitor(Call* call, BaseSession* session);
~CurrentSpeakerMonitor();
BaseSession* session() const { return session_; }
void Start();
void Stop();
// Used by tests. Note that the actual minimum time between switches
// enforced by the monitor will be the given value plus or minus the
// resolution of the system clock.
void set_min_time_between_switches(uint32 min_time_between_switches);
// This is fired when the current speaker changes, and provides his audio
// SSRC. This only fires after the audio monitor on the underlying Call has
// been started.
sigslot::signal2<CurrentSpeakerMonitor*, uint32> SignalUpdate;
private:
void OnAudioMonitor(Call* call, const AudioInfo& info);
void OnMediaStreamsUpdate(Call* call,
Session* session,
const MediaStreams& added,
const MediaStreams& removed);
// These are states that a participant will pass through so that we gradually
// recognize that they have started and stopped speaking. This avoids
// "twitchiness".
enum SpeakingState {
SS_NOT_SPEAKING,
SS_MIGHT_BE_SPEAKING,
SS_SPEAKING,
SS_WAS_SPEAKING_RECENTLY1,
SS_WAS_SPEAKING_RECENTLY2
};
bool started_;
Call* call_;
BaseSession* session_;
std::map<uint32, SpeakingState> ssrc_to_speaking_state_map_;
uint32 current_speaker_ssrc_;
// To prevent overswitching, switching is disabled for some time after a
// switch is made. This gives us the earliest time a switch is permitted.
uint32 earliest_permitted_switch_time_;
uint32 min_time_between_switches_;
};
}
#endif // TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_