webrtc_m130/talk/session/media/currentspeakermonitor.cc
kjellander a96e2d77cb Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.

The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.

The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL in order to not
break Git history.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
  webrtc/base/testutils.cc
  webrtc/base/testutils.h

The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.

I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/

BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1587193006

Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-05 07:52:35 +00:00

219 lines
7.8 KiB
C++

/*
* libjingle
* Copyright 2011 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/currentspeakermonitor.h"
#include "talk/session/media/audiomonitor.h"
#include "webrtc/base/logging.h"
#include "webrtc/media/base/streamparams.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(
uint32_t 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.
uint32_t now = rtc::Time();
if (earliest_permitted_switch_time_ <= now &&
current_speaker_ssrc_ != loudest_speaker_ssrc) {
current_speaker_ssrc_ = loudest_speaker_ssrc;
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