28e2075280
trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
101 lines
3.7 KiB
C++
101 lines
3.7 KiB
C++
/*
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* libjingle
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* Copyright 2011 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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// CurrentSpeakerMonitor monitors the audio levels for a session and determines
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// which participant is currently speaking.
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#ifndef TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
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#define TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
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#include <map>
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#include "talk/base/basictypes.h"
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#include "talk/base/sigslot.h"
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namespace cricket {
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class BaseSession;
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class Call;
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class Session;
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struct AudioInfo;
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struct MediaStreams;
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// Note that the call's audio monitor must be started before this is started.
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// It's recommended that the audio monitor be started with a 100 ms period.
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class CurrentSpeakerMonitor : public sigslot::has_slots<> {
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public:
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CurrentSpeakerMonitor(Call* call, BaseSession* session);
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~CurrentSpeakerMonitor();
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BaseSession* session() const { return session_; }
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void Start();
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void Stop();
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// Used by tests. Note that the actual minimum time between switches
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// enforced by the monitor will be the given value plus or minus the
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// resolution of the system clock.
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void set_min_time_between_switches(uint32 min_time_between_switches);
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// This is fired when the current speaker changes, and provides his audio
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// SSRC. This only fires after the audio monitor on the underlying Call has
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// been started.
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sigslot::signal2<CurrentSpeakerMonitor*, uint32> SignalUpdate;
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private:
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void OnAudioMonitor(Call* call, const AudioInfo& info);
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void OnMediaStreamsUpdate(Call* call,
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Session* session,
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const MediaStreams& added,
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const MediaStreams& removed);
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// These are states that a participant will pass through so that we gradually
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// recognize that they have started and stopped speaking. This avoids
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// "twitchiness".
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enum SpeakingState {
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SS_NOT_SPEAKING,
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SS_MIGHT_BE_SPEAKING,
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SS_SPEAKING,
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SS_WAS_SPEAKING_RECENTLY1,
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SS_WAS_SPEAKING_RECENTLY2
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};
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bool started_;
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Call* call_;
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BaseSession* session_;
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std::map<uint32, SpeakingState> ssrc_to_speaking_state_map_;
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uint32 current_speaker_ssrc_;
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// To prevent overswitching, switching is disabled for some time after a
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// switch is made. This gives us the earliest time a switch is permitted.
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uint32 earliest_permitted_switch_time_;
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uint32 min_time_between_switches_;
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};
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}
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#endif // TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
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