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| 1 /* | |
| 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 "webrtc/base/basictypes.h" | |
| 37 #include "webrtc/base/sigslot.h" | |
| 38 | |
| 39 namespace cricket { | |
| 40 | |
| 41 struct AudioInfo; | |
| 42 struct MediaStreams; | |
| 43 | |
| 44 class AudioSourceContext { | |
| 45 public: | |
| 46 sigslot::signal2<AudioSourceContext*, const cricket::AudioInfo&> | |
| 47 SignalAudioMonitor; | |
| 48 sigslot::signal1<AudioSourceContext*> SignalMediaStreamsReset; | |
| 49 sigslot::signal3<AudioSourceContext*, | |
| 50 const cricket::MediaStreams&, | |
| 51 const cricket::MediaStreams&> SignalMediaStreamsUpdate; | |
| 52 }; | |
| 53 | |
| 54 // CurrentSpeakerMonitor can be used to monitor the audio-levels from | |
| 55 // many audio-sources and report on changes in the loudest audio-source. | |
| 56 // Its a generic type and relies on an AudioSourceContext which is aware of | |
| 57 // the audio-sources. AudioSourceContext needs to provide two signals namely | |
| 58 // SignalAudioInfoMonitor - provides audio info of the all current speakers. | |
| 59 // SignalMediaSourcesUpdated - provides updates when a speaker leaves or joins. | |
| 60 // Note that the AudioSourceContext's audio monitor must be started | |
| 61 // before this is started. | |
| 62 // It's recommended that the audio monitor be started with a 100 ms period. | |
| 63 class CurrentSpeakerMonitor : public sigslot::has_slots<> { | |
| 64 public: | |
| 65 CurrentSpeakerMonitor(AudioSourceContext* audio_source_context); | |
| 66 ~CurrentSpeakerMonitor(); | |
| 67 | |
| 68 void Start(); | |
| 69 void Stop(); | |
| 70 | |
| 71 // Used by tests. Note that the actual minimum time between switches | |
| 72 // enforced by the monitor will be the given value plus or minus the | |
| 73 // resolution of the system clock. | |
| 74 void set_min_time_between_switches(uint32_t min_time_between_switches); | |
| 75 | |
| 76 // This is fired when the current speaker changes, and provides his audio | |
| 77 // SSRC. This only fires after the audio monitor on the underlying | |
| 78 // AudioSourceContext has been started. | |
| 79 sigslot::signal2<CurrentSpeakerMonitor*, uint32_t> SignalUpdate; | |
| 80 | |
| 81 private: | |
| 82 void OnAudioMonitor(AudioSourceContext* audio_source_context, | |
| 83 const AudioInfo& info); | |
| 84 void OnMediaStreamsUpdate(AudioSourceContext* audio_source_context, | |
| 85 const MediaStreams& added, | |
| 86 const MediaStreams& removed); | |
| 87 void OnMediaStreamsReset(AudioSourceContext* audio_source_context); | |
| 88 | |
| 89 // These are states that a participant will pass through so that we gradually | |
| 90 // recognize that they have started and stopped speaking. This avoids | |
| 91 // "twitchiness". | |
| 92 enum SpeakingState { | |
| 93 SS_NOT_SPEAKING, | |
| 94 SS_MIGHT_BE_SPEAKING, | |
| 95 SS_SPEAKING, | |
| 96 SS_WAS_SPEAKING_RECENTLY1, | |
| 97 SS_WAS_SPEAKING_RECENTLY2 | |
| 98 }; | |
| 99 | |
| 100 bool started_; | |
| 101 AudioSourceContext* audio_source_context_; | |
| 102 std::map<uint32_t, SpeakingState> ssrc_to_speaking_state_map_; | |
| 103 uint32_t current_speaker_ssrc_; | |
| 104 // To prevent overswitching, switching is disabled for some time after a | |
| 105 // switch is made. This gives us the earliest time a switch is permitted. | |
| 106 uint32_t earliest_permitted_switch_time_; | |
| 107 uint32_t min_time_between_switches_; | |
| 108 }; | |
| 109 | |
| 110 } | |
| 111 | |
| 112 #endif // TALK_SESSION_MEDIA_CURRENTSPEAKERMONITOR_H_ | |
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Chromium Code Reviews
Issue 1691463002:
Move talk/session/media -> webrtc/pc (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master