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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2011 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28// CurrentSpeakerMonitor monitors the audio levels for a session and determines
29// which participant is currently speaking.
30
31#ifndef TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
32#define TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_
33
34#include <map>
35
36#include "talk/base/basictypes.h"
37#include "talk/base/sigslot.h"
38
39namespace cricket {
40
41class BaseSession;
42class Call;
43class Session;
44struct AudioInfo;
45struct MediaStreams;
46
buildbot@webrtc.orgca272362014-05-08 23:10:23 +000047class AudioSourceContext {
48 public:
49 sigslot::signal2<AudioSourceContext*, const cricket::AudioInfo&>
50 SignalAudioMonitor;
51 sigslot::signal4<AudioSourceContext*, cricket::Session*,
52 const cricket::MediaStreams&, const cricket::MediaStreams&>
53 SignalMediaStreamsUpdate;
54};
55
56// CurrentSpeakerMonitor can be used to monitor the audio-levels from
57// many audio-sources and report on changes in the loudest audio-source.
58// Its a generic type and relies on an AudioSourceContext which is aware of
59// the audio-sources. AudioSourceContext needs to provide two signals namely
60// SignalAudioInfoMonitor - provides audio info of the all current speakers.
61// SignalMediaSourcesUpdated - provides updates when a speaker leaves or joins.
62// Note that the AudioSourceContext's audio monitor must be started
63// before this is started.
henrike@webrtc.org28e20752013-07-10 00:45:36 +000064// It's recommended that the audio monitor be started with a 100 ms period.
65class CurrentSpeakerMonitor : public sigslot::has_slots<> {
66 public:
buildbot@webrtc.orgca272362014-05-08 23:10:23 +000067 CurrentSpeakerMonitor(AudioSourceContext* call, BaseSession* session);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000068 ~CurrentSpeakerMonitor();
69
70 BaseSession* session() const { return session_; }
71
72 void Start();
73 void Stop();
74
75 // Used by tests. Note that the actual minimum time between switches
76 // enforced by the monitor will be the given value plus or minus the
77 // resolution of the system clock.
78 void set_min_time_between_switches(uint32 min_time_between_switches);
79
80 // This is fired when the current speaker changes, and provides his audio
81 // SSRC. This only fires after the audio monitor on the underlying Call has
82 // been started.
83 sigslot::signal2<CurrentSpeakerMonitor*, uint32> SignalUpdate;
84
85 private:
buildbot@webrtc.orgca272362014-05-08 23:10:23 +000086 void OnAudioMonitor(AudioSourceContext* call, const AudioInfo& info);
87 void OnMediaStreamsUpdate(AudioSourceContext* call,
henrike@webrtc.org28e20752013-07-10 00:45:36 +000088 Session* session,
89 const MediaStreams& added,
90 const MediaStreams& removed);
91
92 // These are states that a participant will pass through so that we gradually
93 // recognize that they have started and stopped speaking. This avoids
94 // "twitchiness".
95 enum SpeakingState {
96 SS_NOT_SPEAKING,
97 SS_MIGHT_BE_SPEAKING,
98 SS_SPEAKING,
99 SS_WAS_SPEAKING_RECENTLY1,
100 SS_WAS_SPEAKING_RECENTLY2
101 };
102
103 bool started_;
buildbot@webrtc.orgca272362014-05-08 23:10:23 +0000104 AudioSourceContext* audio_source_context_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000105 BaseSession* session_;
106 std::map<uint32, SpeakingState> ssrc_to_speaking_state_map_;
107 uint32 current_speaker_ssrc_;
108 // To prevent overswitching, switching is disabled for some time after a
109 // switch is made. This gives us the earliest time a switch is permitted.
110 uint32 earliest_permitted_switch_time_;
111 uint32 min_time_between_switches_;
112};
113
114}
115
116#endif // TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_