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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
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
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000030#include "webrtc/base/logging.h"
buildbot@webrtc.org117afee2014-06-16 07:11:01 +000031#include "talk/media/base/streamparams.h"
32#include "talk/session/media/audiomonitor.h"
33#include "talk/session/media/mediamessages.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000034
35namespace cricket {
36
37namespace {
38const int kMaxAudioLevel = 9;
39// To avoid overswitching, we disable switching for a period of time after a
40// switch is done.
41const int kDefaultMinTimeBetweenSwitches = 1000;
42}
43
buildbot@webrtc.orgca272362014-05-08 23:10:23 +000044CurrentSpeakerMonitor::CurrentSpeakerMonitor(
45 AudioSourceContext* audio_source_context, BaseSession* session)
henrike@webrtc.org28e20752013-07-10 00:45:36 +000046 : started_(false),
buildbot@webrtc.orgca272362014-05-08 23:10:23 +000047 audio_source_context_(audio_source_context),
henrike@webrtc.org28e20752013-07-10 00:45:36 +000048 session_(session),
49 current_speaker_ssrc_(0),
50 earliest_permitted_switch_time_(0),
51 min_time_between_switches_(kDefaultMinTimeBetweenSwitches) {
52}
53
54CurrentSpeakerMonitor::~CurrentSpeakerMonitor() {
55 Stop();
56}
57
58void CurrentSpeakerMonitor::Start() {
59 if (!started_) {
buildbot@webrtc.orgca272362014-05-08 23:10:23 +000060 audio_source_context_->SignalAudioMonitor.connect(
henrike@webrtc.org28e20752013-07-10 00:45:36 +000061 this, &CurrentSpeakerMonitor::OnAudioMonitor);
buildbot@webrtc.orgca272362014-05-08 23:10:23 +000062 audio_source_context_->SignalMediaStreamsUpdate.connect(
henrike@webrtc.org28e20752013-07-10 00:45:36 +000063 this, &CurrentSpeakerMonitor::OnMediaStreamsUpdate);
buildbot@webrtc.org49a6a272014-05-21 00:24:54 +000064 audio_source_context_->SignalMediaStreamsReset.connect(
65 this, &CurrentSpeakerMonitor::OnMediaStreamsReset);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000066
67 started_ = true;
68 }
69}
70
71void CurrentSpeakerMonitor::Stop() {
72 if (started_) {
buildbot@webrtc.orgca272362014-05-08 23:10:23 +000073 audio_source_context_->SignalAudioMonitor.disconnect(this);
74 audio_source_context_->SignalMediaStreamsUpdate.disconnect(this);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075
76 started_ = false;
77 ssrc_to_speaking_state_map_.clear();
78 current_speaker_ssrc_ = 0;
79 earliest_permitted_switch_time_ = 0;
80 }
81}
82
83void CurrentSpeakerMonitor::set_min_time_between_switches(
84 uint32 min_time_between_switches) {
85 min_time_between_switches_ = min_time_between_switches;
86}
87
buildbot@webrtc.orgca272362014-05-08 23:10:23 +000088void CurrentSpeakerMonitor::OnAudioMonitor(
89 AudioSourceContext* audio_source_context, const AudioInfo& info) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000090 std::map<uint32, int> active_ssrc_to_level_map;
91 cricket::AudioInfo::StreamList::const_iterator stream_list_it;
92 for (stream_list_it = info.active_streams.begin();
93 stream_list_it != info.active_streams.end(); ++stream_list_it) {
94 uint32 ssrc = stream_list_it->first;
95 active_ssrc_to_level_map[ssrc] = stream_list_it->second;
96
97 // It's possible we haven't yet added this source to our map. If so,
98 // add it now with a "not speaking" state.
99 if (ssrc_to_speaking_state_map_.find(ssrc) ==
100 ssrc_to_speaking_state_map_.end()) {
101 ssrc_to_speaking_state_map_[ssrc] = SS_NOT_SPEAKING;
102 }
103 }
104
105 int max_level = 0;
106 uint32 loudest_speaker_ssrc = 0;
107
108 // Update the speaking states of all participants based on the new audio
109 // level information. Also retain loudest speaker.
110 std::map<uint32, SpeakingState>::iterator state_it;
111 for (state_it = ssrc_to_speaking_state_map_.begin();
112 state_it != ssrc_to_speaking_state_map_.end(); ++state_it) {
113 bool is_previous_speaker = current_speaker_ssrc_ == state_it->first;
114
115 // This uses a state machine in order to gradually identify
116 // members as having started or stopped speaking. Matches the
117 // algorithm used by the hangouts js code.
118
119 std::map<uint32, int>::const_iterator level_it =
120 active_ssrc_to_level_map.find(state_it->first);
121 // Note that the stream map only contains streams with non-zero audio
122 // levels.
123 int level = (level_it != active_ssrc_to_level_map.end()) ?
124 level_it->second : 0;
125 switch (state_it->second) {
126 case SS_NOT_SPEAKING:
127 if (level > 0) {
128 // Reset level because we don't think they're really speaking.
129 level = 0;
130 state_it->second = SS_MIGHT_BE_SPEAKING;
131 } else {
132 // State unchanged.
133 }
134 break;
135 case SS_MIGHT_BE_SPEAKING:
136 if (level > 0) {
137 state_it->second = SS_SPEAKING;
138 } else {
139 state_it->second = SS_NOT_SPEAKING;
140 }
141 break;
142 case SS_SPEAKING:
143 if (level > 0) {
144 // State unchanged.
145 } else {
146 state_it->second = SS_WAS_SPEAKING_RECENTLY1;
147 if (is_previous_speaker) {
148 // Assume this is an inter-word silence and assign him the highest
149 // volume.
150 level = kMaxAudioLevel;
151 }
152 }
153 break;
154 case SS_WAS_SPEAKING_RECENTLY1:
155 if (level > 0) {
156 state_it->second = SS_SPEAKING;
157 } else {
158 state_it->second = SS_WAS_SPEAKING_RECENTLY2;
159 if (is_previous_speaker) {
160 // Assume this is an inter-word silence and assign him the highest
161 // volume.
162 level = kMaxAudioLevel;
163 }
164 }
165 break;
166 case SS_WAS_SPEAKING_RECENTLY2:
167 if (level > 0) {
168 state_it->second = SS_SPEAKING;
169 } else {
170 state_it->second = SS_NOT_SPEAKING;
171 }
172 break;
173 }
174
175 if (level > max_level) {
176 loudest_speaker_ssrc = state_it->first;
177 max_level = level;
178 } else if (level > 0 && level == max_level && is_previous_speaker) {
179 // Favor continuity of loudest speakers if audio levels are equal.
180 loudest_speaker_ssrc = state_it->first;
181 }
182 }
183
184 // We avoid over-switching by disabling switching for a period of time after
185 // a switch is done.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000186 uint32 now = rtc::Time();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000187 if (earliest_permitted_switch_time_ <= now &&
188 current_speaker_ssrc_ != loudest_speaker_ssrc) {
189 current_speaker_ssrc_ = loudest_speaker_ssrc;
190 LOG(LS_INFO) << "Current speaker changed to " << current_speaker_ssrc_;
191 earliest_permitted_switch_time_ = now + min_time_between_switches_;
192 SignalUpdate(this, current_speaker_ssrc_);
193 }
194}
195
buildbot@webrtc.orgca272362014-05-08 23:10:23 +0000196void CurrentSpeakerMonitor::OnMediaStreamsUpdate(
buildbot@webrtc.org49a6a272014-05-21 00:24:54 +0000197 AudioSourceContext* audio_source_context, BaseSession* session,
buildbot@webrtc.orgca272362014-05-08 23:10:23 +0000198 const MediaStreams& added, const MediaStreams& removed) {
buildbot@webrtc.org49a6a272014-05-21 00:24:54 +0000199
buildbot@webrtc.orgca272362014-05-08 23:10:23 +0000200 if (audio_source_context == audio_source_context_ && session == session_) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000201 // Update the speaking state map based on added and removed streams.
202 for (std::vector<cricket::StreamParams>::const_iterator
buildbot@webrtc.org49a6a272014-05-21 00:24:54 +0000203 it = removed.audio().begin(); it != removed.audio().end(); ++it) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000204 ssrc_to_speaking_state_map_.erase(it->first_ssrc());
205 }
206
207 for (std::vector<cricket::StreamParams>::const_iterator
buildbot@webrtc.org49a6a272014-05-21 00:24:54 +0000208 it = added.audio().begin(); it != added.audio().end(); ++it) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000209 ssrc_to_speaking_state_map_[it->first_ssrc()] = SS_NOT_SPEAKING;
210 }
211 }
212}
213
buildbot@webrtc.org49a6a272014-05-21 00:24:54 +0000214void CurrentSpeakerMonitor::OnMediaStreamsReset(
215 AudioSourceContext* audio_source_context, BaseSession* session) {
216 if (audio_source_context == audio_source_context_ && session == session_) {
217 ssrc_to_speaking_state_map_.clear();
218 }
219}
220
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000221} // namespace cricket