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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 #include "talk/session/media/currentspeakermonitor.h" | |
| 29 | |
| 30 #include "talk/session/media/audiomonitor.h" | |
| 31 #include "webrtc/base/logging.h" | |
| 32 #include "webrtc/media/base/streamparams.h" | |
| 33 | |
| 34 namespace cricket { | |
| 35 | |
| 36 namespace { | |
| 37 const int kMaxAudioLevel = 9; | |
| 38 // To avoid overswitching, we disable switching for a period of time after a | |
| 39 // switch is done. | |
| 40 const int kDefaultMinTimeBetweenSwitches = 1000; | |
| 41 } | |
| 42 | |
| 43 CurrentSpeakerMonitor::CurrentSpeakerMonitor( | |
| 44 AudioSourceContext* audio_source_context) | |
| 45 : started_(false), | |
| 46 audio_source_context_(audio_source_context), | |
| 47 current_speaker_ssrc_(0), | |
| 48 earliest_permitted_switch_time_(0), | |
| 49 min_time_between_switches_(kDefaultMinTimeBetweenSwitches) {} | |
| 50 | |
| 51 CurrentSpeakerMonitor::~CurrentSpeakerMonitor() { | |
| 52 Stop(); | |
| 53 } | |
| 54 | |
| 55 void CurrentSpeakerMonitor::Start() { | |
| 56 if (!started_) { | |
| 57 audio_source_context_->SignalAudioMonitor.connect( | |
| 58 this, &CurrentSpeakerMonitor::OnAudioMonitor); | |
| 59 audio_source_context_->SignalMediaStreamsUpdate.connect( | |
| 60 this, &CurrentSpeakerMonitor::OnMediaStreamsUpdate); | |
| 61 audio_source_context_->SignalMediaStreamsReset.connect( | |
| 62 this, &CurrentSpeakerMonitor::OnMediaStreamsReset); | |
| 63 | |
| 64 started_ = true; | |
| 65 } | |
| 66 } | |
| 67 | |
| 68 void CurrentSpeakerMonitor::Stop() { | |
| 69 if (started_) { | |
| 70 audio_source_context_->SignalAudioMonitor.disconnect(this); | |
| 71 audio_source_context_->SignalMediaStreamsUpdate.disconnect(this); | |
| 72 | |
| 73 started_ = false; | |
| 74 ssrc_to_speaking_state_map_.clear(); | |
| 75 current_speaker_ssrc_ = 0; | |
| 76 earliest_permitted_switch_time_ = 0; | |
| 77 } | |
| 78 } | |
| 79 | |
| 80 void CurrentSpeakerMonitor::set_min_time_between_switches( | |
| 81 uint32_t min_time_between_switches) { | |
| 82 min_time_between_switches_ = min_time_between_switches; | |
| 83 } | |
| 84 | |
| 85 void CurrentSpeakerMonitor::OnAudioMonitor( | |
| 86 AudioSourceContext* audio_source_context, const AudioInfo& info) { | |
| 87 std::map<uint32_t, int> active_ssrc_to_level_map; | |
| 88 cricket::AudioInfo::StreamList::const_iterator stream_list_it; | |
| 89 for (stream_list_it = info.active_streams.begin(); | |
| 90 stream_list_it != info.active_streams.end(); ++stream_list_it) { | |
| 91 uint32_t ssrc = stream_list_it->first; | |
| 92 active_ssrc_to_level_map[ssrc] = stream_list_it->second; | |
| 93 | |
| 94 // It's possible we haven't yet added this source to our map. If so, | |
| 95 // add it now with a "not speaking" state. | |
| 96 if (ssrc_to_speaking_state_map_.find(ssrc) == | |
| 97 ssrc_to_speaking_state_map_.end()) { | |
| 98 ssrc_to_speaking_state_map_[ssrc] = SS_NOT_SPEAKING; | |
| 99 } | |
| 100 } | |
| 101 | |
| 102 int max_level = 0; | |
| 103 uint32_t loudest_speaker_ssrc = 0; | |
| 104 | |
| 105 // Update the speaking states of all participants based on the new audio | |
| 106 // level information. Also retain loudest speaker. | |
| 107 std::map<uint32_t, SpeakingState>::iterator state_it; | |
| 108 for (state_it = ssrc_to_speaking_state_map_.begin(); | |
| 109 state_it != ssrc_to_speaking_state_map_.end(); ++state_it) { | |
| 110 bool is_previous_speaker = current_speaker_ssrc_ == state_it->first; | |
| 111 | |
| 112 // This uses a state machine in order to gradually identify | |
| 113 // members as having started or stopped speaking. Matches the | |
| 114 // algorithm used by the hangouts js code. | |
| 115 | |
| 116 std::map<uint32_t, int>::const_iterator level_it = | |
| 117 active_ssrc_to_level_map.find(state_it->first); | |
| 118 // Note that the stream map only contains streams with non-zero audio | |
| 119 // levels. | |
| 120 int level = (level_it != active_ssrc_to_level_map.end()) ? | |
| 121 level_it->second : 0; | |
| 122 switch (state_it->second) { | |
| 123 case SS_NOT_SPEAKING: | |
| 124 if (level > 0) { | |
| 125 // Reset level because we don't think they're really speaking. | |
| 126 level = 0; | |
| 127 state_it->second = SS_MIGHT_BE_SPEAKING; | |
| 128 } else { | |
| 129 // State unchanged. | |
| 130 } | |
| 131 break; | |
| 132 case SS_MIGHT_BE_SPEAKING: | |
| 133 if (level > 0) { | |
| 134 state_it->second = SS_SPEAKING; | |
| 135 } else { | |
| 136 state_it->second = SS_NOT_SPEAKING; | |
| 137 } | |
| 138 break; | |
| 139 case SS_SPEAKING: | |
| 140 if (level > 0) { | |
| 141 // State unchanged. | |
| 142 } else { | |
| 143 state_it->second = SS_WAS_SPEAKING_RECENTLY1; | |
| 144 if (is_previous_speaker) { | |
| 145 // Assume this is an inter-word silence and assign him the highest | |
| 146 // volume. | |
| 147 level = kMaxAudioLevel; | |
| 148 } | |
| 149 } | |
| 150 break; | |
| 151 case SS_WAS_SPEAKING_RECENTLY1: | |
| 152 if (level > 0) { | |
| 153 state_it->second = SS_SPEAKING; | |
| 154 } else { | |
| 155 state_it->second = SS_WAS_SPEAKING_RECENTLY2; | |
| 156 if (is_previous_speaker) { | |
| 157 // Assume this is an inter-word silence and assign him the highest | |
| 158 // volume. | |
| 159 level = kMaxAudioLevel; | |
| 160 } | |
| 161 } | |
| 162 break; | |
| 163 case SS_WAS_SPEAKING_RECENTLY2: | |
| 164 if (level > 0) { | |
| 165 state_it->second = SS_SPEAKING; | |
| 166 } else { | |
| 167 state_it->second = SS_NOT_SPEAKING; | |
| 168 } | |
| 169 break; | |
| 170 } | |
| 171 | |
| 172 if (level > max_level) { | |
| 173 loudest_speaker_ssrc = state_it->first; | |
| 174 max_level = level; | |
| 175 } else if (level > 0 && level == max_level && is_previous_speaker) { | |
| 176 // Favor continuity of loudest speakers if audio levels are equal. | |
| 177 loudest_speaker_ssrc = state_it->first; | |
| 178 } | |
| 179 } | |
| 180 | |
| 181 // We avoid over-switching by disabling switching for a period of time after | |
| 182 // a switch is done. | |
| 183 uint32_t now = rtc::Time(); | |
| 184 if (earliest_permitted_switch_time_ <= now && | |
| 185 current_speaker_ssrc_ != loudest_speaker_ssrc) { | |
| 186 current_speaker_ssrc_ = loudest_speaker_ssrc; | |
| 187 LOG(LS_INFO) << "Current speaker changed to " << current_speaker_ssrc_; | |
| 188 earliest_permitted_switch_time_ = now + min_time_between_switches_; | |
| 189 SignalUpdate(this, current_speaker_ssrc_); | |
| 190 } | |
| 191 } | |
| 192 | |
| 193 void CurrentSpeakerMonitor::OnMediaStreamsUpdate( | |
| 194 AudioSourceContext* audio_source_context, | |
| 195 const MediaStreams& added, | |
| 196 const MediaStreams& removed) { | |
| 197 if (audio_source_context == audio_source_context_) { | |
| 198 // Update the speaking state map based on added and removed streams. | |
| 199 for (std::vector<cricket::StreamParams>::const_iterator | |
| 200 it = removed.audio().begin(); it != removed.audio().end(); ++it) { | |
| 201 ssrc_to_speaking_state_map_.erase(it->first_ssrc()); | |
| 202 } | |
| 203 | |
| 204 for (std::vector<cricket::StreamParams>::const_iterator | |
| 205 it = added.audio().begin(); it != added.audio().end(); ++it) { | |
| 206 ssrc_to_speaking_state_map_[it->first_ssrc()] = SS_NOT_SPEAKING; | |
| 207 } | |
| 208 } | |
| 209 } | |
| 210 | |
| 211 void CurrentSpeakerMonitor::OnMediaStreamsReset( | |
| 212 AudioSourceContext* audio_source_context) { | |
| 213 if (audio_source_context == audio_source_context_) { | |
| 214 ssrc_to_speaking_state_map_.clear(); | |
| 215 } | |
| 216 } | |
| 217 | |
| 218 } // namespace cricket | |
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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