Delete unused class CurrentSpeakerMonitor.
Bug: webrtc:8760
Change-Id: Ib2f84c7d74f1f3187f02dcf697e9c16a4d5f10e3
Reviewed-on: https://webrtc-review.googlesource.com/34652
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23336}diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 0681564..e117bb3 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -29,13 +29,10 @@
visibility = [ "*" ]
defines = []
sources = [
- "audiomonitor.h",
"channel.cc",
"channel.h",
"channelmanager.cc",
"channelmanager.h",
- "currentspeakermonitor.cc",
- "currentspeakermonitor.h",
"dtlssrtptransport.cc",
"dtlssrtptransport.h",
"externalhmac.cc",
@@ -283,7 +280,6 @@
sources = [
"channel_unittest.cc",
"channelmanager_unittest.cc",
- "currentspeakermonitor_unittest.cc",
"dtlssrtptransport_unittest.cc",
"jseptransport_unittest.cc",
"jseptransportcontroller_unittest.cc",
diff --git a/pc/audiomonitor.h b/pc/audiomonitor.h
deleted file mode 100644
index 61edd32..0000000
--- a/pc/audiomonitor.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Copyright 2004 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.
- */
-
-#ifndef PC_AUDIOMONITOR_H_
-#define PC_AUDIOMONITOR_H_
-
-#include <vector>
-#include <utility>
-
-// For ConnectionInfo/ConnectionInfos
-#include "p2p/base/port.h"
-
-namespace cricket {
-
-struct AudioInfo {
- int input_level;
- int output_level;
- typedef std::vector<std::pair<uint32_t, int> > StreamList;
- StreamList active_streams; // ssrcs contributing to output_level
-};
-
-} // namespace cricket
-
-#endif // PC_AUDIOMONITOR_H_
diff --git a/pc/channel.h b/pc/channel.h
index cb068a2..9532ee4 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -29,7 +29,6 @@
#include "media/base/streamparams.h"
#include "p2p/base/dtlstransportinternal.h"
#include "p2p/base/packettransportinternal.h"
-#include "pc/audiomonitor.h"
#include "pc/dtlssrtptransport.h"
#include "pc/mediasession.h"
#include "pc/rtptransport.h"
diff --git a/pc/currentspeakermonitor.cc b/pc/currentspeakermonitor.cc
deleted file mode 100644
index 2b584196..0000000
--- a/pc/currentspeakermonitor.cc
+++ /dev/null
@@ -1,204 +0,0 @@
-/*
- * 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.
- */
-
-#include "pc/currentspeakermonitor.h"
-
-#include <vector>
-
-#include "media/base/streamparams.h"
-#include "pc/audiomonitor.h"
-#include "rtc_base/logging.h"
-#include "rtc_base/timeutils.h"
-
-namespace cricket {
-
-namespace {
-const int kMaxAudioLevel = 9;
-// To avoid overswitching, we disable switching for a period of time after a
-// switch is done.
-const int kDefaultMinTimeBetweenSwitches = 1000;
-}
-
-CurrentSpeakerMonitor::CurrentSpeakerMonitor(
- AudioSourceContext* audio_source_context)
- : started_(false),
- audio_source_context_(audio_source_context),
- current_speaker_ssrc_(0),
- earliest_permitted_switch_time_(0),
- min_time_between_switches_(kDefaultMinTimeBetweenSwitches) {}
-
-CurrentSpeakerMonitor::~CurrentSpeakerMonitor() {
- Stop();
-}
-
-void CurrentSpeakerMonitor::Start() {
- if (!started_) {
- audio_source_context_->SignalAudioMonitor.connect(
- this, &CurrentSpeakerMonitor::OnAudioMonitor);
- audio_source_context_->SignalMediaStreamsUpdate.connect(
- this, &CurrentSpeakerMonitor::OnMediaStreamsUpdate);
- audio_source_context_->SignalMediaStreamsReset.connect(
- this, &CurrentSpeakerMonitor::OnMediaStreamsReset);
-
- started_ = true;
- }
-}
-
-void CurrentSpeakerMonitor::Stop() {
- if (started_) {
- audio_source_context_->SignalAudioMonitor.disconnect(this);
- audio_source_context_->SignalMediaStreamsUpdate.disconnect(this);
-
- started_ = false;
- ssrc_to_speaking_state_map_.clear();
- current_speaker_ssrc_ = 0;
- earliest_permitted_switch_time_ = 0;
- }
-}
-
-void CurrentSpeakerMonitor::set_min_time_between_switches(
- int min_time_between_switches) {
- min_time_between_switches_ = min_time_between_switches;
-}
-
-void CurrentSpeakerMonitor::OnAudioMonitor(
- AudioSourceContext* audio_source_context, const AudioInfo& info) {
- std::map<uint32_t, int> active_ssrc_to_level_map;
- cricket::AudioInfo::StreamList::const_iterator stream_list_it;
- for (stream_list_it = info.active_streams.begin();
- stream_list_it != info.active_streams.end(); ++stream_list_it) {
- uint32_t ssrc = stream_list_it->first;
- active_ssrc_to_level_map[ssrc] = stream_list_it->second;
-
- // It's possible we haven't yet added this source to our map. If so,
- // add it now with a "not speaking" state.
- if (ssrc_to_speaking_state_map_.find(ssrc) ==
- ssrc_to_speaking_state_map_.end()) {
- ssrc_to_speaking_state_map_[ssrc] = SS_NOT_SPEAKING;
- }
- }
-
- int max_level = 0;
- uint32_t loudest_speaker_ssrc = 0;
-
- // Update the speaking states of all participants based on the new audio
- // level information. Also retain loudest speaker.
- std::map<uint32_t, SpeakingState>::iterator state_it;
- for (state_it = ssrc_to_speaking_state_map_.begin();
- state_it != ssrc_to_speaking_state_map_.end(); ++state_it) {
- bool is_previous_speaker = current_speaker_ssrc_ == state_it->first;
-
- // This uses a state machine in order to gradually identify
- // members as having started or stopped speaking. Matches the
- // algorithm used by the hangouts js code.
-
- std::map<uint32_t, int>::const_iterator level_it =
- active_ssrc_to_level_map.find(state_it->first);
- // Note that the stream map only contains streams with non-zero audio
- // levels.
- int level = (level_it != active_ssrc_to_level_map.end()) ?
- level_it->second : 0;
- switch (state_it->second) {
- case SS_NOT_SPEAKING:
- if (level > 0) {
- // Reset level because we don't think they're really speaking.
- level = 0;
- state_it->second = SS_MIGHT_BE_SPEAKING;
- } else {
- // State unchanged.
- }
- break;
- case SS_MIGHT_BE_SPEAKING:
- if (level > 0) {
- state_it->second = SS_SPEAKING;
- } else {
- state_it->second = SS_NOT_SPEAKING;
- }
- break;
- case SS_SPEAKING:
- if (level > 0) {
- // State unchanged.
- } else {
- state_it->second = SS_WAS_SPEAKING_RECENTLY1;
- if (is_previous_speaker) {
- // Assume this is an inter-word silence and assign him the highest
- // volume.
- level = kMaxAudioLevel;
- }
- }
- break;
- case SS_WAS_SPEAKING_RECENTLY1:
- if (level > 0) {
- state_it->second = SS_SPEAKING;
- } else {
- state_it->second = SS_WAS_SPEAKING_RECENTLY2;
- if (is_previous_speaker) {
- // Assume this is an inter-word silence and assign him the highest
- // volume.
- level = kMaxAudioLevel;
- }
- }
- break;
- case SS_WAS_SPEAKING_RECENTLY2:
- if (level > 0) {
- state_it->second = SS_SPEAKING;
- } else {
- state_it->second = SS_NOT_SPEAKING;
- }
- break;
- }
-
- if (level > max_level) {
- loudest_speaker_ssrc = state_it->first;
- max_level = level;
- } else if (level > 0 && level == max_level && is_previous_speaker) {
- // Favor continuity of loudest speakers if audio levels are equal.
- loudest_speaker_ssrc = state_it->first;
- }
- }
-
- // We avoid over-switching by disabling switching for a period of time after
- // a switch is done.
- int64_t now = rtc::TimeMillis();
- if (earliest_permitted_switch_time_ <= now &&
- current_speaker_ssrc_ != loudest_speaker_ssrc) {
- current_speaker_ssrc_ = loudest_speaker_ssrc;
- RTC_LOG(LS_INFO) << "Current speaker changed to " << current_speaker_ssrc_;
- earliest_permitted_switch_time_ = now + min_time_between_switches_;
- SignalUpdate(this, current_speaker_ssrc_);
- }
-}
-
-void CurrentSpeakerMonitor::OnMediaStreamsUpdate(
- AudioSourceContext* audio_source_context,
- const MediaStreams& added,
- const MediaStreams& removed) {
- if (audio_source_context == audio_source_context_) {
- // Update the speaking state map based on added and removed streams.
- for (std::vector<cricket::StreamParams>::const_iterator
- it = removed.audio().begin(); it != removed.audio().end(); ++it) {
- ssrc_to_speaking_state_map_.erase(it->first_ssrc());
- }
-
- for (std::vector<cricket::StreamParams>::const_iterator
- it = added.audio().begin(); it != added.audio().end(); ++it) {
- ssrc_to_speaking_state_map_[it->first_ssrc()] = SS_NOT_SPEAKING;
- }
- }
-}
-
-void CurrentSpeakerMonitor::OnMediaStreamsReset(
- AudioSourceContext* audio_source_context) {
- if (audio_source_context == audio_source_context_) {
- ssrc_to_speaking_state_map_.clear();
- }
-}
-
-} // namespace cricket
diff --git a/pc/currentspeakermonitor.h b/pc/currentspeakermonitor.h
deleted file mode 100644
index 589fb1b..0000000
--- a/pc/currentspeakermonitor.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * 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 PC_CURRENTSPEAKERMONITOR_H_
-#define PC_CURRENTSPEAKERMONITOR_H_
-
-#include <stdint.h>
-
-#include <map>
-
-#include "rtc_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 // PC_CURRENTSPEAKERMONITOR_H_
diff --git a/pc/currentspeakermonitor_unittest.cc b/pc/currentspeakermonitor_unittest.cc
deleted file mode 100644
index fa3f0cb..0000000
--- a/pc/currentspeakermonitor_unittest.cc
+++ /dev/null
@@ -1,217 +0,0 @@
-/*
- * Copyright 2004 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.
- */
-
-#include <utility>
-
-#include "pc/currentspeakermonitor.h"
-#include "pc/audiomonitor.h"
-#include "rtc_base/gunit.h"
-#include "rtc_base/thread.h"
-
-namespace cricket {
-
-static const uint32_t kSsrc1 = 1001;
-static const uint32_t kSsrc2 = 1002;
-static const uint32_t kMinTimeBetweenSwitches = 10;
-// Due to limited system clock resolution, the CurrentSpeakerMonitor may
-// actually require more or less time between switches than that specified
-// in the call to set_min_time_between_switches. To be safe, we sleep for
-// 90 ms more than the min time between switches before checking for a switch.
-// I am assuming system clocks do not have a coarser resolution than 90 ms.
-static const uint32_t kSleepTimeBetweenSwitches = 100;
-
-class CurrentSpeakerMonitorTest : public testing::Test,
- public sigslot::has_slots<> {
- public:
- CurrentSpeakerMonitorTest() {
- monitor_ = new CurrentSpeakerMonitor(&source_);
- // Shrink the minimum time betweeen switches to 10 ms so we don't have to
- // slow down our tests.
- monitor_->set_min_time_between_switches(kMinTimeBetweenSwitches);
- monitor_->SignalUpdate.connect(this, &CurrentSpeakerMonitorTest::OnUpdate);
- current_speaker_ = 0;
- num_changes_ = 0;
- monitor_->Start();
- }
-
- ~CurrentSpeakerMonitorTest() {
- delete monitor_;
- }
-
- void SignalAudioMonitor(const AudioInfo& info) {
- source_.SignalAudioMonitor(&source_, info);
- }
-
- protected:
- AudioSourceContext source_;
- CurrentSpeakerMonitor* monitor_;
- int num_changes_;
- uint32_t current_speaker_;
-
- void OnUpdate(CurrentSpeakerMonitor* monitor, uint32_t current_speaker) {
- current_speaker_ = current_speaker;
- num_changes_++;
- }
-};
-
-static void InitAudioInfo(AudioInfo* info, int input_level, int output_level) {
- info->input_level = input_level;
- info->output_level = output_level;
-}
-
-TEST_F(CurrentSpeakerMonitorTest, NoActiveStreams) {
- AudioInfo info;
- InitAudioInfo(&info, 0, 0);
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, 0U);
- EXPECT_EQ(num_changes_, 0);
-}
-
-TEST_F(CurrentSpeakerMonitorTest, MultipleActiveStreams) {
- AudioInfo info;
- InitAudioInfo(&info, 0, 0);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 7));
- SignalAudioMonitor(info);
-
- // No speaker recognized because the initial sample is treated as possibly
- // just noise and disregarded.
- EXPECT_EQ(current_speaker_, 0U);
- EXPECT_EQ(num_changes_, 0);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 7));
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, kSsrc2);
- EXPECT_EQ(num_changes_, 1);
-}
-
-// See: https://code.google.com/p/webrtc/issues/detail?id=2409
-TEST_F(CurrentSpeakerMonitorTest, DISABLED_RapidSpeakerChange) {
- AudioInfo info;
- InitAudioInfo(&info, 0, 0);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 7));
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, 0U);
- EXPECT_EQ(num_changes_, 0);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 7));
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, kSsrc2);
- EXPECT_EQ(num_changes_, 1);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 9));
- info.active_streams.push_back(std::make_pair(kSsrc2, 1));
- SignalAudioMonitor(info);
-
- // We expect no speaker change because of the rapid change.
- EXPECT_EQ(current_speaker_, kSsrc2);
- EXPECT_EQ(num_changes_, 1);
-}
-
-// Flaky on iOS: webrtc:7057.
-#if defined(WEBRTC_IOS)
-#define MAYBE_SpeakerChange DISABLED_SpeakerChange
-#else
-#define MAYBE_SpeakerChange SpeakerChange
-#endif
-TEST_F(CurrentSpeakerMonitorTest, MAYBE_SpeakerChange) {
- AudioInfo info;
- InitAudioInfo(&info, 0, 0);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 7));
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, 0U);
- EXPECT_EQ(num_changes_, 0);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 7));
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, kSsrc2);
- EXPECT_EQ(num_changes_, 1);
-
- // Wait so the changes don't come so rapidly.
- rtc::Thread::SleepMs(kSleepTimeBetweenSwitches);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 9));
- info.active_streams.push_back(std::make_pair(kSsrc2, 1));
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, kSsrc1);
- EXPECT_EQ(num_changes_, 2);
-}
-
-TEST_F(CurrentSpeakerMonitorTest, InterwordSilence) {
- AudioInfo info;
- InitAudioInfo(&info, 0, 0);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 7));
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, 0U);
- EXPECT_EQ(num_changes_, 0);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 7));
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, kSsrc2);
- EXPECT_EQ(num_changes_, 1);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 7));
- SignalAudioMonitor(info);
-
- EXPECT_EQ(current_speaker_, kSsrc2);
- EXPECT_EQ(num_changes_, 1);
-
- // Wait so the changes don't come so rapidly.
- rtc::Thread::SleepMs(kSleepTimeBetweenSwitches);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 0));
- SignalAudioMonitor(info);
-
- // Current speaker shouldn't have changed because we treat this as an inter-
- // word silence.
- EXPECT_EQ(current_speaker_, kSsrc2);
- EXPECT_EQ(num_changes_, 1);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 0));
- SignalAudioMonitor(info);
-
- // Current speaker shouldn't have changed because we treat this as an inter-
- // word silence.
- EXPECT_EQ(current_speaker_, kSsrc2);
- EXPECT_EQ(num_changes_, 1);
-
- info.active_streams.push_back(std::make_pair(kSsrc1, 3));
- info.active_streams.push_back(std::make_pair(kSsrc2, 0));
- SignalAudioMonitor(info);
-
- // At this point, we should have concluded that SSRC2 stopped speaking.
- EXPECT_EQ(current_speaker_, kSsrc1);
- EXPECT_EQ(num_changes_, 2);
-}
-
-} // namespace cricket
