| /* |
| * 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. |
| |
| #ifndef WEBRTC_PC_CURRENTSPEAKERMONITOR_H_ |
| #define WEBRTC_PC_CURRENTSPEAKERMONITOR_H_ |
| |
| #include <map> |
| |
| #include "webrtc/base/basictypes.h" |
| #include "webrtc/base/sigslot.h" |
| |
| namespace cricket { |
| |
| struct AudioInfo; |
| struct MediaStreams; |
| |
| class AudioSourceContext { |
| public: |
| sigslot::signal2<AudioSourceContext*, const cricket::AudioInfo&> |
| SignalAudioMonitor; |
| sigslot::signal1<AudioSourceContext*> SignalMediaStreamsReset; |
| sigslot::signal3<AudioSourceContext*, |
| 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<> { |
| public: |
| explicit CurrentSpeakerMonitor(AudioSourceContext* audio_source_context); |
| ~CurrentSpeakerMonitor(); |
| |
| 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; |
| |
| private: |
| 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 { |
| SS_NOT_SPEAKING, |
| SS_MIGHT_BE_SPEAKING, |
| SS_SPEAKING, |
| SS_WAS_SPEAKING_RECENTLY1, |
| SS_WAS_SPEAKING_RECENTLY2 |
| }; |
| |
| 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 |
| |
| #endif // WEBRTC_PC_CURRENTSPEAKERMONITOR_H_ |