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