From 5872e037e801654d59f37e32f672db84c456d587 Mon Sep 17 00:00:00 2001
From: Roxie Linden <roxie@lindenlab.com>
Date: Fri, 29 Sep 2023 19:13:03 -0700
Subject: Improve reconnection logic and allow device setting when connected or
not connected
---
indra/llwebrtc/llwebrtc.cpp | 268 +++++++++++++++++++++++++----------------
indra/llwebrtc/llwebrtc.h | 5 +-
indra/llwebrtc/llwebrtc_impl.h | 17 ++-
3 files changed, 182 insertions(+), 108 deletions(-)
(limited to 'indra/llwebrtc')
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index 481d5b940d..95f87ed65b 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -42,15 +42,15 @@ const float VOLUME_SCALE_WEBRTC = 3.0f;
LLAudioDeviceObserver::LLAudioDeviceObserver() : mMicrophoneEnergy(0.0), mSumVector {0} {}
-double LLAudioDeviceObserver::getMicrophoneEnergy() { return mMicrophoneEnergy; }
+float LLAudioDeviceObserver::getMicrophoneEnergy() { return mMicrophoneEnergy; }
void LLAudioDeviceObserver::OnCaptureData(const void *audio_samples,
- const size_t num_samples,
- const size_t bytes_per_sample,
- const size_t num_channels,
- const uint32_t samples_per_sec)
+ const size_t num_samples,
+ const size_t bytes_per_sample,
+ const size_t num_channels,
+ const uint32_t samples_per_sec)
{
- float energy = 0;
+ float energy = 0;
const short *samples = (const short *) audio_samples;
for (size_t index = 0; index < num_samples * num_channels; index++)
{
@@ -60,8 +60,8 @@ void LLAudioDeviceObserver::OnCaptureData(const void *audio_samples,
// smooth it.
size_t buffer_size = sizeof(mSumVector) / sizeof(mSumVector[0]);
- float totalSum = 0;
- int i;
+ float totalSum = 0;
+ int i;
for (i = 0; i < (buffer_size - 1); i++)
{
mSumVector[i] = mSumVector[i + 1];
@@ -73,16 +73,17 @@ void LLAudioDeviceObserver::OnCaptureData(const void *audio_samples,
}
void LLAudioDeviceObserver::OnRenderData(const void *audio_samples,
- const size_t num_samples,
- const size_t bytes_per_sample,
- const size_t num_channels,
- const uint32_t samples_per_sec)
+ const size_t num_samples,
+ const size_t bytes_per_sample,
+ const size_t num_channels,
+ const uint32_t samples_per_sec)
{
}
-
void LLWebRTCImpl::init()
{
+ mPlayoutDevice = -1;
+ mRecordingDevice = -1;
mAnswerReceived = false;
rtc::InitializeSSL();
mTaskQueueFactory = webrtc::CreateDefaultTaskQueueFactory();
@@ -100,20 +101,21 @@ void LLWebRTCImpl::init()
mWorkerThread->PostTask(
[this]()
{
- mAudioDeviceObserver = new LLAudioDeviceObserver;
- mDeviceModule = webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
- mTaskQueueFactory.get(),
- std::unique_ptr<webrtc::AudioDeviceDataObserver>(mAudioDeviceObserver));
- mDeviceModule->Init();
- mDeviceModule->SetStereoRecording(false);
- mDeviceModule->SetStereoPlayout(true);
- mDeviceModule->EnableBuiltInAEC(false);
+ mTuningAudioDeviceObserver = new LLAudioDeviceObserver;
+ mTuningDeviceModule =
+ webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
+ mTaskQueueFactory.get(),
+ std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
+ mTuningDeviceModule->Init();
+ mTuningDeviceModule->SetStereoRecording(false);
+ mTuningDeviceModule->SetStereoPlayout(true);
+ mTuningDeviceModule->EnableBuiltInAEC(false);
updateDevices();
});
}
-void LLWebRTCImpl::terminate()
-{
+void LLWebRTCImpl::terminate()
+{
mSignalingThread->BlockingCall(
[this]()
{
@@ -127,13 +129,17 @@ void LLWebRTCImpl::terminate()
mWorkerThread->BlockingCall(
[this]()
{
- if (mDeviceModule)
+ if (mTuningDeviceModule)
{
- mDeviceModule->Terminate();
+ mTuningDeviceModule->Terminate();
}
- mDeviceModule = nullptr;
- mTaskQueueFactory = nullptr;
-
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->Terminate();
+ }
+ mTuningDeviceModule = nullptr;
+ mPeerDeviceModule = nullptr;
+ mTaskQueueFactory = nullptr;
});
mNetworkThread->BlockingCall(
[this]()
@@ -146,7 +152,6 @@ void LLWebRTCImpl::terminate()
});
}
-
void LLWebRTCImpl::refreshDevices()
{
mWorkerThread->PostTask([this]() { updateDevices(); });
@@ -169,32 +174,50 @@ void LLWebRTCImpl::setCaptureDevice(const std::string &id)
mWorkerThread->PostTask(
[this, id]()
{
- bool was_recording = mDeviceModule->Recording();
+ bool was_tuning_recording = mTuningDeviceModule->Recording();
- if (was_recording)
+ if (was_tuning_recording)
{
- mDeviceModule->StopRecording();
+ mTuningDeviceModule->StopRecording();
}
- int16_t captureDeviceCount = mDeviceModule->RecordingDevices();
- int16_t index = 0; /* default to first one if no match */
+ bool was_peer_recording = false;
+ if (mPeerDeviceModule)
+ {
+ was_peer_recording = mPeerDeviceModule->Recording();
+ if (was_peer_recording)
+ {
+ mPeerDeviceModule->StopRecording();
+ }
+ }
+ int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
for (int16_t i = 0; i < captureDeviceCount; i++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
- mDeviceModule->RecordingDeviceName(i, name, guid);
- if (id == guid || id == "Default")
+ mTuningDeviceModule->RecordingDeviceName(i, name, guid);
+ if (id == guid || id == "Default") // first one in list is default
{
RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
- index = i;
+ mRecordingDevice = i;
break;
}
}
- mDeviceModule->SetRecordingDevice(index);
- mDeviceModule->InitMicrophone();
- mDeviceModule->InitRecording();
- if (was_recording)
+ mTuningDeviceModule->SetRecordingDevice(mRecordingDevice);
+ mTuningDeviceModule->InitMicrophone();
+ mTuningDeviceModule->InitRecording();
+ if (was_tuning_recording)
{
- mDeviceModule->StartRecording();
+ mTuningDeviceModule->StartRecording();
+ }
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
+ mPeerDeviceModule->InitMicrophone();
+ mPeerDeviceModule->InitRecording();
+ if (was_peer_recording)
+ {
+ mPeerDeviceModule->StartRecording();
+ }
}
});
}
@@ -204,54 +227,72 @@ void LLWebRTCImpl::setRenderDevice(const std::string &id)
mWorkerThread->PostTask(
[this, id]()
{
- bool was_playing = mDeviceModule->Playing();
- if (was_playing)
+ bool was_tuning_playing = mTuningDeviceModule->Playing();
+ if (was_tuning_playing)
{
- mDeviceModule->StopPlayout();
+ mTuningDeviceModule->StopPlayout();
+ }
+ bool was_peer_playing = false;
+ if (mPeerDeviceModule)
+ {
+ was_peer_playing = mPeerDeviceModule->Playing();
+ if (was_tuning_playing)
+ {
+ mPeerDeviceModule->StopPlayout();
+ }
}
- int16_t renderDeviceCount = mDeviceModule->PlayoutDevices();
- int16_t index = 0; /* default to first one if no match */
+ int16_t renderDeviceCount = mTuningDeviceModule->PlayoutDevices();
for (int16_t i = 0; i < renderDeviceCount; i++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
- mDeviceModule->PlayoutDeviceName(i, name, guid);
+ mTuningDeviceModule->PlayoutDeviceName(i, name, guid);
if (id == guid || id == "Default")
{
RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
- index = i;
+ mPlayoutDevice = i;
break;
}
}
- mDeviceModule->SetPlayoutDevice(index);
- mDeviceModule->InitSpeaker();
- mDeviceModule->InitPlayout();
- if (was_playing)
+ mTuningDeviceModule->SetPlayoutDevice(mPlayoutDevice);
+ mTuningDeviceModule->InitSpeaker();
+ mTuningDeviceModule->InitPlayout();
+ if (was_tuning_playing)
{
- mDeviceModule->StartPlayout();
+ mTuningDeviceModule->StartPlayout();
+ }
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
+ mPeerDeviceModule->InitSpeaker();
+ mPeerDeviceModule->InitPlayout();
+ if (was_peer_playing)
+ {
+ mPeerDeviceModule->StartPlayout();
+ }
}
});
}
void LLWebRTCImpl::updateDevices()
{
- int16_t renderDeviceCount = mDeviceModule->PlayoutDevices();
+ int16_t renderDeviceCount = mTuningDeviceModule->PlayoutDevices();
LLWebRTCVoiceDeviceList renderDeviceList;
for (int16_t index = 0; index < renderDeviceCount; index++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
- mDeviceModule->PlayoutDeviceName(index, name, guid);
+ mTuningDeviceModule->PlayoutDeviceName(index, name, guid);
renderDeviceList.emplace_back(name, guid);
}
- int16_t captureDeviceCount = mDeviceModule->RecordingDevices();
+ int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
LLWebRTCVoiceDeviceList captureDeviceList;
for (int16_t index = 0; index < captureDeviceCount; index++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
- mDeviceModule->RecordingDeviceName(index, name, guid);
+ mTuningDeviceModule->RecordingDeviceName(index, name, guid);
captureDeviceList.emplace_back(name, guid);
}
for (auto &observer : mVoiceDevicesObserverList)
@@ -267,14 +308,22 @@ void LLWebRTCImpl::setTuningMode(bool enable)
{
if (enable)
{
- mDeviceModule->InitMicrophone();
- mDeviceModule->InitRecording();
- mDeviceModule->StartRecording();
- mDeviceModule->SetMicrophoneMute(false);
+ mTuningDeviceModule->StartRecording();
+ mTuningDeviceModule->SetMicrophoneMute(false);
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->StopRecording();
+ mPeerDeviceModule->StopPlayout();
+ }
}
else
{
- mDeviceModule->StopRecording();
+ mTuningDeviceModule->StopRecording();
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->StartPlayout();
+ mPeerDeviceModule->StartRecording();
+ }
}
});
}
@@ -295,42 +344,58 @@ void LLWebRTCImpl::unsetSignalingObserver(LLWebRTCSignalingObserver *observer)
}
}
-
bool LLWebRTCImpl::initializeConnection()
{
RTC_DCHECK(!mPeerConnection);
RTC_DCHECK(mPeerConnectionFactory);
- mAnswerReceived = false;
+ mAnswerReceived = false;
mSignalingThread->PostTask([this]() { initializeConnectionThreaded(); });
return true;
}
-
-
bool LLWebRTCImpl::initializeConnectionThreaded()
{
rtc::scoped_refptr<webrtc::AudioProcessing> apm = webrtc::AudioProcessingBuilder().Create();
- webrtc::AudioProcessing::Config apm_config;
- apm_config.echo_canceller.enabled = false;
- apm_config.echo_canceller.mobile_mode = false;
- apm_config.gain_controller1.enabled = true;
- apm_config.gain_controller1.mode =
- webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
- apm_config.gain_controller2.enabled = true;
- apm_config.high_pass_filter.enabled = true;
- apm_config.noise_suppression.enabled = true;
- apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kVeryHigh;
- apm_config.transient_suppression.enabled = true;
- apm_config.pipeline.multi_channel_render = true;
+ webrtc::AudioProcessing::Config apm_config;
+ apm_config.echo_canceller.enabled = false;
+ apm_config.echo_canceller.mobile_mode = false;
+ apm_config.gain_controller1.enabled = true;
+ apm_config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+ apm_config.gain_controller2.enabled = true;
+ apm_config.high_pass_filter.enabled = true;
+ apm_config.noise_suppression.enabled = true;
+ apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kVeryHigh;
+ apm_config.transient_suppression.enabled = true;
+ apm_config.pipeline.multi_channel_render = true;
apm_config.pipeline.multi_channel_capture = true;
//
apm->ApplyConfig(apm_config);
+ mWorkerThread->BlockingCall(
+ [this]()
+ {
+ mPeerAudioDeviceObserver = new LLAudioDeviceObserver;
+ mPeerDeviceModule =
+ webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
+ mTaskQueueFactory.get(),
+ std::unique_ptr<webrtc::AudioDeviceDataObserver>(mPeerAudioDeviceObserver));
+ mPeerDeviceModule->Init();
+ mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
+ mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
+ mPeerDeviceModule->SetStereoRecording(false);
+ mPeerDeviceModule->SetStereoPlayout(true);
+ mPeerDeviceModule->EnableBuiltInAEC(false);
+ mPeerDeviceModule->InitMicrophone();
+ mPeerDeviceModule->InitSpeaker();
+ mPeerDeviceModule->InitRecording();
+ mPeerDeviceModule->InitPlayout();
+ });
+
mPeerConnectionFactory = webrtc::CreatePeerConnectionFactory(mNetworkThread.get(),
mWorkerThread.get(),
mSignalingThread.get(),
- mDeviceModule,
+ mPeerDeviceModule,
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
nullptr /* video_encoder_factory */,
@@ -406,9 +471,9 @@ bool LLWebRTCImpl::initializeConnectionThreaded()
}
auto receivers = mPeerConnection->GetReceivers();
- for (auto& receiver : receivers)
+ for (auto &receiver : receivers)
{
- webrtc::RtpParameters params;
+ webrtc::RtpParameters params;
webrtc::RtpCodecParameters codecparam;
codecparam.name = "opus";
codecparam.kind = cricket::MEDIA_TYPE_AUDIO;
@@ -423,19 +488,23 @@ bool LLWebRTCImpl::initializeConnectionThreaded()
mPeerConnection->CreateOffer(this, offerOptions);
RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
-
+
return true;
}
void LLWebRTCImpl::shutdownConnection()
{
+ mDataChannel->Close();
+ mDataChannel = nullptr;
+ mPeerConnection->Close();
+ mPeerDeviceModule->Terminate();
+ mPeerDeviceModule = nullptr;
mPeerConnection = nullptr;
mPeerConnectionFactory = nullptr;
}
void LLWebRTCImpl::AnswerAvailable(const std::string &sdp)
{
-
RTC_LOG(LS_INFO) << __FUNCTION__ << " Remote SDP: " << sdp;
mSignalingThread->PostTask(
@@ -450,7 +519,7 @@ void LLWebRTCImpl::AnswerAvailable(const std::string &sdp)
for (auto &candidate : mCachedIceCandidates)
{
LLWebRTCIceCandidate ice_candidate;
- ice_candidate.candidate = candidate->candidate().ToString();
+ ice_candidate.candidate = candidate->candidate().ToString();
ice_candidate.mline_index = candidate->sdp_mline_index();
ice_candidate.sdp_mid = candidate->sdp_mid();
observer->OnIceCandidate(ice_candidate);
@@ -470,14 +539,13 @@ void LLWebRTCImpl::AnswerAvailable(const std::string &sdp)
void LLWebRTCImpl::setMute(bool mute)
{
mSignalingThread->PostTask(
- [this,mute]()
+ [this, mute]()
{
auto senders = mPeerConnection->GetSenders();
- RTC_LOG(LS_INFO) << __FUNCTION__ << (mute ? "disabling" : "enabling") << " streams count "
- << senders.size();
+ RTC_LOG(LS_INFO) << __FUNCTION__ << (mute ? "disabling" : "enabling") << " streams count " << senders.size();
- for (auto& sender : senders)
+ for (auto &sender : senders)
{
sender->track()->set_enabled(!mute);
}
@@ -497,19 +565,15 @@ void LLWebRTCImpl::setSpeakerVolume(float volume)
webrtc::MediaStreamTrackInterface *track = receiver->track().get();
if (track->kind() == webrtc::MediaStreamTrackInterface::kAudioKind)
{
- webrtc::AudioTrackInterface* audio_track = static_cast<webrtc::AudioTrackInterface*>(track);
- webrtc::AudioSourceInterface* source = audio_track->GetSource();
+ webrtc::AudioTrackInterface *audio_track = static_cast<webrtc::AudioTrackInterface *>(track);
+ webrtc::AudioSourceInterface *source = audio_track->GetSource();
source->SetVolume(VOLUME_SCALE_WEBRTC * volume);
-
}
}
});
}
-double LLWebRTCImpl::getAudioLevel()
-{
- return 20*mAudioDeviceObserver->getMicrophoneEnergy();
-}
+float LLWebRTCImpl::getTuningAudioLevel() { return 20 * mTuningAudioDeviceObserver->getMicrophoneEnergy(); }
//
// PeerConnectionObserver implementation.
@@ -519,7 +583,7 @@ void LLWebRTCImpl::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>
const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> &streams)
{
RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
- webrtc::RtpParameters params;
+ webrtc::RtpParameters params;
webrtc::RtpCodecParameters codecparam;
codecparam.name = "opus";
codecparam.kind = cricket::MEDIA_TYPE_AUDIO;
@@ -539,10 +603,9 @@ void LLWebRTCImpl::OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface
void LLWebRTCImpl::OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> channel)
{
mDataChannel = channel;
- channel->RegisterObserver(this);
+ channel->RegisterObserver(this);
}
-
void LLWebRTCImpl::OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state)
{
LLWebRTCSignalingObserver::IceGatheringState webrtc_new_state = LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_NEW;
@@ -584,8 +647,8 @@ void LLWebRTCImpl::OnConnectionChange(webrtc::PeerConnectionInterface::PeerConne
if (new_state == webrtc::PeerConnectionInterface::PeerConnectionState::kConnected)
{
mWorkerThread->PostTask([this]() {
- mDeviceModule->StartRecording();
- mDeviceModule->StartPlayout();
+ mPeerDeviceModule->StartRecording();
+ mPeerDeviceModule->StartPlayout();
for (auto &observer : mSignalingObserverList)
{
observer->OnAudioEstablished(this);
@@ -595,6 +658,7 @@ void LLWebRTCImpl::OnConnectionChange(webrtc::PeerConnectionInterface::PeerConne
break;
}
case webrtc::PeerConnectionInterface::PeerConnectionState::kFailed:
+ case webrtc::PeerConnectionInterface::PeerConnectionState::kDisconnected:
{
for (auto &observer : mSignalingObserverList)
{
@@ -738,6 +802,8 @@ void LLWebRTCImpl::unsetAudioObserver(LLWebRTCAudioObserver *observer)
}
}
+float LLWebRTCImpl::getAudioLevel() { return 20 * mPeerAudioDeviceObserver->getMicrophoneEnergy(); }
+
//
// DataChannelObserver implementation
//
diff --git a/indra/llwebrtc/llwebrtc.h b/indra/llwebrtc/llwebrtc.h
index 30c0d63c55..9224e88dfa 100644
--- a/indra/llwebrtc/llwebrtc.h
+++ b/indra/llwebrtc/llwebrtc.h
@@ -86,7 +86,7 @@ class LLWebRTCDeviceInterface
virtual void unsetDevicesObserver(LLWebRTCDevicesObserver *observer) = 0;
virtual void setTuningMode(bool enable) = 0;
- virtual double getAudioLevel() = 0;
+ virtual float getTuningAudioLevel() = 0;
};
class LLWebRTCAudioObserver
@@ -100,7 +100,8 @@ class LLWebRTCAudioInterface
virtual void setAudioObserver(LLWebRTCAudioObserver *observer) = 0;
virtual void unsetAudioObserver(LLWebRTCAudioObserver *observer) = 0;
virtual void setMute(bool mute) = 0;
- virtual void setSpeakerVolume(float volume) = 0; // volume between 0.0 and 1.0 = 0;
+ virtual void setSpeakerVolume(float volume) = 0; // volume between 0.0 and 1.0
+ virtual float getAudioLevel() = 0;
};
class LLWebRTCDataObserver
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h
index 70586f9013..947b417923 100644
--- a/indra/llwebrtc/llwebrtc_impl.h
+++ b/indra/llwebrtc/llwebrtc_impl.h
@@ -68,7 +68,7 @@ class LLAudioDeviceObserver : public webrtc::AudioDeviceDataObserver
public:
LLAudioDeviceObserver();
- double getMicrophoneEnergy();
+ float getMicrophoneEnergy();
void OnCaptureData(const void *audio_samples,
const size_t num_samples,
@@ -100,7 +100,7 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface,
{
public:
LLWebRTCImpl() :
- mAudioDeviceObserver(nullptr)
+ mTuningAudioDeviceObserver(nullptr), mPeerAudioDeviceObserver(nullptr)
{
}
~LLWebRTCImpl() {}
@@ -121,7 +121,7 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface,
void setRenderDevice(const std::string& id) override;
void setTuningMode(bool enable) override;
- double getAudioLevel() override;
+ float getTuningAudioLevel() override;
//
// LLWebRTCSignalInterface
@@ -141,6 +141,7 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface,
void unsetAudioObserver(LLWebRTCAudioObserver *observer) override;
void setMute(bool mute) override;
void setSpeakerVolume(float folume) override; // range 0.0-1.0
+ float getAudioLevel() override;
//
// LLWebRTCDataInterface
@@ -201,10 +202,16 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface,
// Devices
void updateDevices();
- rtc::scoped_refptr<webrtc::AudioDeviceModule> mDeviceModule;
+ rtc::scoped_refptr<webrtc::AudioDeviceModule> mTuningDeviceModule;
+ rtc::scoped_refptr<webrtc::AudioDeviceModule> mPeerDeviceModule;
std::vector<LLWebRTCDevicesObserver *> mVoiceDevicesObserverList;
- LLAudioDeviceObserver * mAudioDeviceObserver;
+ // accessors in webrtc aren't apparently implemented yet.
+ int32_t mPlayoutDevice;
+ int32_t mRecordingDevice;
+
+ LLAudioDeviceObserver * mTuningAudioDeviceObserver;
+ LLAudioDeviceObserver * mPeerAudioDeviceObserver;
// signaling
std::vector<LLWebRTCSignalingObserver *> mSignalingObserverList;
--
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