96 lines
3.5 KiB
C
96 lines
3.5 KiB
C
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/*
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* Copyright 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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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 WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
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#define WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
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#include <map>
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/sigslot.h"
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namespace cricket {
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struct AudioInfo;
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struct MediaStreams;
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class AudioSourceContext {
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public:
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sigslot::signal2<AudioSourceContext*, const cricket::AudioInfo&>
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SignalAudioMonitor;
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sigslot::signal1<AudioSourceContext*> SignalMediaStreamsReset;
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sigslot::signal3<AudioSourceContext*,
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const cricket::MediaStreams&,
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const cricket::MediaStreams&> SignalMediaStreamsUpdate;
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};
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// CurrentSpeakerMonitor can be used to monitor the audio-levels from
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// many audio-sources and report on changes in the loudest audio-source.
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// Its a generic type and relies on an AudioSourceContext which is aware of
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// the audio-sources. AudioSourceContext needs to provide two signals namely
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// SignalAudioInfoMonitor - provides audio info of the all current speakers.
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// SignalMediaSourcesUpdated - provides updates when a speaker leaves or joins.
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// Note that the AudioSourceContext's audio monitor must be started
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// 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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explicit CurrentSpeakerMonitor(AudioSourceContext* audio_source_context);
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~CurrentSpeakerMonitor();
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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(int 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
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// AudioSourceContext has been started.
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sigslot::signal2<CurrentSpeakerMonitor*, uint32_t> SignalUpdate;
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private:
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void OnAudioMonitor(AudioSourceContext* audio_source_context,
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const AudioInfo& info);
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void OnMediaStreamsUpdate(AudioSourceContext* audio_source_context,
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const MediaStreams& added,
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const MediaStreams& removed);
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void OnMediaStreamsReset(AudioSourceContext* audio_source_context);
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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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AudioSourceContext* audio_source_context_;
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std::map<uint32_t, SpeakingState> ssrc_to_speaking_state_map_;
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uint32_t 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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int64_t earliest_permitted_switch_time_;
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int min_time_between_switches_;
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};
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} // namespace cricket
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#endif // WEBRTC_PC_CURRENTSPEAKERMONITOR_H_
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