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Merge pull request #118 from secondlife/roxie/webrtc-voice 

Various WebRTC Fixes and Improvements
meow-7.2.2
Roxanne Skelly 2024-03-15 11:12:41 -07:00 committed by GitHub
parent b32364ba49 dbbbbc55af
commit 23ecf64b24
No known key found for this signature in database
GPG Key ID: B5690EEEBB952194
8 changed files with 215 additions and 179 deletions
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319
indra/llwebrtc/llwebrtc.cpp
View File

@ -84,10 +84,7 @@ void LLAudioDeviceObserver::OnRenderData(const void *audio_samples,
{
}
LLCustomProcessor::LLCustomProcessor() :
mSampleRateHz(0),
mNumChannels(0),
mMicrophoneEnergy(0.0)
LLCustomProcessor::LLCustomProcessor() : mSampleRateHz(0), mNumChannels(0), mMicrophoneEnergy(0.0)
{
memset(mSumVector, 0, sizeof(mSumVector));
}
@ -95,7 +92,7 @@ LLCustomProcessor::LLCustomProcessor() :
void LLCustomProcessor::Initialize(int sample_rate_hz, int num_channels)
{
mSampleRateHz = sample_rate_hz;
mNumChannels = num_channels;
mNumChannels = num_channels;
memset(mSumVector, 0, sizeof(mSumVector));
}
@ -105,7 +102,7 @@ void LLCustomProcessor::Process(webrtc::AudioBuffer *audio_in)
stream_config.set_sample_rate_hz(mSampleRateHz);
stream_config.set_num_channels(mNumChannels);
std::vector<float *> frame;
std::vector<float> frame_samples;
std::vector<float> frame_samples;
if (audio_in->num_channels() < 1 || audio_in->num_frames() < 480)
{
@ -123,7 +120,7 @@ void LLCustomProcessor::Process(webrtc::AudioBuffer *audio_in)
audio_in->CopyTo(stream_config, &frame[0]);
// calculate the energy
float energy = 0;
float energy = 0;
for (size_t index = 0; index < stream_config.num_samples(); index++)
{
float sample = frame_samples[index];
@ -151,6 +148,7 @@ void LLCustomProcessor::Process(webrtc::AudioBuffer *audio_in)
LLWebRTCImpl::LLWebRTCImpl() :
mPeerCustomProcessor(nullptr),
mMute(true),
mTuningMode(false),
mPlayoutDevice(0),
mRecordingDevice(0),
mTuningAudioDeviceObserver(nullptr)
@ -160,7 +158,7 @@ LLWebRTCImpl::LLWebRTCImpl() :
void LLWebRTCImpl::init()
{
RTC_DCHECK(mPeerConnectionFactory);
mPlayoutDevice = 0;
mPlayoutDevice = 0;
mRecordingDevice = 0;
rtc::InitializeSSL();
@ -183,43 +181,41 @@ void LLWebRTCImpl::init()
mTuningAudioDeviceObserver = new LLAudioDeviceObserver;
mWorkerThread->PostTask(
[this]()
{
// Initialize the audio devices on the Worker Thread
mTuningDeviceModule = webrtc::CreateAudioDeviceWithDataObserver(
webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
mTaskQueueFactory.get(),
std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
[this]()
{
// Initialize the audio devices on the Worker Thread
mTuningDeviceModule =
webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
mTaskQueueFactory.get(),
std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
mTuningDeviceModule->Init();
mTuningDeviceModule->SetStereoRecording(true);
mTuningDeviceModule->SetStereoPlayout(true);
mTuningDeviceModule->EnableBuiltInAEC(false);
mTuningDeviceModule->SetAudioDeviceSink(this);
updateDevices();
});
mTuningDeviceModule->Init();
mTuningDeviceModule->SetStereoRecording(true);
mTuningDeviceModule->SetStereoPlayout(true);
mTuningDeviceModule->EnableBuiltInAEC(false);
mTuningDeviceModule->SetAudioDeviceSink(this);
updateDevices();
});
mWorkerThread->BlockingCall(
[this]()
{
// the peer device module doesn't need an observer
// as we pull peer data after audio processing.
mPeerDeviceModule =
webrtc::CreateAudioDeviceWithDataObserver(
webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
mTaskQueueFactory.get(),
nullptr);
mPeerDeviceModule->Init();
mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
mPeerDeviceModule->SetStereoRecording(true);
mPeerDeviceModule->SetStereoPlayout(true);
mPeerDeviceModule->EnableBuiltInAEC(false);
mPeerDeviceModule->InitMicrophone();
mPeerDeviceModule->InitSpeaker();
mPeerDeviceModule->InitRecording();
mPeerDeviceModule->InitPlayout();
});
[this]()
{
// the peer device module doesn't need an observer
// as we pull peer data after audio processing.
mPeerDeviceModule = webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
mTaskQueueFactory.get(),
nullptr);
mPeerDeviceModule->Init();
mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
mPeerDeviceModule->SetStereoRecording(true);
mPeerDeviceModule->SetStereoPlayout(true);
mPeerDeviceModule->EnableBuiltInAEC(false);
mPeerDeviceModule->InitMicrophone();
mPeerDeviceModule->InitSpeaker();
mPeerDeviceModule->InitRecording();
mPeerDeviceModule->InitPlayout();
});
// The custom processor allows us to retrieve audio data (and levels)
// from after other audio processing such as AEC, AGC, etc.
@ -230,7 +226,7 @@ void LLWebRTCImpl::init()
// TODO: wire some of these to the primary interface and ultimately
// to the UI to allow user config.
webrtc::AudioProcessing::Config apm_config;
webrtc::AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = false;
apm_config.gain_controller1.enabled = true;
@ -268,12 +264,7 @@ void LLWebRTCImpl::init()
nullptr /* audio_mixer */,
apm);
mWorkerThread->BlockingCall(
[this]()
{
mPeerDeviceModule->StartPlayout();
});
mWorkerThread->BlockingCall([this]() { mPeerDeviceModule->StartPlayout(); });
}
void LLWebRTCImpl::terminate()
@ -337,128 +328,115 @@ void LLWebRTCImpl::setDevicesObserver(LLWebRTCDevicesObserver *observer) { mVoic
void LLWebRTCImpl::unsetDevicesObserver(LLWebRTCDevicesObserver *observer)
{
std::vector<LLWebRTCDevicesObserver *>::iterator it =
std::find(mVoiceDevicesObserverList.begin(), mVoiceDevicesObserverList.end(), observer);
std::find(mVoiceDevicesObserverList.begin(), mVoiceDevicesObserverList.end(), observer);
if (it != mVoiceDevicesObserverList.end())
{
mVoiceDevicesObserverList.erase(it);
}
}
// TODO: There's potential for shared code here as the patterns
// are similar.
static int16_t ll_get_device_module_capture_device(rtc::scoped_refptr<webrtc::AudioDeviceModule> device_module, const std::string &id)
{
int16_t recordingDevice = 0;
int16_t captureDeviceCount = device_module->RecordingDevices();
for (int16_t i = 0; i < captureDeviceCount; i++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
device_module->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;
recordingDevice = i;
break;
}
}
return recordingDevice;
}
void ll_set_device_module_capture_device(rtc::scoped_refptr<webrtc::AudioDeviceModule> device_module, int16_t device)
{
device_module->StopRecording();
device_module->SetRecordingDevice(device);
device_module->InitMicrophone();
device_module->SetStereoRecording(false);
device_module->InitRecording();
device_module->StartRecording();
}
void LLWebRTCImpl::setCaptureDevice(const std::string &id)
{
mWorkerThread->PostTask(
[this, id]()
{
int16_t tuningRecordingDevice = 0;
int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
for (int16_t i = 0; i < captureDeviceCount; i++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
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;
tuningRecordingDevice = i;
break;
}
}
mTuningDeviceModule->StopRecording();
mTuningDeviceModule->SetRecordingDevice(tuningRecordingDevice);
mTuningDeviceModule->InitMicrophone();
mTuningDeviceModule->InitRecording();
mTuningDeviceModule->StartRecording();
if (mPeerDeviceModule)
{
int16_t captureDeviceCount = mPeerDeviceModule->RecordingDevices();
for (int16_t i = 0; i < captureDeviceCount; i++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
mPeerDeviceModule->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;
mRecordingDevice = i;
break;
}
}
bool was_peer_recording = mPeerDeviceModule->Recording();
if (was_peer_recording)
{
mPeerDeviceModule->StopRecording();
}
mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
mPeerDeviceModule->InitMicrophone();
mPeerDeviceModule->InitRecording();
if (was_peer_recording)
{
mPeerDeviceModule->StartRecording();
}
}
});
[this, id]()
{
int16_t recordingDevice = ll_get_device_module_capture_device(mTuningDeviceModule, id);
if (recordingDevice != mRecordingDevice)
{
mRecordingDevice = recordingDevice;
if (mTuningMode)
{
ll_set_device_module_capture_device(mTuningDeviceModule, recordingDevice);
}
else
{
ll_set_device_module_capture_device(mPeerDeviceModule, recordingDevice);
}
}
});
}
static int16_t ll_get_device_module_render_device(
rtc::scoped_refptr<webrtc::AudioDeviceModule> device_module,
const std::string &id)
{
int16_t playoutDevice = 0;
int16_t playoutDeviceCount = device_module->PlayoutDevices();
for (int16_t i = 0; i < playoutDeviceCount; i++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
device_module->PlayoutDeviceName(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;
playoutDevice = i;
break;
}
}
return playoutDevice;
}
void ll_set_device_module_render_device(rtc::scoped_refptr<webrtc::AudioDeviceModule> device_module, int16_t device)
{
device_module->StopPlayout();
device_module->SetPlayoutDevice(device);
device_module->InitSpeaker();
device_module->SetStereoPlayout(false);
device_module->InitPlayout();
device_module->StartPlayout();
}
void LLWebRTCImpl::setRenderDevice(const std::string &id)
{
mWorkerThread->PostTask(
[this, id]()
{
int16_t renderDeviceCount = mTuningDeviceModule->PlayoutDevices();
int16_t tuningPlayoutDevice = 0;
for (int16_t i = 0; i < renderDeviceCount; i++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
mTuningDeviceModule->PlayoutDeviceName(i, name, guid);
if (id == guid || id == "Default")
{
RTC_LOG(LS_INFO) << __FUNCTION__ << "Set playout device to " << name << " " << guid << " " << i;
tuningPlayoutDevice = i;
break;
}
}
bool was_tuning_playing = mTuningDeviceModule->Playing();
if (was_tuning_playing)
{
mTuningDeviceModule->StopPlayout();
}
mTuningDeviceModule->SetPlayoutDevice(tuningPlayoutDevice);
mTuningDeviceModule->InitSpeaker();
mTuningDeviceModule->InitPlayout();
if (was_tuning_playing)
{
mTuningDeviceModule->StartPlayout();
}
if (mPeerDeviceModule)
{
renderDeviceCount = mPeerDeviceModule->PlayoutDevices();
for (int16_t i = 0; i < renderDeviceCount; i++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
mPeerDeviceModule->PlayoutDeviceName(i, name, guid);
if (id == guid || id == "Default")
{
RTC_LOG(LS_INFO)
<< __FUNCTION__ << "Set playout device to " << name << " " << guid << " " << i;
mPlayoutDevice = i;
break;
}
}
mPeerDeviceModule->StopPlayout();
mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
mPeerDeviceModule->InitSpeaker();
mPeerDeviceModule->InitPlayout();
mPeerDeviceModule->StartPlayout();
}
});
[this, id]()
{
int16_t playoutDevice = ll_get_device_module_render_device(mTuningDeviceModule, id);
if (playoutDevice != mPlayoutDevice)
{
mPlayoutDevice = playoutDevice;
if (mTuningMode)
{
ll_set_device_module_render_device(mTuningDeviceModule, playoutDevice);
}
else
{
ll_set_device_module_render_device(mPeerDeviceModule, playoutDevice);
}
}
});
}
// updateDevices needs to happen on the worker thread.
@ -473,7 +451,7 @@ void LLWebRTCImpl::updateDevices()
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
mTuningDeviceModule->PlayoutDeviceName(index, name, guid);
renderDeviceList.emplace_back(name, guid, index == currentRenderDeviceIndex);
renderDeviceList.emplace_back(name, guid);
}
int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
@ -485,7 +463,7 @@ void LLWebRTCImpl::updateDevices()
char name[webrtc::kAdmMaxDeviceNameSize];
char guid[webrtc::kAdmMaxGuidSize];
mTuningDeviceModule->RecordingDeviceName(index, name, guid);
captureDeviceList.emplace_back(name, guid, index == currentCaptureDeviceIndex);
captureDeviceList.emplace_back(name, guid);
}
for (auto &observer : mVoiceDevicesObserverList)
{
@ -502,6 +480,29 @@ void LLWebRTCImpl::OnDevicesUpdated()
void LLWebRTCImpl::setTuningMode(bool enable)
{
mTuningMode = enable;
mWorkerThread->PostTask(
[this, enable] {
if (enable)
{
mPeerDeviceModule->StopRecording();
mPeerDeviceModule->StopPlayout();
ll_set_device_module_render_device(mTuningDeviceModule, mPlayoutDevice);
ll_set_device_module_capture_device(mTuningDeviceModule, mRecordingDevice);
mTuningDeviceModule->StartRecording();
mTuningDeviceModule->StartPlayout();
}
else
{
mTuningDeviceModule->StopRecording();
mTuningDeviceModule->StopPlayout();
ll_set_device_module_render_device(mPeerDeviceModule, mPlayoutDevice);
ll_set_device_module_capture_device(mPeerDeviceModule, mRecordingDevice);
mPeerDeviceModule->StartRecording();
mPeerDeviceModule->StartPlayout();
}
}
);
mSignalingThread->PostTask(
[this, enable]
{

6
indra/llwebrtc/llwebrtc.h
View File

@ -74,12 +74,10 @@ class LLWebRTCVoiceDevice
public:
std::string mDisplayName; // friendly name for user interface purposes
std::string mID; // internal value for selection
bool mCurrent; // current device
LLWebRTCVoiceDevice(const std::string &display_name, const std::string &id, bool current) :
LLWebRTCVoiceDevice(const std::string &display_name, const std::string &id) :
mDisplayName(display_name),
mID(id),
mCurrent(current)
mID(id)
{};
};

8
indra/llwebrtc/llwebrtc_impl.h
View File

@ -40,8 +40,11 @@
#include "llwebrtc.h"
// WebRTC Includes
#ifdef WEBRTC_WIN
#pragma warning(disable : 4996)
#pragma warning(disable : 4068)
#pragma warning(disable : 4996) // ignore 'deprecated.' We don't use the functions marked
// deprecated in the webrtc headers, but msvc complains anyway.
// Clang doesn't, and that's generally what webrtc uses.
#pragma warning(disable : 4068) // ignore 'invalid pragma.' There are clang pragma's in
// the webrtc headers, which msvc doesn't recognize.
#endif // WEBRTC_WIN
#include "api/scoped_refptr.h"
@ -236,6 +239,7 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS
std::vector<LLWebRTCDevicesObserver *> mVoiceDevicesObserverList;
// accessors in native webrtc for devices aren't apparently implemented yet.
bool mTuningMode;
int32_t mPlayoutDevice;
int32_t mRecordingDevice;
bool mMute;

5
indra/newview/llimview.cpp
View File

@ -87,7 +87,10 @@ const S32 XL8_PADDING = 3; // XL8_START_TAG.size() + XL8_END_TAG.size()
/** Timeout of outgoing session initialization (in seconds) */
const static U32 SESSION_INITIALIZATION_TIMEOUT = 30;
// This enum corresponds to the sim's and adds P2P_CHAT_SESSION,
// as webrtc uses the multiagent chat mechanism for p2p calls,
// instead of relying on vivox calling.
// Don't change this without consulting a server developer.
enum EMultiAgentChatSessionType
{
GROUP_CHAT_SESSION = 0,

5
indra/newview/llvoicechannel.cpp
View File

@ -418,7 +418,7 @@ void LLVoiceChannelGroup::activate()
{
// we have the channel info, just need to use it now
LLVoiceClient::getInstance()->setNonSpatialChannel(mChannelInfo,
mCallDirection == OUTGOING_CALL,
mIsP2P && (mCallDirection == OUTGOING_CALL),
mIsP2P);
if (mIsP2P)
@ -732,7 +732,7 @@ void LLVoiceChannelProximal::handleError(EStatusType status)
LLNotificationsUtil::add(notify, mNotifyArgs);
}
LLVoiceChannel::handleError(status);
// proximal voice remains up and the provider will try to reconnect.
}
void LLVoiceChannelProximal::deactivate()
@ -741,6 +741,7 @@ void LLVoiceChannelProximal::deactivate()
{
setState(STATE_HUNG_UP);
}
LLVoiceClient::getInstance()->activateSpatialChannel(false);
}

4
indra/newview/llvoiceclient.cpp
View File

@ -382,14 +382,14 @@ void LLVoiceClient::refreshDeviceLists(bool clearCurrentList)
void LLVoiceClient::setCaptureDevice(const std::string& name)
{
LLVivoxVoiceClient::getInstance()->setCaptureDevice(name);
LLWebRTCVoiceClient::getInstance()->setCaptureDevice(name);
LLVivoxVoiceClient::getInstance()->setCaptureDevice(name);
}
void LLVoiceClient::setRenderDevice(const std::string& name)
{
LLVivoxVoiceClient::getInstance()->setRenderDevice(name);
LLWebRTCVoiceClient::getInstance()->setRenderDevice(name);
LLVivoxVoiceClient::getInstance()->setRenderDevice(name);
}
const LLVoiceDeviceList& LLVoiceClient::getCaptureDevices()

41
indra/newview/llvoicewebrtc.cpp
View File

@ -92,6 +92,7 @@ namespace {
// Don't send positional updates more frequently than this:
const F32 UPDATE_THROTTLE_SECONDS = 0.1f;
const F32 MAX_RETRY_WAIT_SECONDS = 10.0f;
// Cosine of a "trivially" small angle
const F32 FOUR_DEGREES = 4.0f * (F_PI / 180.0f);
@ -590,14 +591,15 @@ void LLWebRTCVoiceClient::OnDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceLi
std::string inputDevice = gSavedSettings.getString("VoiceInputAudioDevice");
std::string outputDevice = gSavedSettings.getString("VoiceOutputAudioDevice");
LL_DEBUGS("Voice") << "Setting devices to-input: '" << inputDevice << "' output: '" << outputDevice << "'" << LL_ENDL;
clearRenderDevices();
bool renderDeviceSet = false;
for (auto &device : render_devices)
{
addRenderDevice(LLVoiceDevice(device.mDisplayName, device.mID));
if (device.mCurrent && outputDevice == device.mID)
LL_DEBUGS("Voice") << "Checking render device" << "'" << device.mID << "'" << LL_ENDL;
if (outputDevice == device.mID)
{
setRenderDevice(outputDevice);
renderDeviceSet = true;
}
}
@ -610,10 +612,11 @@ void LLWebRTCVoiceClient::OnDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceLi
bool captureDeviceSet = false;
for (auto &device : capture_devices)
{
LL_DEBUGS("Voice") << "Checking capture device:'" << device.mID << "'" << LL_ENDL;
addCaptureDevice(LLVoiceDevice(device.mDisplayName, device.mID));
if (device.mCurrent && inputDevice == device.mID)
if (inputDevice == device.mID)
{
setCaptureDevice(outputDevice);
captureDeviceSet = true;
}
}
@ -1988,8 +1991,13 @@ LLVoiceWebRTCConnection::LLVoiceWebRTCConnection(const LLUUID &regionID, const s
mMicGain(0.0),
mOutstandingRequests(0),
mChannelID(channelID),
mRegionID(regionID)
mRegionID(regionID),
mRetryWaitPeriod(0)
{
// retries wait a short period...randomize it so
// all clients don't try to reconnect at once.
mRetryWaitSecs = ((F32) rand() / (RAND_MAX)) + 0.5;
mWebRTCPeerConnectionInterface = llwebrtc::newPeerConnection();
mWebRTCPeerConnectionInterface->setSignalingObserver(this);
}
@ -2404,7 +2412,7 @@ bool LLVoiceWebRTCSpatialConnection::requestVoiceConnection()
{
body["parcel_local_id"] = mParcelLocalID;
}
body["channel_type"] = "local";
body["voice_server_type"] = WEBRTC_VOICE_SERVER_TYPE;
LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
@ -2567,6 +2575,9 @@ bool LLVoiceWebRTCConnection::connectionStateMachine()
case VOICE_STATE_SESSION_UP:
{
mRetryWaitPeriod = 0;
mRetryWaitSecs = ((F32) rand() / (RAND_MAX)) + 0.5;
// we'll stay here as long as the session remains up.
if (mShutDown)
{
@ -2576,9 +2587,21 @@ bool LLVoiceWebRTCConnection::connectionStateMachine()
}
case VOICE_STATE_SESSION_RETRY:
// something went wrong, so notify that the connection has failed.
LLWebRTCVoiceClient::getInstance()->OnConnectionFailure(mChannelID, mRegionID);
setVoiceConnectionState(VOICE_STATE_DISCONNECT);
// only retry ever 'n' seconds
if (mRetryWaitPeriod++ * UPDATE_THROTTLE_SECONDS > mRetryWaitSecs)
{
// something went wrong, so notify that the connection has failed.
LLWebRTCVoiceClient::getInstance()->OnConnectionFailure(mChannelID, mRegionID);
setVoiceConnectionState(VOICE_STATE_DISCONNECT);
mRetryWaitPeriod = 0;
if (mRetryWaitSecs < MAX_RETRY_WAIT_SECONDS)
{
// back off the retry period, and do it by a small random
// bit so all clients don't reconnect at once.
mRetryWaitSecs += ((F32) rand() / (RAND_MAX)) + 0.5;
mRetryWaitPeriod = 0;
}
}
break;
case VOICE_STATE_DISCONNECT:

6
indra/newview/llvoicewebrtc.h
View File

@ -694,6 +694,12 @@ class LLVoiceWebRTCConnection :
bool mShutDown;
S32 mOutstandingRequests;
S32 mRetryWaitPeriod; // number of UPDATE_THROTTLE_SECONDS we've
// waited since our last attempt to connect.
F32 mRetryWaitSecs; // number of seconds to wait before next retry
std::vector<llwebrtc::LLWebRTCIceCandidate> mIceCandidates;
bool mIceCompleted;
bool mTrickling;