summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
Diffstat
-rwxr-xr-xetc/message.xml11
-rw-r--r--indra/CMakeLists.txt1
-rw-r--r--indra/cmake/CMakeLists.txt1
-rw-r--r--indra/cmake/LLWebRTC.cmake1
-rw-r--r--indra/cmake/WebRTC.cmake43
-rw-r--r--indra/llwebrtc/CMakeLists.txt55
-rw-r--r--indra/llwebrtc/llwebrtc.cpp486
-rw-r--r--indra/llwebrtc/llwebrtc.h127
-rw-r--r--indra/llwebrtc/llwebrtc_impl.h177
-rw-r--r--indra/newview/CMakeLists.txt6
-rw-r--r--indra/newview/app_settings/settings.xml2
-rw-r--r--indra/newview/llvieweraudio.cpp4
-rwxr-xr-xindra/newview/llviewerregion.cpp1
-rw-r--r--indra/newview/llvoiceclient.cpp10
-rw-r--r--indra/newview/llvoiceclient.h1
-rw-r--r--indra/newview/llvoicewebrtc.cpp7299
-rw-r--r--indra/newview/llvoicewebrtc.h1102
-rwxr-xr-xscripts/messages/message_template.msg.sha12
18 files changed, 9322 insertions, 7 deletions
diff --git a/etc/message.xml b/etc/message.xml
index b444fe6c11..7d524b5eb8 100755
--- a/etc/message.xml
+++ b/etc/message.xml
@@ -506,6 +506,14 @@
<boolean>false</boolean>
</map>
+ <key>VoiceSignalingRequest</key>
+ <map>
+ <key>flavor</key>
+ <string>llsd</string>
+ <key>trusted-sender</key>
+ <boolean>false</boolean>
+ </map>
+
<!-- Server to client -->
<key>RequiredVoiceVersion</key>
<map>
@@ -688,6 +696,9 @@
<key>ProvisionVoiceAccountRequest</key>
<boolean>false</boolean>
+
+ <key>VoiceSignalingRequest</key>
+ <boolean>false</boolean>
<key>RemoteParcelRequest</key>
<boolean>false</boolean>
diff --git a/indra/CMakeLists.txt b/indra/CMakeLists.txt
index 500ffa3e8b..422927704a 100644
--- a/indra/CMakeLists.txt
+++ b/indra/CMakeLists.txt
@@ -54,6 +54,7 @@ add_subdirectory(${LIBS_OPEN_PREFIX}llmessage)
add_subdirectory(${LIBS_OPEN_PREFIX}llprimitive)
add_subdirectory(${LIBS_OPEN_PREFIX}llrender)
add_subdirectory(${LIBS_OPEN_PREFIX}llfilesystem)
+add_subdirectory(${LIBS_OPEN_PREFIX}llwebrtc)
add_subdirectory(${LIBS_OPEN_PREFIX}llwindow)
add_subdirectory(${LIBS_OPEN_PREFIX}llxml)
diff --git a/indra/cmake/CMakeLists.txt b/indra/cmake/CMakeLists.txt
index 1fd83eadff..8a77d0b882 100644
--- a/indra/cmake/CMakeLists.txt
+++ b/indra/cmake/CMakeLists.txt
@@ -63,6 +63,7 @@ set(cmake_SOURCE_FILES
ViewerMiscLibs.cmake
VisualLeakDetector.cmake
LibVLCPlugin.cmake
+ WebRTC.cmake
XmlRpcEpi.cmake
xxHash.cmake
ZLIBNG.cmake
diff --git a/indra/cmake/LLWebRTC.cmake b/indra/cmake/LLWebRTC.cmake
new file mode 100644
index 0000000000..913e28c2ff
--- /dev/null
+++ b/indra/cmake/LLWebRTC.cmake
@@ -0,0 +1 @@
+# -*- cmake -*- \ No newline at end of file
diff --git a/indra/cmake/WebRTC.cmake b/indra/cmake/WebRTC.cmake
new file mode 100644
index 0000000000..f8ce9c8104
--- /dev/null
+++ b/indra/cmake/WebRTC.cmake
@@ -0,0 +1,43 @@
+# -*- cmake -*-
+include(CMakeCopyIfDifferent)
+
+include(Linking)
+
+include_guard()
+
+set(WEBRTC_ROOT ${CMAKE_BINARY_DIR}/../../webrtc/src)
+file(COPY ${WEBRTC_ROOT}/out/Default/obj/webrtc.lib
+ DESTINATION ${CMAKE_BINARY_DIR}/packages/lib/release
+)
+set(WEBRTC_INCLUDE_DIR ${CMAKE_BINARY_DIR}/packages/include/webrtc)
+file(MAKE_DIRECTORY ${WEBRTC_INCLUDE_DIR})
+
+file(COPY ${WEBRTC_ROOT}/api
+ ${WEBRTC_ROOT}/media/base
+ ${WEBRTC_ROOT}/media/engine
+ ${WEBRTC_ROOT}/rtc_base
+ ${WEBRTC_ROOT}/pc
+ ${WEBRTC_ROOT}/p2p
+ ${WEBRTC_ROOT}/call
+ ${WEBRTC_ROOT}/media
+ ${WEBRTC_ROOT}/system_wrappers
+ ${WEBRTC_ROOT}/common_video
+ ${WEBRTC_ROOT}/video
+ ${WEBRTC_ROOT}/common_audio
+ ${WEBRTC_ROOT}/logging
+ ${WEBRTC_ROOT}/third_party/abseil-cpp/absl
+ DESTINATION ${WEBRTC_INCLUDE_DIR}
+ FILES_MATCHING PATTERN "*.h"
+)
+
+add_library(ll::webrtc STATIC IMPORTED)
+
+if (LINUX)
+ target_link_libraries( ll::webrtc INTERFACE ../webrtc/src/obj/Default/webrtc)
+elseif (DARWIN)
+ target_link_libraries( ll::webrtc INTERFACE ../webrtc/src/obj/Default/webrtc)
+elseif (WINDOWS)
+ set_target_properties( ll::webrtc PROPERTIES IMPORTED_LOCATION ${CMAKE_BINARY_DIR}/packages/lib/release/webrtc.lib)
+ target_link_libraries( ll::webrtc INTERFACE ${CMAKE_BINARY_DIR}/packages/lib/release/webrtc.lib)
+endif (LINUX)
+target_include_directories( ll::webrtc INTERFACE "${WEBRTC_INCLUDE_DIR}")
diff --git a/indra/llwebrtc/CMakeLists.txt b/indra/llwebrtc/CMakeLists.txt
new file mode 100644
index 0000000000..c0e2520a22
--- /dev/null
+++ b/indra/llwebrtc/CMakeLists.txt
@@ -0,0 +1,55 @@
+# -*- cmake -*-
+
+# some webrtc headers require C++ 20
+set(CMAKE_CXX_STANDARD 20)
+set(CMAKE_CXX_STANDARD_REQUIRED ON)
+
+
+set(CMAKE_GENERATOR_TOOLSET "clang_cl_x64")
+
+include(00-Common)
+include(Linking)
+include(WebRTC)
+
+project(llwebrtc)
+
+message(STATUS "C Compiler executable: ${CMAKE_C_COMPILER}")
+message(STATUS "CXX Compiler executable: ${CMAKE_CXX_COMPILER}")
+message(STATUS "Linker executable: ${CMAKE_LINKER}")
+message(STATUS "SharedLib: ${SHARED_LIB_STAGING_DIR}")
+
+set(CMAKE_EXE_LINKER_FLAGS "${CMAKE_EXE_LINKER_FLAGS} -stdlib=libc++ -lc++abi")
+
+set(llwebrtc_SOURCE_FILES
+ llwebrtc.cpp
+ )
+
+set(llwebrtc_HEADER_FILES
+ CMakeLists.txt
+ llwebrtc.h
+ llwebrtc_impl.h
+ )
+
+list(APPEND llwebrtc_SOURCE_FILES ${llwebrtc_HEADER_FILES})
+
+add_library (llwebrtc SHARED ${llwebrtc_SOURCE_FILES})
+
+set_target_properties(llwebrtc PROPERTIES PUBLIC_HEADER llwebrtc.h)
+
+target_link_libraries(llwebrtc PRIVATE ll::webrtc
+ secur32
+ winmm
+ dmoguids
+ wmcodecdspuuid
+ msdmo
+ strmiids
+ iphlpapi)
+target_include_directories( llwebrtc INTERFACE ${CMAKE_CURRENT_SOURCE_DIR})
+
+set_property(TARGET llwebrtc PROPERTY MSVC_RUNTIME_LIBRARY "MultiThreadedDebug")
+
+install(TARGETS llwebrtc RUNTIME DESTINATION "${CMAKE_BINARY_DIR}/sharedlibs/RelWithDebInfo")
+
+# Add tests
+if (LL_TESTS)
+endif (LL_TESTS)
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
new file mode 100644
index 0000000000..5e71a00b60
--- /dev/null
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -0,0 +1,486 @@
+/**
+ * @file llaccordionctrl.cpp
+ * @brief Accordion panel implementation
+ *
+ * $LicenseInfo:firstyear=2023&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2023, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+#include "llwebrtc_impl.h"
+#include <algorithm>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/media_stream_interface.h"
+#include "api/media_stream_track.h"
+
+namespace llwebrtc
+{
+
+void LLWebRTCImpl::init()
+{
+ mAnswerReceived = false;
+ rtc::InitializeSSL();
+ mTaskQueueFactory = webrtc::CreateDefaultTaskQueueFactory();
+
+ mNetworkThread = rtc::Thread::CreateWithSocketServer();
+ mNetworkThread->SetName("WebRTCNetworkThread", nullptr);
+ mNetworkThread->Start();
+ mWorkerThread = rtc::Thread::Create();
+ mWorkerThread->SetName("WebRTCWorkerThread", nullptr);
+ mWorkerThread->Start();
+ mSignalingThread = rtc::Thread::Create();
+ mSignalingThread->SetName("WebRTCSignalingThread", nullptr);
+ mSignalingThread->Start();
+
+ mSignalingThread->PostTask(
+ [this]()
+ {
+ mDeviceModule = webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
+ mTaskQueueFactory.get(),
+ std::unique_ptr<webrtc::AudioDeviceDataObserver>(this));
+ mDeviceModule->Init();
+ updateDevices();
+ });
+}
+
+void LLWebRTCImpl::refreshDevices()
+{
+ mSignalingThread->PostTask([this]() { updateDevices(); });
+}
+
+void LLWebRTCImpl::setDevicesObserver(LLWebRTCDevicesObserver *observer) { mVoiceDevicesObserverList.emplace_back(observer); }
+
+void LLWebRTCImpl::unsetDevicesObserver(LLWebRTCDevicesObserver *observer)
+{
+ std::vector<LLWebRTCDevicesObserver *>::iterator it =
+ std::find(mVoiceDevicesObserverList.begin(), mVoiceDevicesObserverList.end(), observer);
+ if (it != mVoiceDevicesObserverList.end())
+ {
+ mVoiceDevicesObserverList.erase(it);
+ }
+}
+
+void LLWebRTCImpl::setCaptureDevice(const std::string &id)
+{
+ mSignalingThread->PostTask(
+ [this, id]()
+ {
+ int16_t captureDeviceCount = mDeviceModule->RecordingDevices();
+ for (int16_t index = 0; index < captureDeviceCount; index++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mDeviceModule->RecordingDeviceName(index, name, guid);
+ if (id == guid || id == name)
+ {
+ mDeviceModule->SetRecordingDevice(index);
+ break;
+ }
+ }
+ });
+}
+
+void LLWebRTCImpl::setRenderDevice(const std::string &id)
+{
+ mSignalingThread->PostTask(
+ [this, id]()
+ {
+ int16_t renderDeviceCount = mDeviceModule->RecordingDevices();
+ for (int16_t index = 0; index < renderDeviceCount; index++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mDeviceModule->PlayoutDeviceName(index, name, guid);
+ if (id == guid || id == name)
+ {
+ mDeviceModule->SetPlayoutDevice(index);
+ break;
+ }
+ }
+ });
+}
+
+void LLWebRTCImpl::updateDevices()
+{
+ int16_t renderDeviceCount = mDeviceModule->PlayoutDevices();
+ LLWebRTCVoiceDeviceList renderDeviceList;
+ for (int16_t index = 0; index < renderDeviceCount; index++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mDeviceModule->PlayoutDeviceName(index, name, guid);
+ renderDeviceList.emplace_back(name, guid);
+ }
+ for (auto &observer : mVoiceDevicesObserverList)
+ {
+ observer->OnRenderDevicesChanged(renderDeviceList);
+ }
+
+ int16_t captureDeviceCount = mDeviceModule->RecordingDevices();
+ LLWebRTCVoiceDeviceList captureDeviceList;
+ for (int16_t index = 0; index < captureDeviceCount; index++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mDeviceModule->RecordingDeviceName(index, name, guid);
+ captureDeviceList.emplace_back(name, guid);
+ }
+ for (auto &observer : mVoiceDevicesObserverList)
+ {
+ observer->OnCaptureDevicesChanged(captureDeviceList);
+ }
+}
+
+void LLWebRTCImpl::setTuningMode(bool enable)
+{
+ mSignalingThread->PostTask(
+ [this, enable]()
+ {
+ if (enable)
+ {
+ mDeviceModule->InitMicrophone();
+ mDeviceModule->InitRecording();
+ mDeviceModule->StartRecording();
+ mDeviceModule->SetMicrophoneMute(false);
+ }
+ else
+ {
+ mDeviceModule->StopRecording();
+ }
+ });
+}
+
+double LLWebRTCImpl::getTuningMicrophoneEnergy() { return mTuningEnergy; }
+
+void LLWebRTCImpl::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)
+{
+ if (bytes_per_sample != 4)
+ {
+ return;
+ }
+
+ double energy = 0;
+ const short *samples = (const short *) audio_samples;
+ for (size_t index = 0; index < num_samples * num_channels; index++)
+ {
+ double sample = (static_cast<double>(samples[index]) / (double) 32768);
+ energy += sample * sample;
+ }
+ mTuningEnergy = std::sqrt(energy);
+}
+
+void LLWebRTCImpl::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)
+{
+}
+
+//
+// LLWebRTCSignalInterface
+//
+
+void LLWebRTCImpl::setSignalingObserver(LLWebRTCSignalingObserver *observer) { mSignalingObserverList.emplace_back(observer); }
+
+void LLWebRTCImpl::unsetSignalingObserver(LLWebRTCSignalingObserver *observer)
+{
+ std::vector<LLWebRTCSignalingObserver *>::iterator it =
+ std::find(mSignalingObserverList.begin(), mSignalingObserverList.end(), observer);
+ if (it != mSignalingObserverList.end())
+ {
+ mSignalingObserverList.erase(it);
+ }
+}
+
+bool LLWebRTCImpl::initializeConnection()
+{
+ RTC_DCHECK(!mPeerConnection);
+ RTC_DCHECK(!mPeerConnectionFactory);
+ mAnswerReceived = false;
+ mPeerConnectionFactory = webrtc::CreatePeerConnectionFactory(mNetworkThread.get(),
+ mWorkerThread.get(),
+ mSignalingThread.get(),
+ nullptr /* default_adm */,
+ webrtc::CreateBuiltinAudioEncoderFactory(),
+ webrtc::CreateBuiltinAudioDecoderFactory(),
+ nullptr /* video_encoder_factory */,
+ nullptr /* video_decoder_factory */,
+ nullptr /* audio_mixer */,
+ nullptr /* audio_processing */);
+
+ if (!mPeerConnectionFactory)
+ {
+ shutdownConnection();
+ return false;
+ }
+
+ webrtc::PeerConnectionInterface::RTCConfiguration config;
+ config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+ webrtc::PeerConnectionInterface::IceServer server;
+ server.uri = "stun:stun.l.google.com:19302";
+ // config.servers.push_back(server);
+ // server.uri = "stun:stun1.l.google.com:19302";
+ // config.servers.push_back(server);
+ // server.uri = "stun:stun2.l.google.com:19302";
+ // config.servers.push_back(server);
+ // server.uri = "stun:stun3.l.google.com:19302";
+ // config.servers.push_back(server);
+ // server.uri = "stun:stun4.l.google.com:19302";
+ // config.servers.push_back(server);
+
+ webrtc::PeerConnectionDependencies pc_dependencies(this);
+ auto error_or_peer_connection = mPeerConnectionFactory->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
+ if (error_or_peer_connection.ok())
+ {
+ mPeerConnection = std::move(error_or_peer_connection.value());
+ }
+ else
+ {
+ shutdownConnection();
+ return false;
+ }
+
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
+
+ cricket::AudioOptions audioOptions;
+ audioOptions.auto_gain_control = true;
+ audioOptions.echo_cancellation = true;
+ audioOptions.noise_suppression = true;
+
+ rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream");
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ mPeerConnectionFactory->CreateAudioTrack("SLAudio", mPeerConnectionFactory->CreateAudioSource(cricket::AudioOptions()).get()));
+ audio_track->set_enabled(true);
+ stream->AddTrack(audio_track);
+
+ mPeerConnection->AddTrack(audio_track, {"SLStream"});
+ mPeerConnection->SetLocalDescription(rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this));
+
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
+
+ return mPeerConnection != nullptr;
+}
+
+void LLWebRTCImpl::shutdownConnection()
+{
+ mPeerConnection = nullptr;
+ mPeerConnectionFactory = nullptr;
+}
+
+void LLWebRTCImpl::AnswerAvailable(const std::string &sdp)
+{
+ mSignalingThread->PostTask(
+ [this, sdp]()
+ {
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->peer_connection_state();
+ mPeerConnection->SetRemoteDescription(webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp),
+ rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
+ mAnswerReceived = true;
+ for (auto &observer : mSignalingObserverList)
+ {
+ for (auto &candidate : mCachedIceCandidates)
+ {
+ LLWebRTCIceCandidate ice_candidate;
+ 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);
+ }
+ mCachedIceCandidates.clear();
+ if (mPeerConnection->ice_gathering_state() == webrtc::PeerConnectionInterface::IceGatheringState::kIceGatheringComplete)
+ {
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnIceGatheringState(llwebrtc::LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_COMPLETE);
+ }
+ }
+ }
+ });
+}
+
+void LLWebRTCImpl::setMute(bool mute)
+{
+ mSignalingThread->PostTask(
+ [this,mute]()
+ {
+ auto senders = mPeerConnection->GetSenders();
+
+ RTC_LOG(LS_INFO) << __FUNCTION__ << (mute ? "disabling" : "enabling") << " streams count "
+ << senders.size();
+
+ for (auto& sender : senders)
+ {
+ sender->track()->set_enabled(!mute);
+ }
+ });
+}
+
+//
+// PeerConnectionObserver implementation.
+//
+
+void LLWebRTCImpl::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
+ const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> &streams)
+{
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
+}
+
+void LLWebRTCImpl::OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver)
+{
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
+}
+
+void LLWebRTCImpl::OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state)
+{
+ LLWebRTCSignalingObserver::IceGatheringState webrtc_new_state = LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_NEW;
+ switch (new_state)
+ {
+ case webrtc::PeerConnectionInterface::IceGatheringState::kIceGatheringNew:
+ webrtc_new_state = LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_NEW;
+ break;
+ case webrtc::PeerConnectionInterface::IceGatheringState::kIceGatheringGathering:
+ webrtc_new_state = LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_GATHERING;
+ break;
+ case webrtc::PeerConnectionInterface::IceGatheringState::kIceGatheringComplete:
+ webrtc_new_state = LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_COMPLETE;
+ break;
+ default:
+ RTC_LOG(LS_ERROR) << __FUNCTION__ << " Bad Ice Gathering State" << new_state;
+ webrtc_new_state = LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_NEW;
+ return;
+ }
+
+ if (mAnswerReceived)
+ {
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnIceGatheringState(webrtc_new_state);
+ }
+ }
+}
+
+// Called any time the PeerConnectionState changes.
+void LLWebRTCImpl::OnConnectionChange(webrtc::PeerConnectionInterface::PeerConnectionState new_state)
+{
+ RTC_LOG(LS_ERROR) << __FUNCTION__ << " Peer Connection State Change " << new_state;
+}
+
+void LLWebRTCImpl::OnIceCandidate(const webrtc::IceCandidateInterface *candidate)
+{
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();
+
+ if (!candidate)
+ {
+ RTC_LOG(LS_ERROR) << __FUNCTION__ << " No Ice Candidate Given";
+ return;
+ }
+ if (mAnswerReceived)
+ {
+ for (auto &observer : mSignalingObserverList)
+ {
+ LLWebRTCIceCandidate ice_candidate;
+ 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);
+ }
+ }
+ else
+ {
+ mCachedIceCandidates.push_back(
+ webrtc::CreateIceCandidate(candidate->sdp_mid(), candidate->sdp_mline_index(), candidate->candidate()));
+ }
+}
+
+//
+// CreateSessionDescriptionObserver implementation.
+//
+void LLWebRTCImpl::OnSuccess(webrtc::SessionDescriptionInterface *desc)
+{
+ std::string sdp;
+ desc->ToString(&sdp);
+ RTC_LOG(LS_INFO) << sdp;
+
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnOfferAvailable(sdp);
+ }
+}
+
+void LLWebRTCImpl::OnFailure(webrtc::RTCError error) { RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message(); }
+
+//
+// SetRemoteDescriptionObserverInterface implementation.
+//
+void LLWebRTCImpl::OnSetRemoteDescriptionComplete(webrtc::RTCError error)
+{
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
+ if (!error.ok())
+ {
+ RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
+ return;
+ }
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnAudioEstablished(this);
+ }
+}
+
+//
+// SetLocalDescriptionObserverInterface implementation.
+//
+void LLWebRTCImpl::OnSetLocalDescriptionComplete(webrtc::RTCError error)
+{
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
+ if (!error.ok())
+ {
+ RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
+ return;
+ }
+ auto desc = mPeerConnection->pending_local_description();
+ std::string sdp;
+ desc->ToString(&sdp);
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnOfferAvailable(sdp);
+ }
+}
+
+rtc::RefCountedObject<LLWebRTCImpl> *gWebRTCImpl = nullptr;
+LLWebRTCDeviceInterface *getDeviceInterface() { return gWebRTCImpl; }
+LLWebRTCSignalInterface *getSignalingInterface() { return gWebRTCImpl; }
+
+void init()
+{
+ gWebRTCImpl = new rtc::RefCountedObject<LLWebRTCImpl>();
+ gWebRTCImpl->AddRef();
+ gWebRTCImpl->init();
+}
+} // namespace llwebrtc \ No newline at end of file
diff --git a/indra/llwebrtc/llwebrtc.h b/indra/llwebrtc/llwebrtc.h
new file mode 100644
index 0000000000..121efaab86
--- /dev/null
+++ b/indra/llwebrtc/llwebrtc.h
@@ -0,0 +1,127 @@
+/**
+ * @file llaccordionctrl.cpp
+ * @brief Accordion panel implementation
+ *
+ * $LicenseInfo:firstyear=2023&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2023, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+#ifndef LLWEBRTC_H
+#define LLWEBRTC_H
+
+#include <string>
+#include <vector>
+
+#ifdef LL_MAKEDLL
+#ifdef WEBRTC_WIN
+#define LLSYMEXPORT __declspec(dllexport)
+#elif WEBRTC_LINUX
+#define LLSYMEXPORT __attribute__((visibility("default")))
+#else
+#define LLSYMEXPORT /**/
+#endif
+#else
+#define LLSYMEXPORT /**/
+#endif // LL_MAKEDLL
+
+namespace llwebrtc
+{
+LLSYMEXPORT void init();
+
+struct LLWebRTCIceCandidate
+{
+ std::string candidate;
+ std::string sdp_mid;
+ int mline_index;
+};
+
+class LLWebRTCVoiceDevice
+{
+ public:
+ std::string display_name; // friendly value for the user
+ std::string id; // internal value for selection
+
+ LLWebRTCVoiceDevice(const std::string &display_name, const std::string &id) :
+ display_name(display_name),
+ id(id) {};
+};
+
+typedef std::vector<LLWebRTCVoiceDevice> LLWebRTCVoiceDeviceList;
+
+class LLWebRTCDevicesObserver
+{
+ public:
+ virtual void OnRenderDevicesChanged(const LLWebRTCVoiceDeviceList &render_devices) = 0;
+ virtual void OnCaptureDevicesChanged(const LLWebRTCVoiceDeviceList &capture_devices) = 0;
+};
+
+class LLWebRTCDeviceInterface
+{
+ public:
+
+ virtual void refreshDevices() = 0;
+
+ virtual void setCaptureDevice(const std::string& id) = 0;
+ virtual void setRenderDevice(const std::string& id) = 0;
+
+ virtual void setDevicesObserver(LLWebRTCDevicesObserver *observer) = 0;
+ virtual void unsetDevicesObserver(LLWebRTCDevicesObserver *observer) = 0;
+
+ virtual void setTuningMode(bool enable) = 0;
+ virtual double getTuningMicrophoneEnergy() = 0;
+};
+
+class LLWebRTCAudioInterface
+{
+ public:
+ virtual void setMute(bool mute) = 0;
+};
+
+class LLWebRTCSignalingObserver
+{
+ public:
+ enum IceGatheringState{
+ ICE_GATHERING_NEW,
+ ICE_GATHERING_GATHERING,
+ ICE_GATHERING_COMPLETE
+ };
+ virtual void OnIceGatheringState(IceGatheringState state) = 0;
+ virtual void OnIceCandidate(const LLWebRTCIceCandidate& candidate) = 0;
+ virtual void OnOfferAvailable(const std::string& sdp) = 0;
+ virtual void OnAudioEstablished(LLWebRTCAudioInterface *audio_interface) = 0;
+};
+
+class LLWebRTCSignalInterface
+{
+ public:
+ virtual void setSignalingObserver(LLWebRTCSignalingObserver* observer) = 0;
+ virtual void unsetSignalingObserver(LLWebRTCSignalingObserver* observer) = 0;
+
+ virtual bool initializeConnection() = 0;
+ virtual void shutdownConnection() = 0;
+ virtual void AnswerAvailable(const std::string &sdp) = 0;
+};
+
+LLSYMEXPORT LLWebRTCDeviceInterface* getDeviceInterface();
+LLSYMEXPORT LLWebRTCSignalInterface* getSignalingInterface();
+}
+
+#endif // LLWEBRTC_H \ No newline at end of file
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h
new file mode 100644
index 0000000000..10916e5a25
--- /dev/null
+++ b/indra/llwebrtc/llwebrtc_impl.h
@@ -0,0 +1,177 @@
+/**
+ * @file llaccordionctrl.cpp
+ * @brief Accordion panel implementation
+ *
+ * $LicenseInfo:firstyear=2023&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2023, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+
+#ifndef LLWEBRTC_IMPL_H
+#define LLWEBRTC_IMPL_H
+
+#define LL_MAKEDLL
+#define WEBRTC_WIN 1
+#include "llwebrtc.h"
+// WebRTC Includes
+#ifdef WEBRTC_WIN
+#pragma warning(disable : 4996)
+#endif // WEBRTC_WIN
+#include "api/scoped_refptr.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/ref_counted_object.h"
+#include "rtc_base/ssl_adapter.h"
+#include "rtc_base/thread.h"
+#include "api/peer_connection_interface.h"
+#include "api/media_stream_interface.h"
+#include "api/create_peerconnection_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_data_observer.h"
+#include "rtc_base/task_queue.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+
+namespace llwebrtc
+{
+
+class LLWebRTCImpl : public LLWebRTCDeviceInterface,
+ public LLWebRTCSignalInterface,
+ public LLWebRTCAudioInterface,
+ public webrtc::AudioDeviceDataObserver,
+ public webrtc::PeerConnectionObserver,
+ public webrtc::CreateSessionDescriptionObserver,
+ public webrtc::SetRemoteDescriptionObserverInterface,
+ public webrtc::SetLocalDescriptionObserverInterface
+
+{
+ public:
+ LLWebRTCImpl() :
+ mTuningEnergy(0.0)
+ {
+ }
+ ~LLWebRTCImpl() {}
+
+ void init();
+
+ //
+ // LLWebRTCDeviceInterface
+ //
+
+ void refreshDevices() override;
+
+ void setDevicesObserver(LLWebRTCDevicesObserver *observer) override;
+ void unsetDevicesObserver(LLWebRTCDevicesObserver *observer) override;
+ void setCaptureDevice(const std::string& id) override;
+
+ void setRenderDevice(const std::string& id) override;
+
+ void setTuningMode(bool enable) override;
+ double getTuningMicrophoneEnergy() override;
+
+
+ void 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) override;
+
+ void 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) override;
+
+ //
+ // LLWebRTCSignalInterface
+ //
+
+ void setSignalingObserver(LLWebRTCSignalingObserver *observer) override;
+ void unsetSignalingObserver(LLWebRTCSignalingObserver *observer) override;
+ bool initializeConnection() override;
+ void shutdownConnection() override;
+ void AnswerAvailable(const std::string &sdp) override;
+
+
+ //
+ // LLWebRTCAudioInterface
+ //
+ void setMute(bool mute) override;
+
+ //
+ // PeerConnectionObserver implementation.
+ //
+
+ void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState new_state) override {}
+ void OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
+ const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> &streams) override;
+ void OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) override;
+ void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override {}
+ void OnRenegotiationNeeded() override {}
+ void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState new_state) override {};
+ void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
+ void OnIceCandidate(const webrtc::IceCandidateInterface *candidate) override;
+ void OnIceConnectionReceivingChange(bool receiving) override {}
+ void OnConnectionChange(webrtc::PeerConnectionInterface::PeerConnectionState new_state) override;
+
+ //
+ // CreateSessionDescriptionObserver implementation.
+ //
+ void OnSuccess(webrtc::SessionDescriptionInterface *desc) override;
+ void OnFailure(webrtc::RTCError error) override;
+
+ //
+ // SetRemoteDescriptionObserverInterface implementation.
+ //
+ void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override;
+
+ //
+ // SetLocalDescriptionObserverInterface implementation.
+ //
+ void OnSetLocalDescriptionComplete(webrtc::RTCError error) override;
+
+ protected:
+ std::unique_ptr<rtc::Thread> mNetworkThread;
+ std::unique_ptr<rtc::Thread> mWorkerThread;
+ std::unique_ptr<rtc::Thread> mSignalingThread;
+ rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> mPeerConnectionFactory;
+ webrtc::PeerConnectionInterface::RTCConfiguration mConfiguration;
+ std::unique_ptr<webrtc::TaskQueueFactory> mTaskQueueFactory;
+
+
+ // Devices
+ void updateDevices();
+ rtc::scoped_refptr<webrtc::AudioDeviceModule> mDeviceModule;
+ std::vector<LLWebRTCDevicesObserver *> mVoiceDevicesObserverList;
+
+ double mTuningEnergy;
+
+ // signaling
+ std::vector<LLWebRTCSignalingObserver *> mSignalingObserverList;
+ std::vector<std::unique_ptr<webrtc::IceCandidateInterface>> mCachedIceCandidates;
+ bool mAnswerReceived;
+
+ rtc::scoped_refptr<webrtc::PeerConnectionInterface> mPeerConnection;
+};
+
+}
+
+#endif // LLWEBRTC_IMPL_H \ No newline at end of file
diff --git a/indra/newview/CMakeLists.txt b/indra/newview/CMakeLists.txt
index 7a70d0b6e6..6d9b0ab2dc 100644
--- a/indra/newview/CMakeLists.txt
+++ b/indra/newview/CMakeLists.txt
@@ -30,6 +30,7 @@ include(LLKDU)
include(LLPhysicsExtensions)
include(LLPrimitive)
include(LLWindow)
+include(LLWebRTC)
include(NDOF)
include(NVAPI)
include(OPENAL)
@@ -689,6 +690,7 @@ set(viewer_SOURCE_FILES
llvoiceclient.cpp
llvoicevisualizer.cpp
llvoicevivox.cpp
+ llvoicewebrtc.cpp
llvoinventorylistener.cpp
llvopartgroup.cpp
llvosky.cpp
@@ -1333,6 +1335,7 @@ set(viewer_HEADER_FILES
llvoiceclient.h
llvoicevisualizer.h
llvoicevivox.h
+ llvoicewebrtc.h
llvoinventorylistener.h
llvopartgroup.h
llvosky.h
@@ -1433,6 +1436,7 @@ if (LINUX)
endif (LINUX)
if (WINDOWS)
+
list(APPEND viewer_SOURCE_FILES
llappviewerwin32.cpp
llwindebug.cpp
@@ -1724,6 +1728,7 @@ if (WINDOWS)
${SHARED_LIB_STAGING_DIR}/openjp2.dll
${SHARED_LIB_STAGING_DIR}/libhunspell.dll
${SHARED_LIB_STAGING_DIR}/uriparser.dll
+ ${CMAKE_BINARY_DIR}/llwebrtc/Release/llwebrtc.dll
#${SHARED_LIB_STAGING_DIR}/${LL_INTDIR}/SLVoice.exe
#${SHARED_LIB_STAGING_DIR}/${LL_INTDIR}/libsndfile-1.dll
#${SHARED_LIB_STAGING_DIR}/${LL_INTDIR}/vivoxoal.dll
@@ -1920,6 +1925,7 @@ target_link_libraries(${VIEWER_BINARY_NAME}
llcorehttp
llcommon
llmeshoptimizer
+ llwebrtc
ll::ndof
lllogin
llprimitive
diff --git a/indra/newview/app_settings/settings.xml b/indra/newview/app_settings/settings.xml
index 00b59f9a4d..a91726917d 100644
--- a/indra/newview/app_settings/settings.xml
+++ b/indra/newview/app_settings/settings.xml
@@ -15128,7 +15128,7 @@
<key>Type</key>
<string>String</string>
<key>Value</key>
- <string>vivox</string>
+ <string>webrtc</string>
</map>
<key>WLSkyDetail</key>
<map>
diff --git a/indra/newview/llvieweraudio.cpp b/indra/newview/llvieweraudio.cpp
index 6a0edbecb1..184b6d8e93 100644
--- a/indra/newview/llvieweraudio.cpp
+++ b/indra/newview/llvieweraudio.cpp
@@ -480,11 +480,11 @@ void audio_update_volume(bool force_update)
if (!gViewerWindow->getActive() && (gSavedSettings.getBOOL("MuteWhenMinimized")))
{
- voice_inst->setMuteMic(true);
+ //voice_inst->setMuteMic(true);
}
else
{
- voice_inst->setMuteMic(false);
+ //voice_inst->setMuteMic(false);
}
}
}
diff --git a/indra/newview/llviewerregion.cpp b/indra/newview/llviewerregion.cpp
index 452dcdd8fd..84b007eaa4 100755
--- a/indra/newview/llviewerregion.cpp
+++ b/indra/newview/llviewerregion.cpp
@@ -3136,6 +3136,7 @@ void LLViewerRegionImpl::buildCapabilityNames(LLSD& capabilityNames)
capabilityNames.append("ParcelVoiceInfoRequest");
capabilityNames.append("ProductInfoRequest");
capabilityNames.append("ProvisionVoiceAccountRequest");
+ capabilityNames.append("VoiceSignalingRequest");
capabilityNames.append("ReadOfflineMsgs"); // Requires to respond reliably: AcceptFriendship, AcceptGroupInvite, DeclineFriendship, DeclineGroupInvite
capabilityNames.append("RegionObjects");
capabilityNames.append("RemoteParcelRequest");
diff --git a/indra/newview/llvoiceclient.cpp b/indra/newview/llvoiceclient.cpp
index 68d9f4ffab..294ae0c9ad 100644
--- a/indra/newview/llvoiceclient.cpp
+++ b/indra/newview/llvoiceclient.cpp
@@ -24,13 +24,13 @@
* $/LicenseInfo$
*/
-#include "llviewerprecompiledheaders.h"
#include "llvoiceclient.h"
-#include "llviewercontrol.h"
-#include "llviewerwindow.h"
#include "llvoicevivox.h"
+#include "llvoicewebrtc.h"
#include "llviewernetwork.h"
+#include "llviewercontrol.h"
#include "llcommandhandler.h"
+#include "lldir.h"
#include "llhttpnode.h"
#include "llnotificationsutil.h"
#include "llsdserialize.h"
@@ -161,6 +161,10 @@ void LLVoiceClient::userAuthorized(const std::string& user_id, const LLUUID &age
{
mVoiceModule = (LLVoiceModuleInterface *)LLVivoxVoiceClient::getInstance();
}
+ if (voice_server == "webrtc")
+ {
+ mVoiceModule = (LLVoiceModuleInterface *) LLWebRTCVoiceClient::getInstance();
+ }
else
{
mVoiceModule = NULL;
diff --git a/indra/newview/llvoiceclient.h b/indra/newview/llvoiceclient.h
index aa67502908..1e8ff21b4b 100644
--- a/indra/newview/llvoiceclient.h
+++ b/indra/newview/llvoiceclient.h
@@ -34,6 +34,7 @@ class LLVOAvatar;
#include "lliosocket.h"
#include "v3math.h"
#include "llframetimer.h"
+#include "llsingleton.h"
#include "llcallingcard.h" // for LLFriendObserver
#include "llsecapi.h"
#include "llcontrol.h"
diff --git a/indra/newview/llvoicewebrtc.cpp b/indra/newview/llvoicewebrtc.cpp
new file mode 100644
index 0000000000..6e68ca7e4f
--- /dev/null
+++ b/indra/newview/llvoicewebrtc.cpp
@@ -0,0 +1,7299 @@
+ /**
+ * @file LLWebRTCVoiceClient.cpp
+ * @brief Implementation of LLWebRTCVoiceClient class which is the interface to the voice client process.
+ *
+ * $LicenseInfo:firstyear=2001&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2023, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+#include <algorithm>
+#include "llvoicewebrtc.h"
+
+#include "llsdutil.h"
+
+// Linden library includes
+#include "llavatarnamecache.h"
+#include "llvoavatarself.h"
+#include "llbufferstream.h"
+#include "llfile.h"
+#include "llmenugl.h"
+#ifdef LL_USESYSTEMLIBS
+# include "expat.h"
+#else
+# include "expat/expat.h"
+#endif
+#include "llcallbacklist.h"
+#include "llviewernetwork.h" // for gGridChoice
+#include "llbase64.h"
+#include "llviewercontrol.h"
+#include "llappviewer.h" // for gDisconnected, gDisableVoice
+#include "llprocess.h"
+
+// Viewer includes
+#include "llmutelist.h" // to check for muted avatars
+#include "llagent.h"
+#include "llcachename.h"
+#include "llimview.h" // for LLIMMgr
+#include "llparcel.h"
+#include "llviewerparcelmgr.h"
+#include "llfirstuse.h"
+#include "llspeakers.h"
+#include "lltrans.h"
+#include "llrand.h"
+#include "llviewerwindow.h"
+#include "llviewercamera.h"
+#include "llversioninfo.h"
+
+#include "llviewernetwork.h"
+#include "llnotificationsutil.h"
+
+#include "llcorehttputil.h"
+#include "lleventfilter.h"
+
+#include "stringize.h"
+
+#include "llwebrtc.h"
+
+// for base64 decoding
+#include "apr_base64.h"
+
+#define USE_SESSION_GROUPS 0
+#define VX_NULL_POSITION -2147483648.0 /*The Silence*/
+
+extern LLMenuBarGL* gMenuBarView;
+extern void handle_voice_morphing_subscribe();
+
+namespace {
+ const F32 VOLUME_SCALE_WEBRTC = 0.01f;
+
+ const F32 SPEAKING_TIMEOUT = 1.f;
+
+ static const std::string VOICE_SERVER_TYPE = "WebRTC";
+
+ // Don't retry connecting to the daemon more frequently than this:
+ const F32 DAEMON_CONNECT_THROTTLE_SECONDS = 1.0f;
+ const int DAEMON_CONNECT_RETRY_MAX = 3;
+
+ // Don't send positional updates more frequently than this:
+ const F32 UPDATE_THROTTLE_SECONDS = 0.5f;
+
+ // Timeout for connection to WebRTC
+ const F32 CONNECT_ATTEMPT_TIMEOUT = 300.0f;
+ const F32 CONNECT_DNS_TIMEOUT = 5.0f;
+ const int CONNECT_RETRY_MAX = 3;
+
+ const F32 LOGIN_ATTEMPT_TIMEOUT = 30.0f;
+ const F32 LOGOUT_ATTEMPT_TIMEOUT = 5.0f;
+ const int LOGIN_RETRY_MAX = 3;
+
+ const F32 PROVISION_RETRY_TIMEOUT = 2.0;
+ const int PROVISION_RETRY_MAX = 5;
+
+ // Cosine of a "trivially" small angle
+ const F32 FOUR_DEGREES = 4.0f * (F_PI / 180.0f);
+ const F32 MINUSCULE_ANGLE_COS = (F32) cos(0.5f * FOUR_DEGREES);
+
+ const F32 SESSION_JOIN_TIMEOUT = 30.0f;
+
+ // Defines the maximum number of times(in a row) "stateJoiningSession" case for spatial channel is reached in stateMachine()
+ // which is treated as normal. The is the number of frames to wait for a channel join before giving up. This was changed
+ // from the original count of 50 for two reason. Modern PCs have higher frame rates and sometimes the SLVoice process
+ // backs up processing join requests. There is a log statement that records when channel joins take longer than 100 frames.
+ const int MAX_NORMAL_JOINING_SPATIAL_NUM = 1500;
+
+ // How often to check for expired voice fonts in seconds
+ const F32 VOICE_FONT_EXPIRY_INTERVAL = 10.f;
+ // Time of day at which WebRTC expires voice font subscriptions.
+ // Used to replace the time portion of received expiry timestamps.
+ static const std::string VOICE_FONT_EXPIRY_TIME = "T05:00:00Z";
+
+ // Maximum length of capture buffer recordings in seconds.
+ const F32 CAPTURE_BUFFER_MAX_TIME = 10.f;
+
+ const int ERROR_WebRTC_OBJECT_NOT_FOUND = 1001;
+ const int ERROR_WebRTC_NOT_LOGGED_IN = 1007;
+}
+
+static int scale_mic_volume(float volume)
+{
+ // incoming volume has the range [0.0 ... 2.0], with 1.0 as the default.
+ // Map it to WebRTC levels as follows: 0.0 -> 30, 1.0 -> 50, 2.0 -> 70
+ return 30 + (int)(volume * 20.0f);
+}
+
+static int scale_speaker_volume(float volume)
+{
+ // incoming volume has the range [0.0 ... 1.0], with 0.5 as the default.
+ // Map it to WebRTC levels as follows: 0.0 -> 30, 0.5 -> 50, 1.0 -> 70
+ return 30 + (int)(volume * 40.0f);
+
+}
+
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+
+class LLWebRTCVoiceClientMuteListObserver : public LLMuteListObserver
+{
+ /* virtual */ void onChange() { LLWebRTCVoiceClient::getInstance()->muteListChanged();}
+};
+
+
+void LLVoiceWebRTCStats::reset()
+{
+ mStartTime = -1.0f;
+ mConnectCycles = 0;
+ mConnectTime = -1.0f;
+ mConnectAttempts = 0;
+ mProvisionTime = -1.0f;
+ mProvisionAttempts = 0;
+ mEstablishTime = -1.0f;
+ mEstablishAttempts = 0;
+}
+
+LLVoiceWebRTCStats::LLVoiceWebRTCStats()
+{
+ reset();
+}
+
+LLVoiceWebRTCStats::~LLVoiceWebRTCStats()
+{
+}
+
+void LLVoiceWebRTCStats::connectionAttemptStart()
+{
+ if (!mConnectAttempts)
+ {
+ mStartTime = LLTimer::getTotalTime();
+ mConnectCycles++;
+ }
+ mConnectAttempts++;
+}
+
+void LLVoiceWebRTCStats::connectionAttemptEnd(bool success)
+{
+ if ( success )
+ {
+ mConnectTime = (LLTimer::getTotalTime() - mStartTime) / USEC_PER_SEC;
+ }
+}
+
+void LLVoiceWebRTCStats::provisionAttemptStart()
+{
+ if (!mProvisionAttempts)
+ {
+ mStartTime = LLTimer::getTotalTime();
+ }
+ mProvisionAttempts++;
+}
+
+void LLVoiceWebRTCStats::provisionAttemptEnd(bool success)
+{
+ if ( success )
+ {
+ mProvisionTime = (LLTimer::getTotalTime() - mStartTime) / USEC_PER_SEC;
+ }
+}
+
+void LLVoiceWebRTCStats::establishAttemptStart()
+{
+ if (!mEstablishAttempts)
+ {
+ mStartTime = LLTimer::getTotalTime();
+ }
+ mEstablishAttempts++;
+}
+
+void LLVoiceWebRTCStats::establishAttemptEnd(bool success)
+{
+ if ( success )
+ {
+ mEstablishTime = (LLTimer::getTotalTime() - mStartTime) / USEC_PER_SEC;
+ }
+}
+
+LLSD LLVoiceWebRTCStats::read()
+{
+ LLSD stats(LLSD::emptyMap());
+
+ stats["connect_cycles"] = LLSD::Integer(mConnectCycles);
+ stats["connect_attempts"] = LLSD::Integer(mConnectAttempts);
+ stats["connect_time"] = LLSD::Real(mConnectTime);
+
+ stats["provision_attempts"] = LLSD::Integer(mProvisionAttempts);
+ stats["provision_time"] = LLSD::Real(mProvisionTime);
+
+ stats["establish_attempts"] = LLSD::Integer(mEstablishAttempts);
+ stats["establish_time"] = LLSD::Real(mEstablishTime);
+
+ return stats;
+}
+
+static LLWebRTCVoiceClientMuteListObserver mutelist_listener;
+static bool sMuteListListener_listening = false;
+
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+
+bool LLWebRTCVoiceClient::sShuttingDown = false;
+bool LLWebRTCVoiceClient::sConnected = false;
+LLPumpIO *LLWebRTCVoiceClient::sPump = nullptr;
+
+LLWebRTCVoiceClient::LLWebRTCVoiceClient() :
+ mSessionTerminateRequested(false),
+ mRelogRequested(false),
+ mTerminateDaemon(false),
+ mSpatialJoiningNum(0),
+
+ mTuningMode(false),
+ mTuningEnergy(0.0f),
+ mTuningMicVolume(0),
+ mTuningMicVolumeDirty(true),
+ mTuningSpeakerVolume(50), // Set to 50 so the user can hear himself when he sets his mic volume
+ mTuningSpeakerVolumeDirty(true),
+ mDevicesListUpdated(false),
+
+ mAreaVoiceDisabled(false),
+ mAudioSession(), // TBD - should be NULL
+ mAudioSessionChanged(false),
+ mNextAudioSession(),
+
+ mCurrentParcelLocalID(0),
+ mConnectorEstablished(false),
+ mAccountLoggedIn(false),
+ mNumberOfAliases(0),
+ mCommandCookie(0),
+ mLoginRetryCount(0),
+
+ mBuddyListMapPopulated(false),
+ mBlockRulesListReceived(false),
+ mAutoAcceptRulesListReceived(false),
+
+ mSpatialCoordsDirty(false),
+ mIsInitialized(false),
+
+ mMuteMic(false),
+ mMuteMicDirty(false),
+ mFriendsListDirty(true),
+
+ mEarLocation(0),
+ mSpeakerVolumeDirty(true),
+ mSpeakerMuteDirty(true),
+ mMicVolume(0),
+ mMicVolumeDirty(true),
+
+ mVoiceEnabled(false),
+ mWriteInProgress(false),
+
+ mLipSyncEnabled(false),
+
+ mVoiceFontsReceived(false),
+ mVoiceFontsNew(false),
+ mVoiceFontListDirty(false),
+
+ mCaptureBufferMode(false),
+ mCaptureBufferRecording(false),
+ mCaptureBufferRecorded(false),
+ mCaptureBufferPlaying(false),
+ mShutdownComplete(true),
+ mPlayRequestCount(0),
+
+ mAvatarNameCacheConnection(),
+ mIsInTuningMode(false),
+ mIsInChannel(false),
+ mIsJoiningSession(false),
+ mIsWaitingForFonts(false),
+ mIsLoggingIn(false),
+ mIsLoggedIn(false),
+ mIsProcessingChannels(false),
+ mIsCoroutineActive(false),
+ mWebRTCPump("WebRTCClientPump"),
+ mWebRTCDeviceInterface(nullptr),
+ mWebRTCSignalingInterface(nullptr),
+ mWebRTCAudioInterface(nullptr)
+{
+ sShuttingDown = false;
+ sConnected = false;
+ sPump = nullptr;
+
+ mSpeakerVolume = scale_speaker_volume(0);
+
+ mVoiceVersion.serverVersion = "";
+ mVoiceVersion.serverType = VOICE_SERVER_TYPE;
+
+ // gMuteListp isn't set up at this point, so we defer this until later.
+// gMuteListp->addObserver(&mutelist_listener);
+
+
+#if LL_DARWIN || LL_LINUX
+ // HACK: THIS DOES NOT BELONG HERE
+ // When the WebRTC daemon dies, the next write attempt on our socket generates a SIGPIPE, which kills us.
+ // This should cause us to ignore SIGPIPE and handle the error through proper channels.
+ // This should really be set up elsewhere. Where should it go?
+ signal(SIGPIPE, SIG_IGN);
+
+ // Since we're now launching the gateway with fork/exec instead of system(), we need to deal with zombie processes.
+ // Ignoring SIGCHLD should prevent zombies from being created. Alternately, we could use wait(), but I'd rather not do that.
+ signal(SIGCHLD, SIG_IGN);
+#endif
+
+
+ gIdleCallbacks.addFunction(idle, this);
+}
+
+//---------------------------------------------------
+
+LLWebRTCVoiceClient::~LLWebRTCVoiceClient()
+{
+ if (mAvatarNameCacheConnection.connected())
+ {
+ mAvatarNameCacheConnection.disconnect();
+ }
+ sShuttingDown = true;
+}
+
+//---------------------------------------------------
+
+void LLWebRTCVoiceClient::init(LLPumpIO *pump)
+{
+ // constructor will set up LLVoiceClient::getInstance()
+ sPump = pump;
+
+// LLCoros::instance().launch("LLWebRTCVoiceClient::voiceControlCoro",
+// boost::bind(&LLWebRTCVoiceClient::voiceControlCoro, LLWebRTCVoiceClient::getInstance()));
+ llwebrtc::init();
+
+ mWebRTCDeviceInterface = llwebrtc::getDeviceInterface();
+ mWebRTCDeviceInterface->setDevicesObserver(this);
+
+ mWebRTCSignalingInterface = llwebrtc::getSignalingInterface();
+ mWebRTCSignalingInterface->setSignalingObserver(this);
+}
+
+void LLWebRTCVoiceClient::terminate()
+{
+ if (sShuttingDown)
+ {
+ return;
+ }
+
+ // needs to be done manually here since we will not get another pass in
+ // coroutines... that mechanism is long since gone.
+ if (mIsLoggedIn)
+ {
+ logoutOfWebRTC(false);
+ }
+
+ if(sConnected)
+ {
+ breakVoiceConnection(false);
+ sConnected = false;
+ }
+ else
+ {
+ mRelogRequested = false;
+ }
+
+ sShuttingDown = true;
+ sPump = NULL;
+}
+
+//---------------------------------------------------
+
+void LLWebRTCVoiceClient::cleanUp()
+{
+ LL_DEBUGS("Voice") << LL_ENDL;
+
+ deleteAllSessions();
+ deleteAllVoiceFonts();
+ deleteVoiceFontTemplates();
+ LL_DEBUGS("Voice") << "exiting" << LL_ENDL;
+}
+
+//---------------------------------------------------
+
+const LLVoiceVersionInfo& LLWebRTCVoiceClient::getVersion()
+{
+ return mVoiceVersion;
+}
+
+//---------------------------------------------------
+
+void LLWebRTCVoiceClient::updateSettings()
+{
+ setVoiceEnabled(voiceEnabled());
+ setEarLocation(gSavedSettings.getS32("VoiceEarLocation"));
+
+ std::string inputDevice = gSavedSettings.getString("VoiceInputAudioDevice");
+ setCaptureDevice(inputDevice);
+ std::string outputDevice = gSavedSettings.getString("VoiceOutputAudioDevice");
+ setRenderDevice(outputDevice);
+ F32 mic_level = gSavedSettings.getF32("AudioLevelMic");
+ setMicGain(mic_level);
+ setLipSyncEnabled(gSavedSettings.getBOOL("LipSyncEnabled"));
+}
+
+/////////////////////////////
+// utility functions
+
+bool LLWebRTCVoiceClient::writeString(const std::string &str)
+{
+ bool result = false;
+ LL_DEBUGS("LowVoice") << "sending:\n" << str << LL_ENDL;
+
+ if(sConnected)
+ {
+ apr_status_t err;
+ apr_size_t size = (apr_size_t)str.size();
+ apr_size_t written = size;
+
+ //MARK: Turn this on to log outgoing XML
+ // LL_DEBUGS("Voice") << "sending: " << str << LL_ENDL;
+
+ // check return code - sockets will fail (broken, etc.)
+ err = apr_socket_send(
+ mSocket->getSocket(),
+ (const char*)str.data(),
+ &written);
+
+ if(err == 0 && written == size)
+ {
+ // Success.
+ result = true;
+ }
+ else if (err == 0 && written != size) {
+ // Did a short write, log it for now
+ LL_WARNS("Voice") << ") short write on socket sending data to WebRTC daemon." << "Sent " << written << "bytes instead of " << size <<LL_ENDL;
+ }
+ else if(APR_STATUS_IS_EAGAIN(err))
+ {
+ char buf[MAX_STRING];
+ LL_WARNS("Voice") << "EAGAIN error " << err << " (" << apr_strerror(err, buf, MAX_STRING) << ") sending data to WebRTC daemon." << LL_ENDL;
+ }
+ else
+ {
+ // Assume any socket error means something bad. For now, just close the socket.
+ char buf[MAX_STRING];
+ LL_WARNS("Voice") << "apr error " << err << " ("<< apr_strerror(err, buf, MAX_STRING) << ") sending data to WebRTC daemon." << LL_ENDL;
+ daemonDied();
+ }
+ }
+
+ return result;
+}
+
+
+/////////////////////////////
+// session control messages
+void LLWebRTCVoiceClient::connectorCreate()
+{
+
+}
+
+void LLWebRTCVoiceClient::connectorShutdown()
+{
+
+ mShutdownComplete = true;
+}
+
+void LLWebRTCVoiceClient::userAuthorized(const std::string& user_id, const LLUUID &agentID)
+{
+
+ mAccountDisplayName = user_id;
+
+ LL_INFOS("Voice") << "name \"" << mAccountDisplayName << "\" , ID " << agentID << LL_ENDL;
+
+ mAccountName = nameFromID(agentID);
+}
+
+void LLWebRTCVoiceClient::setLoginInfo(
+ const std::string& account_name,
+ const std::string& password,
+ const std::string& channel_sdp)
+{
+ mRemoteChannelSDP = channel_sdp;
+ mWebRTCSignalingInterface->AnswerAvailable(channel_sdp);
+
+ if(mAccountLoggedIn)
+ {
+ // Already logged in.
+ LL_WARNS("Voice") << "Called while already logged in." << LL_ENDL;
+
+ // Don't process another login.
+ return;
+ }
+ else if ( account_name != mAccountName )
+ {
+ LL_WARNS("Voice") << "Mismatched account name! " << account_name
+ << " instead of " << mAccountName << LL_ENDL;
+ }
+ else
+ {
+ mAccountPassword = password;
+ }
+}
+
+void LLWebRTCVoiceClient::idle(void* user_data)
+{
+}
+
+//=========================================================================
+// the following are methods to support the coroutine implementation of the
+// voice connection and processing. They should only be called in the context
+// of a coroutine.
+//
+//
+
+typedef enum e_voice_control_coro_state
+{
+ VOICE_STATE_ERROR = -1,
+ VOICE_STATE_DONE = 0,
+ VOICE_STATE_TP_WAIT, // entry point
+ VOICE_STATE_START_DAEMON,
+ VOICE_STATE_PROVISION_ACCOUNT,
+ VOICE_STATE_SESSION_PROVISION_WAIT,
+ VOICE_STATE_START_SESSION,
+ VOICE_STATE_WAIT_FOR_SESSION_START,
+ VOICE_STATE_SESSION_RETRY,
+ VOICE_STATE_SESSION_ESTABLISHED,
+ VOICE_STATE_WAIT_FOR_CHANNEL,
+ VOICE_STATE_DISCONNECT,
+ VOICE_STATE_WAIT_FOR_EXIT,
+} EVoiceControlCoroState;
+
+void LLWebRTCVoiceClient::voiceControlCoro()
+{
+ int state = 0;
+ try
+ {
+ // state is passed as a reference instead of being
+ // a member due to unresolved issues with coroutine
+ // surviving longer than LLWebRTCVoiceClient
+ voiceControlStateMachine();
+ }
+ catch (const LLCoros::Stop&)
+ {
+ LL_DEBUGS("LLWebRTCVoiceClient") << "Received a shutdown exception" << LL_ENDL;
+ }
+ catch (const LLContinueError&)
+ {
+ LOG_UNHANDLED_EXCEPTION("LLWebRTCVoiceClient");
+ }
+ catch (...)
+ {
+ // Ideally for Windows need to log SEH exception instead or to set SEH
+ // handlers but bugsplat shows local variables for windows, which should
+ // be enough
+ LL_WARNS("Voice") << "voiceControlStateMachine crashed in state " << state << LL_ENDL;
+ throw;
+ }
+}
+
+void LLWebRTCVoiceClient::voiceControlStateMachine()
+{
+ if (sShuttingDown)
+ {
+ return;
+ }
+
+ LL_DEBUGS("Voice") << "starting" << LL_ENDL;
+ mIsCoroutineActive = true;
+ LLCoros::set_consuming(true);
+
+ U32 retry = 0;
+
+ mVoiceControlState = VOICE_STATE_TP_WAIT;
+
+ do
+ {
+ if (sShuttingDown)
+ {
+ // WebRTC singleton performed the exit, logged out,
+ // cleaned sockets, gateway and no longer cares
+ // about state of coroutine, so just stop
+ return;
+ }
+
+ processIceUpdates();
+
+ switch (mVoiceControlState)
+ {
+ case VOICE_STATE_TP_WAIT:
+ // starting point for voice
+ if (gAgent.getTeleportState() != LLAgent::TELEPORT_NONE)
+ {
+ LL_DEBUGS("Voice") << "Suspending voiceControlCoro() momentarily for teleport. Tuning: " << mTuningMode << ". Relog: " << mRelogRequested << LL_ENDL;
+ llcoro::suspendUntilTimeout(1.0);
+ }
+ else
+ {
+ mVoiceControlState = VOICE_STATE_START_SESSION;
+ }
+ break;
+
+ case VOICE_STATE_START_SESSION:
+ if (establishVoiceConnection())
+ {
+ mVoiceControlState = VOICE_STATE_WAIT_FOR_SESSION_START;
+ }
+ else
+ {
+ mVoiceControlState = VOICE_STATE_SESSION_RETRY;
+ }
+ break;
+
+ case VOICE_STATE_WAIT_FOR_SESSION_START:
+ llcoro::suspendUntilTimeout(1.0);
+ if (!mChannelSDP.empty())
+ {
+ mVoiceControlState = VOICE_STATE_PROVISION_ACCOUNT;
+ }
+ break;
+
+ case VOICE_STATE_PROVISION_ACCOUNT:
+ if (!provisionVoiceAccount())
+ {
+ mVoiceControlState = VOICE_STATE_SESSION_RETRY;
+ }
+ else
+ {
+ mVoiceControlState = VOICE_STATE_SESSION_PROVISION_WAIT;
+ }
+ break;
+ case VOICE_STATE_SESSION_PROVISION_WAIT:
+ llcoro::suspendUntilTimeout(1.0);
+ break;
+
+
+ case VOICE_STATE_SESSION_RETRY:
+ giveUp(); // cleans sockets and session
+ if (mRelogRequested)
+ {
+ // We failed to connect, give it a bit time before retrying.
+ retry++;
+ F32 full_delay = llmin(5.f * (F32)retry, 60.f);
+ F32 current_delay = 0.f;
+ LL_INFOS("Voice") << "Voice failed to establish session after " << retry
+ << " tries. Will attempt to reconnect in " << full_delay
+ << " seconds" << LL_ENDL;
+ while (current_delay < full_delay && !sShuttingDown)
+ {
+ // Assuming that a second has passed is not accurate,
+ // but we don't need accurancy here, just to make sure
+ // that some time passed and not to outlive voice itself
+ current_delay++;
+ llcoro::suspendUntilTimeout(1.f);
+ }
+ mVoiceControlState = VOICE_STATE_WAIT_FOR_EXIT;
+ }
+ else
+ {
+ mVoiceControlState = VOICE_STATE_DONE;
+ }
+ break;
+
+ case VOICE_STATE_SESSION_ESTABLISHED:
+ {
+ if (mTuningMode)
+ {
+ performMicTuning();
+ }
+
+ mVoiceControlState = VOICE_STATE_WAIT_FOR_CHANNEL;
+ }
+ break;
+
+ case VOICE_STATE_WAIT_FOR_CHANNEL:
+ waitForChannel(); // todo: split into more states like login/fonts
+ mVoiceControlState = VOICE_STATE_DISCONNECT;
+ break;
+
+ case VOICE_STATE_DISCONNECT:
+ LL_DEBUGS("Voice") << "lost channel RelogRequested=" << mRelogRequested << LL_ENDL;
+ endAndDisconnectSession();
+ retry = 0; // Connected without issues
+ mVoiceControlState = VOICE_STATE_WAIT_FOR_EXIT;
+ break;
+
+ case VOICE_STATE_WAIT_FOR_EXIT:
+ if (mRelogRequested && mVoiceEnabled)
+ {
+ LL_INFOS("Voice") << "will attempt to reconnect to voice" << LL_ENDL;
+ mVoiceControlState = VOICE_STATE_TP_WAIT;
+ }
+ else
+ {
+ mVoiceControlState = VOICE_STATE_DONE;
+ }
+ break;
+
+ case VOICE_STATE_DONE:
+ break;
+ }
+ } while (mVoiceControlState > 0);
+
+ if (sShuttingDown)
+ {
+ // LLWebRTCVoiceClient might be already dead
+ return;
+ }
+
+ mIsCoroutineActive = false;
+ LL_INFOS("Voice") << "exiting" << LL_ENDL;
+}
+
+bool LLWebRTCVoiceClient::endAndDisconnectSession()
+{
+ LL_DEBUGS("Voice") << LL_ENDL;
+
+ breakVoiceConnection(true);
+
+ return true;
+}
+
+bool LLWebRTCVoiceClient::callbackEndDaemon(const LLSD& data)
+{
+ if (!sShuttingDown && mVoiceEnabled)
+ {
+ LL_WARNS("Voice") << "SLVoice terminated " << ll_stream_notation_sd(data) << LL_ENDL;
+ terminateAudioSession(false);
+ closeSocket();
+ cleanUp();
+ LLVoiceClient::getInstance()->setUserPTTState(false);
+ gAgent.setVoiceConnected(false);
+ mRelogRequested = true;
+ }
+ return false;
+}
+
+
+bool LLWebRTCVoiceClient::provisionVoiceAccount()
+{
+ LL_INFOS("Voice") << "Provisioning voice account." << LL_ENDL;
+
+ while ((!gAgent.getRegion() || !gAgent.getRegion()->capabilitiesReceived()) && !sShuttingDown)
+ {
+ LL_DEBUGS("Voice") << "no capabilities for voice provisioning; waiting " << LL_ENDL;
+ // *TODO* Pump a message for wake up.
+ llcoro::suspend();
+ }
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+
+ std::string url = gAgent.getRegionCapability("ProvisionVoiceAccountRequest");
+
+ LL_DEBUGS("Voice") << "region ready for voice provisioning; url=" << url << LL_ENDL;
+
+ LLVoiceWebRTCStats::getInstance()->provisionAttemptStart();
+ LLSD body;
+ LLSD jsep;
+ jsep["type"] = "offer";
+ jsep["sdp"] = mChannelSDP;
+ body["jsep"] = jsep;
+
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+ url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLWebRTCVoiceClient::OnVoiceAccountProvisioned, this, _1),
+ boost::bind(&LLWebRTCVoiceClient::OnVoiceAccountProvisionFailure, this, url, 3, body, _1));
+ return true;
+}
+
+void LLWebRTCVoiceClient::OnVoiceAccountProvisioned(const LLSD& result)
+{
+ mVoiceControlState = VOICE_STATE_SESSION_ESTABLISHED;
+ LLVoiceWebRTCStats::getInstance()->provisionAttemptEnd(true);
+ std::string channelSDP;
+ if (result.has("jsep") &&
+ result["jsep"].has("type") &&
+ result["jsep"]["type"] == "answer" &&
+ result["jsep"].has("sdp"))
+ {
+ channelSDP = result["jsep"]["sdp"];
+ }
+ std::string voiceAccountServerUri;
+ std::string voiceUserName = gAgent.getID().asString();
+ std::string voicePassword = ""; // no password for now.
+
+ LL_DEBUGS("Voice") << "ProvisionVoiceAccountRequest response"
+ << " user " << (voiceUserName.empty() ? "not set" : "set") << " password "
+ << (voicePassword.empty() ? "not set" : "set") << " channel sdp " << channelSDP << LL_ENDL;
+
+ setLoginInfo(voiceUserName, voicePassword, channelSDP);
+}
+
+void LLWebRTCVoiceClient::OnVoiceAccountProvisionFailure(std::string url, int retries, LLSD body, const LLSD& result)
+{
+ if (sShuttingDown)
+ {
+ return;
+ }
+ if (retries >= 0)
+ {
+
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+ url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLWebRTCVoiceClient::OnVoiceAccountProvisioned, this, _1),
+ boost::bind(&LLWebRTCVoiceClient::OnVoiceAccountProvisionFailure, this, url, retries - 1, body, _1));
+ }
+ else
+ {
+ LL_WARNS("Voice") << "Unable to complete ice trickling voice account, retrying." << LL_ENDL;
+ }
+}
+
+bool LLWebRTCVoiceClient::establishVoiceConnection()
+{
+ LL_INFOS("Voice") << "Ice Gathering voice account." << LL_ENDL;
+ while ((!gAgent.getRegion() || !gAgent.getRegion()->capabilitiesReceived()) && !sShuttingDown)
+ {
+ LL_DEBUGS("Voice") << "no capabilities for voice provisioning; waiting " << LL_ENDL;
+ // *TODO* Pump a message for wake up.
+ llcoro::suspend();
+ return false;
+ }
+
+ if (!mVoiceEnabled && mIsInitialized)
+ {
+ LL_WARNS("Voice") << "cannot establish connection; enabled "<<mVoiceEnabled<<" initialized "<<mIsInitialized<<LL_ENDL;
+ return false;
+ }
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+ return mWebRTCSignalingInterface->initializeConnection();
+}
+
+bool LLWebRTCVoiceClient::breakVoiceConnection(bool corowait)
+{
+ LL_DEBUGS("Voice") << "( wait=" << corowait << ")" << LL_ENDL;
+ bool retval(true);
+
+ mShutdownComplete = false;
+ connectorShutdown();
+
+ if (corowait)
+ {
+ LLSD timeoutResult(LLSDMap("connector", "timeout"));
+
+ LLSD result = llcoro::suspendUntilEventOnWithTimeout(mWebRTCPump, LOGOUT_ATTEMPT_TIMEOUT, timeoutResult);
+ LL_DEBUGS("Voice") << "event=" << ll_stream_notation_sd(result) << LL_ENDL;
+
+ retval = result.has("connector");
+ }
+ else
+ {
+ mRelogRequested = false; //stop the control coro
+ // If we are not doing a corowait then we must sleep until the connector has responded
+ // otherwise we may very well close the socket too early.
+#if LL_WINDOWS
+ if (!mShutdownComplete)
+ {
+ // The situation that brings us here is a call from ::terminate()
+ // At this point message system is already down so we can't wait for
+ // the message, yet we need to receive "connector shutdown response".
+ // Either wait a bit and emulate it or check gMessageSystem for specific message
+ _sleep(1000);
+ if (sConnected)
+ {
+ sConnected = false;
+ LLSD WebRTCevent(LLSDMap("connector", LLSD::Boolean(false)));
+ mWebRTCPump.post(WebRTCevent);
+ }
+ mShutdownComplete = true;
+ }
+#endif
+ }
+
+ LL_DEBUGS("Voice") << "closing SLVoice socket" << LL_ENDL;
+ closeSocket(); // Need to do this now -- bad things happen if the destructor does it later.
+ cleanUp();
+ sConnected = false;
+
+ return retval;
+}
+
+bool LLWebRTCVoiceClient::loginToWebRTC()
+{
+
+
+ mRelogRequested = false;
+ mIsLoggedIn = true;
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LOGGED_IN);
+
+ // Set up the mute list observer if it hasn't been set up already.
+ if ((!sMuteListListener_listening))
+ {
+ LLMuteList::getInstance()->addObserver(&mutelist_listener);
+ sMuteListListener_listening = true;
+ }
+
+ // Set the initial state of mic mute, local speaker volume, etc.
+ sendLocalAudioUpdates();
+ mIsLoggingIn = false;
+
+ return true;
+}
+
+void LLWebRTCVoiceClient::logoutOfWebRTC(bool wait)
+{
+ if (mIsLoggedIn)
+ {
+ // Ensure that we'll re-request provisioning before logging in again
+ mAccountPassword.clear();
+
+ logoutSendMessage();
+
+ if (wait)
+ {
+ LLSD timeoutResult(LLSDMap("logout", "timeout"));
+ LLSD result;
+
+ do
+ {
+ LL_DEBUGS("Voice")
+ << "waiting for logout response on "
+ << mWebRTCPump.getName()
+ << LL_ENDL;
+
+ result = llcoro::suspendUntilEventOnWithTimeout(mWebRTCPump, LOGOUT_ATTEMPT_TIMEOUT, timeoutResult);
+
+ if (sShuttingDown)
+ {
+ break;
+ }
+
+ LL_DEBUGS("Voice") << "event=" << ll_stream_notation_sd(result) << LL_ENDL;
+ // Don't get confused by prior queued events -- note that it's
+ // very important that mWebRTCPump is an LLEventMailDrop, which
+ // does queue events.
+ } while (! result["logout"]);
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "not waiting for logout" << LL_ENDL;
+ }
+
+ mIsLoggedIn = false;
+ }
+}
+
+bool LLWebRTCVoiceClient::requestParcelVoiceInfo()
+{
+ //_INFOS("Voice") << "Requesting voice info for Parcel" << LL_ENDL;
+
+ LLViewerRegion * region = gAgent.getRegion();
+ if (region == NULL || !region->capabilitiesReceived())
+ {
+ LL_DEBUGS("Voice") << "ParcelVoiceInfoRequest capability not yet available, deferring" << LL_ENDL;
+ return false;
+ }
+
+ // grab the cap.
+ std::string url = gAgent.getRegion()->getCapability("ParcelVoiceInfoRequest");
+ if (url.empty())
+ {
+ // Region dosn't have the cap. Stop probing.
+ LL_DEBUGS("Voice") << "ParcelVoiceInfoRequest capability not available in this region" << LL_ENDL;
+ return false;
+ }
+
+ // update the parcel
+ checkParcelChanged(true);
+
+ LL_DEBUGS("Voice") << "sending ParcelVoiceInfoRequest (" << mCurrentRegionName << ", " << mCurrentParcelLocalID << ")" << LL_ENDL;
+
+ LLCore::HttpRequest::policy_t httpPolicy(LLCore::HttpRequest::DEFAULT_POLICY_ID);
+ LLCoreHttpUtil::HttpCoroutineAdapter::ptr_t
+ httpAdapter(new LLCoreHttpUtil::HttpCoroutineAdapter("parcelVoiceInfoRequest", httpPolicy));
+ LLCore::HttpRequest::ptr_t httpRequest(new LLCore::HttpRequest);
+
+ LLSD result = httpAdapter->postAndSuspend(httpRequest, url, LLSD());
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+
+ LLSD httpResults = result[LLCoreHttpUtil::HttpCoroutineAdapter::HTTP_RESULTS];
+ LLCore::HttpStatus status = LLCoreHttpUtil::HttpCoroutineAdapter::getStatusFromLLSD(httpResults);
+
+ if (mSessionTerminateRequested || (!mVoiceEnabled && mIsInitialized))
+ {
+ // if a terminate request has been received,
+ // bail and go to the stateSessionTerminated
+ // state. If the cap request is still pending,
+ // the responder will check to see if we've moved
+ // to a new session and won't change any state.
+ LL_DEBUGS("Voice") << "terminate requested " << mSessionTerminateRequested
+ << " enabled " << mVoiceEnabled
+ << " initialized " << mIsInitialized
+ << LL_ENDL;
+ terminateAudioSession(true);
+ return false;
+ }
+
+ if ((!status) || (mSessionTerminateRequested || (!mVoiceEnabled && mIsInitialized)))
+ {
+ if (mSessionTerminateRequested || (!mVoiceEnabled && mIsInitialized))
+ {
+ LL_WARNS("Voice") << "Session terminated." << LL_ENDL;
+ }
+
+ LL_WARNS("Voice") << "No voice on parcel" << LL_ENDL;
+ sessionTerminate();
+ return false;
+ }
+
+ std::string uri;
+ std::string credentials;
+
+ if (result.has("voice_credentials"))
+ {
+ LLSD voice_credentials = result["voice_credentials"];
+ if (voice_credentials.has("channel_uri"))
+ {
+ LL_DEBUGS("Voice") << "got voice channel uri" << LL_ENDL;
+ uri = voice_credentials["channel_uri"].asString();
+ }
+ else
+ {
+ LL_WARNS("Voice") << "No voice channel uri" << LL_ENDL;
+ }
+
+ if (voice_credentials.has("channel_credentials"))
+ {
+ LL_DEBUGS("Voice") << "got voice channel credentials" << LL_ENDL;
+ credentials =
+ voice_credentials["channel_credentials"].asString();
+ }
+ else
+ {
+ LLVoiceChannel* channel = LLVoiceChannel::getCurrentVoiceChannel();
+ if (channel != NULL)
+ {
+ if (channel->getSessionName().empty() && channel->getSessionID().isNull())
+ {
+ if (LLViewerParcelMgr::getInstance()->allowAgentVoice())
+ {
+ LL_WARNS("Voice") << "No channel credentials for default channel" << LL_ENDL;
+ }
+ }
+ else
+ {
+ LL_WARNS("Voice") << "No voice channel credentials" << LL_ENDL;
+ }
+ }
+ }
+ }
+ else
+ {
+ if (LLViewerParcelMgr::getInstance()->allowAgentVoice())
+ {
+ LL_WARNS("Voice") << "No voice credentials" << LL_ENDL;
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "No voice credentials" << LL_ENDL;
+ }
+ }
+
+ // set the spatial channel. If no voice credentials or uri are
+ // available, then we simply drop out of voice spatially.
+ return !setSpatialChannel(uri, credentials);
+}
+
+bool LLWebRTCVoiceClient::addAndJoinSession(const sessionStatePtr_t &nextSession)
+{
+ mIsJoiningSession = true;
+
+ sessionStatePtr_t oldSession = mAudioSession;
+
+ LL_INFOS("Voice") << "Adding or joining voice session " << nextSession->mHandle << LL_ENDL;
+
+ mAudioSession = nextSession;
+ mAudioSessionChanged = true;
+ if (!mAudioSession || !mAudioSession->mReconnect)
+ {
+ mNextAudioSession.reset();
+ }
+
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_JOINING);
+
+ llcoro::suspend();
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+
+ LLSD result;
+
+ if (mSpatialJoiningNum == MAX_NORMAL_JOINING_SPATIAL_NUM)
+ {
+ // Notify observers to let them know there is problem with voice
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_VOICE_DISABLED);
+ LL_WARNS() << "There seems to be problem with connection to voice server. Disabling voice chat abilities." << LL_ENDL;
+ }
+
+ // Increase mSpatialJoiningNum only for spatial sessions- it's normal to reach this case for
+ // example for p2p many times while waiting for response, so it can't be used to detect errors
+ if (mAudioSession && mAudioSession->mIsSpatial)
+ {
+ mSpatialJoiningNum++;
+ }
+
+ if (!mVoiceEnabled && mIsInitialized)
+ {
+ LL_DEBUGS("Voice") << "Voice no longer enabled. Exiting"
+ << " enabled " << mVoiceEnabled
+ << " initialized " << mIsInitialized
+ << LL_ENDL;
+ mIsJoiningSession = false;
+ // User bailed out during connect -- jump straight to teardown.
+ terminateAudioSession(true);
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_VOICE_DISABLED);
+ return false;
+ }
+ else if (mSessionTerminateRequested)
+ {
+ LL_DEBUGS("Voice") << "Terminate requested" << LL_ENDL;
+ if (mAudioSession && !mAudioSession->mHandle.empty())
+ {
+ // Only allow direct exits from this state in p2p calls (for cancelling an invite).
+ // Terminating a half-connected session on other types of calls seems to break something in the WebRTC gateway.
+ if (mAudioSession->mIsP2P)
+ {
+ terminateAudioSession(true);
+ mIsJoiningSession = false;
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LEFT_CHANNEL);
+ return false;
+ }
+ }
+ }
+
+ LLSD timeoutResult(LLSDMap("session", "timeout"));
+
+ // We are about to start a whole new session. Anything that MIGHT still be in our
+ // maildrop is going to be stale and cause us much wailing and gnashing of teeth.
+ // Just flush it all out and start new.
+ mWebRTCPump.discard();
+
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_JOINED);
+
+ return true;
+}
+
+bool LLWebRTCVoiceClient::terminateAudioSession(bool wait)
+{
+
+ if (mAudioSession)
+ {
+ LL_INFOS("Voice") << "terminateAudioSession(" << wait << ") Terminating current voice session " << mAudioSession->mHandle << LL_ENDL;
+
+ if (mIsLoggedIn)
+ {
+ if (!mAudioSession->mHandle.empty())
+ {
+
+#if RECORD_EVERYTHING
+ // HACK: for testing only
+ // Save looped recording
+ std::string savepath("/tmp/WebRTCrecording");
+ {
+ time_t now = time(NULL);
+ const size_t BUF_SIZE = 64;
+ char time_str[BUF_SIZE]; /* Flawfinder: ignore */
+
+ strftime(time_str, BUF_SIZE, "%Y-%m-%dT%H:%M:%SZ", gmtime(&now));
+ savepath += time_str;
+ }
+ recordingLoopSave(savepath);
+#endif
+
+ sessionMediaDisconnectSendMessage(mAudioSession);
+
+ if (wait)
+ {
+ LLSD result;
+ do
+ {
+ LLSD timeoutResult(LLSDMap("session", "timeout"));
+
+ result = llcoro::suspendUntilEventOnWithTimeout(mWebRTCPump, LOGOUT_ATTEMPT_TIMEOUT, timeoutResult);
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+
+ LL_DEBUGS("Voice") << "event=" << ll_stream_notation_sd(result) << LL_ENDL;
+ if (result.has("session"))
+ {
+ if (result.has("handle"))
+ {
+ if (result["handle"] != mAudioSession->mHandle)
+ {
+ LL_WARNS("Voice") << "Message for session handle \"" << result["handle"] << "\" while waiting for \"" << mAudioSession->mHandle << "\"." << LL_ENDL;
+ continue;
+ }
+ }
+
+ std::string message = result["session"].asString();
+ if (message == "removed" || message == "timeout")
+ break;
+ }
+ } while (true);
+
+ }
+ }
+ else
+ {
+ LL_WARNS("Voice") << "called with no session handle" << LL_ENDL;
+ }
+ }
+ else
+ {
+ LL_WARNS("Voice") << "Session " << mAudioSession->mHandle << " already terminated by logout." << LL_ENDL;
+ }
+
+ sessionStatePtr_t oldSession = mAudioSession;
+
+ mAudioSession.reset();
+ // We just notified status observers about this change. Don't do it again.
+ mAudioSessionChanged = false;
+
+ // The old session may now need to be deleted.
+ reapSession(oldSession);
+ }
+ else
+ {
+ LL_WARNS("Voice") << "terminateAudioSession(" << wait << ") with NULL mAudioSession" << LL_ENDL;
+ }
+
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LEFT_CHANNEL);
+
+ // Always reset the terminate request flag when we get here.
+ // Some slower PCs have a race condition where they can switch to an incoming P2P call faster than the state machine leaves
+ // the region chat.
+ mSessionTerminateRequested = false;
+
+ bool status=((mVoiceEnabled || !mIsInitialized) && !mRelogRequested && !sShuttingDown);
+ LL_DEBUGS("Voice") << "exiting"
+ << " VoiceEnabled " << mVoiceEnabled
+ << " IsInitialized " << mIsInitialized
+ << " RelogRequested " << mRelogRequested
+ << " ShuttingDown " << (sShuttingDown ? "TRUE" : "FALSE")
+ << " returning " << status
+ << LL_ENDL;
+ return status;
+}
+
+
+typedef enum e_voice_wait_for_channel_state
+{
+ VOICE_CHANNEL_STATE_LOGIN = 0, // entry point
+ VOICE_CHANNEL_STATE_CHECK_EFFECTS,
+ VOICE_CHANNEL_STATE_START_CHANNEL_PROCESSING,
+ VOICE_CHANNEL_STATE_PROCESS_CHANNEL,
+ VOICE_CHANNEL_STATE_NEXT_CHANNEL_DELAY,
+ VOICE_CHANNEL_STATE_NEXT_CHANNEL_CHECK,
+ VOICE_CHANNEL_STATE_LOGOUT,
+ VOICE_CHANNEL_STATE_RELOG,
+ VOICE_CHANNEL_STATE_DONE,
+} EVoiceWaitForChannelState;
+
+bool LLWebRTCVoiceClient::waitForChannel()
+{
+ LL_INFOS("Voice") << "Waiting for channel" << LL_ENDL;
+
+ EVoiceWaitForChannelState state = VOICE_CHANNEL_STATE_LOGIN;
+
+ do
+ {
+ if (sShuttingDown)
+ {
+ // terminate() forcefully disconects voice, no need for cleanup
+ return false;
+ }
+
+ processIceUpdates();
+ switch (state)
+ {
+ case VOICE_CHANNEL_STATE_LOGIN:
+ if (!loginToWebRTC())
+ {
+ return false;
+ }
+ state = VOICE_CHANNEL_STATE_START_CHANNEL_PROCESSING;
+ break;
+
+ case VOICE_CHANNEL_STATE_START_CHANNEL_PROCESSING:
+ mIsProcessingChannels = true;
+ llcoro::suspend();
+ state = VOICE_CHANNEL_STATE_PROCESS_CHANNEL;
+ break;
+
+ case VOICE_CHANNEL_STATE_PROCESS_CHANNEL:
+ if (mTuningMode)
+ {
+ performMicTuning();
+ }
+ else if (mCaptureBufferMode)
+ {
+ recordingAndPlaybackMode();
+ }
+ else if (checkParcelChanged() || (mNextAudioSession == NULL))
+ {
+ // the parcel is changed, or we have no pending audio sessions,
+ // so try to request the parcel voice info
+ // if we have the cap, we move to the appropriate state
+ requestParcelVoiceInfo(); //suspends for http reply
+ }
+ else if (sessionNeedsRelog(mNextAudioSession))
+ {
+ LL_INFOS("Voice") << "Session requesting reprovision and login." << LL_ENDL;
+ requestRelog();
+ break;
+ }
+ else if (mNextAudioSession)
+ {
+ sessionStatePtr_t joinSession = mNextAudioSession;
+ mNextAudioSession.reset();
+ if (!runSession(joinSession)) //suspends
+ {
+ LL_DEBUGS("Voice") << "runSession returned false; leaving inner loop" << LL_ENDL;
+ break;
+ }
+ else
+ {
+ LL_DEBUGS("Voice")
+ << "runSession returned true to inner loop"
+ << " RelogRequested=" << mRelogRequested
+ << " VoiceEnabled=" << mVoiceEnabled
+ << LL_ENDL;
+ }
+ }
+
+ state = VOICE_CHANNEL_STATE_NEXT_CHANNEL_DELAY;
+ break;
+
+ case VOICE_CHANNEL_STATE_NEXT_CHANNEL_DELAY:
+ if (!mNextAudioSession)
+ {
+ llcoro::suspendUntilTimeout(1.0);
+ }
+ state = VOICE_CHANNEL_STATE_NEXT_CHANNEL_CHECK;
+ break;
+
+ case VOICE_CHANNEL_STATE_NEXT_CHANNEL_CHECK:
+ if (mVoiceEnabled && !mRelogRequested)
+ {
+ state = VOICE_CHANNEL_STATE_START_CHANNEL_PROCESSING;
+ break;
+ }
+ else
+ {
+ mIsProcessingChannels = false;
+ LL_DEBUGS("Voice")
+ << "leaving inner waitForChannel loop"
+ << " RelogRequested=" << mRelogRequested
+ << " VoiceEnabled=" << mVoiceEnabled
+ << LL_ENDL;
+ state = VOICE_CHANNEL_STATE_LOGOUT;
+ break;
+ }
+
+ case VOICE_CHANNEL_STATE_LOGOUT:
+ logoutOfWebRTC(true /*bool wait*/);
+ if (mRelogRequested)
+ {
+ state = VOICE_CHANNEL_STATE_RELOG;
+ }
+ else
+ {
+ state = VOICE_CHANNEL_STATE_DONE;
+ }
+ break;
+
+ case VOICE_CHANNEL_STATE_RELOG:
+ LL_DEBUGS("Voice") << "Relog Requested, restarting provisioning" << LL_ENDL;
+ if (!provisionVoiceAccount())
+ {
+ if (sShuttingDown)
+ {
+ return false;
+ }
+ LL_WARNS("Voice") << "provisioning voice failed; giving up" << LL_ENDL;
+ giveUp();
+ return false;
+ }
+ if (mVoiceEnabled && mRelogRequested)
+ {
+ state = VOICE_CHANNEL_STATE_LOGIN;
+ }
+ else
+ {
+ state = VOICE_CHANNEL_STATE_DONE;
+ }
+ break;
+ case VOICE_CHANNEL_STATE_DONE:
+ LL_DEBUGS("Voice")
+ << "exiting"
+ << " RelogRequested=" << mRelogRequested
+ << " VoiceEnabled=" << mVoiceEnabled
+ << LL_ENDL;
+ return !sShuttingDown;
+ }
+ } while (true);
+}
+
+bool LLWebRTCVoiceClient::runSession(const sessionStatePtr_t &session)
+{
+ LL_INFOS("Voice") << "running new voice session " << session->mHandle << LL_ENDL;
+
+ bool joined_session = addAndJoinSession(session);
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+
+ if (!joined_session)
+ {
+ notifyStatusObservers(LLVoiceClientStatusObserver::ERROR_UNKNOWN);
+
+ if (mSessionTerminateRequested)
+ {
+ LL_DEBUGS("Voice") << "runSession terminate requested " << LL_ENDL;
+ terminateAudioSession(true);
+ }
+ // if a relog has been requested then addAndJoineSession
+ // failed in a spectacular way and we need to back out.
+ // If this is not the case then we were simply trying to
+ // make a call and the other party rejected it.
+ return !mRelogRequested;
+ }
+
+ notifyParticipantObservers();
+ notifyVoiceFontObservers();
+
+ LLSD timeoutEvent(LLSDMap("timeout", LLSD::Boolean(true)));
+
+ mIsInChannel = true;
+ mMuteMicDirty = true;
+
+ while (!sShuttingDown
+ && mVoiceEnabled
+ && !mSessionTerminateRequested
+ && !mTuningMode)
+ {
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+
+ if (mSessionTerminateRequested)
+ {
+ break;
+ }
+
+ if (mAudioSession && mAudioSession->mParticipantsChanged)
+ {
+ mAudioSession->mParticipantsChanged = false;
+ notifyParticipantObservers();
+ }
+
+ if (!inSpatialChannel())
+ {
+ // When in a non-spatial channel, never send positional updates.
+ mSpatialCoordsDirty = false;
+ }
+ else
+ {
+ updatePosition();
+
+ if (checkParcelChanged())
+ {
+ // *RIDER: I think I can just return here if the parcel has changed
+ // and grab the new voice channel from the outside loop.
+ //
+ // if the parcel has changed, attempted to request the
+ // cap for the parcel voice info. If we can't request it
+ // then we don't have the cap URL so we do nothing and will
+ // recheck next time around
+ if (requestParcelVoiceInfo()) // suspends
+ { // The parcel voice URI has changed.. break out and reconnect.
+ break;
+ }
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+ }
+ // Do the calculation that enforces the listener<->speaker tether (and also updates the real camera position)
+ enforceTether();
+ }
+ sendPositionAndVolumeUpdate();
+
+ // Do notifications for expiring Voice Fonts.
+ if (mVoiceFontExpiryTimer.hasExpired())
+ {
+ expireVoiceFonts();
+ mVoiceFontExpiryTimer.setTimerExpirySec(VOICE_FONT_EXPIRY_INTERVAL);
+ }
+
+ // send any requests to adjust mic and speaker settings if they have changed
+ sendLocalAudioUpdates();
+
+ mIsInitialized = true;
+ LLSD result = llcoro::suspendUntilEventOnWithTimeout(mWebRTCPump, UPDATE_THROTTLE_SECONDS, timeoutEvent);
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+
+ if (!result.has("timeout")) // logging the timeout event spams the log
+ {
+ LL_DEBUGS("Voice") << "event=" << ll_stream_notation_sd(result) << LL_ENDL;
+ }
+ if (result.has("session"))
+ {
+ if (result.has("handle"))
+ {
+ if (!mAudioSession)
+ {
+ LL_WARNS("Voice") << "Message for session handle \"" << result["handle"] << "\" while session is not initiated." << LL_ENDL;
+ continue;
+ }
+ if (result["handle"] != mAudioSession->mHandle)
+ {
+ LL_WARNS("Voice") << "Message for session handle \"" << result["handle"] << "\" while waiting for \"" << mAudioSession->mHandle << "\"." << LL_ENDL;
+ continue;
+ }
+ }
+
+ std::string message = result["session"];
+
+ if (message == "removed")
+ {
+ LL_DEBUGS("Voice") << "session removed" << LL_ENDL;
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LEFT_CHANNEL);
+ break;
+ }
+ }
+ else if (result.has("login"))
+ {
+ std::string message = result["login"];
+ if (message == "account_logout")
+ {
+ LL_DEBUGS("Voice") << "logged out" << LL_ENDL;
+ mIsLoggedIn = false;
+ mRelogRequested = true;
+ break;
+ }
+ }
+ }
+
+ if (sShuttingDown)
+ {
+ return false;
+ }
+
+ mIsInChannel = false;
+ LL_DEBUGS("Voice") << "terminating at end of runSession" << LL_ENDL;
+ terminateAudioSession(true);
+
+ return true;
+}
+
+void LLWebRTCVoiceClient::recordingAndPlaybackMode()
+{
+ LL_INFOS("Voice") << "In voice capture/playback mode." << LL_ENDL;
+
+ while (true)
+ {
+ LLSD command;
+ do
+ {
+ command = llcoro::suspendUntilEventOn(mWebRTCPump);
+ LL_DEBUGS("Voice") << "event=" << ll_stream_notation_sd(command) << LL_ENDL;
+ } while (!command.has("recplay"));
+
+ if (command["recplay"].asString() == "quit")
+ {
+ mCaptureBufferMode = false;
+ break;
+ }
+ else if (command["recplay"].asString() == "record")
+ {
+ voiceRecordBuffer();
+ }
+ else if (command["recplay"].asString() == "playback")
+ {
+ voicePlaybackBuffer();
+ }
+ }
+
+ LL_INFOS("Voice") << "Leaving capture/playback mode." << LL_ENDL;
+ mCaptureBufferRecording = false;
+ mCaptureBufferRecorded = false;
+ mCaptureBufferPlaying = false;
+
+ return;
+}
+
+int LLWebRTCVoiceClient::voiceRecordBuffer()
+{
+ LLSD timeoutResult(LLSDMap("recplay", "stop"));
+
+ LL_INFOS("Voice") << "Recording voice buffer" << LL_ENDL;
+
+ LLSD result;
+
+ captureBufferRecordStartSendMessage();
+ notifyVoiceFontObservers();
+
+ do
+ {
+ result = llcoro::suspendUntilEventOnWithTimeout(mWebRTCPump, CAPTURE_BUFFER_MAX_TIME, timeoutResult);
+ LL_DEBUGS("Voice") << "event=" << ll_stream_notation_sd(result) << LL_ENDL;
+ } while (!result.has("recplay"));
+
+ mCaptureBufferRecorded = true;
+
+ captureBufferRecordStopSendMessage();
+ mCaptureBufferRecording = false;
+
+ // Update UI, should really use a separate callback.
+ notifyVoiceFontObservers();
+
+ return true;
+ /*TODO expand return to move directly into play*/
+}
+
+int LLWebRTCVoiceClient::voicePlaybackBuffer()
+{
+ LLSD timeoutResult(LLSDMap("recplay", "stop"));
+
+ LL_INFOS("Voice") << "Playing voice buffer" << LL_ENDL;
+
+ LLSD result;
+
+ do
+ {
+ captureBufferPlayStartSendMessage(mPreviewVoiceFont);
+
+ // Store the voice font being previewed, so that we know to restart if it changes.
+ mPreviewVoiceFontLast = mPreviewVoiceFont;
+
+ do
+ {
+ // Update UI, should really use a separate callback.
+ notifyVoiceFontObservers();
+
+ result = llcoro::suspendUntilEventOnWithTimeout(mWebRTCPump, CAPTURE_BUFFER_MAX_TIME, timeoutResult);
+ LL_DEBUGS("Voice") << "event=" << ll_stream_notation_sd(result) << LL_ENDL;
+ } while (!result.has("recplay"));
+
+ if (result["recplay"] == "playback")
+ continue; // restart playback... May be a font change.
+
+ break;
+ } while (true);
+
+ // Stop playing.
+ captureBufferPlayStopSendMessage();
+ mCaptureBufferPlaying = false;
+
+ // Update UI, should really use a separate callback.
+ notifyVoiceFontObservers();
+
+ return true;
+}
+
+
+bool LLWebRTCVoiceClient::performMicTuning()
+{
+ LL_INFOS("Voice") << "Entering voice tuning mode." << LL_ENDL;
+
+ mIsInTuningMode = false;
+
+ //---------------------------------------------------------------------
+ return true;
+}
+
+//=========================================================================
+
+void LLWebRTCVoiceClient::closeSocket(void)
+{
+ mSocket.reset();
+ sConnected = false;
+ mConnectorEstablished = false;
+ mAccountLoggedIn = false;
+}
+
+void LLWebRTCVoiceClient::loginSendMessage()
+{
+ std::ostringstream stream;
+
+ bool autoPostCrashDumps = gSavedSettings.getBOOL("WebRTCAutoPostCrashDumps");
+
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Account.Login.1\">"
+ << "<ConnectorHandle>" << LLWebRTCSecurity::getInstance()->connectorHandle() << "</ConnectorHandle>"
+ << "<AccountName>" << mAccountName << "</AccountName>"
+ << "<AccountPassword>" << mAccountPassword << "</AccountPassword>"
+ << "<AccountHandle>" << LLWebRTCSecurity::getInstance()->accountHandle() << "</AccountHandle>"
+ << "<AudioSessionAnswerMode>VerifyAnswer</AudioSessionAnswerMode>"
+ << "<EnableBuddiesAndPresence>false</EnableBuddiesAndPresence>"
+ << "<EnablePresencePersistence>0</EnablePresencePersistence>"
+ << "<BuddyManagementMode>Application</BuddyManagementMode>"
+ << "<ParticipantPropertyFrequency>5</ParticipantPropertyFrequency>"
+ << (autoPostCrashDumps?"<AutopostCrashDumps>true</AutopostCrashDumps>":"")
+ << "</Request>\n\n\n";
+
+ LL_INFOS("Voice") << "Attempting voice login" << LL_ENDL;
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::logout()
+{
+ // Ensure that we'll re-request provisioning before logging in again
+ mAccountPassword.clear();
+
+ logoutSendMessage();
+}
+
+void LLWebRTCVoiceClient::logoutSendMessage()
+{
+ if(mAccountLoggedIn)
+ {
+ LL_INFOS("Voice") << "Attempting voice logout" << LL_ENDL;
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Account.Logout.1\">"
+ << "<AccountHandle>" << LLWebRTCSecurity::getInstance()->accountHandle() << "</AccountHandle>"
+ << "</Request>"
+ << "\n\n\n";
+
+ mAccountLoggedIn = false;
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::sessionGroupCreateSendMessage()
+{
+ if(mAccountLoggedIn)
+ {
+ std::ostringstream stream;
+
+ LL_DEBUGS("Voice") << "creating session group" << LL_ENDL;
+
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"SessionGroup.Create.1\">"
+ << "<AccountHandle>" << LLWebRTCSecurity::getInstance()->accountHandle() << "</AccountHandle>"
+ << "<Type>Normal</Type>"
+ << "</Request>"
+ << "\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::sessionCreateSendMessage(const sessionStatePtr_t &session, bool startAudio, bool startText)
+{
+ S32 font_index = getVoiceFontIndex(session->mVoiceFontID);
+ LL_DEBUGS("Voice") << "Requesting create: " << session->mSIPURI
+ << " with voice font: " << session->mVoiceFontID << " (" << font_index << ")"
+ << LL_ENDL;
+
+ session->mCreateInProgress = true;
+ if(startAudio)
+ {
+ session->mMediaConnectInProgress = true;
+ }
+
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << session->mSIPURI << "\" action=\"Session.Create.1\">"
+ << "<AccountHandle>" << LLWebRTCSecurity::getInstance()->accountHandle() << "</AccountHandle>"
+ << "<URI>" << session->mSIPURI << "</URI>";
+
+ static const std::string allowed_chars =
+ "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz"
+ "0123456789"
+ "-._~";
+
+ if(!session->mHash.empty())
+ {
+ stream
+ << "<Password>" << LLURI::escape(session->mHash, allowed_chars) << "</Password>"
+ << "<PasswordHashAlgorithm>SHA1UserName</PasswordHashAlgorithm>";
+ }
+
+ stream
+ << "<ConnectAudio>" << (startAudio?"true":"false") << "</ConnectAudio>"
+ << "<ConnectText>" << (startText?"true":"false") << "</ConnectText>"
+ << "<VoiceFontID>" << font_index << "</VoiceFontID>"
+ << "<Name>" << mChannelName << "</Name>"
+ << "</Request>\n\n\n";
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::sessionGroupAddSessionSendMessage(const sessionStatePtr_t &session, bool startAudio, bool startText)
+{
+ LL_DEBUGS("Voice") << "Requesting create: " << session->mSIPURI << LL_ENDL;
+
+ S32 font_index = getVoiceFontIndex(session->mVoiceFontID);
+ LL_DEBUGS("Voice") << "With voice font: " << session->mVoiceFontID << " (" << font_index << ")" << LL_ENDL;
+
+ session->mCreateInProgress = true;
+ if(startAudio)
+ {
+ session->mMediaConnectInProgress = true;
+ }
+
+ std::string password;
+ if(!session->mHash.empty())
+ {
+ static const std::string allowed_chars =
+ "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz"
+ "0123456789"
+ "-._~"
+ ;
+ password = LLURI::escape(session->mHash, allowed_chars);
+ }
+
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << session->mSIPURI << "\" action=\"SessionGroup.AddSession.1\">"
+ << "<SessionGroupHandle>" << session->mGroupHandle << "</SessionGroupHandle>"
+ << "<URI>" << session->mSIPURI << "</URI>"
+ << "<Name>" << mChannelName << "</Name>"
+ << "<ConnectAudio>" << (startAudio?"true":"false") << "</ConnectAudio>"
+ << "<ConnectText>" << (startText?"true":"false") << "</ConnectText>"
+ << "<VoiceFontID>" << font_index << "</VoiceFontID>"
+ << "<Password>" << password << "</Password>"
+ << "<PasswordHashAlgorithm>SHA1UserName</PasswordHashAlgorithm>"
+ << "</Request>\n\n\n"
+ ;
+
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::sessionMediaConnectSendMessage(const sessionStatePtr_t &session)
+{
+ S32 font_index = getVoiceFontIndex(session->mVoiceFontID);
+ LL_DEBUGS("Voice") << "Connecting audio to session handle: " << session->mHandle
+ << " with voice font: " << session->mVoiceFontID << " (" << font_index << ")"
+ << LL_ENDL;
+
+ session->mMediaConnectInProgress = true;
+
+ std::ostringstream stream;
+
+ stream
+ << "<Request requestId=\"" << session->mHandle << "\" action=\"Session.MediaConnect.1\">"
+ << "<SessionGroupHandle>" << session->mGroupHandle << "</SessionGroupHandle>"
+ << "<SessionHandle>" << session->mHandle << "</SessionHandle>"
+ << "<VoiceFontID>" << font_index << "</VoiceFontID>"
+ << "<Media>Audio</Media>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::sessionTextConnectSendMessage(const sessionStatePtr_t &session)
+{
+ LL_DEBUGS("Voice") << "connecting text to session handle: " << session->mHandle << LL_ENDL;
+
+ std::ostringstream stream;
+
+ stream
+ << "<Request requestId=\"" << session->mHandle << "\" action=\"Session.TextConnect.1\">"
+ << "<SessionGroupHandle>" << session->mGroupHandle << "</SessionGroupHandle>"
+ << "<SessionHandle>" << session->mHandle << "</SessionHandle>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::sessionTerminate()
+{
+ mSessionTerminateRequested = true;
+}
+
+void LLWebRTCVoiceClient::requestRelog()
+{
+ mSessionTerminateRequested = true;
+ mRelogRequested = true;
+}
+
+
+void LLWebRTCVoiceClient::leaveAudioSession()
+{
+ if(mAudioSession)
+ {
+ LL_DEBUGS("Voice") << "leaving session: " << mAudioSession->mSIPURI << LL_ENDL;
+
+ if(!mAudioSession->mHandle.empty())
+ {
+
+#if RECORD_EVERYTHING
+ // HACK: for testing only
+ // Save looped recording
+ std::string savepath("/tmp/WebRTCrecording");
+ {
+ time_t now = time(NULL);
+ const size_t BUF_SIZE = 64;
+ char time_str[BUF_SIZE]; /* Flawfinder: ignore */
+
+ strftime(time_str, BUF_SIZE, "%Y-%m-%dT%H:%M:%SZ", gmtime(&now));
+ savepath += time_str;
+ }
+ recordingLoopSave(savepath);
+#endif
+
+ sessionMediaDisconnectSendMessage(mAudioSession);
+ }
+ else
+ {
+ LL_WARNS("Voice") << "called with no session handle" << LL_ENDL;
+ }
+ }
+ else
+ {
+ LL_WARNS("Voice") << "called with no active session" << LL_ENDL;
+ }
+ sessionTerminate();
+}
+
+void LLWebRTCVoiceClient::sessionTerminateSendMessage(const sessionStatePtr_t &session)
+{
+ std::ostringstream stream;
+
+ sessionGroupTerminateSendMessage(session);
+ return;
+ /*
+ LL_DEBUGS("Voice") << "Sending Session.Terminate with handle " << session->mHandle << LL_ENDL;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Session.Terminate.1\">"
+ << "<SessionHandle>" << session->mHandle << "</SessionHandle>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+ */
+}
+
+void LLWebRTCVoiceClient::sessionGroupTerminateSendMessage(const sessionStatePtr_t &session)
+{
+ std::ostringstream stream;
+
+ LL_DEBUGS("Voice") << "Sending SessionGroup.Terminate with handle " << session->mGroupHandle << LL_ENDL;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"SessionGroup.Terminate.1\">"
+ << "<SessionGroupHandle>" << session->mGroupHandle << "</SessionGroupHandle>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::sessionMediaDisconnectSendMessage(const sessionStatePtr_t &session)
+{
+ std::ostringstream stream;
+ sessionGroupTerminateSendMessage(session);
+ return;
+ /*
+ LL_DEBUGS("Voice") << "Sending Session.MediaDisconnect with handle " << session->mHandle << LL_ENDL;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Session.MediaDisconnect.1\">"
+ << "<SessionGroupHandle>" << session->mGroupHandle << "</SessionGroupHandle>"
+ << "<SessionHandle>" << session->mHandle << "</SessionHandle>"
+ << "<Media>Audio</Media>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+ */
+
+}
+
+void LLWebRTCVoiceClient::OnCaptureDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceList& render_devices)
+{
+ clearCaptureDevices();
+ for (auto &device : render_devices)
+ {
+ LLWebRTCVoiceClient::addCaptureDevice(LLVoiceDevice(device.display_name, device.id));
+ }
+ LLWebRTCVoiceClient::setDevicesListUpdated(true);
+}
+
+void LLWebRTCVoiceClient::clearCaptureDevices()
+{
+ LL_DEBUGS("Voice") << "called" << LL_ENDL;
+ mCaptureDevices.clear();
+}
+
+void LLWebRTCVoiceClient::addCaptureDevice(const LLVoiceDevice& device)
+{
+ LL_DEBUGS("Voice") << "display: '" << device.display_name << "' device: '" << device.full_name << "'" << LL_ENDL;
+ mCaptureDevices.push_back(device);
+}
+
+LLVoiceDeviceList& LLWebRTCVoiceClient::getCaptureDevices()
+{
+ return mCaptureDevices;
+}
+
+void LLWebRTCVoiceClient::setCaptureDevice(const std::string& name)
+{
+ bool inTuningMode = mIsInTuningMode;
+ if (inTuningMode)
+ {
+ tuningStop();
+ }
+ mWebRTCDeviceInterface->setCaptureDevice(name);
+ if (inTuningMode)
+ {
+ tuningStart();
+ }
+}
+void LLWebRTCVoiceClient::setDevicesListUpdated(bool state)
+{
+ mDevicesListUpdated = state;
+}
+
+void LLWebRTCVoiceClient::OnRenderDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceList &render_devices)
+{
+ clearRenderDevices();
+ for (auto &device : render_devices)
+ {
+ addRenderDevice(LLVoiceDevice(device.display_name, device.id));
+ }
+ setDevicesListUpdated(true);
+}
+
+void LLWebRTCVoiceClient::clearRenderDevices()
+{
+ LL_DEBUGS("Voice") << "called" << LL_ENDL;
+ mRenderDevices.clear();
+}
+
+void LLWebRTCVoiceClient::addRenderDevice(const LLVoiceDevice& device)
+{
+ LL_DEBUGS("Voice") << "display: '" << device.display_name << "' device: '" << device.full_name << "'" << LL_ENDL;
+ mRenderDevices.push_back(device);
+
+}
+
+LLVoiceDeviceList& LLWebRTCVoiceClient::getRenderDevices()
+{
+ return mRenderDevices;
+}
+
+void LLWebRTCVoiceClient::setRenderDevice(const std::string& name)
+{
+ mWebRTCDeviceInterface->setRenderDevice(name);
+}
+
+void LLWebRTCVoiceClient::tuningStart()
+{
+ if (!mIsInTuningMode)
+ {
+ mWebRTCDeviceInterface->setTuningMode(true);
+ mIsInTuningMode = true;
+ }
+}
+
+void LLWebRTCVoiceClient::tuningStop()
+{
+ if (mIsInTuningMode)
+ {
+ mWebRTCDeviceInterface->setTuningMode(false);
+ mIsInTuningMode = false;
+ }
+}
+
+bool LLWebRTCVoiceClient::inTuningMode()
+{
+ return mIsInTuningMode;
+}
+
+void LLWebRTCVoiceClient::tuningRenderStartSendMessage(const std::string& name, bool loop)
+{
+ mTuningAudioFile = name;
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Aux.RenderAudioStart.1\">"
+ << "<SoundFilePath>" << mTuningAudioFile << "</SoundFilePath>"
+ << "<Loop>" << (loop?"1":"0") << "</Loop>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::tuningRenderStopSendMessage()
+{
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Aux.RenderAudioStop.1\">"
+ << "<SoundFilePath>" << mTuningAudioFile << "</SoundFilePath>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::tuningCaptureStartSendMessage(int loop)
+{
+ LL_DEBUGS("Voice") << "sending CaptureAudioStart" << LL_ENDL;
+
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Aux.CaptureAudioStart.1\">"
+ << "<Duration>-1</Duration>"
+ << "<LoopToRenderDevice>" << loop << "</LoopToRenderDevice>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::tuningCaptureStopSendMessage()
+{
+ LL_DEBUGS("Voice") << "sending CaptureAudioStop" << LL_ENDL;
+
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Aux.CaptureAudioStop.1\">"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+
+ mTuningEnergy = 0.0f;
+}
+
+void LLWebRTCVoiceClient::tuningSetMicVolume(float volume)
+{
+ int scaled_volume = scale_mic_volume(volume);
+
+ if(scaled_volume != mTuningMicVolume)
+ {
+ mTuningMicVolume = scaled_volume;
+ mTuningMicVolumeDirty = true;
+ }
+}
+
+void LLWebRTCVoiceClient::tuningSetSpeakerVolume(float volume)
+{
+ int scaled_volume = scale_speaker_volume(volume);
+
+ if(scaled_volume != mTuningSpeakerVolume)
+ {
+ mTuningSpeakerVolume = scaled_volume;
+ mTuningSpeakerVolumeDirty = true;
+ }
+}
+
+float LLWebRTCVoiceClient::tuningGetEnergy(void)
+{
+ return mWebRTCDeviceInterface->getTuningMicrophoneEnergy();
+}
+
+bool LLWebRTCVoiceClient::deviceSettingsAvailable()
+{
+ bool result = true;
+
+ if(mRenderDevices.empty())
+ result = false;
+
+ return result;
+}
+bool LLWebRTCVoiceClient::deviceSettingsUpdated()
+{
+ bool updated = mDevicesListUpdated;
+ if (mDevicesListUpdated)
+ {
+ // a hot swap event or a polling of the audio devices has been parsed since the last redraw of the input and output device panel.
+ mDevicesListUpdated = false; // toggle the setting
+ }
+ return updated;
+}
+
+void LLWebRTCVoiceClient::refreshDeviceLists(bool clearCurrentList)
+{
+ if(clearCurrentList)
+ {
+ clearCaptureDevices();
+ clearRenderDevices();
+ }
+ mWebRTCDeviceInterface->refreshDevices();
+}
+
+void LLWebRTCVoiceClient::daemonDied()
+{
+ // The daemon died, so the connection is gone. Reset everything and start over.
+ LL_WARNS("Voice") << "Connection to WebRTC daemon lost. Resetting state."<< LL_ENDL;
+
+ //TODO: Try to relaunch the daemon
+}
+
+void LLWebRTCVoiceClient::giveUp()
+{
+ // All has failed. Clean up and stop trying.
+ LL_WARNS("Voice") << "Terminating Voice Service" << LL_ENDL;
+ closeSocket();
+ cleanUp();
+}
+
+static void oldSDKTransform (LLVector3 &left, LLVector3 &up, LLVector3 &at, LLVector3d &pos, LLVector3 &vel)
+{
+ F32 nat[3], nup[3], nl[3]; // the new at, up, left vectors and the new position and velocity
+// F32 nvel[3];
+ F64 npos[3];
+
+ // The original XML command was sent like this:
+ /*
+ << "<Position>"
+ << "<X>" << pos[VX] << "</X>"
+ << "<Y>" << pos[VZ] << "</Y>"
+ << "<Z>" << pos[VY] << "</Z>"
+ << "</Position>"
+ << "<Velocity>"
+ << "<X>" << mAvatarVelocity[VX] << "</X>"
+ << "<Y>" << mAvatarVelocity[VZ] << "</Y>"
+ << "<Z>" << mAvatarVelocity[VY] << "</Z>"
+ << "</Velocity>"
+ << "<AtOrientation>"
+ << "<X>" << l.mV[VX] << "</X>"
+ << "<Y>" << u.mV[VX] << "</Y>"
+ << "<Z>" << a.mV[VX] << "</Z>"
+ << "</AtOrientation>"
+ << "<UpOrientation>"
+ << "<X>" << l.mV[VZ] << "</X>"
+ << "<Y>" << u.mV[VY] << "</Y>"
+ << "<Z>" << a.mV[VZ] << "</Z>"
+ << "</UpOrientation>"
+ << "<LeftOrientation>"
+ << "<X>" << l.mV [VY] << "</X>"
+ << "<Y>" << u.mV [VZ] << "</Y>"
+ << "<Z>" << a.mV [VY] << "</Z>"
+ << "</LeftOrientation>";
+ */
+
+#if 1
+ // This was the original transform done when building the XML command
+ nat[0] = left.mV[VX];
+ nat[1] = up.mV[VX];
+ nat[2] = at.mV[VX];
+
+ nup[0] = left.mV[VZ];
+ nup[1] = up.mV[VY];
+ nup[2] = at.mV[VZ];
+
+ nl[0] = left.mV[VY];
+ nl[1] = up.mV[VZ];
+ nl[2] = at.mV[VY];
+
+ npos[0] = pos.mdV[VX];
+ npos[1] = pos.mdV[VZ];
+ npos[2] = pos.mdV[VY];
+
+// nvel[0] = vel.mV[VX];
+// nvel[1] = vel.mV[VZ];
+// nvel[2] = vel.mV[VY];
+
+ for(int i=0;i<3;++i) {
+ at.mV[i] = nat[i];
+ up.mV[i] = nup[i];
+ left.mV[i] = nl[i];
+ pos.mdV[i] = npos[i];
+ }
+
+ // This was the original transform done in the SDK
+ nat[0] = at.mV[2];
+ nat[1] = 0; // y component of at vector is always 0, this was up[2]
+ nat[2] = -1 * left.mV[2];
+
+ // We override whatever the application gives us
+ nup[0] = 0; // x component of up vector is always 0
+ nup[1] = 1; // y component of up vector is always 1
+ nup[2] = 0; // z component of up vector is always 0
+
+ nl[0] = at.mV[0];
+ nl[1] = 0; // y component of left vector is always zero, this was up[0]
+ nl[2] = -1 * left.mV[0];
+
+ npos[2] = pos.mdV[2] * -1.0;
+ npos[1] = pos.mdV[1];
+ npos[0] = pos.mdV[0];
+
+ for(int i=0;i<3;++i) {
+ at.mV[i] = nat[i];
+ up.mV[i] = nup[i];
+ left.mV[i] = nl[i];
+ pos.mdV[i] = npos[i];
+ }
+#else
+ // This is the compose of the two transforms (at least, that's what I'm trying for)
+ nat[0] = at.mV[VX];
+ nat[1] = 0; // y component of at vector is always 0, this was up[2]
+ nat[2] = -1 * up.mV[VZ];
+
+ // We override whatever the application gives us
+ nup[0] = 0; // x component of up vector is always 0
+ nup[1] = 1; // y component of up vector is always 1
+ nup[2] = 0; // z component of up vector is always 0
+
+ nl[0] = left.mV[VX];
+ nl[1] = 0; // y component of left vector is always zero, this was up[0]
+ nl[2] = -1 * left.mV[VY];
+
+ npos[0] = pos.mdV[VX];
+ npos[1] = pos.mdV[VZ];
+ npos[2] = pos.mdV[VY] * -1.0;
+
+ nvel[0] = vel.mV[VX];
+ nvel[1] = vel.mV[VZ];
+ nvel[2] = vel.mV[VY];
+
+ for(int i=0;i<3;++i) {
+ at.mV[i] = nat[i];
+ up.mV[i] = nup[i];
+ left.mV[i] = nl[i];
+ pos.mdV[i] = npos[i];
+ }
+
+#endif
+}
+
+void LLWebRTCVoiceClient::setHidden(bool hidden)
+{
+ mHidden = hidden;
+
+ if (mHidden && inSpatialChannel())
+ {
+ // get out of the channel entirely
+ leaveAudioSession();
+ }
+ else
+ {
+ sendPositionAndVolumeUpdate();
+ }
+}
+
+void LLWebRTCVoiceClient::sendPositionAndVolumeUpdate(void)
+{
+ std::ostringstream stream;
+
+ if (mSpatialCoordsDirty && inSpatialChannel())
+ {
+ LLVector3 l, u, a, vel;
+ LLVector3d pos;
+
+ mSpatialCoordsDirty = false;
+
+ // Always send both speaker and listener positions together.
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Session.Set3DPosition.1\">"
+ << "<SessionHandle>" << getAudioSessionHandle() << "</SessionHandle>";
+
+ stream << "<SpeakerPosition>";
+
+ LLMatrix3 avatarRot = mAvatarRot.getMatrix3();
+
+// LL_DEBUGS("Voice") << "Sending speaker position " << mAvatarPosition << LL_ENDL;
+ l = avatarRot.getLeftRow();
+ u = avatarRot.getUpRow();
+ a = avatarRot.getFwdRow();
+
+ pos = mAvatarPosition;
+ vel = mAvatarVelocity;
+
+ // SLIM SDK: the old SDK was doing a transform on the passed coordinates that the new one doesn't do anymore.
+ // The old transform is replicated by this function.
+ oldSDKTransform(l, u, a, pos, vel);
+
+ if (mHidden)
+ {
+ for (int i=0;i<3;++i)
+ {
+ pos.mdV[i] = VX_NULL_POSITION;
+ }
+ }
+
+ stream
+ << "<Position>"
+ << "<X>" << pos.mdV[VX] << "</X>"
+ << "<Y>" << pos.mdV[VY] << "</Y>"
+ << "<Z>" << pos.mdV[VZ] << "</Z>"
+ << "</Position>"
+ << "<Velocity>"
+ << "<X>" << vel.mV[VX] << "</X>"
+ << "<Y>" << vel.mV[VY] << "</Y>"
+ << "<Z>" << vel.mV[VZ] << "</Z>"
+ << "</Velocity>"
+ << "<AtOrientation>"
+ << "<X>" << a.mV[VX] << "</X>"
+ << "<Y>" << a.mV[VY] << "</Y>"
+ << "<Z>" << a.mV[VZ] << "</Z>"
+ << "</AtOrientation>"
+ << "<UpOrientation>"
+ << "<X>" << u.mV[VX] << "</X>"
+ << "<Y>" << u.mV[VY] << "</Y>"
+ << "<Z>" << u.mV[VZ] << "</Z>"
+ << "</UpOrientation>"
+ << "<LeftOrientation>"
+ << "<X>" << l.mV [VX] << "</X>"
+ << "<Y>" << l.mV [VY] << "</Y>"
+ << "<Z>" << l.mV [VZ] << "</Z>"
+ << "</LeftOrientation>"
+ ;
+
+ stream << "</SpeakerPosition>";
+
+ stream << "<ListenerPosition>";
+
+ LLVector3d earPosition;
+ LLVector3 earVelocity;
+ LLMatrix3 earRot;
+
+ switch(mEarLocation)
+ {
+ case earLocCamera:
+ default:
+ earPosition = mCameraPosition;
+ earVelocity = mCameraVelocity;
+ earRot = mCameraRot;
+ break;
+
+ case earLocAvatar:
+ earPosition = mAvatarPosition;
+ earVelocity = mAvatarVelocity;
+ earRot = avatarRot;
+ break;
+
+ case earLocMixed:
+ earPosition = mAvatarPosition;
+ earVelocity = mAvatarVelocity;
+ earRot = mCameraRot;
+ break;
+ }
+
+ l = earRot.getLeftRow();
+ u = earRot.getUpRow();
+ a = earRot.getFwdRow();
+
+ pos = earPosition;
+ vel = earVelocity;
+
+
+ oldSDKTransform(l, u, a, pos, vel);
+
+ if (mHidden)
+ {
+ for (int i=0;i<3;++i)
+ {
+ pos.mdV[i] = VX_NULL_POSITION;
+ }
+ }
+
+ stream
+ << "<Position>"
+ << "<X>" << pos.mdV[VX] << "</X>"
+ << "<Y>" << pos.mdV[VY] << "</Y>"
+ << "<Z>" << pos.mdV[VZ] << "</Z>"
+ << "</Position>"
+ << "<Velocity>"
+ << "<X>" << vel.mV[VX] << "</X>"
+ << "<Y>" << vel.mV[VY] << "</Y>"
+ << "<Z>" << vel.mV[VZ] << "</Z>"
+ << "</Velocity>"
+ << "<AtOrientation>"
+ << "<X>" << a.mV[VX] << "</X>"
+ << "<Y>" << a.mV[VY] << "</Y>"
+ << "<Z>" << a.mV[VZ] << "</Z>"
+ << "</AtOrientation>"
+ << "<UpOrientation>"
+ << "<X>" << u.mV[VX] << "</X>"
+ << "<Y>" << u.mV[VY] << "</Y>"
+ << "<Z>" << u.mV[VZ] << "</Z>"
+ << "</UpOrientation>"
+ << "<LeftOrientation>"
+ << "<X>" << l.mV [VX] << "</X>"
+ << "<Y>" << l.mV [VY] << "</Y>"
+ << "<Z>" << l.mV [VZ] << "</Z>"
+ << "</LeftOrientation>"
+ ;
+
+ stream << "</ListenerPosition>";
+
+ stream << "<ReqDispositionType>1</ReqDispositionType>"; //do not generate responses for update requests
+ stream << "</Request>\n\n\n";
+ }
+
+ if(mAudioSession && (mAudioSession->mVolumeDirty || mAudioSession->mMuteDirty))
+ {
+ participantMap::iterator iter = mAudioSession->mParticipantsByURI.begin();
+
+ mAudioSession->mVolumeDirty = false;
+ mAudioSession->mMuteDirty = false;
+
+ for(; iter != mAudioSession->mParticipantsByURI.end(); iter++)
+ {
+ participantStatePtr_t p(iter->second);
+
+ if(p->mVolumeDirty)
+ {
+ // Can't set volume/mute for yourself
+ if(!p->mIsSelf)
+ {
+ // scale from the range 0.0-1.0 to WebRTC volume in the range 0-100
+ S32 volume = ll_round(p->mVolume / VOLUME_SCALE_WEBRTC);
+ bool mute = p->mOnMuteList;
+
+ if(mute)
+ {
+ // SetParticipantMuteForMe doesn't work in p2p sessions.
+ // If we want the user to be muted, set their volume to 0 as well.
+ // This isn't perfect, but it will at least reduce their volume to a minimum.
+ volume = 0;
+ // Mark the current volume level as set to prevent incoming events
+ // changing it to 0, so that we can return to it when unmuting.
+ p->mVolumeSet = true;
+ }
+
+ if(volume == 0)
+ {
+ mute = true;
+ }
+
+ LL_DEBUGS("Voice") << "Setting volume/mute for avatar " << p->mAvatarID << " to " << volume << (mute?"/true":"/false") << LL_ENDL;
+
+ // SLIM SDK: Send both volume and mute commands.
+
+ // Send a "volume for me" command for the user.
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Session.SetParticipantVolumeForMe.1\">"
+ << "<SessionHandle>" << getAudioSessionHandle() << "</SessionHandle>"
+ << "<ParticipantURI>" << p->mURI << "</ParticipantURI>"
+ << "<Volume>" << volume << "</Volume>"
+ << "</Request>\n\n\n";
+
+ if(!mAudioSession->mIsP2P)
+ {
+ // Send a "mute for me" command for the user
+ // Doesn't work in P2P sessions
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Session.SetParticipantMuteForMe.1\">"
+ << "<SessionHandle>" << getAudioSessionHandle() << "</SessionHandle>"
+ << "<ParticipantURI>" << p->mURI << "</ParticipantURI>"
+ << "<Mute>" << (mute?"1":"0") << "</Mute>"
+ << "<Scope>Audio</Scope>"
+ << "</Request>\n\n\n";
+ }
+ }
+
+ p->mVolumeDirty = false;
+ }
+ }
+ }
+
+ std::string update(stream.str());
+ if(!update.empty())
+ {
+ LL_DEBUGS("VoiceUpdate") << "sending update " << update << LL_ENDL;
+ writeString(update);
+ }
+
+}
+
+void LLWebRTCVoiceClient::sendLocalAudioUpdates()
+{
+ // Check all of the dirty states and then send messages to those needing to be changed.
+ // Tuningmode hands its own mute settings.
+ std::ostringstream stream;
+
+ if (mMuteMicDirty && !mTuningMode)
+ {
+ mMuteMicDirty = false;
+
+ // Send a local mute command.
+
+ LL_INFOS("Voice") << "Sending MuteLocalMic command with parameter " << (mMuteMic ? "true" : "false") << LL_ENDL;
+
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Connector.MuteLocalMic.1\">"
+ << "<ConnectorHandle>" << LLWebRTCSecurity::getInstance()->connectorHandle() << "</ConnectorHandle>"
+ << "<Value>" << (mMuteMic ? "true" : "false") << "</Value>"
+ << "</Request>\n\n\n";
+
+ }
+
+ if (mSpeakerMuteDirty && !mTuningMode)
+ {
+ const char *muteval = ((mSpeakerVolume <= scale_speaker_volume(0)) ? "true" : "false");
+
+ mSpeakerMuteDirty = false;
+
+ LL_INFOS("Voice") << "Setting speaker mute to " << muteval << LL_ENDL;
+
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Connector.MuteLocalSpeaker.1\">"
+ << "<ConnectorHandle>" << LLWebRTCSecurity::getInstance()->connectorHandle() << "</ConnectorHandle>"
+ << "<Value>" << muteval << "</Value>"
+ << "</Request>\n\n\n";
+
+ }
+
+ if (mSpeakerVolumeDirty)
+ {
+ mSpeakerVolumeDirty = false;
+
+ LL_INFOS("Voice") << "Setting speaker volume to " << mSpeakerVolume << LL_ENDL;
+
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Connector.SetLocalSpeakerVolume.1\">"
+ << "<ConnectorHandle>" << LLWebRTCSecurity::getInstance()->connectorHandle() << "</ConnectorHandle>"
+ << "<Value>" << mSpeakerVolume << "</Value>"
+ << "</Request>\n\n\n";
+
+ }
+
+ if (mMicVolumeDirty)
+ {
+ mMicVolumeDirty = false;
+
+ LL_INFOS("Voice") << "Setting mic volume to " << mMicVolume << LL_ENDL;
+
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Connector.SetLocalMicVolume.1\">"
+ << "<ConnectorHandle>" << LLWebRTCSecurity::getInstance()->connectorHandle() << "</ConnectorHandle>"
+ << "<Value>" << mMicVolume << "</Value>"
+ << "</Request>\n\n\n";
+ }
+
+
+ if (!stream.str().empty())
+ {
+ writeString(stream.str());
+ }
+}
+
+/**
+ * Because of the recurring voice cutout issues (SL-15072) we are going to try
+ * to disable the automatic VAD (Voice Activity Detection) and set the associated
+ * parameters directly. We will expose them via Debug Settings and that should
+ * let us iterate on a collection of values that work for us. Hopefully!
+ *
+ * From the WebRTC Docs:
+ *
+ * VadAuto: A flag indicating if the automatic VAD is enabled (1) or disabled (0)
+ *
+ * VadHangover: The time (in milliseconds) that it takes
+ * for the VAD to switch back to silence from speech mode after the last speech
+ * frame has been detected.
+ *
+ * VadNoiseFloor: A dimensionless value between 0 and
+ * 20000 (default 576) that controls the maximum level at which the noise floor
+ * may be set at by the VAD's noise tracking. Too low of a value will make noise
+ * tracking ineffective (A value of 0 disables noise tracking and the VAD then
+ * relies purely on the sensitivity property). Too high of a value will make
+ * long speech classifiable as noise.
+ *
+ * VadSensitivity: A dimensionless value between 0 and
+ * 100, indicating the 'sensitivity of the VAD'. Increasing this value corresponds
+ * to decreasing the sensitivity of the VAD (i.e. '0' is most sensitive,
+ * while 100 is 'least sensitive')
+ */
+void LLWebRTCVoiceClient::setupVADParams(unsigned int vad_auto,
+ unsigned int vad_hangover,
+ unsigned int vad_noise_floor,
+ unsigned int vad_sensitivity)
+{
+ std::ostringstream stream;
+
+ LL_INFOS("Voice") << "Setting the automatic VAD to "
+ << (vad_auto ? "True" : "False")
+ << " and discrete values to"
+ << " VadHangover = " << vad_hangover
+ << ", VadSensitivity = " << vad_sensitivity
+ << ", VadNoiseFloor = " << vad_noise_floor
+ << LL_ENDL;
+
+ // Create a request to set the VAD parameters:
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Aux.SetVadProperties.1\">"
+ << "<VadAuto>" << vad_auto << "</VadAuto>"
+ << "<VadHangover>" << vad_hangover << "</VadHangover>"
+ << "<VadSensitivity>" << vad_sensitivity << "</VadSensitivity>"
+ << "<VadNoiseFloor>" << vad_noise_floor << "</VadNoiseFloor>"
+ << "</Request>\n\n\n";
+
+ if (!stream.str().empty())
+ {
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::onVADSettingsChange()
+{
+ // pick up the VAD variables (one of which was changed)
+ unsigned int vad_auto = gSavedSettings.getU32("WebRTCVadAuto");
+ unsigned int vad_hangover = gSavedSettings.getU32("WebRTCVadHangover");
+ unsigned int vad_noise_floor = gSavedSettings.getU32("WebRTCVadNoiseFloor");
+ unsigned int vad_sensitivity = gSavedSettings.getU32("WebRTCVadSensitivity");
+
+ // build a VAD params change request and send it to SLVoice
+ setupVADParams(vad_auto, vad_hangover, vad_noise_floor, vad_sensitivity);
+}
+
+/////////////////////////////
+// WebRTC Signaling Handlers
+void LLWebRTCVoiceClient::OnIceGatheringState(llwebrtc::LLWebRTCSignalingObserver::IceGatheringState state)
+{
+ LL_INFOS("Voice") << "Ice Gathering voice account. " << state << LL_ENDL;
+
+ if (state != llwebrtc::LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_COMPLETE)
+ {
+ return;
+ }
+ mIceCompleted = true;
+}
+
+void LLWebRTCVoiceClient::OnIceCandidate(const llwebrtc::LLWebRTCIceCandidate &candidate) {
+ mIceCandidates.push_back(candidate);
+}
+
+void LLWebRTCVoiceClient::processIceUpdates()
+{
+ LL_INFOS("Voice") << "Ice Gathering voice account." << LL_ENDL;
+ while ((!gAgent.getRegion() || !gAgent.getRegion()->capabilitiesReceived()) && !sShuttingDown)
+ {
+ LL_DEBUGS("Voice") << "no capabilities for voice provisioning; waiting " << LL_ENDL;
+ // *TODO* Pump a message for wake up.
+ llcoro::suspend();
+ }
+
+ if (sShuttingDown)
+ {
+ return;
+ }
+
+ std::string url = gAgent.getRegionCapability("VoiceSignalingRequest");
+
+ LL_DEBUGS("Voice") << "region ready to complete voice signaling; url=" << url << LL_ENDL;
+
+ LLCore::HttpRequest::policy_t httpPolicy(LLCore::HttpRequest::DEFAULT_POLICY_ID);
+ LLCoreHttpUtil::HttpCoroutineAdapter::ptr_t httpAdapter(new LLCoreHttpUtil::HttpCoroutineAdapter("voiceAccountProvision", httpPolicy));
+ LLCore::HttpRequest::ptr_t httpRequest(new LLCore::HttpRequest);
+ LLCore::HttpOptions::ptr_t httpOpts = LLCore::HttpOptions::ptr_t(new LLCore::HttpOptions);
+
+ LLSD body;
+ if (mIceCandidates.size())
+ {
+ LLSD body;
+
+ for (auto &ice_candidate : mIceCandidates)
+ {
+ LLSD body_candidate;
+ body_candidate["sdpMid"] = ice_candidate.sdp_mid;
+ body_candidate["sdpMLineIndex"] = ice_candidate.mline_index;
+ body_candidate["candidate"] = ice_candidate.candidate;
+ body["candidates"].append(body_candidate);
+ }
+ mIceCandidates.clear();
+ }
+ else if (mIceCompleted)
+ {
+ LLSD body_candidate;
+ body_candidate["completed"] = true;
+ body["candidate"] = body_candidate;
+ mIceCompleted = false;
+ }
+ else
+ {
+ return;
+ }
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLWebRTCVoiceClient::onIceUpdateComplete, this, _1),
+ boost::bind(&LLWebRTCVoiceClient::onIceUpdateError, this, 3, url, body, _1));
+}
+
+void LLWebRTCVoiceClient::onIceUpdateComplete(const LLSD& result)
+{
+ if (sShuttingDown)
+ {
+ return;
+ }
+}
+
+void LLWebRTCVoiceClient::onIceUpdateError(int retries, std::string url, LLSD body, const LLSD& result)
+{
+ if (sShuttingDown)
+ {
+ return;
+ }
+ LLCore::HttpRequest::policy_t httpPolicy(LLCore::HttpRequest::DEFAULT_POLICY_ID);
+ LLCoreHttpUtil::HttpCoroutineAdapter::ptr_t httpAdapter(new LLCoreHttpUtil::HttpCoroutineAdapter("voiceAccountProvision", httpPolicy));
+
+ if (retries >= 0)
+ {
+ LL_WARNS("Voice") << "Unable to complete ice trickling voice account, retrying." << LL_ENDL;
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLWebRTCVoiceClient::onIceUpdateComplete, this, _1),
+ boost::bind(&LLWebRTCVoiceClient::onIceUpdateError, this, retries - 1, url, body, _1));
+ }
+ else
+ {
+ LL_WARNS("Voice") << "Unable to complete ice trickling voice account, retrying." << LL_ENDL;
+ }
+}
+
+void LLWebRTCVoiceClient::OnOfferAvailable(const std::string &sdp)
+{
+ LL_INFOS("Voice") << "On Offer Available." << LL_ENDL;
+ mChannelSDP = sdp;
+}
+
+void LLWebRTCVoiceClient::OnAudioEstablished(llwebrtc::LLWebRTCAudioInterface * audio_interface)
+{
+ LL_INFOS("Voice") << "On AudioEstablished." << LL_ENDL;
+ mWebRTCAudioInterface = audio_interface;
+ audio_interface->setMute(true);
+}
+
+/////////////////////////////
+// Response/Event handlers
+
+void LLWebRTCVoiceClient::connectorCreateResponse(int statusCode, std::string &statusString, std::string &connectorHandle, std::string &versionID)
+{
+ LLSD result = LLSD::emptyMap();
+
+ if(statusCode == 0)
+ {
+ // Connector created, move forward.
+ if (connectorHandle == LLWebRTCSecurity::getInstance()->connectorHandle())
+ {
+ LL_INFOS("Voice") << "Voice connector succeeded, WebRTC SDK version is " << versionID << " connector handle " << connectorHandle << LL_ENDL;
+ mVoiceVersion.serverVersion = versionID;
+ mConnectorEstablished = true;
+ mTerminateDaemon = false;
+
+ result["connector"] = LLSD::Boolean(true);
+ }
+ else
+ {
+ // This shouldn't happen - we are somehow out of sync with SLVoice
+ // or possibly there are two things trying to run SLVoice at once
+ // or someone is trying to hack into it.
+ LL_WARNS("Voice") << "Connector returned wrong handle "
+ << "(" << connectorHandle << ")"
+ << " expected (" << LLWebRTCSecurity::getInstance()->connectorHandle() << ")"
+ << LL_ENDL;
+ result["connector"] = LLSD::Boolean(false);
+ // Give up.
+ mTerminateDaemon = true;
+ }
+ }
+ else if (statusCode == 10028) // web request timeout prior to login
+ {
+ // this is usually fatal, but a long timeout might work
+ result["connector"] = LLSD::Boolean(false);
+ result["retry"] = LLSD::Real(CONNECT_ATTEMPT_TIMEOUT);
+
+ LL_WARNS("Voice") << "Voice connection failed" << LL_ENDL;
+ }
+ else if (statusCode == 10006) // name resolution failure - a shorter retry may work
+ {
+ // some networks have slower DNS, but a short timeout might let it catch up
+ result["connector"] = LLSD::Boolean(false);
+ result["retry"] = LLSD::Real(CONNECT_DNS_TIMEOUT);
+
+ LL_WARNS("Voice") << "Voice connection DNS lookup failed" << LL_ENDL;
+ }
+ else // unknown failure - give up
+ {
+ LL_WARNS("Voice") << "Voice connection failure ("<< statusCode << "): " << statusString << LL_ENDL;
+ mTerminateDaemon = true;
+ result["connector"] = LLSD::Boolean(false);
+ }
+
+ mWebRTCPump.post(result);
+}
+
+void LLWebRTCVoiceClient::loginResponse(int statusCode, std::string &statusString, std::string &accountHandle, int numberOfAliases)
+{
+ LLSD result = LLSD::emptyMap();
+
+ LL_DEBUGS("Voice") << "Account.Login response (" << statusCode << "): " << statusString << LL_ENDL;
+
+ // Status code of 20200 means "bad password". We may want to special-case that at some point.
+
+ if ( statusCode == HTTP_UNAUTHORIZED )
+ {
+ // Login failure which is probably caused by the delay after a user's password being updated.
+ LL_INFOS("Voice") << "Account.Login response failure (" << statusCode << "): " << statusString << LL_ENDL;
+ result["login"] = LLSD::String("retry");
+ }
+ else if(statusCode != 0)
+ {
+ LL_WARNS("Voice") << "Account.Login response failure (" << statusCode << "): " << statusString << LL_ENDL;
+ result["login"] = LLSD::String("failed");
+ }
+ else
+ {
+ // Login succeeded, move forward.
+ mAccountLoggedIn = true;
+ mNumberOfAliases = numberOfAliases;
+ result["login"] = LLSD::String("response_ok");
+ }
+
+ mWebRTCPump.post(result);
+
+}
+
+void LLWebRTCVoiceClient::sessionCreateResponse(std::string &requestId, int statusCode, std::string &statusString, std::string &sessionHandle)
+{
+ sessionStatePtr_t session(findSessionBeingCreatedByURI(requestId));
+
+ if(session)
+ {
+ session->mCreateInProgress = false;
+ }
+
+ if(statusCode != 0)
+ {
+ LL_WARNS("Voice") << "Session.Create response failure (" << statusCode << "): " << statusString << LL_ENDL;
+ if(session)
+ {
+ session->mErrorStatusCode = statusCode;
+ session->mErrorStatusString = statusString;
+ if(session == mAudioSession)
+ {
+ LLSD WebRTCevent(LLSDMap("handle", LLSD::String(sessionHandle))
+ ("session", "failed")
+ ("reason", LLSD::Integer(statusCode)));
+
+ mWebRTCPump.post(WebRTCevent);
+ }
+ else
+ {
+ reapSession(session);
+ }
+ }
+ }
+ else
+ {
+ LL_INFOS("Voice") << "Session.Create response received (success), session handle is " << sessionHandle << LL_ENDL;
+ if(session)
+ {
+ setSessionHandle(session, sessionHandle);
+ }
+ LLSD WebRTCevent(LLSDMap("handle", LLSD::String(sessionHandle))
+ ("session", "created"));
+
+ mWebRTCPump.post(WebRTCevent);
+ }
+}
+
+void LLWebRTCVoiceClient::sessionGroupAddSessionResponse(std::string &requestId, int statusCode, std::string &statusString, std::string &sessionHandle)
+{
+ sessionStatePtr_t session(findSessionBeingCreatedByURI(requestId));
+
+ if(session)
+ {
+ session->mCreateInProgress = false;
+ }
+
+ if(statusCode != 0)
+ {
+ LL_WARNS("Voice") << "SessionGroup.AddSession response failure (" << statusCode << "): " << statusString << LL_ENDL;
+ if(session)
+ {
+ session->mErrorStatusCode = statusCode;
+ session->mErrorStatusString = statusString;
+ if(session == mAudioSession)
+ {
+ LLSD WebRTCevent(LLSDMap("handle", LLSD::String(sessionHandle))
+ ("session", "failed"));
+
+ mWebRTCPump.post(WebRTCevent);
+ }
+ else
+ {
+ reapSession(session);
+ }
+ }
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "SessionGroup.AddSession response received (success), session handle is " << sessionHandle << LL_ENDL;
+ if(session)
+ {
+ setSessionHandle(session, sessionHandle);
+ }
+
+ LLSD WebRTCevent(LLSDMap("handle", LLSD::String(sessionHandle))
+ ("session", "added"));
+
+ mWebRTCPump.post(WebRTCevent);
+
+ }
+}
+
+void LLWebRTCVoiceClient::sessionConnectResponse(std::string &requestId, int statusCode, std::string &statusString)
+{
+ sessionStatePtr_t session(findSession(requestId));
+ // 1026 is session already has media, somehow mediaconnect was called twice on the same session.
+ // set the session info to reflect that the user is already connected.
+ if (statusCode == 1026)
+ {
+ session->mVoiceActive = true;
+ session->mMediaConnectInProgress = false;
+ session->mMediaStreamState = streamStateConnected;
+ //session->mTextStreamState = streamStateConnected;
+ session->mErrorStatusCode = 0;
+ }
+ else if (statusCode != 0)
+ {
+ LL_WARNS("Voice") << "Session.Connect response failure (" << statusCode << "): " << statusString << LL_ENDL;
+ if (session)
+ {
+ session->mMediaConnectInProgress = false;
+ session->mErrorStatusCode = statusCode;
+ session->mErrorStatusString = statusString;
+ }
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "Session.Connect response received (success)" << LL_ENDL;
+ }
+}
+
+void LLWebRTCVoiceClient::logoutResponse(int statusCode, std::string &statusString)
+{
+ if(statusCode != 0)
+ {
+ LL_WARNS("Voice") << "Account.Logout response failure: " << statusString << LL_ENDL;
+ // Should this ever fail? do we care if it does?
+ }
+ LLSD WebRTCevent(LLSDMap("logout", LLSD::Boolean(true)));
+
+ mWebRTCPump.post(WebRTCevent);
+}
+
+void LLWebRTCVoiceClient::connectorShutdownResponse(int statusCode, std::string &statusString)
+{
+ if(statusCode != 0)
+ {
+ LL_WARNS("Voice") << "Connector.InitiateShutdown response failure: " << statusString << LL_ENDL;
+ // Should this ever fail? do we care if it does?
+ }
+
+ sConnected = false;
+ mShutdownComplete = true;
+
+ LLSD WebRTCevent(LLSDMap("connector", LLSD::Boolean(false)));
+
+ mWebRTCPump.post(WebRTCevent);
+}
+
+void LLWebRTCVoiceClient::sessionAddedEvent(
+ std::string &uriString,
+ std::string &alias,
+ std::string &sessionHandle,
+ std::string &sessionGroupHandle,
+ bool isChannel,
+ bool incoming,
+ std::string &nameString,
+ std::string &applicationString)
+{
+ sessionStatePtr_t session;
+
+ LL_INFOS("Voice") << "session " << uriString << ", alias " << alias << ", name " << nameString << " handle " << sessionHandle << LL_ENDL;
+
+ session = addSession(uriString, sessionHandle);
+ if(session)
+ {
+ session->mGroupHandle = sessionGroupHandle;
+ session->mIsChannel = isChannel;
+ session->mIncoming = incoming;
+ session->mAlias = alias;
+
+ // Generate a caller UUID -- don't need to do this for channels
+ if(!session->mIsChannel)
+ {
+ if(IDFromName(session->mSIPURI, session->mCallerID))
+ {
+ // Normal URI(base64-encoded UUID)
+ }
+ else if(!session->mAlias.empty() && IDFromName(session->mAlias, session->mCallerID))
+ {
+ // Wrong URI, but an alias is available. Stash the incoming URI as an alternate
+ session->mAlternateSIPURI = session->mSIPURI;
+
+ // and generate a proper URI from the ID.
+ setSessionURI(session, sipURIFromID(session->mCallerID));
+ }
+ else
+ {
+ LL_INFOS("Voice") << "Could not generate caller id from uri, using hash of uri " << session->mSIPURI << LL_ENDL;
+ session->mCallerID.generate(session->mSIPURI);
+ session->mSynthesizedCallerID = true;
+
+ // Can't look up the name in this case -- we have to extract it from the URI.
+ std::string namePortion = nameString;
+
+ // Some incoming names may be separated with an underscore instead of a space. Fix this.
+ LLStringUtil::replaceChar(namePortion, '_', ' ');
+
+ // Act like we just finished resolving the name (this stores it in all the right places)
+ avatarNameResolved(session->mCallerID, namePortion);
+ }
+
+ LL_INFOS("Voice") << "caller ID: " << session->mCallerID << LL_ENDL;
+
+ if(!session->mSynthesizedCallerID)
+ {
+ // If we got here, we don't have a proper name. Initiate a lookup.
+ lookupName(session->mCallerID);
+ }
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::sessionGroupAddedEvent(std::string &sessionGroupHandle)
+{
+ LL_DEBUGS("Voice") << "handle " << sessionGroupHandle << LL_ENDL;
+
+#if USE_SESSION_GROUPS
+ if(mMainSessionGroupHandle.empty())
+ {
+ // This is the first (i.e. "main") session group. Save its handle.
+ mMainSessionGroupHandle = sessionGroupHandle;
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "Already had a session group handle " << mMainSessionGroupHandle << LL_ENDL;
+ }
+#endif
+}
+
+void LLWebRTCVoiceClient::joinedAudioSession(const sessionStatePtr_t &session)
+{
+ LL_DEBUGS("Voice") << "Joined Audio Session" << LL_ENDL;
+ if(mAudioSession != session)
+ {
+ sessionStatePtr_t oldSession = mAudioSession;
+
+ mAudioSession = session;
+ mAudioSessionChanged = true;
+
+ // The old session may now need to be deleted.
+ reapSession(oldSession);
+ }
+
+ // This is the session we're joining.
+ if(mIsJoiningSession)
+ {
+ LLSD WebRTCevent(LLSDMap("handle", LLSD::String(session->mHandle))
+ ("session", "joined"));
+
+ mWebRTCPump.post(WebRTCevent);
+
+ if(!session->mIsChannel)
+ {
+ // this is a p2p session. Make sure the other end is added as a participant.
+ participantStatePtr_t participant(session->addParticipant(session->mSIPURI));
+ if(participant)
+ {
+ if(participant->mAvatarIDValid)
+ {
+ lookupName(participant->mAvatarID);
+ }
+ else if(!session->mName.empty())
+ {
+ participant->mDisplayName = session->mName;
+ avatarNameResolved(participant->mAvatarID, session->mName);
+ }
+
+ // TODO: Question: Do we need to set up mAvatarID/mAvatarIDValid here?
+ LL_INFOS("Voice") << "added caller as participant \"" << participant->mAccountName
+ << "\" (" << participant->mAvatarID << ")"<< LL_ENDL;
+ }
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::sessionRemovedEvent(
+ std::string &sessionHandle,
+ std::string &sessionGroupHandle)
+{
+ LL_INFOS("Voice") << "handle " << sessionHandle << LL_ENDL;
+
+ sessionStatePtr_t session(findSession(sessionHandle));
+ if(session)
+ {
+ leftAudioSession(session);
+
+ // This message invalidates the session's handle. Set it to empty.
+ clearSessionHandle(session);
+
+ // This also means that the session's session group is now empty.
+ // Terminate the session group so it doesn't leak.
+ sessionGroupTerminateSendMessage(session);
+
+ // Reset the media state (we now have no info)
+ session->mMediaStreamState = streamStateUnknown;
+ //session->mTextStreamState = streamStateUnknown;
+
+ // Conditionally delete the session
+ reapSession(session);
+ }
+ else
+ {
+ // Already reaped this session.
+ LL_DEBUGS("Voice") << "unknown session " << sessionHandle << " removed" << LL_ENDL;
+ }
+
+}
+
+void LLWebRTCVoiceClient::reapSession(const sessionStatePtr_t &session)
+{
+ if(session)
+ {
+
+ if(session->mCreateInProgress)
+ {
+ LL_DEBUGS("Voice") << "NOT deleting session " << session->mSIPURI << " (create in progress)" << LL_ENDL;
+ }
+ else if(session->mMediaConnectInProgress)
+ {
+ LL_DEBUGS("Voice") << "NOT deleting session " << session->mSIPURI << " (connect in progress)" << LL_ENDL;
+ }
+ else if(session == mAudioSession)
+ {
+ LL_DEBUGS("Voice") << "NOT deleting session " << session->mSIPURI << " (it's the current session)" << LL_ENDL;
+ }
+ else if(session == mNextAudioSession)
+ {
+ LL_DEBUGS("Voice") << "NOT deleting session " << session->mSIPURI << " (it's the next session)" << LL_ENDL;
+ }
+ else
+ {
+ // We don't have a reason to keep tracking this session, so just delete it.
+ LL_DEBUGS("Voice") << "deleting session " << session->mSIPURI << LL_ENDL;
+ deleteSession(session);
+ }
+ }
+ else
+ {
+// LL_DEBUGS("Voice") << "session is NULL" << LL_ENDL;
+ }
+}
+
+// Returns true if the session seems to indicate we've moved to a region on a different voice server
+bool LLWebRTCVoiceClient::sessionNeedsRelog(const sessionStatePtr_t &session)
+{
+ bool result = false;
+
+ if(session)
+ {
+ // Only make this check for spatial channels (so it won't happen for group or p2p calls)
+ if(session->mIsSpatial)
+ {
+ std::string::size_type atsign;
+
+ atsign = session->mSIPURI.find("@");
+
+ if(atsign != std::string::npos)
+ {
+ std::string urihost = session->mSIPURI.substr(atsign + 1);
+ }
+ }
+ }
+
+ return result;
+}
+
+void LLWebRTCVoiceClient::leftAudioSession(const sessionStatePtr_t &session)
+{
+ if (mAudioSession == session)
+ {
+ LLSD WebRTCevent(LLSDMap("handle", LLSD::String(session->mHandle))
+ ("session", "removed"));
+
+ mWebRTCPump.post(WebRTCevent);
+ }
+}
+
+void LLWebRTCVoiceClient::accountLoginStateChangeEvent(
+ std::string &accountHandle,
+ int statusCode,
+ std::string &statusString,
+ int state)
+{
+ LLSD levent = LLSD::emptyMap();
+
+ /*
+ According to Mike S., status codes for this event are:
+ login_state_logged_out=0,
+ login_state_logged_in = 1,
+ login_state_logging_in = 2,
+ login_state_logging_out = 3,
+ login_state_resetting = 4,
+ login_state_error=100
+ */
+
+ LL_DEBUGS("Voice") << "state change event: " << state << LL_ENDL;
+ switch(state)
+ {
+ case 1:
+ levent["login"] = LLSD::String("account_login");
+
+ mWebRTCPump.post(levent);
+ break;
+ case 2:
+ break;
+
+ case 3:
+ levent["login"] = LLSD::String("account_loggingOut");
+
+ mWebRTCPump.post(levent);
+ break;
+
+ case 4:
+ break;
+
+ case 100:
+ LL_WARNS("Voice") << "account state event error" << LL_ENDL;
+ break;
+
+ case 0:
+ levent["login"] = LLSD::String("account_logout");
+
+ mWebRTCPump.post(levent);
+ break;
+
+ default:
+ //Used to be a commented out warning
+ LL_WARNS("Voice") << "unknown account state event: " << state << LL_ENDL;
+ break;
+ }
+}
+
+void LLWebRTCVoiceClient::mediaCompletionEvent(std::string &sessionGroupHandle, std::string &mediaCompletionType)
+{
+ LLSD result;
+
+ if (mediaCompletionType == "AuxBufferAudioCapture")
+ {
+ mCaptureBufferRecording = false;
+ result["recplay"] = "end";
+ }
+ else if (mediaCompletionType == "AuxBufferAudioRender")
+ {
+ // Ignore all but the last stop event
+ if (--mPlayRequestCount <= 0)
+ {
+ mCaptureBufferPlaying = false;
+ result["recplay"] = "end";
+// result["recplay"] = "done";
+ }
+ }
+ else
+ {
+ LL_WARNS("Voice") << "Unknown MediaCompletionType: " << mediaCompletionType << LL_ENDL;
+ }
+
+ if (!result.isUndefined())
+ mWebRTCPump.post(result);
+}
+
+void LLWebRTCVoiceClient::mediaStreamUpdatedEvent(
+ std::string &sessionHandle,
+ std::string &sessionGroupHandle,
+ int statusCode,
+ std::string &statusString,
+ int state,
+ bool incoming)
+{
+ sessionStatePtr_t session(findSession(sessionHandle));
+
+ LL_DEBUGS("Voice") << "session " << sessionHandle << ", status code " << statusCode << ", string \"" << statusString << "\"" << LL_ENDL;
+
+ if(session)
+ {
+ // We know about this session
+
+ // Save the state for later use
+ session->mMediaStreamState = state;
+
+ switch(statusCode)
+ {
+ case 0:
+ case HTTP_OK:
+ // generic success
+ // Don't change the saved error code (it may have been set elsewhere)
+ break;
+ default:
+ // save the status code for later
+ session->mErrorStatusCode = statusCode;
+ break;
+ }
+
+ switch(state)
+ {
+ case streamStateDisconnecting:
+ case streamStateIdle:
+ // Standard "left audio session", WebRTC state 'disconnected'
+ session->mVoiceActive = false;
+ session->mMediaConnectInProgress = false;
+ leftAudioSession(session);
+ break;
+
+ case streamStateConnected:
+ session->mVoiceActive = true;
+ session->mMediaConnectInProgress = false;
+ joinedAudioSession(session);
+ case streamStateConnecting: // do nothing, but prevents a warning getting into the logs.
+ break;
+
+ case streamStateRinging:
+ if(incoming)
+ {
+ // Send the voice chat invite to the GUI layer
+ // TODO: Question: Should we correlate with the mute list here?
+ session->mIMSessionID = LLIMMgr::computeSessionID(IM_SESSION_P2P_INVITE, session->mCallerID);
+ session->mVoiceInvitePending = true;
+ if(session->mName.empty())
+ {
+ lookupName(session->mCallerID);
+ }
+ else
+ {
+ // Act like we just finished resolving the name
+ avatarNameResolved(session->mCallerID, session->mName);
+ }
+ }
+ break;
+
+ default:
+ LL_WARNS("Voice") << "unknown state " << state << LL_ENDL;
+ break;
+
+ }
+
+ }
+ else
+ {
+ // session disconnectintg and disconnected events arriving after we have already left the session.
+ LL_DEBUGS("Voice") << "session " << sessionHandle << " not found"<< LL_ENDL;
+ }
+}
+
+void LLWebRTCVoiceClient::participantAddedEvent(
+ std::string &sessionHandle,
+ std::string &sessionGroupHandle,
+ std::string &uriString,
+ std::string &alias,
+ std::string &nameString,
+ std::string &displayNameString,
+ int participantType)
+{
+ sessionStatePtr_t session(findSession(sessionHandle));
+ if(session)
+ {
+ participantStatePtr_t participant(session->addParticipant(uriString));
+ if(participant)
+ {
+ participant->mAccountName = nameString;
+
+ LL_DEBUGS("Voice") << "added participant \"" << participant->mAccountName
+ << "\" (" << participant->mAvatarID << ")"<< LL_ENDL;
+
+ if(participant->mAvatarIDValid)
+ {
+ // Initiate a lookup
+ lookupName(participant->mAvatarID);
+ }
+ else
+ {
+ // If we don't have a valid avatar UUID, we need to fill in the display name to make the active speakers floater work.
+ std::string namePortion = displayNameString;
+
+ if(namePortion.empty())
+ {
+ // Problems with both of the above, fall back to the account name
+ namePortion = nameString;
+ }
+
+ // Set the display name (which is a hint to the active speakers window not to do its own lookup)
+ participant->mDisplayName = namePortion;
+ avatarNameResolved(participant->mAvatarID, namePortion);
+ }
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::participantRemovedEvent(
+ std::string &sessionHandle,
+ std::string &sessionGroupHandle,
+ std::string &uriString,
+ std::string &alias,
+ std::string &nameString)
+{
+ sessionStatePtr_t session(findSession(sessionHandle));
+ if(session)
+ {
+ participantStatePtr_t participant(session->findParticipant(uriString));
+ if(participant)
+ {
+ session->removeParticipant(participant);
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "unknown participant " << uriString << LL_ENDL;
+ }
+ }
+ else
+ {
+ // a late arriving event on a session we have already left.
+ LL_DEBUGS("Voice") << "unknown session " << sessionHandle << LL_ENDL;
+ }
+}
+
+
+void LLWebRTCVoiceClient::participantUpdatedEvent(
+ std::string &sessionHandle,
+ std::string &sessionGroupHandle,
+ std::string &uriString,
+ std::string &alias,
+ bool isModeratorMuted,
+ bool isSpeaking,
+ int volume,
+ F32 energy)
+{
+ sessionStatePtr_t session(findSession(sessionHandle));
+ if(session)
+ {
+ participantStatePtr_t participant(session->findParticipant(uriString));
+
+ if(participant)
+ {
+ //LL_INFOS("Voice") << "Participant Update for " << participant->mDisplayName << LL_ENDL;
+
+ participant->mIsSpeaking = isSpeaking;
+ participant->mIsModeratorMuted = isModeratorMuted;
+
+ // SLIM SDK: convert range: ensure that energy is set to zero if is_speaking is false
+ if (isSpeaking)
+ {
+ participant->mSpeakingTimeout.reset();
+ participant->mPower = energy;
+ }
+ else
+ {
+ participant->mPower = 0.0f;
+ }
+
+ // Ignore incoming volume level if it has been explicitly set, or there
+ // is a volume or mute change pending.
+ if ( !participant->mVolumeSet && !participant->mVolumeDirty)
+ {
+ participant->mVolume = (F32)volume * VOLUME_SCALE_WEBRTC;
+ }
+
+ // *HACK: mantipov: added while working on EXT-3544
+ /*
+ Sometimes LLVoiceClient::participantUpdatedEvent callback is called BEFORE
+ LLViewerChatterBoxSessionAgentListUpdates::post() sometimes AFTER.
+
+ participantUpdatedEvent updates voice participant state in particular participantState::mIsModeratorMuted
+ Originally we wanted to update session Speaker Manager to fire LLSpeakerVoiceModerationEvent to fix the EXT-3544 bug.
+ Calling of the LLSpeakerMgr::update() method was added into LLIMMgr::processAgentListUpdates.
+
+ But in case participantUpdatedEvent() is called after LLViewerChatterBoxSessionAgentListUpdates::post()
+ voice participant mIsModeratorMuted is changed after speakers are updated in Speaker Manager
+ and event is not fired.
+
+ So, we have to call LLSpeakerMgr::update() here.
+ */
+ LLVoiceChannel* voice_cnl = LLVoiceChannel::getCurrentVoiceChannel();
+
+ // ignore session ID of local chat
+ if (voice_cnl && voice_cnl->getSessionID().notNull())
+ {
+ LLSpeakerMgr* speaker_manager = LLIMModel::getInstance()->getSpeakerManager(voice_cnl->getSessionID());
+ if (speaker_manager)
+ {
+ speaker_manager->update(true);
+
+ // also initialize voice moderate_mode depend on Agent's participant. See EXT-6937.
+ // *TODO: remove once a way to request the current voice channel moderation mode is implemented.
+ if (gAgent.getID() == participant->mAvatarID)
+ {
+ speaker_manager->initVoiceModerateMode();
+ }
+ }
+ }
+ }
+ else
+ {
+ LL_WARNS("Voice") << "unknown participant: " << uriString << LL_ENDL;
+ }
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "unknown session " << sessionHandle << LL_ENDL;
+ }
+}
+
+void LLWebRTCVoiceClient::messageEvent(
+ std::string &sessionHandle,
+ std::string &uriString,
+ std::string &alias,
+ std::string &messageHeader,
+ std::string &messageBody,
+ std::string &applicationString)
+{
+ LL_DEBUGS("Voice") << "Message event, session " << sessionHandle << " from " << uriString << LL_ENDL;
+// LL_DEBUGS("Voice") << " header " << messageHeader << ", body: \n" << messageBody << LL_ENDL;
+
+ LL_INFOS("Voice") << "WebRTC raw message:" << std::endl << messageBody << LL_ENDL;
+
+ if(messageHeader.find(HTTP_CONTENT_TEXT_HTML) != std::string::npos)
+ {
+ std::string message;
+
+ {
+ const std::string startMarker = "<body";
+ const std::string startMarker2 = ">";
+ const std::string endMarker = "</body>";
+ const std::string startSpan = "<span";
+ const std::string endSpan = "</span>";
+ std::string::size_type start;
+ std::string::size_type end;
+
+ // Default to displaying the raw string, so the message gets through.
+ message = messageBody;
+
+ // Find the actual message text within the XML fragment
+ start = messageBody.find(startMarker);
+ start = messageBody.find(startMarker2, start);
+ end = messageBody.find(endMarker);
+
+ if(start != std::string::npos)
+ {
+ start += startMarker2.size();
+
+ if(end != std::string::npos)
+ end -= start;
+
+ message.assign(messageBody, start, end);
+ }
+ else
+ {
+ // Didn't find a <body>, try looking for a <span> instead.
+ start = messageBody.find(startSpan);
+ start = messageBody.find(startMarker2, start);
+ end = messageBody.find(endSpan);
+
+ if(start != std::string::npos)
+ {
+ start += startMarker2.size();
+
+ if(end != std::string::npos)
+ end -= start;
+
+ message.assign(messageBody, start, end);
+ }
+ }
+ }
+
+// LL_DEBUGS("Voice") << " raw message = \n" << message << LL_ENDL;
+
+ // strip formatting tags
+ {
+ std::string::size_type start;
+ std::string::size_type end;
+
+ while((start = message.find('<')) != std::string::npos)
+ {
+ if((end = message.find('>', start + 1)) != std::string::npos)
+ {
+ // Strip out the tag
+ message.erase(start, (end + 1) - start);
+ }
+ else
+ {
+ // Avoid an infinite loop
+ break;
+ }
+ }
+ }
+
+ // Decode ampersand-escaped chars
+ {
+ std::string::size_type mark = 0;
+
+ // The text may contain text encoded with &lt;, &gt;, and &amp;
+ mark = 0;
+ while((mark = message.find("&lt;", mark)) != std::string::npos)
+ {
+ message.replace(mark, 4, "<");
+ mark += 1;
+ }
+
+ mark = 0;
+ while((mark = message.find("&gt;", mark)) != std::string::npos)
+ {
+ message.replace(mark, 4, ">");
+ mark += 1;
+ }
+
+ mark = 0;
+ while((mark = message.find("&amp;", mark)) != std::string::npos)
+ {
+ message.replace(mark, 5, "&");
+ mark += 1;
+ }
+ }
+
+ // strip leading/trailing whitespace (since we always seem to get a couple newlines)
+ LLStringUtil::trim(message);
+
+// LL_DEBUGS("Voice") << " stripped message = \n" << message << LL_ENDL;
+
+ sessionStatePtr_t session(findSession(sessionHandle));
+ if(session)
+ {
+ bool is_do_not_disturb = gAgent.isDoNotDisturb();
+ bool is_muted = LLMuteList::getInstance()->isMuted(session->mCallerID, session->mName, LLMute::flagTextChat);
+ bool is_linden = LLMuteList::isLinden(session->mName);
+ LLChat chat;
+
+ chat.mMuted = is_muted && !is_linden;
+
+ if(!chat.mMuted)
+ {
+ chat.mFromID = session->mCallerID;
+ chat.mFromName = session->mName;
+ chat.mSourceType = CHAT_SOURCE_AGENT;
+
+ if(is_do_not_disturb && !is_linden)
+ {
+ // TODO: Question: Return do not disturb mode response here? Or maybe when session is started instead?
+ }
+
+ LL_DEBUGS("Voice") << "adding message, name " << session->mName << " session " << session->mIMSessionID << ", target " << session->mCallerID << LL_ENDL;
+ LLIMMgr::getInstance()->addMessage(session->mIMSessionID,
+ session->mCallerID,
+ session->mName.c_str(),
+ message.c_str(),
+ false,
+ LLStringUtil::null, // default arg
+ IM_NOTHING_SPECIAL, // default arg
+ 0, // default arg
+ LLUUID::null, // default arg
+ LLVector3::zero); // default arg
+ }
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::sessionNotificationEvent(std::string &sessionHandle, std::string &uriString, std::string &notificationType)
+{
+ sessionStatePtr_t session(findSession(sessionHandle));
+
+ if(session)
+ {
+ participantStatePtr_t participant(session->findParticipant(uriString));
+ if(participant)
+ {
+ if (!stricmp(notificationType.c_str(), "Typing"))
+ {
+ // Other end started typing
+ // TODO: The proper way to add a typing notification seems to be LLIMMgr::processIMTypingStart().
+ // It requires some info for the message, which we don't have here.
+ }
+ else if (!stricmp(notificationType.c_str(), "NotTyping"))
+ {
+ // Other end stopped typing
+ // TODO: The proper way to remove a typing notification seems to be LLIMMgr::processIMTypingStop().
+ // It requires some info for the message, which we don't have here.
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "Unknown notification type " << notificationType << "for participant " << uriString << " in session " << session->mSIPURI << LL_ENDL;
+ }
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "Unknown participant " << uriString << " in session " << session->mSIPURI << LL_ENDL;
+ }
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "Unknown session handle " << sessionHandle << LL_ENDL;
+ }
+}
+
+void LLWebRTCVoiceClient::voiceServiceConnectionStateChangedEvent(int statusCode, std::string &statusString, std::string &build_id)
+{
+ // We don't generally need to process this. However, one occurence is when we first connect, and so it is the
+ // earliest opportunity to learn what we're connected to.
+ if (statusCode)
+ {
+ LL_WARNS("Voice") << "VoiceServiceConnectionStateChangedEvent statusCode: " << statusCode <<
+ "statusString: " << statusString << LL_ENDL;
+ return;
+ }
+ if (build_id.empty())
+ {
+ return;
+ }
+ mVoiceVersion.mBuildVersion = build_id;
+}
+
+void LLWebRTCVoiceClient::auxAudioPropertiesEvent(F32 energy)
+{
+ LL_DEBUGS("VoiceEnergy") << "got energy " << energy << LL_ENDL;
+ mTuningEnergy = energy;
+}
+
+void LLWebRTCVoiceClient::muteListChanged()
+{
+ // The user's mute list has been updated. Go through the current participant list and sync it with the mute list.
+ if(mAudioSession)
+ {
+ participantMap::iterator iter = mAudioSession->mParticipantsByURI.begin();
+
+ for(; iter != mAudioSession->mParticipantsByURI.end(); iter++)
+ {
+ participantStatePtr_t p(iter->second);
+
+ // Check to see if this participant is on the mute list already
+ if(p->updateMuteState())
+ mAudioSession->mVolumeDirty = true;
+ }
+ }
+}
+
+/////////////////////////////
+// Managing list of participants
+LLWebRTCVoiceClient::participantState::participantState(const std::string &uri) :
+ mURI(uri),
+ mPTT(false),
+ mIsSpeaking(false),
+ mIsModeratorMuted(false),
+ mLastSpokeTimestamp(0.f),
+ mPower(0.f),
+ mVolume(LLVoiceClient::VOLUME_DEFAULT),
+ mUserVolume(0),
+ mOnMuteList(false),
+ mVolumeSet(false),
+ mVolumeDirty(false),
+ mAvatarIDValid(false),
+ mIsSelf(false)
+{
+}
+
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::sessionState::addParticipant(const std::string &uri)
+{
+ participantStatePtr_t result;
+ bool useAlternateURI = false;
+
+ // Note: this is mostly the body of LLWebRTCVoiceClient::sessionState::findParticipant(), but since we need to know if it
+ // matched the alternate SIP URI (so we can add it properly), we need to reproduce it here.
+ {
+ participantMap::iterator iter = mParticipantsByURI.find(uri);
+
+ if(iter == mParticipantsByURI.end())
+ {
+ if(!mAlternateSIPURI.empty() && (uri == mAlternateSIPURI))
+ {
+ // This is a p2p session (probably with the SLIM client) with an alternate URI for the other participant.
+ // Use mSIPURI instead, since it will be properly encoded.
+ iter = mParticipantsByURI.find(mSIPURI);
+ useAlternateURI = true;
+ }
+ }
+
+ if(iter != mParticipantsByURI.end())
+ {
+ result = iter->second;
+ }
+ }
+
+ if(!result)
+ {
+ // participant isn't already in one list or the other.
+ result.reset(new participantState(useAlternateURI?mSIPURI:uri));
+ mParticipantsByURI.insert(participantMap::value_type(result->mURI, result));
+ mParticipantsChanged = true;
+
+ // Try to do a reverse transform on the URI to get the GUID back.
+ {
+ LLUUID id;
+ if(LLWebRTCVoiceClient::getInstance()->IDFromName(result->mURI, id))
+ {
+ result->mAvatarIDValid = true;
+ result->mAvatarID = id;
+ }
+ else
+ {
+ // Create a UUID by hashing the URI, but do NOT set mAvatarIDValid.
+ // This indicates that the ID will not be in the name cache.
+ result->mAvatarID.generate(uri);
+ }
+ }
+
+ if(result->updateMuteState())
+ {
+ mMuteDirty = true;
+ }
+
+ mParticipantsByUUID.insert(participantUUIDMap::value_type(result->mAvatarID, result));
+
+ if (LLSpeakerVolumeStorage::getInstance()->getSpeakerVolume(result->mAvatarID, result->mVolume))
+ {
+ result->mVolumeDirty = true;
+ mVolumeDirty = true;
+ }
+
+ LL_DEBUGS("Voice") << "participant \"" << result->mURI << "\" added." << LL_ENDL;
+ }
+
+ return result;
+}
+
+bool LLWebRTCVoiceClient::participantState::updateMuteState()
+{
+ bool result = false;
+
+ bool isMuted = LLMuteList::getInstance()->isMuted(mAvatarID, LLMute::flagVoiceChat);
+ if(mOnMuteList != isMuted)
+ {
+ mOnMuteList = isMuted;
+ mVolumeDirty = true;
+ result = true;
+ }
+ return result;
+}
+
+bool LLWebRTCVoiceClient::participantState::isAvatar()
+{
+ return mAvatarIDValid;
+}
+
+void LLWebRTCVoiceClient::sessionState::removeParticipant(const LLWebRTCVoiceClient::participantStatePtr_t &participant)
+{
+ if(participant)
+ {
+ participantMap::iterator iter = mParticipantsByURI.find(participant->mURI);
+ participantUUIDMap::iterator iter2 = mParticipantsByUUID.find(participant->mAvatarID);
+
+ LL_DEBUGS("Voice") << "participant \"" << participant->mURI << "\" (" << participant->mAvatarID << ") removed." << LL_ENDL;
+
+ if(iter == mParticipantsByURI.end())
+ {
+ LL_WARNS("Voice") << "Internal error: participant " << participant->mURI << " not in URI map" << LL_ENDL;
+ }
+ else if(iter2 == mParticipantsByUUID.end())
+ {
+ LL_WARNS("Voice") << "Internal error: participant ID " << participant->mAvatarID << " not in UUID map" << LL_ENDL;
+ }
+ else if(iter->second != iter2->second)
+ {
+ LL_WARNS("Voice") << "Internal error: participant mismatch!" << LL_ENDL;
+ }
+ else
+ {
+ mParticipantsByURI.erase(iter);
+ mParticipantsByUUID.erase(iter2);
+
+ mParticipantsChanged = true;
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::sessionState::removeAllParticipants()
+{
+ LL_DEBUGS("Voice") << "called" << LL_ENDL;
+
+ while(!mParticipantsByURI.empty())
+ {
+ removeParticipant(mParticipantsByURI.begin()->second);
+ }
+
+ if(!mParticipantsByUUID.empty())
+ {
+ LL_WARNS("Voice") << "Internal error: empty URI map, non-empty UUID map" << LL_ENDL;
+ }
+}
+
+/*static*/
+void LLWebRTCVoiceClient::sessionState::VerifySessions()
+{
+ std::set<wptr_t>::iterator it = mSession.begin();
+ while (it != mSession.end())
+ {
+ if ((*it).expired())
+ {
+ LL_WARNS("Voice") << "Expired session found! removing" << LL_ENDL;
+ it = mSession.erase(it);
+ }
+ else
+ ++it;
+ }
+}
+
+
+void LLWebRTCVoiceClient::getParticipantList(std::set<LLUUID> &participants)
+{
+ if(mAudioSession)
+ {
+ for(participantUUIDMap::iterator iter = mAudioSession->mParticipantsByUUID.begin();
+ iter != mAudioSession->mParticipantsByUUID.end();
+ iter++)
+ {
+ participants.insert(iter->first);
+ }
+ }
+}
+
+bool LLWebRTCVoiceClient::isParticipant(const LLUUID &speaker_id)
+{
+ if(mAudioSession)
+ {
+ return (mAudioSession->mParticipantsByUUID.find(speaker_id) != mAudioSession->mParticipantsByUUID.end());
+ }
+ return false;
+}
+
+
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::sessionState::findParticipant(const std::string &uri)
+{
+ participantStatePtr_t result;
+
+ participantMap::iterator iter = mParticipantsByURI.find(uri);
+
+ if(iter == mParticipantsByURI.end())
+ {
+ if(!mAlternateSIPURI.empty() && (uri == mAlternateSIPURI))
+ {
+ // This is a p2p session (probably with the SLIM client) with an alternate URI for the other participant.
+ // Look up the other URI
+ iter = mParticipantsByURI.find(mSIPURI);
+ }
+ }
+
+ if(iter != mParticipantsByURI.end())
+ {
+ result = iter->second;
+ }
+
+ return result;
+}
+
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::sessionState::findParticipantByID(const LLUUID& id)
+{
+ participantStatePtr_t result;
+ participantUUIDMap::iterator iter = mParticipantsByUUID.find(id);
+
+ if(iter != mParticipantsByUUID.end())
+ {
+ result = iter->second;
+ }
+
+ return result;
+}
+
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::findParticipantByID(const LLUUID& id)
+{
+ participantStatePtr_t result;
+
+ if(mAudioSession)
+ {
+ result = mAudioSession->findParticipantByID(id);
+ }
+
+ return result;
+}
+
+
+
+// Check for parcel boundary crossing
+bool LLWebRTCVoiceClient::checkParcelChanged(bool update)
+{
+ LLViewerRegion *region = gAgent.getRegion();
+ LLParcel *parcel = LLViewerParcelMgr::getInstance()->getAgentParcel();
+
+ if(region && parcel)
+ {
+ S32 parcelLocalID = parcel->getLocalID();
+ std::string regionName = region->getName();
+
+ // LL_DEBUGS("Voice") << "Region name = \"" << regionName << "\", parcel local ID = " << parcelLocalID << ", cap URI = \"" << capURI << "\"" << LL_ENDL;
+
+ // The region name starts out empty and gets filled in later.
+ // Also, the cap gets filled in a short time after the region cross, but a little too late for our purposes.
+ // If either is empty, wait for the next time around.
+ if(!regionName.empty())
+ {
+ if((parcelLocalID != mCurrentParcelLocalID) || (regionName != mCurrentRegionName))
+ {
+ // We have changed parcels. Initiate a parcel channel lookup.
+ if (update)
+ {
+ mCurrentParcelLocalID = parcelLocalID;
+ mCurrentRegionName = regionName;
+ }
+ return true;
+ }
+ }
+ }
+ return false;
+}
+
+bool LLWebRTCVoiceClient::switchChannel(
+ std::string uri,
+ bool spatial,
+ bool no_reconnect,
+ bool is_p2p,
+ std::string hash)
+{
+ bool needsSwitch = !mIsInChannel;
+
+ if (mIsInChannel)
+ {
+ if (mSessionTerminateRequested)
+ {
+ // If a terminate has been requested, we need to compare against where the URI we're already headed to.
+ if(mNextAudioSession)
+ {
+ if(mNextAudioSession->mSIPURI != uri)
+ needsSwitch = true;
+ }
+ else
+ {
+ // mNextAudioSession is null -- this probably means we're on our way back to spatial.
+ if(!uri.empty())
+ {
+ // We do want to process a switch in this case.
+ needsSwitch = true;
+ }
+ }
+ }
+ else
+ {
+ // Otherwise, compare against the URI we're in now.
+ if(mAudioSession)
+ {
+ if(mAudioSession->mSIPURI != uri)
+ {
+ needsSwitch = true;
+ }
+ }
+ else
+ {
+ if(!uri.empty())
+ {
+ // mAudioSession is null -- it's not clear what case would cause this.
+ // For now, log it as a warning and see if it ever crops up.
+ LL_WARNS("Voice") << "No current audio session... Forcing switch" << LL_ENDL;
+ needsSwitch = true;
+ }
+ }
+ }
+ }
+
+ if(needsSwitch)
+ {
+ if(uri.empty())
+ {
+ // Leave any channel we may be in
+ LL_DEBUGS("Voice") << "leaving channel" << LL_ENDL;
+
+ sessionStatePtr_t oldSession = mNextAudioSession;
+ mNextAudioSession.reset();
+
+ // The old session may now need to be deleted.
+ reapSession(oldSession);
+
+ // If voice was on, turn it off
+ if (LLVoiceClient::getInstance()->getUserPTTState())
+ {
+ LLVoiceClient::getInstance()->setUserPTTState(false);
+ }
+
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_VOICE_DISABLED);
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "switching to channel " << uri << LL_ENDL;
+
+ mNextAudioSession = addSession(uri);
+ mNextAudioSession->mHash = hash;
+ mNextAudioSession->mIsSpatial = spatial;
+ mNextAudioSession->mReconnect = !no_reconnect;
+ mNextAudioSession->mIsP2P = is_p2p;
+ }
+
+ if (mIsInChannel)
+ {
+ // If we're already in a channel, or if we're joining one, terminate
+ // so we can rejoin with the new session data.
+ sessionTerminate();
+ }
+ }
+
+ return needsSwitch;
+}
+
+void LLWebRTCVoiceClient::joinSession(const sessionStatePtr_t &session)
+{
+ mNextAudioSession = session;
+
+ if (mIsInChannel)
+ {
+ // If we're already in a channel, or if we're joining one, terminate
+ // so we can rejoin with the new session data.
+ sessionTerminate();
+ }
+}
+
+void LLWebRTCVoiceClient::setNonSpatialChannel(
+ const std::string &uri,
+ const std::string &credentials)
+{
+ switchChannel(uri, false, false, false, credentials);
+}
+
+bool LLWebRTCVoiceClient::setSpatialChannel(
+ const std::string &uri,
+ const std::string &credentials)
+{
+ mSpatialSessionURI = uri;
+ mSpatialSessionCredentials = credentials;
+ mAreaVoiceDisabled = mSpatialSessionURI.empty();
+
+ LL_DEBUGS("Voice") << "got spatial channel uri: \"" << uri << "\"" << LL_ENDL;
+
+ if((mIsInChannel && mAudioSession && !(mAudioSession->mIsSpatial)) || (mNextAudioSession && !(mNextAudioSession->mIsSpatial)))
+ {
+ // User is in a non-spatial chat or joining a non-spatial chat. Don't switch channels.
+ LL_INFOS("Voice") << "in non-spatial chat, not switching channels" << LL_ENDL;
+ return false;
+ }
+ else
+ {
+ return switchChannel(mSpatialSessionURI, true, false, false, mSpatialSessionCredentials);
+ }
+}
+
+void LLWebRTCVoiceClient::callUser(const LLUUID &uuid)
+{
+ std::string userURI = sipURIFromID(uuid);
+
+ switchChannel(userURI, false, true, true);
+}
+
+#if 0
+// WebRTC text IMs are not in use.
+LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::startUserIMSession(const LLUUID &uuid)
+{
+ // Figure out if a session with the user already exists
+ sessionStatePtr_t session(findSession(uuid));
+ if(!session)
+ {
+ // No session with user, need to start one.
+ std::string uri = sipURIFromID(uuid);
+ session = addSession(uri);
+
+ llassert(session);
+ if (!session)
+ return session;
+
+ session->mIsSpatial = false;
+ session->mReconnect = false;
+ session->mIsP2P = true;
+ session->mCallerID = uuid;
+ }
+
+ if(session->mHandle.empty())
+ {
+ // Session isn't active -- start it up.
+ sessionCreateSendMessage(session, false, false);
+ }
+ else
+ {
+ // Session is already active -- start up text.
+ sessionTextConnectSendMessage(session);
+ }
+
+ return session;
+}
+#endif
+
+void LLWebRTCVoiceClient::endUserIMSession(const LLUUID &uuid)
+{
+#if 0
+ // WebRTC text IMs are not in use.
+
+ // Figure out if a session with the user exists
+ sessionStatePtr_t session(findSession(uuid));
+ if(session)
+ {
+ // found the session
+ if(!session->mHandle.empty())
+ {
+ // sessionTextDisconnectSendMessage(session); // a SLim leftover, not used any more.
+ }
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "Session not found for participant ID " << uuid << LL_ENDL;
+ }
+#endif
+}
+bool LLWebRTCVoiceClient::isValidChannel(std::string &sessionHandle)
+{
+ return(findSession(sessionHandle) != NULL);
+
+}
+bool LLWebRTCVoiceClient::answerInvite(std::string &sessionHandle)
+{
+ // this is only ever used to answer incoming p2p call invites.
+
+ sessionStatePtr_t session(findSession(sessionHandle));
+ if(session)
+ {
+ session->mIsSpatial = false;
+ session->mReconnect = false;
+ session->mIsP2P = true;
+
+ joinSession(session);
+ return true;
+ }
+
+ return false;
+}
+
+bool LLWebRTCVoiceClient::isVoiceWorking() const
+{
+
+ //Added stateSessionTerminated state to avoid problems with call in parcels with disabled voice (EXT-4758)
+ // Condition with joining spatial num was added to take into account possible problems with connection to voice
+ // server(EXT-4313). See bug descriptions and comments for MAX_NORMAL_JOINING_SPATIAL_NUM for more info.
+ return (mSpatialJoiningNum < MAX_NORMAL_JOINING_SPATIAL_NUM) && mIsProcessingChannels;
+// return (mSpatialJoiningNum < MAX_NORMAL_JOINING_SPATIAL_NUM) && (stateLoggedIn <= mState) && (mState <= stateSessionTerminated);
+}
+
+// Returns true if the indicated participant in the current audio session is really an SL avatar.
+// Currently this will be false only for PSTN callers into group chats, and PSTN p2p calls.
+BOOL LLWebRTCVoiceClient::isParticipantAvatar(const LLUUID &id)
+{
+ BOOL result = TRUE;
+ sessionStatePtr_t session(findSession(id));
+
+ if(session)
+ {
+ // this is a p2p session with the indicated caller, or the session with the specified UUID.
+ if(session->mSynthesizedCallerID)
+ result = FALSE;
+ }
+ else
+ {
+ // Didn't find a matching session -- check the current audio session for a matching participant
+ if(mAudioSession)
+ {
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ result = participant->isAvatar();
+ }
+ }
+ }
+
+ return result;
+}
+
+// Returns true if calling back the session URI after the session has closed is possible.
+// Currently this will be false only for PSTN P2P calls.
+BOOL LLWebRTCVoiceClient::isSessionCallBackPossible(const LLUUID &session_id)
+{
+ BOOL result = TRUE;
+ sessionStatePtr_t session(findSession(session_id));
+
+ if(session != NULL)
+ {
+ result = session->isCallBackPossible();
+ }
+
+ return result;
+}
+
+// Returns true if the session can accept text IM's.
+// Currently this will be false only for PSTN P2P calls.
+BOOL LLWebRTCVoiceClient::isSessionTextIMPossible(const LLUUID &session_id)
+{
+ bool result = TRUE;
+ sessionStatePtr_t session(findSession(session_id));
+
+ if(session != NULL)
+ {
+ result = session->isTextIMPossible();
+ }
+
+ return result;
+}
+
+
+void LLWebRTCVoiceClient::declineInvite(std::string &sessionHandle)
+{
+ sessionStatePtr_t session(findSession(sessionHandle));
+ if(session)
+ {
+ sessionMediaDisconnectSendMessage(session);
+ }
+}
+
+void LLWebRTCVoiceClient::leaveNonSpatialChannel()
+{
+ LL_DEBUGS("Voice") << "Request to leave spacial channel." << LL_ENDL;
+
+ // Make sure we don't rejoin the current session.
+ sessionStatePtr_t oldNextSession(mNextAudioSession);
+ mNextAudioSession.reset();
+
+ // Most likely this will still be the current session at this point, but check it anyway.
+ reapSession(oldNextSession);
+
+ verifySessionState();
+
+ sessionTerminate();
+}
+
+std::string LLWebRTCVoiceClient::getCurrentChannel()
+{
+ std::string result;
+
+ if (mIsInChannel && !mSessionTerminateRequested)
+ {
+ result = getAudioSessionURI();
+ }
+
+ return result;
+}
+
+bool LLWebRTCVoiceClient::inProximalChannel()
+{
+ bool result = false;
+
+ if (mIsInChannel && !mSessionTerminateRequested)
+ {
+ result = inSpatialChannel();
+ }
+
+ return result;
+}
+
+std::string LLWebRTCVoiceClient::sipURIFromID(const LLUUID &id)
+{
+ std::string result;
+ result = "sip:";
+ result += nameFromID(id);
+ result += "@";
+
+ return result;
+}
+
+std::string LLWebRTCVoiceClient::nameFromAvatar(LLVOAvatar *avatar)
+{
+ std::string result;
+ if(avatar)
+ {
+ result = nameFromID(avatar->getID());
+ }
+ return result;
+}
+
+std::string LLWebRTCVoiceClient::nameFromID(const LLUUID &uuid)
+{
+ std::string result;
+
+ if (uuid.isNull()) {
+ //WebRTC, the uuid emtpy look for the mURIString and return that instead.
+ //result.assign(uuid.mURIStringName);
+ LLStringUtil::replaceChar(result, '_', ' ');
+ return result;
+ }
+ // Prepending this apparently prevents conflicts with reserved names inside the WebRTC code.
+ result = "x";
+
+ // Base64 encode and replace the pieces of base64 that are less compatible
+ // with e-mail local-parts.
+ // See RFC-4648 "Base 64 Encoding with URL and Filename Safe Alphabet"
+ result += LLBase64::encode(uuid.mData, UUID_BYTES);
+ LLStringUtil::replaceChar(result, '+', '-');
+ LLStringUtil::replaceChar(result, '/', '_');
+
+ // If you need to transform a GUID to this form on the Mac OS X command line, this will do so:
+ // echo -n x && (echo e669132a-6c43-4ee1-a78d-6c82fff59f32 |xxd -r -p |openssl base64|tr '/+' '_-')
+
+ // The reverse transform can be done with:
+ // echo 'x5mkTKmxDTuGnjWyC__WfMg==' |cut -b 2- -|tr '_-' '/+' |openssl base64 -d|xxd -p
+
+ return result;
+}
+
+bool LLWebRTCVoiceClient::IDFromName(const std::string inName, LLUUID &uuid)
+{
+ bool result = false;
+
+ // SLIM SDK: The "name" may actually be a SIP URI such as: "sip:xFnPP04IpREWNkuw1cOXlhw==@bhr.WebRTC.com"
+ // If it is, convert to a bare name before doing the transform.
+ std::string name;
+
+ // Doesn't look like a SIP URI, assume it's an actual name.
+ if(name.empty())
+ name = inName;
+
+ // This will only work if the name is of the proper form.
+ // As an example, the account name for Monroe Linden (UUID 1673cfd3-8229-4445-8d92-ec3570e5e587) is:
+ // "xFnPP04IpREWNkuw1cOXlhw=="
+
+ if((name.size() == 25) && (name[0] == 'x') && (name[23] == '=') && (name[24] == '='))
+ {
+ // The name appears to have the right form.
+
+ // Reverse the transforms done by nameFromID
+ std::string temp = name;
+ LLStringUtil::replaceChar(temp, '-', '+');
+ LLStringUtil::replaceChar(temp, '_', '/');
+
+ U8 rawuuid[UUID_BYTES + 1];
+ int len = apr_base64_decode_binary(rawuuid, temp.c_str() + 1);
+ if(len == UUID_BYTES)
+ {
+ // The decode succeeded. Stuff the bits into the result's UUID
+ memcpy(uuid.mData, rawuuid, UUID_BYTES);
+ result = true;
+ }
+ }
+
+ if(!result)
+ {
+ // WebRTC: not a standard account name, just copy the URI name mURIString field
+ // and hope for the best. bpj
+ uuid.setNull(); // WebRTC, set the uuid field to nulls
+ }
+
+ return result;
+}
+
+std::string LLWebRTCVoiceClient::displayNameFromAvatar(LLVOAvatar *avatar)
+{
+ return avatar->getFullname();
+}
+
+bool LLWebRTCVoiceClient::inSpatialChannel(void)
+{
+ bool result = false;
+
+ if(mAudioSession)
+ {
+ result = mAudioSession->mIsSpatial;
+ }
+
+ return result;
+}
+
+std::string LLWebRTCVoiceClient::getAudioSessionURI()
+{
+ std::string result;
+
+ if(mAudioSession)
+ result = mAudioSession->mSIPURI;
+
+ return result;
+}
+
+std::string LLWebRTCVoiceClient::getAudioSessionHandle()
+{
+ std::string result;
+
+ if(mAudioSession)
+ result = mAudioSession->mHandle;
+
+ return result;
+}
+
+
+/////////////////////////////
+// Sending updates of current state
+
+void LLWebRTCVoiceClient::enforceTether(void)
+{
+ LLVector3d tethered = mCameraRequestedPosition;
+
+ // constrain 'tethered' to within 50m of mAvatarPosition.
+ {
+ F32 max_dist = 50.0f;
+ LLVector3d camera_offset = mCameraRequestedPosition - mAvatarPosition;
+ F32 camera_distance = (F32)camera_offset.magVec();
+ if(camera_distance > max_dist)
+ {
+ tethered = mAvatarPosition +
+ (max_dist / camera_distance) * camera_offset;
+ }
+ }
+
+ if(dist_vec_squared(mCameraPosition, tethered) > 0.01)
+ {
+ mCameraPosition = tethered;
+ mSpatialCoordsDirty = true;
+ }
+}
+
+void LLWebRTCVoiceClient::updatePosition(void)
+{
+
+ LLViewerRegion *region = gAgent.getRegion();
+ if(region && isAgentAvatarValid())
+ {
+ LLMatrix3 rot;
+ LLVector3d pos;
+ LLQuaternion qrot;
+
+ // TODO: If camera and avatar velocity are actually used by the voice system, we could compute them here...
+ // They're currently always set to zero.
+
+ // Send the current camera position to the voice code
+ rot.setRows(LLViewerCamera::getInstance()->getAtAxis(), LLViewerCamera::getInstance()->getLeftAxis (), LLViewerCamera::getInstance()->getUpAxis());
+ pos = gAgent.getRegion()->getPosGlobalFromRegion(LLViewerCamera::getInstance()->getOrigin());
+
+ LLWebRTCVoiceClient::getInstance()->setCameraPosition(
+ pos, // position
+ LLVector3::zero, // velocity
+ rot); // rotation matrix
+
+ // Send the current avatar position to the voice code
+ qrot = gAgentAvatarp->getRootJoint()->getWorldRotation();
+ pos = gAgentAvatarp->getPositionGlobal();
+
+ // TODO: Can we get the head offset from outside the LLVOAvatar?
+ // pos += LLVector3d(mHeadOffset);
+ pos += LLVector3d(0.f, 0.f, 1.f);
+
+ LLWebRTCVoiceClient::getInstance()->setAvatarPosition(
+ pos, // position
+ LLVector3::zero, // velocity
+ qrot); // rotation matrix
+ }
+}
+
+void LLWebRTCVoiceClient::setCameraPosition(const LLVector3d &position, const LLVector3 &velocity, const LLMatrix3 &rot)
+{
+ mCameraRequestedPosition = position;
+
+ if(mCameraVelocity != velocity)
+ {
+ mCameraVelocity = velocity;
+ mSpatialCoordsDirty = true;
+ }
+
+ if(mCameraRot != rot)
+ {
+ mCameraRot = rot;
+ mSpatialCoordsDirty = true;
+ }
+}
+
+void LLWebRTCVoiceClient::setAvatarPosition(const LLVector3d &position, const LLVector3 &velocity, const LLQuaternion &rot)
+{
+ if(dist_vec_squared(mAvatarPosition, position) > 0.01)
+ {
+ mAvatarPosition = position;
+ mSpatialCoordsDirty = true;
+ }
+
+ if(mAvatarVelocity != velocity)
+ {
+ mAvatarVelocity = velocity;
+ mSpatialCoordsDirty = true;
+ }
+
+ // If the two rotations are not exactly equal test their dot product
+ // to get the cos of the angle between them.
+ // If it is too small, don't update.
+ F32 rot_cos_diff = llabs(dot(mAvatarRot, rot));
+ if ((mAvatarRot != rot) && (rot_cos_diff < MINUSCULE_ANGLE_COS))
+ {
+ mAvatarRot = rot;
+ mSpatialCoordsDirty = true;
+ }
+}
+
+bool LLWebRTCVoiceClient::channelFromRegion(LLViewerRegion *region, std::string &name)
+{
+ bool result = false;
+
+ if(region)
+ {
+ name = region->getName();
+ }
+
+ if(!name.empty())
+ result = true;
+
+ return result;
+}
+
+void LLWebRTCVoiceClient::leaveChannel(void)
+{
+ if (mIsInChannel)
+ {
+ LL_DEBUGS("Voice") << "leaving channel for teleport/logout" << LL_ENDL;
+ mChannelName.clear();
+ sessionTerminate();
+ }
+}
+
+void LLWebRTCVoiceClient::setMuteMic(bool muted)
+{
+ if (mWebRTCAudioInterface)
+ {
+ mWebRTCAudioInterface->setMute(muted);
+ }
+}
+
+void LLWebRTCVoiceClient::setVoiceEnabled(bool enabled)
+{
+ LL_DEBUGS("Voice")
+ << "( " << (enabled ? "enabled" : "disabled") << " )"
+ << " was "<< (mVoiceEnabled ? "enabled" : "disabled")
+ << " coro "<< (mIsCoroutineActive ? "active" : "inactive")
+ << LL_ENDL;
+
+ if (enabled != mVoiceEnabled)
+ {
+ // TODO: Refactor this so we don't call into LLVoiceChannel, but simply
+ // use the status observer
+ mVoiceEnabled = enabled;
+ LLVoiceClientStatusObserver::EStatusType status;
+
+ if (enabled)
+ {
+ LL_DEBUGS("Voice") << "enabling" << LL_ENDL;
+ LLVoiceChannel::getCurrentVoiceChannel()->activate();
+ status = LLVoiceClientStatusObserver::STATUS_VOICE_ENABLED;
+
+ if (!mIsCoroutineActive)
+ {
+ LLCoros::instance().launch("LLWebRTCVoiceClient::voiceControlCoro",
+ boost::bind(&LLWebRTCVoiceClient::voiceControlCoro, LLWebRTCVoiceClient::getInstance()));
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "coro should be active.. not launching" << LL_ENDL;
+ }
+ }
+ else
+ {
+ // Turning voice off looses your current channel -- this makes sure the UI isn't out of sync when you re-enable it.
+ LLVoiceChannel::getCurrentVoiceChannel()->deactivate();
+ gAgent.setVoiceConnected(false);
+ status = LLVoiceClientStatusObserver::STATUS_VOICE_DISABLED;
+ }
+
+ notifyStatusObservers(status);
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << " no-op" << LL_ENDL;
+ }
+}
+
+bool LLWebRTCVoiceClient::voiceEnabled()
+{
+ return gSavedSettings.getBOOL("EnableVoiceChat") &&
+ !gSavedSettings.getBOOL("CmdLineDisableVoice") &&
+ !gNonInteractive;
+}
+
+void LLWebRTCVoiceClient::setLipSyncEnabled(BOOL enabled)
+{
+ mLipSyncEnabled = enabled;
+}
+
+BOOL LLWebRTCVoiceClient::lipSyncEnabled()
+{
+
+ if ( mVoiceEnabled )
+ {
+ return mLipSyncEnabled;
+ }
+ else
+ {
+ return FALSE;
+ }
+}
+
+
+void LLWebRTCVoiceClient::setEarLocation(S32 loc)
+{
+ if(mEarLocation != loc)
+ {
+ LL_DEBUGS("Voice") << "Setting mEarLocation to " << loc << LL_ENDL;
+
+ mEarLocation = loc;
+ mSpatialCoordsDirty = true;
+ }
+}
+
+void LLWebRTCVoiceClient::setVoiceVolume(F32 volume)
+{
+ int scaled_volume = scale_speaker_volume(volume);
+
+ if(scaled_volume != mSpeakerVolume)
+ {
+ int min_volume = scale_speaker_volume(0);
+ if((scaled_volume == min_volume) || (mSpeakerVolume == min_volume))
+ {
+ mSpeakerMuteDirty = true;
+ }
+
+ mSpeakerVolume = scaled_volume;
+ mSpeakerVolumeDirty = true;
+ }
+}
+
+void LLWebRTCVoiceClient::setMicGain(F32 volume)
+{
+ int scaled_volume = scale_mic_volume(volume);
+
+ if(scaled_volume != mMicVolume)
+ {
+ mMicVolume = scaled_volume;
+ mMicVolumeDirty = true;
+ }
+}
+
+/////////////////////////////
+// Accessors for data related to nearby speakers
+BOOL LLWebRTCVoiceClient::getVoiceEnabled(const LLUUID& id)
+{
+ BOOL result = FALSE;
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ // I'm not sure what the semantics of this should be.
+ // For now, if we have any data about the user that came through the chat channel, assume they're voice-enabled.
+ result = TRUE;
+ }
+
+ return result;
+}
+
+std::string LLWebRTCVoiceClient::getDisplayName(const LLUUID& id)
+{
+ std::string result;
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ result = participant->mDisplayName;
+ }
+
+ return result;
+}
+
+
+
+BOOL LLWebRTCVoiceClient::getIsSpeaking(const LLUUID& id)
+{
+ BOOL result = FALSE;
+
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ if (participant->mSpeakingTimeout.getElapsedTimeF32() > SPEAKING_TIMEOUT)
+ {
+ participant->mIsSpeaking = FALSE;
+ }
+ result = participant->mIsSpeaking;
+ }
+
+ return result;
+}
+
+BOOL LLWebRTCVoiceClient::getIsModeratorMuted(const LLUUID& id)
+{
+ BOOL result = FALSE;
+
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ result = participant->mIsModeratorMuted;
+ }
+
+ return result;
+}
+
+F32 LLWebRTCVoiceClient::getCurrentPower(const LLUUID& id)
+{
+ F32 result = 0;
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ result = participant->mPower;
+ }
+
+ return result;
+}
+
+BOOL LLWebRTCVoiceClient::getUsingPTT(const LLUUID& id)
+{
+ BOOL result = FALSE;
+
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ // I'm not sure what the semantics of this should be.
+ // Does "using PTT" mean they're configured with a push-to-talk button?
+ // For now, we know there's no PTT mechanism in place, so nobody is using it.
+ }
+
+ return result;
+}
+
+BOOL LLWebRTCVoiceClient::getOnMuteList(const LLUUID& id)
+{
+ BOOL result = FALSE;
+
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ result = participant->mOnMuteList;
+ }
+
+ return result;
+}
+
+// External accessors.
+F32 LLWebRTCVoiceClient::getUserVolume(const LLUUID& id)
+{
+ // Minimum volume will be returned for users with voice disabled
+ F32 result = LLVoiceClient::VOLUME_MIN;
+
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ result = participant->mVolume;
+
+ // Enable this when debugging voice slider issues. It's way to spammy even for debug-level logging.
+ // LL_DEBUGS("Voice") << "mVolume = " << result << " for " << id << LL_ENDL;
+ }
+
+ return result;
+}
+
+void LLWebRTCVoiceClient::setUserVolume(const LLUUID& id, F32 volume)
+{
+ if(mAudioSession)
+ {
+ participantStatePtr_t participant(findParticipantByID(id));
+ if (participant && !participant->mIsSelf)
+ {
+ if (!is_approx_equal(volume, LLVoiceClient::VOLUME_DEFAULT))
+ {
+ // Store this volume setting for future sessions if it has been
+ // changed from the default
+ LLSpeakerVolumeStorage::getInstance()->storeSpeakerVolume(id, volume);
+ }
+ else
+ {
+ // Remove stored volume setting if it is returned to the default
+ LLSpeakerVolumeStorage::getInstance()->removeSpeakerVolume(id);
+ }
+
+ participant->mVolume = llclamp(volume, LLVoiceClient::VOLUME_MIN, LLVoiceClient::VOLUME_MAX);
+ participant->mVolumeDirty = true;
+ mAudioSession->mVolumeDirty = true;
+ }
+ }
+}
+
+std::string LLWebRTCVoiceClient::getGroupID(const LLUUID& id)
+{
+ std::string result;
+
+ participantStatePtr_t participant(findParticipantByID(id));
+ if(participant)
+ {
+ result = participant->mGroupID;
+ }
+
+ return result;
+}
+
+BOOL LLWebRTCVoiceClient::getAreaVoiceDisabled()
+{
+ return mAreaVoiceDisabled;
+}
+
+void LLWebRTCVoiceClient::recordingLoopStart(int seconds, int deltaFramesPerControlFrame)
+{
+// LL_DEBUGS("Voice") << "sending SessionGroup.ControlRecording (Start)" << LL_ENDL;
+
+ if(!mMainSessionGroupHandle.empty())
+ {
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"SessionGroup.ControlRecording.1\">"
+ << "<SessionGroupHandle>" << mMainSessionGroupHandle << "</SessionGroupHandle>"
+ << "<RecordingControlType>Start</RecordingControlType>"
+ << "<DeltaFramesPerControlFrame>" << deltaFramesPerControlFrame << "</DeltaFramesPerControlFrame>"
+ << "<Filename>" << "" << "</Filename>"
+ << "<EnableAudioRecordingEvents>false</EnableAudioRecordingEvents>"
+ << "<LoopModeDurationSeconds>" << seconds << "</LoopModeDurationSeconds>"
+ << "</Request>\n\n\n";
+
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::recordingLoopSave(const std::string& filename)
+{
+// LL_DEBUGS("Voice") << "sending SessionGroup.ControlRecording (Flush)" << LL_ENDL;
+
+ if(mAudioSession != NULL && !mAudioSession->mGroupHandle.empty())
+ {
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"SessionGroup.ControlRecording.1\">"
+ << "<SessionGroupHandle>" << mMainSessionGroupHandle << "</SessionGroupHandle>"
+ << "<RecordingControlType>Flush</RecordingControlType>"
+ << "<Filename>" << filename << "</Filename>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::recordingStop()
+{
+// LL_DEBUGS("Voice") << "sending SessionGroup.ControlRecording (Stop)" << LL_ENDL;
+
+ if(mAudioSession != NULL && !mAudioSession->mGroupHandle.empty())
+ {
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"SessionGroup.ControlRecording.1\">"
+ << "<SessionGroupHandle>" << mMainSessionGroupHandle << "</SessionGroupHandle>"
+ << "<RecordingControlType>Stop</RecordingControlType>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::filePlaybackStart(const std::string& filename)
+{
+// LL_DEBUGS("Voice") << "sending SessionGroup.ControlPlayback (Start)" << LL_ENDL;
+
+ if(mAudioSession != NULL && !mAudioSession->mGroupHandle.empty())
+ {
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"SessionGroup.ControlPlayback.1\">"
+ << "<SessionGroupHandle>" << mMainSessionGroupHandle << "</SessionGroupHandle>"
+ << "<RecordingControlType>Start</RecordingControlType>"
+ << "<Filename>" << filename << "</Filename>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::filePlaybackStop()
+{
+// LL_DEBUGS("Voice") << "sending SessionGroup.ControlPlayback (Stop)" << LL_ENDL;
+
+ if(mAudioSession != NULL && !mAudioSession->mGroupHandle.empty())
+ {
+ std::ostringstream stream;
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"SessionGroup.ControlPlayback.1\">"
+ << "<SessionGroupHandle>" << mMainSessionGroupHandle << "</SessionGroupHandle>"
+ << "<RecordingControlType>Stop</RecordingControlType>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::filePlaybackSetPaused(bool paused)
+{
+ // TODO: Implement once WebRTC gives me a sample
+}
+
+void LLWebRTCVoiceClient::filePlaybackSetMode(bool vox, float speed)
+{
+ // TODO: Implement once WebRTC gives me a sample
+}
+
+//------------------------------------------------------------------------
+std::set<LLWebRTCVoiceClient::sessionState::wptr_t> LLWebRTCVoiceClient::sessionState::mSession;
+
+
+LLWebRTCVoiceClient::sessionState::sessionState() :
+ mErrorStatusCode(0),
+ mMediaStreamState(streamStateUnknown),
+ mCreateInProgress(false),
+ mMediaConnectInProgress(false),
+ mVoiceInvitePending(false),
+ mTextInvitePending(false),
+ mSynthesizedCallerID(false),
+ mIsChannel(false),
+ mIsSpatial(false),
+ mIsP2P(false),
+ mIncoming(false),
+ mVoiceActive(false),
+ mReconnect(false),
+ mVolumeDirty(false),
+ mMuteDirty(false),
+ mParticipantsChanged(false)
+{
+}
+
+/*static*/
+LLWebRTCVoiceClient::sessionState::ptr_t LLWebRTCVoiceClient::sessionState::createSession()
+{
+ sessionState::ptr_t ptr(new sessionState());
+
+ std::pair<std::set<wptr_t>::iterator, bool> result = mSession.insert(ptr);
+
+ if (result.second)
+ ptr->mMyIterator = result.first;
+
+ return ptr;
+}
+
+LLWebRTCVoiceClient::sessionState::~sessionState()
+{
+ LL_INFOS("Voice") << "Destroying session handle=" << mHandle << " SIP=" << mSIPURI << LL_ENDL;
+ if (mMyIterator != mSession.end())
+ mSession.erase(mMyIterator);
+
+ removeAllParticipants();
+}
+
+bool LLWebRTCVoiceClient::sessionState::isCallBackPossible()
+{
+ // This may change to be explicitly specified by WebRTC in the future...
+ // Currently, only PSTN P2P calls cannot be returned.
+ // Conveniently, this is also the only case where we synthesize a caller UUID.
+ return !mSynthesizedCallerID;
+}
+
+bool LLWebRTCVoiceClient::sessionState::isTextIMPossible()
+{
+ // This may change to be explicitly specified by WebRTC in the future...
+ return !mSynthesizedCallerID;
+}
+
+
+/*static*/
+LLWebRTCVoiceClient::sessionState::ptr_t LLWebRTCVoiceClient::sessionState::matchSessionByHandle(const std::string &handle)
+{
+ sessionStatePtr_t result;
+
+ // *TODO: My kingdom for a lambda!
+ std::set<wptr_t>::iterator it = std::find_if(mSession.begin(), mSession.end(), boost::bind(testByHandle, _1, handle));
+
+ if (it != mSession.end())
+ result = (*it).lock();
+
+ return result;
+}
+
+/*static*/
+LLWebRTCVoiceClient::sessionState::ptr_t LLWebRTCVoiceClient::sessionState::matchCreatingSessionByURI(const std::string &uri)
+{
+ sessionStatePtr_t result;
+
+ // *TODO: My kingdom for a lambda!
+ std::set<wptr_t>::iterator it = std::find_if(mSession.begin(), mSession.end(), boost::bind(testByCreatingURI, _1, uri));
+
+ if (it != mSession.end())
+ result = (*it).lock();
+
+ return result;
+}
+
+/*static*/
+LLWebRTCVoiceClient::sessionState::ptr_t LLWebRTCVoiceClient::sessionState::matchSessionByURI(const std::string &uri)
+{
+ sessionStatePtr_t result;
+
+ // *TODO: My kingdom for a lambda!
+ std::set<wptr_t>::iterator it = std::find_if(mSession.begin(), mSession.end(), boost::bind(testBySIPOrAlterateURI, _1, uri));
+
+ if (it != mSession.end())
+ result = (*it).lock();
+
+ return result;
+}
+
+/*static*/
+LLWebRTCVoiceClient::sessionState::ptr_t LLWebRTCVoiceClient::sessionState::matchSessionByParticipant(const LLUUID &participant_id)
+{
+ sessionStatePtr_t result;
+
+ // *TODO: My kingdom for a lambda!
+ std::set<wptr_t>::iterator it = std::find_if(mSession.begin(), mSession.end(), boost::bind(testByCallerId, _1, participant_id));
+
+ if (it != mSession.end())
+ result = (*it).lock();
+
+ return result;
+}
+
+void LLWebRTCVoiceClient::sessionState::for_each(sessionFunc_t func)
+{
+ std::for_each(mSession.begin(), mSession.end(), boost::bind(for_eachPredicate, _1, func));
+}
+
+// simple test predicates.
+// *TODO: These should be made into lambdas when we can pull the trigger on newer C++ features.
+bool LLWebRTCVoiceClient::sessionState::testByHandle(const LLWebRTCVoiceClient::sessionState::wptr_t &a, std::string handle)
+{
+ ptr_t aLock(a.lock());
+
+ return aLock ? aLock->mHandle == handle : false;
+}
+
+bool LLWebRTCVoiceClient::sessionState::testByCreatingURI(const LLWebRTCVoiceClient::sessionState::wptr_t &a, std::string uri)
+{
+ ptr_t aLock(a.lock());
+
+ return aLock ? (aLock->mCreateInProgress && (aLock->mSIPURI == uri)) : false;
+}
+
+bool LLWebRTCVoiceClient::sessionState::testBySIPOrAlterateURI(const LLWebRTCVoiceClient::sessionState::wptr_t &a, std::string uri)
+{
+ ptr_t aLock(a.lock());
+
+ return aLock ? ((aLock->mSIPURI == uri) || (aLock->mAlternateSIPURI == uri)) : false;
+}
+
+
+bool LLWebRTCVoiceClient::sessionState::testByCallerId(const LLWebRTCVoiceClient::sessionState::wptr_t &a, LLUUID participantId)
+{
+ ptr_t aLock(a.lock());
+
+ return aLock ? ((aLock->mCallerID == participantId) || (aLock->mIMSessionID == participantId)) : false;
+}
+
+/*static*/
+void LLWebRTCVoiceClient::sessionState::for_eachPredicate(const LLWebRTCVoiceClient::sessionState::wptr_t &a, sessionFunc_t func)
+{
+ ptr_t aLock(a.lock());
+
+ if (aLock)
+ func(aLock);
+ else
+ {
+ LL_WARNS("Voice") << "Stale handle in session map!" << LL_ENDL;
+ }
+}
+
+
+
+LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::findSession(const std::string &handle)
+{
+ sessionStatePtr_t result;
+ sessionMap::iterator iter = mSessionsByHandle.find(handle);
+ if(iter != mSessionsByHandle.end())
+ {
+ result = iter->second;
+ }
+
+ return result;
+}
+
+LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::findSessionBeingCreatedByURI(const std::string &uri)
+{
+ sessionStatePtr_t result = sessionState::matchCreatingSessionByURI(uri);
+
+ return result;
+}
+
+LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::findSession(const LLUUID &participant_id)
+{
+ sessionStatePtr_t result = sessionState::matchSessionByParticipant(participant_id);
+
+ return result;
+}
+
+LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::addSession(const std::string &uri, const std::string &handle)
+{
+ sessionStatePtr_t result;
+
+ if(handle.empty())
+ {
+ // No handle supplied.
+ // Check whether there's already a session with this URI
+ result = sessionState::matchSessionByURI(uri);
+ }
+ else // (!handle.empty())
+ {
+ // Check for an existing session with this handle
+ sessionMap::iterator iter = mSessionsByHandle.find(handle);
+
+ if(iter != mSessionsByHandle.end())
+ {
+ result = iter->second;
+ }
+ }
+
+ if(!result)
+ {
+ // No existing session found.
+
+ LL_DEBUGS("Voice") << "adding new session: handle \"" << handle << "\" URI " << uri << LL_ENDL;
+ result = sessionState::createSession();
+ result->mSIPURI = uri;
+ result->mHandle = handle;
+
+ if (LLVoiceClient::instance().getVoiceEffectEnabled())
+ {
+ result->mVoiceFontID = LLVoiceClient::instance().getVoiceEffectDefault();
+ }
+
+ if(!result->mHandle.empty())
+ {
+ // *TODO: Rider: This concerns me. There is a path (via switchChannel) where
+ // we do not track the session. In theory this means that we could end up with
+ // a mAuidoSession that does not match the session tracked in mSessionsByHandle
+ mSessionsByHandle.insert(sessionMap::value_type(result->mHandle, result));
+ }
+ }
+ else
+ {
+ // Found an existing session
+
+ if(uri != result->mSIPURI)
+ {
+ // TODO: Should this be an internal error?
+ LL_DEBUGS("Voice") << "changing uri from " << result->mSIPURI << " to " << uri << LL_ENDL;
+ setSessionURI(result, uri);
+ }
+
+ if(handle != result->mHandle)
+ {
+ if(handle.empty())
+ {
+ // There's at least one race condition where where addSession was clearing an existing session handle, which caused things to break.
+ LL_DEBUGS("Voice") << "NOT clearing handle " << result->mHandle << LL_ENDL;
+ }
+ else
+ {
+ // TODO: Should this be an internal error?
+ LL_DEBUGS("Voice") << "changing handle from " << result->mHandle << " to " << handle << LL_ENDL;
+ setSessionHandle(result, handle);
+ }
+ }
+
+ LL_DEBUGS("Voice") << "returning existing session: handle " << handle << " URI " << uri << LL_ENDL;
+ }
+
+ verifySessionState();
+
+ return result;
+}
+
+void LLWebRTCVoiceClient::clearSessionHandle(const sessionStatePtr_t &session)
+{
+ if (session)
+ {
+ if (!session->mHandle.empty())
+ {
+ sessionMap::iterator iter = mSessionsByHandle.find(session->mHandle);
+ if (iter != mSessionsByHandle.end())
+ {
+ mSessionsByHandle.erase(iter);
+ }
+ }
+ else
+ {
+ LL_WARNS("Voice") << "Session has empty handle!" << LL_ENDL;
+ }
+ }
+ else
+ {
+ LL_WARNS("Voice") << "Attempt to clear NULL session!" << LL_ENDL;
+ }
+
+}
+
+void LLWebRTCVoiceClient::setSessionHandle(const sessionStatePtr_t &session, const std::string &handle)
+{
+ // Have to remove the session from the handle-indexed map before changing the handle, or things will break badly.
+
+ if(!session->mHandle.empty())
+ {
+ // Remove session from the map if it should have been there.
+ sessionMap::iterator iter = mSessionsByHandle.find(session->mHandle);
+ if(iter != mSessionsByHandle.end())
+ {
+ if(iter->second != session)
+ {
+ LL_WARNS("Voice") << "Internal error: session mismatch! Session may have been duplicated. Removing version in map." << LL_ENDL;
+ }
+
+ mSessionsByHandle.erase(iter);
+ }
+ else
+ {
+ LL_WARNS("Voice") << "Attempt to remove session with handle " << session->mHandle << " not found in map!" << LL_ENDL;
+ }
+ }
+
+ session->mHandle = handle;
+
+ if(!handle.empty())
+ {
+ mSessionsByHandle.insert(sessionMap::value_type(session->mHandle, session));
+ }
+
+ verifySessionState();
+}
+
+void LLWebRTCVoiceClient::setSessionURI(const sessionStatePtr_t &session, const std::string &uri)
+{
+ // There used to be a map of session URIs to sessions, which made this complex....
+ session->mSIPURI = uri;
+
+ verifySessionState();
+}
+
+void LLWebRTCVoiceClient::deleteSession(const sessionStatePtr_t &session)
+{
+ // Remove the session from the handle map
+ if(!session->mHandle.empty())
+ {
+ sessionMap::iterator iter = mSessionsByHandle.find(session->mHandle);
+ if(iter != mSessionsByHandle.end())
+ {
+ if(iter->second != session)
+ {
+ LL_WARNS("Voice") << "Internal error: session mismatch, removing session in map." << LL_ENDL;
+ }
+ mSessionsByHandle.erase(iter);
+ }
+ }
+
+ // At this point, the session should be unhooked from all lists and all state should be consistent.
+ verifySessionState();
+
+ // If this is the current audio session, clean up the pointer which will soon be dangling.
+ if(mAudioSession == session)
+ {
+ mAudioSession.reset();
+ mAudioSessionChanged = true;
+ }
+
+ // ditto for the next audio session
+ if(mNextAudioSession == session)
+ {
+ mNextAudioSession.reset();
+ }
+
+}
+
+void LLWebRTCVoiceClient::deleteAllSessions()
+{
+ LL_DEBUGS("Voice") << LL_ENDL;
+
+ while (!mSessionsByHandle.empty())
+ {
+ const sessionStatePtr_t session = mSessionsByHandle.begin()->second;
+ deleteSession(session);
+ }
+
+}
+
+void LLWebRTCVoiceClient::verifySessionState(void)
+{
+ LL_DEBUGS("Voice") << "Sessions in handle map=" << mSessionsByHandle.size() << LL_ENDL;
+ sessionState::VerifySessions();
+}
+
+void LLWebRTCVoiceClient::addObserver(LLVoiceClientParticipantObserver* observer)
+{
+ mParticipantObservers.insert(observer);
+}
+
+void LLWebRTCVoiceClient::removeObserver(LLVoiceClientParticipantObserver* observer)
+{
+ mParticipantObservers.erase(observer);
+}
+
+void LLWebRTCVoiceClient::notifyParticipantObservers()
+{
+ for (observer_set_t::iterator it = mParticipantObservers.begin();
+ it != mParticipantObservers.end();
+ )
+ {
+ LLVoiceClientParticipantObserver* observer = *it;
+ observer->onParticipantsChanged();
+ // In case onParticipantsChanged() deleted an entry.
+ it = mParticipantObservers.upper_bound(observer);
+ }
+}
+
+void LLWebRTCVoiceClient::addObserver(LLVoiceClientStatusObserver* observer)
+{
+ mStatusObservers.insert(observer);
+}
+
+void LLWebRTCVoiceClient::removeObserver(LLVoiceClientStatusObserver* observer)
+{
+ mStatusObservers.erase(observer);
+}
+
+void LLWebRTCVoiceClient::notifyStatusObservers(LLVoiceClientStatusObserver::EStatusType status)
+{
+ LL_DEBUGS("Voice") << "( " << LLVoiceClientStatusObserver::status2string(status) << " )"
+ << " mAudioSession=" << mAudioSession
+ << LL_ENDL;
+
+ if(mAudioSession)
+ {
+ if(status == LLVoiceClientStatusObserver::ERROR_UNKNOWN)
+ {
+ switch(mAudioSession->mErrorStatusCode)
+ {
+ case 20713: status = LLVoiceClientStatusObserver::ERROR_CHANNEL_FULL; break;
+ case 20714: status = LLVoiceClientStatusObserver::ERROR_CHANNEL_LOCKED; break;
+ case 20715:
+ //invalid channel, we may be using a set of poorly cached
+ //info
+ status = LLVoiceClientStatusObserver::ERROR_NOT_AVAILABLE;
+ break;
+ case 1009:
+ //invalid username and password
+ status = LLVoiceClientStatusObserver::ERROR_NOT_AVAILABLE;
+ break;
+ }
+
+ // Reset the error code to make sure it won't be reused later by accident.
+ mAudioSession->mErrorStatusCode = 0;
+ }
+ else if(status == LLVoiceClientStatusObserver::STATUS_LEFT_CHANNEL)
+ {
+ switch(mAudioSession->mErrorStatusCode)
+ {
+ case HTTP_NOT_FOUND: // NOT_FOUND
+ // *TODO: Should this be 503?
+ case 480: // TEMPORARILY_UNAVAILABLE
+ case HTTP_REQUEST_TIME_OUT: // REQUEST_TIMEOUT
+ // call failed because other user was not available
+ // treat this as an error case
+ status = LLVoiceClientStatusObserver::ERROR_NOT_AVAILABLE;
+
+ // Reset the error code to make sure it won't be reused later by accident.
+ mAudioSession->mErrorStatusCode = 0;
+ break;
+ }
+ }
+ }
+
+ LL_DEBUGS("Voice")
+ << " " << LLVoiceClientStatusObserver::status2string(status)
+ << ", session URI " << getAudioSessionURI()
+ << ", proximal is " << inSpatialChannel()
+ << LL_ENDL;
+
+ for (status_observer_set_t::iterator it = mStatusObservers.begin();
+ it != mStatusObservers.end();
+ )
+ {
+ LLVoiceClientStatusObserver* observer = *it;
+ observer->onChange(status, getAudioSessionURI(), inSpatialChannel());
+ // In case onError() deleted an entry.
+ it = mStatusObservers.upper_bound(observer);
+ }
+
+ // skipped to avoid speak button blinking
+ if ( status != LLVoiceClientStatusObserver::STATUS_JOINING
+ && status != LLVoiceClientStatusObserver::STATUS_LEFT_CHANNEL
+ && status != LLVoiceClientStatusObserver::STATUS_VOICE_DISABLED)
+ {
+ bool voice_status = LLVoiceClient::getInstance()->voiceEnabled() && LLVoiceClient::getInstance()->isVoiceWorking();
+
+ gAgent.setVoiceConnected(voice_status);
+
+ if (voice_status)
+ {
+ LLFirstUse::speak(true);
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::addObserver(LLFriendObserver* observer)
+{
+ mFriendObservers.insert(observer);
+}
+
+void LLWebRTCVoiceClient::removeObserver(LLFriendObserver* observer)
+{
+ mFriendObservers.erase(observer);
+}
+
+void LLWebRTCVoiceClient::notifyFriendObservers()
+{
+ for (friend_observer_set_t::iterator it = mFriendObservers.begin();
+ it != mFriendObservers.end();
+ )
+ {
+ LLFriendObserver* observer = *it;
+ it++;
+ // The only friend-related thing we notify on is online/offline transitions.
+ observer->changed(LLFriendObserver::ONLINE);
+ }
+}
+
+void LLWebRTCVoiceClient::lookupName(const LLUUID &id)
+{
+ if (mAvatarNameCacheConnection.connected())
+ {
+ mAvatarNameCacheConnection.disconnect();
+ }
+ mAvatarNameCacheConnection = LLAvatarNameCache::get(id, boost::bind(&LLWebRTCVoiceClient::onAvatarNameCache, this, _1, _2));
+}
+
+void LLWebRTCVoiceClient::onAvatarNameCache(const LLUUID& agent_id,
+ const LLAvatarName& av_name)
+{
+ mAvatarNameCacheConnection.disconnect();
+ std::string display_name = av_name.getDisplayName();
+ avatarNameResolved(agent_id, display_name);
+}
+
+void LLWebRTCVoiceClient::predAvatarNameResolution(const LLWebRTCVoiceClient::sessionStatePtr_t &session, LLUUID id, std::string name)
+{
+ participantStatePtr_t participant(session->findParticipantByID(id));
+ if (participant)
+ {
+ // Found -- fill in the name
+ participant->mAccountName = name;
+ // and post a "participants updated" message to listeners later.
+ session->mParticipantsChanged = true;
+ }
+
+ // Check whether this is a p2p session whose caller name just resolved
+ if (session->mCallerID == id)
+ {
+ // this session's "caller ID" just resolved. Fill in the name.
+ session->mName = name;
+ if (session->mTextInvitePending)
+ {
+ session->mTextInvitePending = false;
+
+ // We don't need to call LLIMMgr::getInstance()->addP2PSession() here. The first incoming message will create the panel.
+ }
+ if (session->mVoiceInvitePending)
+ {
+ session->mVoiceInvitePending = false;
+
+ LLIMMgr::getInstance()->inviteToSession(
+ session->mIMSessionID,
+ session->mName,
+ session->mCallerID,
+ session->mName,
+ IM_SESSION_P2P_INVITE,
+ LLIMMgr::INVITATION_TYPE_VOICE,
+ session->mHandle,
+ session->mSIPURI);
+ }
+
+ }
+}
+
+void LLWebRTCVoiceClient::avatarNameResolved(const LLUUID &id, const std::string &name)
+{
+ sessionState::for_each(boost::bind(predAvatarNameResolution, _1, id, name));
+}
+
+bool LLWebRTCVoiceClient::setVoiceEffect(const LLUUID& id)
+{
+ if (!mAudioSession)
+ {
+ return false;
+ }
+
+ if (!id.isNull())
+ {
+ if (mVoiceFontMap.empty())
+ {
+ LL_DEBUGS("Voice") << "Voice fonts not available." << LL_ENDL;
+ return false;
+ }
+ else if (mVoiceFontMap.find(id) == mVoiceFontMap.end())
+ {
+ LL_DEBUGS("Voice") << "Invalid voice font " << id << LL_ENDL;
+ return false;
+ }
+ }
+
+ // *TODO: Check for expired fonts?
+ mAudioSession->mVoiceFontID = id;
+
+ // *TODO: Separate voice font defaults for spatial chat and IM?
+ gSavedPerAccountSettings.setString("VoiceEffectDefault", id.asString());
+
+ sessionSetVoiceFontSendMessage(mAudioSession);
+ notifyVoiceFontObservers();
+
+ return true;
+}
+
+const LLUUID LLWebRTCVoiceClient::getVoiceEffect()
+{
+ return mAudioSession ? mAudioSession->mVoiceFontID : LLUUID::null;
+}
+
+LLSD LLWebRTCVoiceClient::getVoiceEffectProperties(const LLUUID& id)
+{
+ LLSD sd;
+
+ voice_font_map_t::iterator iter = mVoiceFontMap.find(id);
+ if (iter != mVoiceFontMap.end())
+ {
+ sd["template_only"] = false;
+ }
+ else
+ {
+ // Voice effect is not in the voice font map, see if there is a template
+ iter = mVoiceFontTemplateMap.find(id);
+ if (iter == mVoiceFontTemplateMap.end())
+ {
+ LL_WARNS("Voice") << "Voice effect " << id << "not found." << LL_ENDL;
+ return sd;
+ }
+ sd["template_only"] = true;
+ }
+
+ voiceFontEntry *font = iter->second;
+ sd["name"] = font->mName;
+ sd["expiry_date"] = font->mExpirationDate;
+ sd["is_new"] = font->mIsNew;
+
+ return sd;
+}
+
+LLWebRTCVoiceClient::voiceFontEntry::voiceFontEntry(LLUUID& id) :
+ mID(id),
+ mFontIndex(0),
+ mFontType(VOICE_FONT_TYPE_NONE),
+ mFontStatus(VOICE_FONT_STATUS_NONE),
+ mIsNew(false)
+{
+ mExpiryTimer.stop();
+ mExpiryWarningTimer.stop();
+}
+
+LLWebRTCVoiceClient::voiceFontEntry::~voiceFontEntry()
+{
+}
+
+void LLWebRTCVoiceClient::refreshVoiceEffectLists(bool clear_lists)
+{
+ if (clear_lists)
+ {
+ mVoiceFontsReceived = false;
+ deleteAllVoiceFonts();
+ deleteVoiceFontTemplates();
+ }
+
+ accountGetSessionFontsSendMessage();
+ accountGetTemplateFontsSendMessage();
+}
+
+const voice_effect_list_t& LLWebRTCVoiceClient::getVoiceEffectList() const
+{
+ return mVoiceFontList;
+}
+
+const voice_effect_list_t& LLWebRTCVoiceClient::getVoiceEffectTemplateList() const
+{
+ return mVoiceFontTemplateList;
+}
+
+void LLWebRTCVoiceClient::addVoiceFont(const S32 font_index,
+ const std::string &name,
+ const std::string &description,
+ const LLDate &expiration_date,
+ bool has_expired,
+ const S32 font_type,
+ const S32 font_status,
+ const bool template_font)
+{
+ // WebRTC SessionFontIDs are not guaranteed to remain the same between
+ // sessions or grids so use a UUID for the name.
+
+ // If received name is not a UUID, fudge one by hashing the name and type.
+ LLUUID font_id;
+ if (LLUUID::validate(name))
+ {
+ font_id = LLUUID(name);
+ }
+ else
+ {
+ font_id.generate(STRINGIZE(font_type << ":" << name));
+ }
+
+ voiceFontEntry *font = NULL;
+
+ voice_font_map_t& font_map = template_font ? mVoiceFontTemplateMap : mVoiceFontMap;
+ voice_effect_list_t& font_list = template_font ? mVoiceFontTemplateList : mVoiceFontList;
+
+ // Check whether we've seen this font before.
+ voice_font_map_t::iterator iter = font_map.find(font_id);
+ bool new_font = (iter == font_map.end());
+
+ // Override the has_expired flag if we have passed the expiration_date as a double check.
+ if (expiration_date.secondsSinceEpoch() < (LLDate::now().secondsSinceEpoch() + VOICE_FONT_EXPIRY_INTERVAL))
+ {
+ has_expired = true;
+ }
+
+ if (has_expired)
+ {
+ LL_DEBUGS("VoiceFont") << "Expired " << (template_font ? "Template " : "")
+ << expiration_date.asString() << " " << font_id
+ << " (" << font_index << ") " << name << LL_ENDL;
+
+ // Remove existing session fonts that have expired since we last saw them.
+ if (!new_font && !template_font)
+ {
+ deleteVoiceFont(font_id);
+ }
+ return;
+ }
+
+ if (new_font)
+ {
+ // If it is a new font create a new entry.
+ font = new voiceFontEntry(font_id);
+ }
+ else
+ {
+ // Not a new font, update the existing entry
+ font = iter->second;
+ }
+
+ if (font)
+ {
+ font->mFontIndex = font_index;
+ // Use the description for the human readable name if available, as the
+ // "name" may be a UUID.
+ font->mName = description.empty() ? name : description;
+ font->mFontType = font_type;
+ font->mFontStatus = font_status;
+
+ // If the font is new or the expiration date has changed the expiry timers need updating.
+ if (!template_font && (new_font || font->mExpirationDate != expiration_date))
+ {
+ font->mExpirationDate = expiration_date;
+
+ // Set the expiry timer to trigger a notification when the voice font can no longer be used.
+ font->mExpiryTimer.start();
+ font->mExpiryTimer.setExpiryAt(expiration_date.secondsSinceEpoch() - VOICE_FONT_EXPIRY_INTERVAL);
+
+ // Set the warning timer to some interval before actual expiry.
+ S32 warning_time = gSavedSettings.getS32("VoiceEffectExpiryWarningTime");
+ if (warning_time != 0)
+ {
+ font->mExpiryWarningTimer.start();
+ F64 expiry_time = (expiration_date.secondsSinceEpoch() - (F64)warning_time);
+ font->mExpiryWarningTimer.setExpiryAt(expiry_time - VOICE_FONT_EXPIRY_INTERVAL);
+ }
+ else
+ {
+ // Disable the warning timer.
+ font->mExpiryWarningTimer.stop();
+ }
+
+ // Only flag new session fonts after the first time we have fetched the list.
+ if (mVoiceFontsReceived)
+ {
+ font->mIsNew = true;
+ mVoiceFontsNew = true;
+ }
+ }
+
+ LL_DEBUGS("VoiceFont") << (template_font ? "Template " : "")
+ << font->mExpirationDate.asString() << " " << font->mID
+ << " (" << font->mFontIndex << ") " << name << LL_ENDL;
+
+ if (new_font)
+ {
+ font_map.insert(voice_font_map_t::value_type(font->mID, font));
+ font_list.insert(voice_effect_list_t::value_type(font->mName, font->mID));
+ }
+
+ mVoiceFontListDirty = true;
+
+ // Debugging stuff
+
+ if (font_type < VOICE_FONT_TYPE_NONE || font_type >= VOICE_FONT_TYPE_UNKNOWN)
+ {
+ LL_WARNS("VoiceFont") << "Unknown voice font type: " << font_type << LL_ENDL;
+ }
+ if (font_status < VOICE_FONT_STATUS_NONE || font_status >= VOICE_FONT_STATUS_UNKNOWN)
+ {
+ LL_WARNS("VoiceFont") << "Unknown voice font status: " << font_status << LL_ENDL;
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::expireVoiceFonts()
+{
+ // *TODO: If we are selling voice fonts in packs, there are probably
+ // going to be a number of fonts with the same expiration time, so would
+ // be more efficient to just keep a list of expiration times rather
+ // than checking each font individually.
+
+ bool have_expired = false;
+ bool will_expire = false;
+ bool expired_in_use = false;
+
+ LLUUID current_effect = LLVoiceClient::instance().getVoiceEffectDefault();
+
+ voice_font_map_t::iterator iter;
+ for (iter = mVoiceFontMap.begin(); iter != mVoiceFontMap.end(); ++iter)
+ {
+ voiceFontEntry* voice_font = iter->second;
+ LLFrameTimer& expiry_timer = voice_font->mExpiryTimer;
+ LLFrameTimer& warning_timer = voice_font->mExpiryWarningTimer;
+
+ // Check for expired voice fonts
+ if (expiry_timer.getStarted() && expiry_timer.hasExpired())
+ {
+ // Check whether it is the active voice font
+ if (voice_font->mID == current_effect)
+ {
+ // Reset to no voice effect.
+ setVoiceEffect(LLUUID::null);
+ expired_in_use = true;
+ }
+
+ LL_DEBUGS("Voice") << "Voice Font " << voice_font->mName << " has expired." << LL_ENDL;
+ deleteVoiceFont(voice_font->mID);
+ have_expired = true;
+ }
+
+ // Check for voice fonts that will expire in less that the warning time
+ if (warning_timer.getStarted() && warning_timer.hasExpired())
+ {
+ LL_DEBUGS("VoiceFont") << "Voice Font " << voice_font->mName << " will expire soon." << LL_ENDL;
+ will_expire = true;
+ warning_timer.stop();
+ }
+ }
+
+ LLSD args;
+ args["URL"] = LLTrans::getString("voice_morphing_url");
+ args["PREMIUM_URL"] = LLTrans::getString("premium_voice_morphing_url");
+
+ // Give a notification if any voice fonts have expired.
+ if (have_expired)
+ {
+ if (expired_in_use)
+ {
+ LLNotificationsUtil::add("VoiceEffectsExpiredInUse", args);
+ }
+ else
+ {
+ LLNotificationsUtil::add("VoiceEffectsExpired", args);
+ }
+
+ // Refresh voice font lists in the UI.
+ notifyVoiceFontObservers();
+ }
+
+ // Give a warning notification if any voice fonts are due to expire.
+ if (will_expire)
+ {
+ S32Seconds seconds(gSavedSettings.getS32("VoiceEffectExpiryWarningTime"));
+ args["INTERVAL"] = llformat("%d", LLUnit<S32, LLUnits::Days>(seconds).value());
+
+ LLNotificationsUtil::add("VoiceEffectsWillExpire", args);
+ }
+}
+
+void LLWebRTCVoiceClient::deleteVoiceFont(const LLUUID& id)
+{
+ // Remove the entry from the voice font list.
+ voice_effect_list_t::iterator list_iter = mVoiceFontList.begin();
+ while (list_iter != mVoiceFontList.end())
+ {
+ if (list_iter->second == id)
+ {
+ LL_DEBUGS("VoiceFont") << "Removing " << id << " from the voice font list." << LL_ENDL;
+ list_iter = mVoiceFontList.erase(list_iter);
+ mVoiceFontListDirty = true;
+ }
+ else
+ {
+ ++list_iter;
+ }
+ }
+
+ // Find the entry in the voice font map and erase its data.
+ voice_font_map_t::iterator map_iter = mVoiceFontMap.find(id);
+ if (map_iter != mVoiceFontMap.end())
+ {
+ delete map_iter->second;
+ }
+
+ // Remove the entry from the voice font map.
+ mVoiceFontMap.erase(map_iter);
+}
+
+void LLWebRTCVoiceClient::deleteAllVoiceFonts()
+{
+ mVoiceFontList.clear();
+
+ voice_font_map_t::iterator iter;
+ for (iter = mVoiceFontMap.begin(); iter != mVoiceFontMap.end(); ++iter)
+ {
+ delete iter->second;
+ }
+ mVoiceFontMap.clear();
+}
+
+void LLWebRTCVoiceClient::deleteVoiceFontTemplates()
+{
+ mVoiceFontTemplateList.clear();
+
+ voice_font_map_t::iterator iter;
+ for (iter = mVoiceFontTemplateMap.begin(); iter != mVoiceFontTemplateMap.end(); ++iter)
+ {
+ delete iter->second;
+ }
+ mVoiceFontTemplateMap.clear();
+}
+
+S32 LLWebRTCVoiceClient::getVoiceFontIndex(const LLUUID& id) const
+{
+ S32 result = 0;
+ if (!id.isNull())
+ {
+ voice_font_map_t::const_iterator it = mVoiceFontMap.find(id);
+ if (it != mVoiceFontMap.end())
+ {
+ result = it->second->mFontIndex;
+ }
+ else
+ {
+ LL_WARNS("VoiceFont") << "Selected voice font " << id << " is not available." << LL_ENDL;
+ }
+ }
+ return result;
+}
+
+S32 LLWebRTCVoiceClient::getVoiceFontTemplateIndex(const LLUUID& id) const
+{
+ S32 result = 0;
+ if (!id.isNull())
+ {
+ voice_font_map_t::const_iterator it = mVoiceFontTemplateMap.find(id);
+ if (it != mVoiceFontTemplateMap.end())
+ {
+ result = it->second->mFontIndex;
+ }
+ else
+ {
+ LL_WARNS("VoiceFont") << "Selected voice font template " << id << " is not available." << LL_ENDL;
+ }
+ }
+ return result;
+}
+
+void LLWebRTCVoiceClient::accountGetSessionFontsSendMessage()
+{
+ if(mAccountLoggedIn)
+ {
+ std::ostringstream stream;
+
+ LL_DEBUGS("VoiceFont") << "Requesting voice font list." << LL_ENDL;
+
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Account.GetSessionFonts.1\">"
+ << "<AccountHandle>" << LLWebRTCSecurity::getInstance()->accountHandle() << "</AccountHandle>"
+ << "</Request>"
+ << "\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::accountGetTemplateFontsSendMessage()
+{
+ if(mAccountLoggedIn)
+ {
+ std::ostringstream stream;
+
+ LL_DEBUGS("VoiceFont") << "Requesting voice font template list." << LL_ENDL;
+
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Account.GetTemplateFonts.1\">"
+ << "<AccountHandle>" << LLWebRTCSecurity::getInstance()->accountHandle() << "</AccountHandle>"
+ << "</Request>"
+ << "\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::sessionSetVoiceFontSendMessage(const sessionStatePtr_t &session)
+{
+ S32 font_index = getVoiceFontIndex(session->mVoiceFontID);
+ LL_DEBUGS("VoiceFont") << "Requesting voice font: " << session->mVoiceFontID << " (" << font_index << "), session handle: " << session->mHandle << LL_ENDL;
+
+ std::ostringstream stream;
+
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Session.SetVoiceFont.1\">"
+ << "<SessionHandle>" << session->mHandle << "</SessionHandle>"
+ << "<SessionFontID>" << font_index << "</SessionFontID>"
+ << "</Request>\n\n\n";
+
+ writeString(stream.str());
+}
+
+void LLWebRTCVoiceClient::accountGetSessionFontsResponse(int statusCode, const std::string &statusString)
+{
+ if (mIsWaitingForFonts)
+ {
+ // *TODO: We seem to get multiple events of this type. Should figure a way to advance only after
+ // receiving the last one.
+ LLSD result(LLSDMap("voice_fonts", LLSD::Boolean(true)));
+
+ mWebRTCPump.post(result);
+ }
+ notifyVoiceFontObservers();
+ mVoiceFontsReceived = true;
+}
+
+void LLWebRTCVoiceClient::accountGetTemplateFontsResponse(int statusCode, const std::string &statusString)
+{
+ // Voice font list entries were updated via addVoiceFont() during parsing.
+ notifyVoiceFontObservers();
+}
+void LLWebRTCVoiceClient::addObserver(LLVoiceEffectObserver* observer)
+{
+ mVoiceFontObservers.insert(observer);
+}
+
+void LLWebRTCVoiceClient::removeObserver(LLVoiceEffectObserver* observer)
+{
+ mVoiceFontObservers.erase(observer);
+}
+
+// method checks the item in VoiceMorphing menu for appropriate current voice font
+bool LLWebRTCVoiceClient::onCheckVoiceEffect(const std::string& voice_effect_name)
+{
+ LLVoiceEffectInterface * effect_interfacep = LLVoiceClient::instance().getVoiceEffectInterface();
+ if (NULL != effect_interfacep)
+ {
+ const LLUUID& currect_voice_effect_id = effect_interfacep->getVoiceEffect();
+
+ if (currect_voice_effect_id.isNull())
+ {
+ if (voice_effect_name == "NoVoiceMorphing")
+ {
+ return true;
+ }
+ }
+ else
+ {
+ const LLSD& voice_effect_props = effect_interfacep->getVoiceEffectProperties(currect_voice_effect_id);
+ if (voice_effect_props["name"].asString() == voice_effect_name)
+ {
+ return true;
+ }
+ }
+ }
+
+ return false;
+}
+
+// method changes voice font for selected VoiceMorphing menu item
+void LLWebRTCVoiceClient::onClickVoiceEffect(const std::string& voice_effect_name)
+{
+ LLVoiceEffectInterface * effect_interfacep = LLVoiceClient::instance().getVoiceEffectInterface();
+ if (NULL != effect_interfacep)
+ {
+ if (voice_effect_name == "NoVoiceMorphing")
+ {
+ effect_interfacep->setVoiceEffect(LLUUID());
+ return;
+ }
+ const voice_effect_list_t& effect_list = effect_interfacep->getVoiceEffectList();
+ if (!effect_list.empty())
+ {
+ for (voice_effect_list_t::const_iterator it = effect_list.begin(); it != effect_list.end(); ++it)
+ {
+ if (voice_effect_name == it->first)
+ {
+ effect_interfacep->setVoiceEffect(it->second);
+ return;
+ }
+ }
+ }
+ }
+}
+
+// it updates VoiceMorphing menu items in accordance with purchased properties
+void LLWebRTCVoiceClient::updateVoiceMorphingMenu()
+{
+ if (mVoiceFontListDirty)
+ {
+ LLVoiceEffectInterface * effect_interfacep = LLVoiceClient::instance().getVoiceEffectInterface();
+ if (effect_interfacep)
+ {
+ const voice_effect_list_t& effect_list = effect_interfacep->getVoiceEffectList();
+ if (!effect_list.empty())
+ {
+ LLMenuGL * voice_morphing_menup = gMenuBarView->findChildMenuByName("VoiceMorphing", TRUE);
+
+ if (NULL != voice_morphing_menup)
+ {
+ S32 items = voice_morphing_menup->getItemCount();
+ if (items > 0)
+ {
+ voice_morphing_menup->erase(1, items - 3, false);
+
+ S32 pos = 1;
+ for (voice_effect_list_t::const_iterator it = effect_list.begin(); it != effect_list.end(); ++it)
+ {
+ LLMenuItemCheckGL::Params p;
+ p.name = it->first;
+ p.label = it->first;
+ p.on_check.function(boost::bind(&LLWebRTCVoiceClient::onCheckVoiceEffect, this, it->first));
+ p.on_click.function(boost::bind(&LLWebRTCVoiceClient::onClickVoiceEffect, this, it->first));
+ LLMenuItemCheckGL * voice_effect_itemp = LLUICtrlFactory::create<LLMenuItemCheckGL>(p);
+ voice_morphing_menup->insert(pos++, voice_effect_itemp, false);
+ }
+
+ voice_morphing_menup->needsArrange();
+ }
+ }
+ }
+ }
+ }
+}
+void LLWebRTCVoiceClient::notifyVoiceFontObservers()
+{
+ LL_DEBUGS("VoiceFont") << "Notifying voice effect observers. Lists changed: " << mVoiceFontListDirty << LL_ENDL;
+
+ updateVoiceMorphingMenu();
+
+ for (voice_font_observer_set_t::iterator it = mVoiceFontObservers.begin();
+ it != mVoiceFontObservers.end();)
+ {
+ LLVoiceEffectObserver* observer = *it;
+ observer->onVoiceEffectChanged(mVoiceFontListDirty);
+ // In case onVoiceEffectChanged() deleted an entry.
+ it = mVoiceFontObservers.upper_bound(observer);
+ }
+ mVoiceFontListDirty = false;
+
+ // If new Voice Fonts have been added notify the user.
+ if (mVoiceFontsNew)
+ {
+ if (mVoiceFontsReceived)
+ {
+ LLNotificationsUtil::add("VoiceEffectsNew");
+ }
+ mVoiceFontsNew = false;
+ }
+}
+
+void LLWebRTCVoiceClient::enablePreviewBuffer(bool enable)
+{
+ LLSD result;
+ mCaptureBufferMode = enable;
+
+ if (enable)
+ result["recplay"] = "start";
+ else
+ result["recplay"] = "quit";
+
+ mWebRTCPump.post(result);
+
+ if(mCaptureBufferMode && mIsInChannel)
+ {
+ LL_DEBUGS("Voice") << "no channel" << LL_ENDL;
+ sessionTerminate();
+ }
+}
+
+void LLWebRTCVoiceClient::recordPreviewBuffer()
+{
+ if (!mCaptureBufferMode)
+ {
+ LL_DEBUGS("Voice") << "Not in voice effect preview mode, cannot start recording." << LL_ENDL;
+ mCaptureBufferRecording = false;
+ return;
+ }
+
+ mCaptureBufferRecording = true;
+
+ LLSD result(LLSDMap("recplay", "record"));
+ mWebRTCPump.post(result);
+}
+
+void LLWebRTCVoiceClient::playPreviewBuffer(const LLUUID& effect_id)
+{
+ if (!mCaptureBufferMode)
+ {
+ LL_DEBUGS("Voice") << "Not in voice effect preview mode, no buffer to play." << LL_ENDL;
+ mCaptureBufferRecording = false;
+ return;
+ }
+
+ if (!mCaptureBufferRecorded)
+ {
+ // Can't play until we have something recorded!
+ mCaptureBufferPlaying = false;
+ return;
+ }
+
+ mPreviewVoiceFont = effect_id;
+ mCaptureBufferPlaying = true;
+
+ LLSD result(LLSDMap("recplay", "playback"));
+ mWebRTCPump.post(result);
+}
+
+void LLWebRTCVoiceClient::stopPreviewBuffer()
+{
+ mCaptureBufferRecording = false;
+ mCaptureBufferPlaying = false;
+
+ LLSD result(LLSDMap("recplay", "quit"));
+ mWebRTCPump.post(result);
+}
+
+bool LLWebRTCVoiceClient::isPreviewRecording()
+{
+ return (mCaptureBufferMode && mCaptureBufferRecording);
+}
+
+bool LLWebRTCVoiceClient::isPreviewPlaying()
+{
+ return (mCaptureBufferMode && mCaptureBufferPlaying);
+}
+
+void LLWebRTCVoiceClient::captureBufferRecordStartSendMessage()
+{ if(mAccountLoggedIn)
+ {
+ std::ostringstream stream;
+
+ LL_DEBUGS("Voice") << "Starting audio capture to buffer." << LL_ENDL;
+
+ // Start capture
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Aux.StartBufferCapture.1\">"
+ << "</Request>"
+ << "\n\n\n";
+
+ // Unmute the mic
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Connector.MuteLocalMic.1\">"
+ << "<ConnectorHandle>" << LLWebRTCSecurity::getInstance()->connectorHandle() << "</ConnectorHandle>"
+ << "<Value>false</Value>"
+ << "</Request>\n\n\n";
+
+ // Dirty the mute mic state so that it will get reset when we finishing previewing
+ mMuteMicDirty = true;
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::captureBufferRecordStopSendMessage()
+{
+ if(mAccountLoggedIn)
+ {
+ std::ostringstream stream;
+
+ LL_DEBUGS("Voice") << "Stopping audio capture to buffer." << LL_ENDL;
+
+ // Mute the mic. Mic mute state was dirtied at recording start, so will be reset when finished previewing.
+ stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Connector.MuteLocalMic.1\">"
+ << "<ConnectorHandle>" << LLWebRTCSecurity::getInstance()->connectorHandle() << "</ConnectorHandle>"
+ << "<Value>true</Value>"
+ << "</Request>\n\n\n";
+
+ // Stop capture
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Aux.CaptureAudioStop.1\">"
+ << "<AccountHandle>" << LLWebRTCSecurity::getInstance()->accountHandle() << "</AccountHandle>"
+ << "</Request>"
+ << "\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::captureBufferPlayStartSendMessage(const LLUUID& voice_font_id)
+{
+ if(mAccountLoggedIn)
+ {
+ // Track how may play requests are sent, so we know how many stop events to
+ // expect before play actually stops.
+ ++mPlayRequestCount;
+
+ std::ostringstream stream;
+
+ LL_DEBUGS("Voice") << "Starting audio buffer playback." << LL_ENDL;
+
+ S32 font_index = getVoiceFontTemplateIndex(voice_font_id);
+ LL_DEBUGS("Voice") << "With voice font: " << voice_font_id << " (" << font_index << ")" << LL_ENDL;
+
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Aux.PlayAudioBuffer.1\">"
+ << "<AccountHandle>" << LLWebRTCSecurity::getInstance()->accountHandle() << "</AccountHandle>"
+ << "<TemplateFontID>" << font_index << "</TemplateFontID>"
+ << "<FontDelta />"
+ << "</Request>"
+ << "\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+void LLWebRTCVoiceClient::captureBufferPlayStopSendMessage()
+{
+ if(mAccountLoggedIn)
+ {
+ std::ostringstream stream;
+
+ LL_DEBUGS("Voice") << "Stopping audio buffer playback." << LL_ENDL;
+
+ stream
+ << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Aux.RenderAudioStop.1\">"
+ << "<AccountHandle>" << LLWebRTCSecurity::getInstance()->accountHandle() << "</AccountHandle>"
+ << "</Request>"
+ << "\n\n\n";
+
+ writeString(stream.str());
+ }
+}
+
+LLWebRTCProtocolParser::LLWebRTCProtocolParser()
+{
+ parser = XML_ParserCreate(NULL);
+
+ reset();
+}
+
+void LLWebRTCProtocolParser::reset()
+{
+ responseDepth = 0;
+ ignoringTags = false;
+ accumulateText = false;
+ energy = 0.f;
+ hasText = false;
+ hasAudio = false;
+ hasVideo = false;
+ terminated = false;
+ ignoreDepth = 0;
+ isChannel = false;
+ incoming = false;
+ enabled = false;
+ isEvent = false;
+ isLocallyMuted = false;
+ isModeratorMuted = false;
+ isSpeaking = false;
+ participantType = 0;
+ returnCode = -1;
+ state = 0;
+ statusCode = 0;
+ volume = 0;
+ textBuffer.clear();
+ alias.clear();
+ numberOfAliases = 0;
+ applicationString.clear();
+}
+
+//virtual
+LLWebRTCProtocolParser::~LLWebRTCProtocolParser()
+{
+ if (parser)
+ XML_ParserFree(parser);
+}
+
+static LLTrace::BlockTimerStatHandle FTM_WebRTC_PROCESS("WebRTC Process");
+
+// virtual
+LLIOPipe::EStatus LLWebRTCProtocolParser::process_impl(
+ const LLChannelDescriptors& channels,
+ buffer_ptr_t& buffer,
+ bool& eos,
+ LLSD& context,
+ LLPumpIO* pump)
+{
+ LL_RECORD_BLOCK_TIME(FTM_WebRTC_PROCESS);
+ LLBufferStream istr(channels, buffer.get());
+ std::ostringstream ostr;
+ while (istr.good())
+ {
+ char buf[1024];
+ istr.read(buf, sizeof(buf));
+ mInput.append(buf, istr.gcount());
+ }
+
+ // Look for input delimiter(s) in the input buffer. If one is found, send the message to the xml parser.
+ int start = 0;
+ int delim;
+ while((delim = mInput.find("\n\n\n", start)) != std::string::npos)
+ {
+
+ // Reset internal state of the LLWebRTCProtocolParser (no effect on the expat parser)
+ reset();
+
+ XML_ParserReset(parser, NULL);
+ XML_SetElementHandler(parser, ExpatStartTag, ExpatEndTag);
+ XML_SetCharacterDataHandler(parser, ExpatCharHandler);
+ XML_SetUserData(parser, this);
+ XML_Parse(parser, mInput.data() + start, delim - start, false);
+
+ LL_DEBUGS("WebRTCProtocolParser") << "parsing: " << mInput.substr(start, delim - start) << LL_ENDL;
+ start = delim + 3;
+ }
+
+ if(start != 0)
+ mInput = mInput.substr(start);
+
+ LL_DEBUGS("WebRTCProtocolParser") << "at end, mInput is: " << mInput << LL_ENDL;
+
+ if(!LLWebRTCVoiceClient::sConnected)
+ {
+ // If voice has been disabled, we just want to close the socket. This does so.
+ LL_INFOS("Voice") << "returning STATUS_STOP" << LL_ENDL;
+ return STATUS_STOP;
+ }
+
+ return STATUS_OK;
+}
+
+void XMLCALL LLWebRTCProtocolParser::ExpatStartTag(void *data, const char *el, const char **attr)
+{
+ if (data)
+ {
+ LLWebRTCProtocolParser *object = (LLWebRTCProtocolParser*)data;
+ object->StartTag(el, attr);
+ }
+}
+
+// --------------------------------------------------------------------------------
+
+void XMLCALL LLWebRTCProtocolParser::ExpatEndTag(void *data, const char *el)
+{
+ if (data)
+ {
+ LLWebRTCProtocolParser *object = (LLWebRTCProtocolParser*)data;
+ object->EndTag(el);
+ }
+}
+
+// --------------------------------------------------------------------------------
+
+void XMLCALL LLWebRTCProtocolParser::ExpatCharHandler(void *data, const XML_Char *s, int len)
+{
+ if (data)
+ {
+ LLWebRTCProtocolParser *object = (LLWebRTCProtocolParser*)data;
+ object->CharData(s, len);
+ }
+}
+
+// --------------------------------------------------------------------------------
+
+
+void LLWebRTCProtocolParser::StartTag(const char *tag, const char **attr)
+{
+ // Reset the text accumulator. We shouldn't have strings that are inturrupted by new tags
+ textBuffer.clear();
+ // only accumulate text if we're not ignoring tags.
+ accumulateText = !ignoringTags;
+
+ if (responseDepth == 0)
+ {
+ isEvent = !stricmp("Event", tag);
+
+ if (!stricmp("Response", tag) || isEvent)
+ {
+ // Grab the attributes
+ while (*attr)
+ {
+ const char *key = *attr++;
+ const char *value = *attr++;
+
+ if (!stricmp("requestId", key))
+ {
+ requestId = value;
+ }
+ else if (!stricmp("action", key))
+ {
+ actionString = value;
+ }
+ else if (!stricmp("type", key))
+ {
+ eventTypeString = value;
+ }
+ }
+ }
+ LL_DEBUGS("WebRTCProtocolParser") << tag << " (" << responseDepth << ")" << LL_ENDL;
+ }
+ else
+ {
+ if (ignoringTags)
+ {
+ LL_DEBUGS("WebRTCProtocolParser") << "ignoring tag " << tag << " (depth = " << responseDepth << ")" << LL_ENDL;
+ }
+ else
+ {
+ LL_DEBUGS("WebRTCProtocolParser") << tag << " (" << responseDepth << ")" << LL_ENDL;
+
+ // Ignore the InputXml stuff so we don't get confused
+ if (!stricmp("InputXml", tag))
+ {
+ ignoringTags = true;
+ ignoreDepth = responseDepth;
+ accumulateText = false;
+
+ LL_DEBUGS("WebRTCProtocolParser") << "starting ignore, ignoreDepth is " << ignoreDepth << LL_ENDL;
+ }
+ else if (!stricmp("CaptureDevices", tag))
+ {
+ LLWebRTCVoiceClient::getInstance()->clearCaptureDevices();
+ }
+ else if (!stricmp("RenderDevices", tag))
+ {
+ LLWebRTCVoiceClient::getInstance()->clearRenderDevices();
+ }
+ else if (!stricmp("CaptureDevice", tag))
+ {
+ deviceString.clear();
+ }
+ else if (!stricmp("RenderDevice", tag))
+ {
+ deviceString.clear();
+ }
+ else if (!stricmp("SessionFont", tag))
+ {
+ id = 0;
+ nameString.clear();
+ descriptionString.clear();
+ expirationDate = LLDate();
+ hasExpired = false;
+ fontType = 0;
+ fontStatus = 0;
+ }
+ else if (!stricmp("TemplateFont", tag))
+ {
+ id = 0;
+ nameString.clear();
+ descriptionString.clear();
+ expirationDate = LLDate();
+ hasExpired = false;
+ fontType = 0;
+ fontStatus = 0;
+ }
+ else if (!stricmp("MediaCompletionType", tag))
+ {
+ mediaCompletionType.clear();
+ }
+ }
+ }
+ responseDepth++;
+}
+
+// --------------------------------------------------------------------------------
+
+void LLWebRTCProtocolParser::EndTag(const char *tag)
+{
+ const std::string& string = textBuffer;
+
+ responseDepth--;
+
+ if (ignoringTags)
+ {
+ if (ignoreDepth == responseDepth)
+ {
+ LL_DEBUGS("WebRTCProtocolParser") << "end of ignore" << LL_ENDL;
+ ignoringTags = false;
+ }
+ else
+ {
+ LL_DEBUGS("WebRTCProtocolParser") << "ignoring tag " << tag << " (depth = " << responseDepth << ")" << LL_ENDL;
+ }
+ }
+
+ if (!ignoringTags)
+ {
+ LL_DEBUGS("WebRTCProtocolParser") << "processing tag " << tag << " (depth = " << responseDepth << ")" << LL_ENDL;
+
+ // Closing a tag. Finalize the text we've accumulated and reset
+ if (!stricmp("ReturnCode", tag))
+ returnCode = strtol(string.c_str(), NULL, 10);
+ else if (!stricmp("SessionHandle", tag))
+ sessionHandle = string;
+ else if (!stricmp("SessionGroupHandle", tag))
+ sessionGroupHandle = string;
+ else if (!stricmp("StatusCode", tag))
+ statusCode = strtol(string.c_str(), NULL, 10);
+ else if (!stricmp("StatusString", tag))
+ statusString = string;
+ else if (!stricmp("ParticipantURI", tag))
+ uriString = string;
+ else if (!stricmp("Volume", tag))
+ volume = strtol(string.c_str(), NULL, 10);
+ else if (!stricmp("Energy", tag))
+ energy = (F32)strtod(string.c_str(), NULL);
+ else if (!stricmp("IsModeratorMuted", tag))
+ isModeratorMuted = !stricmp(string.c_str(), "true");
+ else if (!stricmp("IsSpeaking", tag))
+ isSpeaking = !stricmp(string.c_str(), "true");
+ else if (!stricmp("Alias", tag))
+ alias = string;
+ else if (!stricmp("NumberOfAliases", tag))
+ numberOfAliases = strtol(string.c_str(), NULL, 10);
+ else if (!stricmp("Application", tag))
+ applicationString = string;
+ else if (!stricmp("ConnectorHandle", tag))
+ connectorHandle = string;
+ else if (!stricmp("VersionID", tag))
+ versionID = string;
+ else if (!stricmp("Version", tag))
+ mBuildID = string;
+ else if (!stricmp("AccountHandle", tag))
+ accountHandle = string;
+ else if (!stricmp("State", tag))
+ state = strtol(string.c_str(), NULL, 10);
+ else if (!stricmp("URI", tag))
+ uriString = string;
+ else if (!stricmp("IsChannel", tag))
+ isChannel = !stricmp(string.c_str(), "true");
+ else if (!stricmp("Incoming", tag))
+ incoming = !stricmp(string.c_str(), "true");
+ else if (!stricmp("Enabled", tag))
+ enabled = !stricmp(string.c_str(), "true");
+ else if (!stricmp("Name", tag))
+ nameString = string;
+ else if (!stricmp("AudioMedia", tag))
+ audioMediaString = string;
+ else if (!stricmp("ChannelName", tag))
+ nameString = string;
+ else if (!stricmp("DisplayName", tag))
+ displayNameString = string;
+ else if (!stricmp("Device", tag))
+ deviceString = string;
+ else if (!stricmp("AccountName", tag))
+ nameString = string;
+ else if (!stricmp("ParticipantType", tag))
+ participantType = strtol(string.c_str(), NULL, 10);
+ else if (!stricmp("IsLocallyMuted", tag))
+ isLocallyMuted = !stricmp(string.c_str(), "true");
+ else if (!stricmp("MicEnergy", tag))
+ energy = (F32)strtod(string.c_str(), NULL);
+ else if (!stricmp("ChannelName", tag))
+ nameString = string;
+ else if (!stricmp("ChannelURI", tag))
+ uriString = string;
+ else if (!stricmp("BuddyURI", tag))
+ uriString = string;
+ else if (!stricmp("Presence", tag))
+ statusString = string;
+ else if (!stricmp("CaptureDevices", tag))
+ {
+ LLWebRTCVoiceClient::getInstance()->setDevicesListUpdated(true);
+ }
+ else if (!stricmp("RenderDevices", tag))
+ {
+ LLWebRTCVoiceClient::getInstance()->setDevicesListUpdated(true);
+ }
+ else if (!stricmp("CaptureDevice", tag))
+ {
+ LLWebRTCVoiceClient::getInstance()->addCaptureDevice(LLVoiceDevice(displayNameString, deviceString));
+ }
+ else if (!stricmp("RenderDevice", tag))
+ {
+ LLWebRTCVoiceClient::getInstance()->addRenderDevice(LLVoiceDevice(displayNameString, deviceString));
+ }
+ else if (!stricmp("BlockMask", tag))
+ blockMask = string;
+ else if (!stricmp("PresenceOnly", tag))
+ presenceOnly = string;
+ else if (!stricmp("AutoAcceptMask", tag))
+ autoAcceptMask = string;
+ else if (!stricmp("AutoAddAsBuddy", tag))
+ autoAddAsBuddy = string;
+ else if (!stricmp("MessageHeader", tag))
+ messageHeader = string;
+ else if (!stricmp("MessageBody", tag))
+ messageBody = string;
+ else if (!stricmp("NotificationType", tag))
+ notificationType = string;
+ else if (!stricmp("HasText", tag))
+ hasText = !stricmp(string.c_str(), "true");
+ else if (!stricmp("HasAudio", tag))
+ hasAudio = !stricmp(string.c_str(), "true");
+ else if (!stricmp("HasVideo", tag))
+ hasVideo = !stricmp(string.c_str(), "true");
+ else if (!stricmp("Terminated", tag))
+ terminated = !stricmp(string.c_str(), "true");
+ else if (!stricmp("SubscriptionHandle", tag))
+ subscriptionHandle = string;
+ else if (!stricmp("SubscriptionType", tag))
+ subscriptionType = string;
+ else if (!stricmp("SessionFont", tag))
+ {
+ LLWebRTCVoiceClient::getInstance()->addVoiceFont(id, nameString, descriptionString, expirationDate, hasExpired, fontType, fontStatus, false);
+ }
+ else if (!stricmp("TemplateFont", tag))
+ {
+ LLWebRTCVoiceClient::getInstance()->addVoiceFont(id, nameString, descriptionString, expirationDate, hasExpired, fontType, fontStatus, true);
+ }
+ else if (!stricmp("ID", tag))
+ {
+ id = strtol(string.c_str(), NULL, 10);
+ }
+ else if (!stricmp("Description", tag))
+ {
+ descriptionString = string;
+ }
+ else if (!stricmp("ExpirationDate", tag))
+ {
+ expirationDate = expiryTimeStampToLLDate(string);
+ }
+ else if (!stricmp("Expired", tag))
+ {
+ hasExpired = !stricmp(string.c_str(), "1");
+ }
+ else if (!stricmp("Type", tag))
+ {
+ fontType = strtol(string.c_str(), NULL, 10);
+ }
+ else if (!stricmp("Status", tag))
+ {
+ fontStatus = strtol(string.c_str(), NULL, 10);
+ }
+ else if (!stricmp("MediaCompletionType", tag))
+ {
+ mediaCompletionType = string;;
+ }
+
+ textBuffer.clear();
+ accumulateText= false;
+
+ if (responseDepth == 0)
+ {
+ // We finished all of the XML, process the data
+ processResponse(tag);
+ }
+ }
+}
+
+// --------------------------------------------------------------------------------
+
+void LLWebRTCProtocolParser::CharData(const char *buffer, int length)
+{
+ /*
+ This method is called for anything that isn't a tag, which can be text you
+ want that lies between tags, and a lot of stuff you don't want like file formatting
+ (tabs, spaces, CR/LF, etc).
+
+ Only copy text if we are in accumulate mode...
+ */
+ if (accumulateText)
+ textBuffer.append(buffer, length);
+}
+
+// --------------------------------------------------------------------------------
+
+LLDate LLWebRTCProtocolParser::expiryTimeStampToLLDate(const std::string& WebRTC_ts)
+{
+ // *HACK: WebRTC reports the time incorrectly. LLDate also only parses a
+ // subset of valid ISO 8601 dates (only handles Z, not offsets).
+ // So just use the date portion and fix the time here.
+ std::string time_stamp = WebRTC_ts.substr(0, 10);
+ time_stamp += VOICE_FONT_EXPIRY_TIME;
+
+ LL_DEBUGS("WebRTCProtocolParser") << "WebRTC timestamp " << WebRTC_ts << " modified to: " << time_stamp << LL_ENDL;
+
+ return LLDate(time_stamp);
+}
+
+// --------------------------------------------------------------------------------
+
+void LLWebRTCProtocolParser::processResponse(std::string tag)
+{
+ LL_DEBUGS("WebRTCProtocolParser") << tag << LL_ENDL;
+
+ // SLIM SDK: the SDK now returns a statusCode of "200" (OK) for success. This is a change vs. previous SDKs.
+ // According to Mike S., "The actual API convention is that responses with return codes of 0 are successful, regardless of the status code returned",
+ // so I believe this will give correct behavior.
+
+ if(returnCode == 0)
+ statusCode = 0;
+
+ if (isEvent)
+ {
+ const char *eventTypeCstr = eventTypeString.c_str();
+ LL_DEBUGS("LowVoice") << eventTypeCstr << LL_ENDL;
+
+ if (!stricmp(eventTypeCstr, "ParticipantUpdatedEvent"))
+ {
+ // These happen so often that logging them is pretty useless.
+ LL_DEBUGS("LowVoice") << "Updated Params: " << sessionHandle << ", " << sessionGroupHandle << ", " << uriString << ", " << alias << ", " << isModeratorMuted << ", " << isSpeaking << ", " << volume << ", " << energy << LL_ENDL;
+ LLWebRTCVoiceClient::getInstance()->participantUpdatedEvent(sessionHandle, sessionGroupHandle, uriString, alias, isModeratorMuted, isSpeaking, volume, energy);
+ }
+ else if (!stricmp(eventTypeCstr, "AccountLoginStateChangeEvent"))
+ {
+ LLWebRTCVoiceClient::getInstance()->accountLoginStateChangeEvent(accountHandle, statusCode, statusString, state);
+ }
+ else if (!stricmp(eventTypeCstr, "SessionAddedEvent"))
+ {
+ /*
+ <Event type="SessionAddedEvent">
+ <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg0</SessionGroupHandle>
+ <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==0</SessionHandle>
+ <Uri>sip:confctl-1408789@bhr.WebRTC.com</Uri>
+ <IsChannel>true</IsChannel>
+ <Incoming>false</Incoming>
+ <ChannelName />
+ </Event>
+ */
+ LLWebRTCVoiceClient::getInstance()->sessionAddedEvent(uriString, alias, sessionHandle, sessionGroupHandle, isChannel, incoming, nameString, applicationString);
+ }
+ else if (!stricmp(eventTypeCstr, "SessionRemovedEvent"))
+ {
+ LLWebRTCVoiceClient::getInstance()->sessionRemovedEvent(sessionHandle, sessionGroupHandle);
+ }
+ else if (!stricmp(eventTypeCstr, "SessionGroupUpdatedEvent"))
+ {
+ //nothng useful to process for this event, but we should not WARN that we have received it.
+ }
+ else if (!stricmp(eventTypeCstr, "SessionGroupAddedEvent"))
+ {
+ LLWebRTCVoiceClient::getInstance()->sessionGroupAddedEvent(sessionGroupHandle);
+ }
+ else if (!stricmp(eventTypeCstr, "MediaStreamUpdatedEvent"))
+ {
+ /*
+ <Event type="MediaStreamUpdatedEvent">
+ <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg0</SessionGroupHandle>
+ <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==0</SessionHandle>
+ <StatusCode>200</StatusCode>
+ <StatusString>OK</StatusString>
+ <State>2</State>
+ <Incoming>false</Incoming>
+ </Event>
+ */
+ LLWebRTCVoiceClient::getInstance()->mediaStreamUpdatedEvent(sessionHandle, sessionGroupHandle, statusCode, statusString, state, incoming);
+ }
+ else if (!stricmp(eventTypeCstr, "MediaCompletionEvent"))
+ {
+ /*
+ <Event type="MediaCompletionEvent">
+ <SessionGroupHandle />
+ <MediaCompletionType>AuxBufferAudioCapture</MediaCompletionType>
+ </Event>
+ */
+ LLWebRTCVoiceClient::getInstance()->mediaCompletionEvent(sessionGroupHandle, mediaCompletionType);
+ }
+ else if (!stricmp(eventTypeCstr, "ParticipantAddedEvent"))
+ {
+ /*
+ <Event type="ParticipantAddedEvent">
+ <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg4</SessionGroupHandle>
+ <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==4</SessionHandle>
+ <ParticipantUri>sip:xI5auBZ60SJWIk606-1JGRQ==@bhr.WebRTC.com</ParticipantUri>
+ <AccountName>xI5auBZ60SJWIk606-1JGRQ==</AccountName>
+ <DisplayName />
+ <ParticipantType>0</ParticipantType>
+ </Event>
+ */
+ LL_DEBUGS("LowVoice") << "Added Params: " << sessionHandle << ", " << sessionGroupHandle << ", " << uriString << ", " << alias << ", " << nameString << ", " << displayNameString << ", " << participantType << LL_ENDL;
+ LLWebRTCVoiceClient::getInstance()->participantAddedEvent(sessionHandle, sessionGroupHandle, uriString, alias, nameString, displayNameString, participantType);
+ }
+ else if (!stricmp(eventTypeCstr, "ParticipantRemovedEvent"))
+ {
+ /*
+ <Event type="ParticipantRemovedEvent">
+ <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg4</SessionGroupHandle>
+ <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==4</SessionHandle>
+ <ParticipantUri>sip:xtx7YNV-3SGiG7rA1fo5Ndw==@bhr.WebRTC.com</ParticipantUri>
+ <AccountName>xtx7YNV-3SGiG7rA1fo5Ndw==</AccountName>
+ </Event>
+ */
+ LL_DEBUGS("LowVoice") << "Removed params:" << sessionHandle << ", " << sessionGroupHandle << ", " << uriString << ", " << alias << ", " << nameString << LL_ENDL;
+
+ LLWebRTCVoiceClient::getInstance()->participantRemovedEvent(sessionHandle, sessionGroupHandle, uriString, alias, nameString);
+ }
+ else if (!stricmp(eventTypeCstr, "AuxAudioPropertiesEvent"))
+ {
+ // These are really spammy in tuning mode
+ LLWebRTCVoiceClient::getInstance()->auxAudioPropertiesEvent(energy);
+ }
+ else if (!stricmp(eventTypeCstr, "MessageEvent"))
+ {
+ //TODO: This probably is not received any more, it was used to support SLim clients
+ LLWebRTCVoiceClient::getInstance()->messageEvent(sessionHandle, uriString, alias, messageHeader, messageBody, applicationString);
+ }
+ else if (!stricmp(eventTypeCstr, "SessionNotificationEvent"))
+ {
+ //TODO: This probably is not received any more, it was used to support SLim clients
+ LLWebRTCVoiceClient::getInstance()->sessionNotificationEvent(sessionHandle, uriString, notificationType);
+ }
+ else if (!stricmp(eventTypeCstr, "SessionUpdatedEvent"))
+ {
+ /*
+ <Event type="SessionUpdatedEvent">
+ <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg0</SessionGroupHandle>
+ <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==0</SessionHandle>
+ <Uri>sip:confctl-9@bhd.WebRTC.com</Uri>
+ <IsMuted>0</IsMuted>
+ <Volume>50</Volume>
+ <TransmitEnabled>1</TransmitEnabled>
+ <IsFocused>0</IsFocused>
+ <SpeakerPosition><Position><X>0</X><Y>0</Y><Z>0</Z></Position></SpeakerPosition>
+ <SessionFontID>0</SessionFontID>
+ </Event>
+ */
+ // We don't need to process this, but we also shouldn't warn on it, since that confuses people.
+ }
+ else if (!stricmp(eventTypeCstr, "SessionGroupRemovedEvent"))
+ {
+ // We don't need to process this, but we also shouldn't warn on it, since that confuses people.
+ }
+ else if (!stricmp(eventTypeCstr, "VoiceServiceConnectionStateChangedEvent"))
+ {
+ LLWebRTCVoiceClient::getInstance()->voiceServiceConnectionStateChangedEvent(statusCode, statusString, mBuildID);
+ }
+ else if (!stricmp(eventTypeCstr, "AudioDeviceHotSwapEvent"))
+ {
+ /*
+ <Event type = "AudioDeviceHotSwapEvent">
+ <EventType>RenderDeviceChanged< / EventType>
+ <RelevantDevice>
+ <Device>Speakers(Turtle Beach P11 Headset)< / Device>
+ <DisplayName>Speakers(Turtle Beach P11 Headset)< / DisplayName>
+ <Type>SpecificDevice< / Type>
+ < / RelevantDevice>
+ < / Event>
+ */
+ // an audio device was removed or added, fetch and update the local list of audio devices.
+ }
+ else
+ {
+ LL_WARNS("WebRTCProtocolParser") << "Unknown event type " << eventTypeString << LL_ENDL;
+ }
+ }
+ else
+ {
+ const char *actionCstr = actionString.c_str();
+ LL_DEBUGS("LowVoice") << actionCstr << LL_ENDL;
+
+ if (!stricmp(actionCstr, "Session.Set3DPosition.1"))
+ {
+ // We don't need to process these
+ }
+ else if (!stricmp(actionCstr, "Connector.Create.1"))
+ {
+ LLWebRTCVoiceClient::getInstance()->connectorCreateResponse(statusCode, statusString, connectorHandle, versionID);
+ }
+ else if (!stricmp(actionCstr, "Account.Login.1"))
+ {
+ LLWebRTCVoiceClient::getInstance()->loginResponse(statusCode, statusString, accountHandle, numberOfAliases);
+ }
+ else if (!stricmp(actionCstr, "Session.Create.1"))
+ {
+ LLWebRTCVoiceClient::getInstance()->sessionCreateResponse(requestId, statusCode, statusString, sessionHandle);
+ }
+ else if (!stricmp(actionCstr, "SessionGroup.AddSession.1"))
+ {
+ LLWebRTCVoiceClient::getInstance()->sessionGroupAddSessionResponse(requestId, statusCode, statusString, sessionHandle);
+ }
+ else if (!stricmp(actionCstr, "Session.Connect.1"))
+ {
+ LLWebRTCVoiceClient::getInstance()->sessionConnectResponse(requestId, statusCode, statusString);
+ }
+ else if (!stricmp(actionCstr, "Account.Logout.1"))
+ {
+ LLWebRTCVoiceClient::getInstance()->logoutResponse(statusCode, statusString);
+ }
+ else if (!stricmp(actionCstr, "Connector.InitiateShutdown.1"))
+ {
+ LLWebRTCVoiceClient::getInstance()->connectorShutdownResponse(statusCode, statusString);
+ }
+ else if (!stricmp(actionCstr, "Account.GetSessionFonts.1"))
+ {
+ LLWebRTCVoiceClient::getInstance()->accountGetSessionFontsResponse(statusCode, statusString);
+ }
+ else if (!stricmp(actionCstr, "Account.GetTemplateFonts.1"))
+ {
+ LLWebRTCVoiceClient::getInstance()->accountGetTemplateFontsResponse(statusCode, statusString);
+ }
+ else if (!stricmp(actionCstr, "Aux.SetVadProperties.1"))
+ {
+ // both values of statusCode (old and more recent) indicate valid requests
+ if (statusCode != 0 && statusCode != 200)
+ {
+ LL_WARNS("Voice") << "Aux.SetVadProperties.1 request failed: "
+ << "statusCode: " << statusCode
+ << " and "
+ << "statusString: " << statusString
+ << LL_ENDL;
+ }
+ }
+ /*
+ else if (!stricmp(actionCstr, "Account.ChannelGetList.1"))
+ {
+ LLVoiceClient::getInstance()->channelGetListResponse(statusCode, statusString);
+ }
+ else if (!stricmp(actionCstr, "Connector.AccountCreate.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Connector.MuteLocalMic.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Connector.MuteLocalSpeaker.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Connector.SetLocalMicVolume.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Connector.SetLocalSpeakerVolume.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Session.ListenerSetPosition.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Session.SpeakerSetPosition.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Session.AudioSourceSetPosition.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Session.GetChannelParticipants.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Account.ChannelCreate.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Account.ChannelUpdate.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Account.ChannelDelete.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Account.ChannelCreateAndInvite.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Account.ChannelFolderCreate.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Account.ChannelFolderUpdate.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Account.ChannelFolderDelete.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Account.ChannelAddModerator.1"))
+ {
+
+ }
+ else if (!stricmp(actionCstr, "Account.ChannelDeleteModerator.1"))
+ {
+
+ }
+ */
+ }
+}
+
+LLWebRTCSecurity::LLWebRTCSecurity()
+{
+ // This size is an arbitrary choice; WebRTC does not care
+ // Use a multiple of three so that there is no '=' padding in the base64 (purely an esthetic choice)
+ #define WebRTC_TOKEN_BYTES 9
+ U8 random_value[WebRTC_TOKEN_BYTES];
+
+ for (int b = 0; b < WebRTC_TOKEN_BYTES; b++)
+ {
+ random_value[b] = ll_rand() & 0xff;
+ }
+ mConnectorHandle = LLBase64::encode(random_value, WebRTC_TOKEN_BYTES);
+
+ for (int b = 0; b < WebRTC_TOKEN_BYTES; b++)
+ {
+ random_value[b] = ll_rand() & 0xff;
+ }
+ mAccountHandle = LLBase64::encode(random_value, WebRTC_TOKEN_BYTES);
+}
+
+LLWebRTCSecurity::~LLWebRTCSecurity()
+{
+}
diff --git a/indra/newview/llvoicewebrtc.h b/indra/newview/llvoicewebrtc.h
new file mode 100644
index 0000000000..56ca74f6e1
--- /dev/null
+++ b/indra/newview/llvoicewebrtc.h
@@ -0,0 +1,1102 @@
+/**
+ * @file llvoicewebrtc.h
+ * @brief Declaration of LLWebRTCVoiceClient class which is the interface to the voice client process.
+ *
+ * $LicenseInfo:firstyear=2001&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2023, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation;
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+#ifndef LL_VOICE_WEBRTC_H
+#define LL_VOICE_WEBRTC_H
+
+class LLVOAvatar;
+class LLWebRTCProtocolParser;
+
+#include "lliopipe.h"
+#include "llpumpio.h"
+#include "llchainio.h"
+#include "lliosocket.h"
+#include "v3math.h"
+#include "llframetimer.h"
+#include "llviewerregion.h"
+#include "llcallingcard.h" // for LLFriendObserver
+#include "lleventcoro.h"
+#include "llcoros.h"
+#include <queue>
+
+#ifdef LL_USESYSTEMLIBS
+# include "expat.h"
+#else
+# include "expat/expat.h"
+#endif
+#include "llvoiceclient.h"
+
+// WebRTC Includes
+#include <llwebrtc.h>
+
+class LLAvatarName;
+class LLWebRTCVoiceClientMuteListObserver;
+
+
+class LLWebRTCVoiceClient : public LLSingleton<LLWebRTCVoiceClient>,
+ virtual public LLVoiceModuleInterface,
+ virtual public LLVoiceEffectInterface,
+ public llwebrtc::LLWebRTCDevicesObserver,
+ public llwebrtc::LLWebRTCSignalingObserver
+{
+ LLSINGLETON(LLWebRTCVoiceClient);
+ LOG_CLASS(LLWebRTCVoiceClient);
+ virtual ~LLWebRTCVoiceClient();
+
+public:
+ /// @name LLVoiceModuleInterface virtual implementations
+ /// @see LLVoiceModuleInterface
+ //@{
+ virtual void init(LLPumpIO *pump); // Call this once at application startup (creates connector)
+ virtual void terminate(); // Call this to clean up during shutdown
+
+ virtual const LLVoiceVersionInfo& getVersion();
+
+ virtual void updateSettings(); // call after loading settings and whenever they change
+
+ // Returns true if WebRTC has successfully logged in and is not in error state
+ virtual bool isVoiceWorking() const;
+
+ /////////////////////
+ /// @name Tuning
+ //@{
+ virtual void tuningStart();
+ virtual void tuningStop();
+ virtual bool inTuningMode();
+
+ virtual void tuningSetMicVolume(float volume);
+ virtual void tuningSetSpeakerVolume(float volume);
+ virtual float tuningGetEnergy(void);
+ //@}
+
+ /////////////////////
+ /// @name Devices
+ //@{
+ // This returns true when it's safe to bring up the "device settings" dialog in the prefs.
+ // i.e. when the daemon is running and connected, and the device lists are populated.
+ virtual bool deviceSettingsAvailable();
+ virtual bool deviceSettingsUpdated(); //return if the list has been updated and never fetched, only to be called from the voicepanel.
+
+ // Requery the WebRTC daemon for the current list of input/output devices.
+ // If you pass true for clearCurrentList, deviceSettingsAvailable() will be false until the query has completed
+ // (use this if you want to know when it's done).
+ // If you pass false, you'll have no way to know when the query finishes, but the device lists will not appear empty in the interim.
+ virtual void refreshDeviceLists(bool clearCurrentList = true);
+
+ virtual void setCaptureDevice(const std::string& name);
+ virtual void setRenderDevice(const std::string& name);
+
+ virtual LLVoiceDeviceList& getCaptureDevices();
+ virtual LLVoiceDeviceList& getRenderDevices();
+ //@}
+
+ virtual void getParticipantList(std::set<LLUUID> &participants);
+ virtual bool isParticipant(const LLUUID& speaker_id);
+
+ // Send a text message to the specified user, initiating the session if necessary.
+ // virtual BOOL sendTextMessage(const LLUUID& participant_id, const std::string& message) const {return false;};
+
+ // close any existing text IM session with the specified user
+ virtual void endUserIMSession(const LLUUID &uuid);
+
+ // Returns true if calling back the session URI after the session has closed is possible.
+ // Currently this will be false only for PSTN P2P calls.
+ // NOTE: this will return true if the session can't be found.
+ virtual BOOL isSessionCallBackPossible(const LLUUID &session_id);
+
+ // Returns true if the session can accepte text IM's.
+ // Currently this will be false only for PSTN P2P calls.
+ // NOTE: this will return true if the session can't be found.
+ virtual BOOL isSessionTextIMPossible(const LLUUID &session_id);
+
+
+ ////////////////////////////
+ /// @name Channel stuff
+ //@{
+ // returns true iff the user is currently in a proximal (local spatial) channel.
+ // Note that gestures should only fire if this returns true.
+ virtual bool inProximalChannel();
+
+ virtual void setNonSpatialChannel(const std::string &uri,
+ const std::string &credentials);
+
+ virtual bool setSpatialChannel(const std::string &uri,
+ const std::string &credentials);
+
+ virtual void leaveNonSpatialChannel();
+
+ virtual void leaveChannel(void);
+
+ // Returns the URI of the current channel, or an empty string if not currently in a channel.
+ // NOTE that it will return an empty string if it's in the process of joining a channel.
+ virtual std::string getCurrentChannel();
+ //@}
+
+
+ //////////////////////////
+ /// @name invitations
+ //@{
+ // start a voice channel with the specified user
+ virtual void callUser(const LLUUID &uuid);
+ virtual bool isValidChannel(std::string &channelHandle);
+ virtual bool answerInvite(std::string &channelHandle);
+ virtual void declineInvite(std::string &channelHandle);
+ //@}
+
+ /////////////////////////
+ /// @name Volume/gain
+ //@{
+ virtual void setVoiceVolume(F32 volume);
+ virtual void setMicGain(F32 volume);
+ //@}
+
+ /////////////////////////
+ /// @name enable disable voice and features
+ //@{
+ virtual bool voiceEnabled();
+ virtual void setVoiceEnabled(bool enabled);
+ virtual BOOL lipSyncEnabled();
+ virtual void setLipSyncEnabled(BOOL enabled);
+ virtual void setMuteMic(bool muted); // Set the mute state of the local mic.
+ //@}
+
+ //////////////////////////
+ /// @name nearby speaker accessors
+ //@{
+ virtual BOOL getVoiceEnabled(const LLUUID& id); // true if we've received data for this avatar
+ virtual std::string getDisplayName(const LLUUID& id);
+ virtual BOOL isParticipantAvatar(const LLUUID &id);
+ virtual BOOL getIsSpeaking(const LLUUID& id);
+ virtual BOOL getIsModeratorMuted(const LLUUID& id);
+ virtual F32 getCurrentPower(const LLUUID& id); // "power" is related to "amplitude" in a defined way. I'm just not sure what the formula is...
+ virtual BOOL getOnMuteList(const LLUUID& id);
+ virtual F32 getUserVolume(const LLUUID& id);
+ virtual void setUserVolume(const LLUUID& id, F32 volume); // set's volume for specified agent, from 0-1 (where .5 is nominal)
+ //@}
+
+ // authorize the user
+ virtual void userAuthorized(const std::string& user_id,
+ const LLUUID &agentID);
+
+ //////////////////////////////
+ /// @name Status notification
+ //@{
+ virtual void addObserver(LLVoiceClientStatusObserver* observer);
+ virtual void removeObserver(LLVoiceClientStatusObserver* observer);
+ virtual void addObserver(LLFriendObserver* observer);
+ virtual void removeObserver(LLFriendObserver* observer);
+ virtual void addObserver(LLVoiceClientParticipantObserver* observer);
+ virtual void removeObserver(LLVoiceClientParticipantObserver* observer);
+ //@}
+
+ virtual std::string sipURIFromID(const LLUUID &id);
+ //@}
+
+ /// @name LLVoiceEffectInterface virtual implementations
+ /// @see LLVoiceEffectInterface
+ //@{
+
+ //////////////////////////
+ /// @name Accessors
+ //@{
+ virtual bool setVoiceEffect(const LLUUID& id);
+ virtual const LLUUID getVoiceEffect();
+ virtual LLSD getVoiceEffectProperties(const LLUUID& id);
+
+ virtual void refreshVoiceEffectLists(bool clear_lists);
+ virtual const voice_effect_list_t& getVoiceEffectList() const;
+ virtual const voice_effect_list_t& getVoiceEffectTemplateList() const;
+ //@}
+
+ //////////////////////////////
+ /// @name Status notification
+ //@{
+ virtual void addObserver(LLVoiceEffectObserver* observer);
+ virtual void removeObserver(LLVoiceEffectObserver* observer);
+ //@}
+
+ //////////////////////////////
+ /// @name Devices change notification
+ // LLWebRTCDevicesObserver
+ //@{
+ void OnRenderDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceList &render_devices) override;
+ void OnCaptureDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceList &render_devices) override;
+ //@}
+
+ //////////////////////////////
+ /// @name Signaling notification
+ // LLWebRTCSignalingObserver
+ //@{
+ void OnIceGatheringState(IceGatheringState state) override;
+ void OnIceCandidate(const llwebrtc::LLWebRTCIceCandidate &candidate) override;
+ void OnOfferAvailable(const std::string &sdp) override;
+ void OnAudioEstablished(llwebrtc::LLWebRTCAudioInterface *audio_interface) override;
+ //@}
+
+ void processIceUpdates();
+ void onIceUpdateComplete(const LLSD& result);
+ void onIceUpdateError(int retries, std::string url, LLSD body, const LLSD& result);
+
+
+ //////////////////////////////
+ /// @name Effect preview buffer
+ //@{
+ virtual void enablePreviewBuffer(bool enable);
+ virtual void recordPreviewBuffer();
+ virtual void playPreviewBuffer(const LLUUID& effect_id = LLUUID::null);
+ virtual void stopPreviewBuffer();
+
+ virtual bool isPreviewRecording();
+ virtual bool isPreviewPlaying();
+ //@}
+
+ //@}
+
+ bool onCheckVoiceEffect(const std::string& voice_effect_name);
+ void onClickVoiceEffect(const std::string& voice_effect_name);
+
+protected:
+ //////////////////////
+ // WebRTC Specific definitions
+
+ friend class LLWebRTCVoiceClientMuteListObserver;
+ friend class LLWebRTCVoiceClientFriendsObserver;
+
+
+ enum streamState
+ {
+ streamStateUnknown = 0,
+ streamStateIdle = 1,
+ streamStateConnected = 2,
+ streamStateRinging = 3,
+ streamStateConnecting = 6, // same as WebRTC session_media_connecting enum
+ streamStateDisconnecting = 7, //Same as WebRTC session_media_disconnecting enum
+ };
+
+ struct participantState
+ {
+ public:
+ participantState(const std::string &uri);
+
+ bool updateMuteState(); // true if mute state has changed
+ bool isAvatar();
+
+ std::string mURI;
+ LLUUID mAvatarID;
+ std::string mAccountName;
+ std::string mDisplayName;
+ LLFrameTimer mSpeakingTimeout;
+ F32 mLastSpokeTimestamp;
+ F32 mPower;
+ F32 mVolume;
+ std::string mGroupID;
+ int mUserVolume;
+ bool mPTT;
+ bool mIsSpeaking;
+ bool mIsModeratorMuted;
+ bool mOnMuteList; // true if this avatar is on the user's mute list (and should be muted)
+ bool mVolumeSet; // true if incoming volume messages should not change the volume
+ bool mVolumeDirty; // true if this participant needs a volume command sent (either mOnMuteList or mUserVolume has changed)
+ bool mAvatarIDValid;
+ bool mIsSelf;
+ };
+ typedef boost::shared_ptr<participantState> participantStatePtr_t;
+ typedef boost::weak_ptr<participantState> participantStateWptr_t;
+
+ typedef std::map<const std::string, participantStatePtr_t> participantMap;
+ typedef std::map<const LLUUID, participantStatePtr_t> participantUUIDMap;
+
+ struct sessionState
+ {
+ public:
+ typedef boost::shared_ptr<sessionState> ptr_t;
+ typedef boost::weak_ptr<sessionState> wptr_t;
+
+ typedef boost::function<void(const ptr_t &)> sessionFunc_t;
+
+ static ptr_t createSession();
+ ~sessionState();
+
+ participantStatePtr_t addParticipant(const std::string &uri);
+ void removeParticipant(const participantStatePtr_t &participant);
+ void removeAllParticipants();
+
+ participantStatePtr_t findParticipant(const std::string &uri);
+ participantStatePtr_t findParticipantByID(const LLUUID& id);
+
+ static ptr_t matchSessionByHandle(const std::string &handle);
+ static ptr_t matchCreatingSessionByURI(const std::string &uri);
+ static ptr_t matchSessionByURI(const std::string &uri);
+ static ptr_t matchSessionByParticipant(const LLUUID &participant_id);
+
+ bool isCallBackPossible();
+ bool isTextIMPossible();
+
+ static void for_each(sessionFunc_t func);
+
+ std::string mHandle;
+ std::string mGroupHandle;
+ std::string mSIPURI;
+ std::string mAlias;
+ std::string mName;
+ std::string mAlternateSIPURI;
+ std::string mHash; // Channel password
+ std::string mErrorStatusString;
+ std::queue<std::string> mTextMsgQueue;
+
+ LLUUID mIMSessionID;
+ LLUUID mCallerID;
+ int mErrorStatusCode;
+ int mMediaStreamState;
+ bool mCreateInProgress; // True if a Session.Create has been sent for this session and no response has been received yet.
+ bool mMediaConnectInProgress; // True if a Session.MediaConnect has been sent for this session and no response has been received yet.
+ bool mVoiceInvitePending; // True if a voice invite is pending for this session (usually waiting on a name lookup)
+ bool mTextInvitePending; // True if a text invite is pending for this session (usually waiting on a name lookup)
+ bool mSynthesizedCallerID; // True if the caller ID is a hash of the SIP URI -- this means we shouldn't do a name lookup.
+ bool mIsChannel; // True for both group and spatial channels (false for p2p, PSTN)
+ bool mIsSpatial; // True for spatial channels
+ bool mIsP2P;
+ bool mIncoming;
+ bool mVoiceActive;
+ bool mReconnect; // Whether we should try to reconnect to this session if it's dropped
+
+ // Set to true when the volume/mute state of someone in the participant list changes.
+ // The code will have to walk the list to find the changed participant(s).
+ bool mVolumeDirty;
+ bool mMuteDirty;
+
+ bool mParticipantsChanged;
+ participantMap mParticipantsByURI;
+ participantUUIDMap mParticipantsByUUID;
+
+ LLUUID mVoiceFontID;
+
+ static void VerifySessions();
+
+ private:
+ sessionState();
+
+ static std::set<wptr_t> mSession; // canonical list of outstanding sessions.
+ std::set<wptr_t>::iterator mMyIterator; // used for delete
+
+ static void for_eachPredicate(const wptr_t &a, sessionFunc_t func);
+
+ static bool testByHandle(const LLWebRTCVoiceClient::sessionState::wptr_t &a, std::string handle);
+ static bool testByCreatingURI(const LLWebRTCVoiceClient::sessionState::wptr_t &a, std::string uri);
+ static bool testBySIPOrAlterateURI(const LLWebRTCVoiceClient::sessionState::wptr_t &a, std::string uri);
+ static bool testByCallerId(const LLWebRTCVoiceClient::sessionState::wptr_t &a, LLUUID participantId);
+
+ };
+ typedef boost::shared_ptr<sessionState> sessionStatePtr_t;
+
+ typedef std::map<std::string, sessionStatePtr_t> sessionMap;
+
+ ///////////////////////////////////////////////////////
+ // Private Member Functions
+ //////////////////////////////////////////////////////
+
+
+
+ //////////////////////////////
+ /// @name TVC/Server management and communication
+ //@{
+ // Call this if the connection to the daemon terminates unexpectedly. It will attempt to reset everything and relaunch.
+ void daemonDied();
+
+ // Call this if we're just giving up on voice (can't provision an account, etc.). It will clean up and go away.
+ void giveUp();
+
+ // write to the tvc
+ bool writeString(const std::string &str);
+
+ void connectorCreate();
+ void connectorShutdown();
+ void closeSocket(void);
+
+// void requestVoiceAccountProvision(S32 retries = 3);
+ void setLoginInfo(
+ const std::string& account_name,
+ const std::string& password,
+ const std::string& channel_sdp);
+ void loginSendMessage();
+ void logout();
+ void logoutSendMessage();
+
+
+ //@}
+
+ //------------------------------------
+ // tuning
+
+ void tuningRenderStartSendMessage(const std::string& name, bool loop);
+ void tuningRenderStopSendMessage();
+
+ void tuningCaptureStartSendMessage(int duration);
+ void tuningCaptureStopSendMessage();
+
+ //----------------------------------
+ // devices
+ void clearCaptureDevices();
+ void addCaptureDevice(const LLVoiceDevice& device);
+
+ void clearRenderDevices();
+ void addRenderDevice(const LLVoiceDevice& device);
+ void setDevicesListUpdated(bool state);
+ void buildSetAudioDevices(std::ostringstream &stream);
+
+ // local audio updates, mic mute, speaker mute, mic volume and speaker volumes
+ void sendLocalAudioUpdates();
+
+ /////////////////////////////
+ // Response/Event handlers
+ void connectorCreateResponse(int statusCode, std::string &statusString, std::string &connectorHandle, std::string &versionID);
+ void loginResponse(int statusCode, std::string &statusString, std::string &accountHandle, int numberOfAliases);
+ void sessionCreateResponse(std::string &requestId, int statusCode, std::string &statusString, std::string &sessionHandle);
+ void sessionGroupAddSessionResponse(std::string &requestId, int statusCode, std::string &statusString, std::string &sessionHandle);
+ void sessionConnectResponse(std::string &requestId, int statusCode, std::string &statusString);
+ void logoutResponse(int statusCode, std::string &statusString);
+ void connectorShutdownResponse(int statusCode, std::string &statusString);
+
+ void accountLoginStateChangeEvent(std::string &accountHandle, int statusCode, std::string &statusString, int state);
+ void mediaCompletionEvent(std::string &sessionGroupHandle, std::string &mediaCompletionType);
+ void mediaStreamUpdatedEvent(std::string &sessionHandle, std::string &sessionGroupHandle, int statusCode, std::string &statusString, int state, bool incoming);
+ void sessionAddedEvent(std::string &uriString, std::string &alias, std::string &sessionHandle, std::string &sessionGroupHandle, bool isChannel, bool incoming, std::string &nameString, std::string &applicationString);
+ void sessionGroupAddedEvent(std::string &sessionGroupHandle);
+ void sessionRemovedEvent(std::string &sessionHandle, std::string &sessionGroupHandle);
+ void participantAddedEvent(std::string &sessionHandle, std::string &sessionGroupHandle, std::string &uriString, std::string &alias, std::string &nameString, std::string &displayNameString, int participantType);
+ void participantRemovedEvent(std::string &sessionHandle, std::string &sessionGroupHandle, std::string &uriString, std::string &alias, std::string &nameString);
+ void participantUpdatedEvent(std::string &sessionHandle, std::string &sessionGroupHandle, std::string &uriString, std::string &alias, bool isModeratorMuted, bool isSpeaking, int volume, F32 energy);
+ void voiceServiceConnectionStateChangedEvent(int statusCode, std::string &statusString, std::string &build_id);
+ void auxAudioPropertiesEvent(F32 energy);
+ void messageEvent(std::string &sessionHandle, std::string &uriString, std::string &alias, std::string &messageHeader, std::string &messageBody, std::string &applicationString);
+ void sessionNotificationEvent(std::string &sessionHandle, std::string &uriString, std::string &notificationType);
+
+ void muteListChanged();
+
+ /////////////////////////////
+ // VAD changes
+ // disable auto-VAD and configure VAD parameters explicitly
+ void setupVADParams(unsigned int vad_auto, unsigned int vad_hangover, unsigned int vad_noise_floor, unsigned int vad_sensitivity);
+ void onVADSettingsChange();
+
+ /////////////////////////////
+ // Sending updates of current state
+ void updatePosition(void);
+ void setCameraPosition(const LLVector3d &position, const LLVector3 &velocity, const LLMatrix3 &rot);
+ void setAvatarPosition(const LLVector3d &position, const LLVector3 &velocity, const LLQuaternion &rot);
+ bool channelFromRegion(LLViewerRegion *region, std::string &name);
+
+ void setEarLocation(S32 loc);
+
+
+ /////////////////////////////
+ // Accessors for data related to nearby speakers
+
+ // MBW -- XXX -- Not sure how to get this data out of the TVC
+ BOOL getUsingPTT(const LLUUID& id);
+ std::string getGroupID(const LLUUID& id); // group ID if the user is in group chat (empty string if not applicable)
+
+ /////////////////////////////
+ BOOL getAreaVoiceDisabled(); // returns true if the area the avatar is in is speech-disabled.
+ // Use this to determine whether to show a "no speech" icon in the menu bar.
+
+
+ /////////////////////////////
+ // Recording controls
+ void recordingLoopStart(int seconds = 3600, int deltaFramesPerControlFrame = 200);
+ void recordingLoopSave(const std::string& filename);
+ void recordingStop();
+
+ // Playback controls
+ void filePlaybackStart(const std::string& filename);
+ void filePlaybackStop();
+ void filePlaybackSetPaused(bool paused);
+ void filePlaybackSetMode(bool vox = false, float speed = 1.0f);
+
+ participantStatePtr_t findParticipantByID(const LLUUID& id);
+
+
+#if 0
+ ////////////////////////////////////////
+ // voice sessions.
+ typedef std::set<sessionStatePtr_t> sessionSet;
+
+ typedef sessionSet::iterator sessionIterator;
+ sessionIterator sessionsBegin(void);
+ sessionIterator sessionsEnd(void);
+#endif
+
+ sessionStatePtr_t findSession(const std::string &handle);
+ sessionStatePtr_t findSessionBeingCreatedByURI(const std::string &uri);
+ sessionStatePtr_t findSession(const LLUUID &participant_id);
+
+ sessionStatePtr_t addSession(const std::string &uri, const std::string &handle = std::string());
+ void clearSessionHandle(const sessionStatePtr_t &session);
+ void setSessionHandle(const sessionStatePtr_t &session, const std::string &handle);
+ void setSessionURI(const sessionStatePtr_t &session, const std::string &uri);
+ void deleteSession(const sessionStatePtr_t &session);
+ void deleteAllSessions(void);
+
+ void verifySessionState(void);
+
+ void joinedAudioSession(const sessionStatePtr_t &session);
+ void leftAudioSession(const sessionStatePtr_t &session);
+
+ // This is called in several places where the session _may_ need to be deleted.
+ // It contains logic for whether to delete the session or keep it around.
+ void reapSession(const sessionStatePtr_t &session);
+
+ // Returns true if the session seems to indicate we've moved to a region on a different voice server
+ bool sessionNeedsRelog(const sessionStatePtr_t &session);
+
+
+ //////////////////////////////////////
+ // buddy list stuff, needed for SLIM later
+ struct buddyListEntry
+ {
+ buddyListEntry(const std::string &uri);
+ std::string mURI;
+ std::string mDisplayName;
+ LLUUID mUUID;
+ bool mOnlineSL;
+ bool mOnlineSLim;
+ bool mCanSeeMeOnline;
+ bool mHasBlockListEntry;
+ bool mHasAutoAcceptListEntry;
+ bool mNameResolved;
+ bool mInSLFriends;
+ bool mInWebRTCBuddies;
+ };
+
+ typedef std::map<std::string, buddyListEntry*> buddyListMap;
+
+ /////////////////////////////
+ // session control messages
+
+ void accountListBlockRulesSendMessage();
+ void accountListAutoAcceptRulesSendMessage();
+
+ void sessionGroupCreateSendMessage();
+ void sessionCreateSendMessage(const sessionStatePtr_t &session, bool startAudio = true, bool startText = false);
+ void sessionGroupAddSessionSendMessage(const sessionStatePtr_t &session, bool startAudio = true, bool startText = false);
+ void sessionMediaConnectSendMessage(const sessionStatePtr_t &session); // just joins the audio session
+ void sessionTextConnectSendMessage(const sessionStatePtr_t &session); // just joins the text session
+ void sessionTerminateSendMessage(const sessionStatePtr_t &session);
+ void sessionGroupTerminateSendMessage(const sessionStatePtr_t &session);
+ void sessionMediaDisconnectSendMessage(const sessionStatePtr_t &session);
+ // void sessionTextDisconnectSendMessage(sessionState *session);
+
+
+
+ // Pokes the state machine to leave the audio session next time around.
+ void sessionTerminate();
+
+ // Pokes the state machine to shut down the connector and restart it.
+ void requestRelog();
+
+ // Does the actual work to get out of the audio session
+ void leaveAudioSession();
+
+ friend class LLWebRTCVoiceClientCapResponder;
+
+
+ void lookupName(const LLUUID &id);
+ void onAvatarNameCache(const LLUUID& id, const LLAvatarName& av_name);
+ void avatarNameResolved(const LLUUID &id, const std::string &name);
+ static void predAvatarNameResolution(const LLWebRTCVoiceClient::sessionStatePtr_t &session, LLUUID id, std::string name);
+
+ boost::signals2::connection mAvatarNameCacheConnection;
+
+ /////////////////////////////
+ // Voice fonts
+
+ void addVoiceFont(const S32 id,
+ const std::string &name,
+ const std::string &description,
+ const LLDate &expiration_date,
+ bool has_expired,
+ const S32 font_type,
+ const S32 font_status,
+ const bool template_font = false);
+ void accountGetSessionFontsResponse(int statusCode, const std::string &statusString);
+ void accountGetTemplateFontsResponse(int statusCode, const std::string &statusString);
+
+private:
+
+ LLVoiceVersionInfo mVoiceVersion;
+
+ // Coroutine support methods
+ //---
+ void voiceControlCoro();
+ void voiceControlStateMachine();
+ int mVoiceControlState;
+
+ bool endAndDisconnectSession();
+
+ bool callbackEndDaemon(const LLSD& data);
+ bool provisionVoiceAccount();
+ void OnVoiceAccountProvisioned(const LLSD& body);
+ void OnVoiceAccountProvisionFailure(std::string url, int retries, LLSD body, const LLSD& result);
+ bool establishVoiceConnection();
+ bool breakVoiceConnection(bool wait);
+ bool loginToWebRTC();
+ void logoutOfWebRTC(bool wait);
+
+ bool requestParcelVoiceInfo();
+
+ bool addAndJoinSession(const sessionStatePtr_t &nextSession);
+ bool terminateAudioSession(bool wait);
+
+ bool waitForChannel();
+ bool runSession(const sessionStatePtr_t &session);
+
+ void recordingAndPlaybackMode();
+ int voiceRecordBuffer();
+ int voicePlaybackBuffer();
+
+ bool performMicTuning();
+ //---
+ /// Clean up objects created during a voice session.
+ void cleanUp();
+
+ bool mSessionTerminateRequested;
+ bool mRelogRequested;
+ // Number of times (in a row) "stateJoiningSession" case for spatial channel is reached in stateMachine().
+ // The larger it is the greater is possibility there is a problem with connection to voice server.
+ // Introduced while fixing EXT-4313.
+ int mSpatialJoiningNum;
+
+ static void idle(void *user_data);
+
+ LLHost mDaemonHost;
+ LLSocket::ptr_t mSocket;
+
+ // We should kill the voice daemon in case of connection alert
+ bool mTerminateDaemon;
+
+ friend class LLWebRTCProtocolParser;
+
+ std::string mAccountName;
+ std::string mAccountPassword;
+ std::string mChannelSDP;
+ std::string mRemoteChannelSDP;
+ std::string mAccountDisplayName;
+
+
+ bool mTuningMode;
+ float mTuningEnergy;
+ std::string mTuningAudioFile;
+ int mTuningMicVolume;
+ bool mTuningMicVolumeDirty;
+ int mTuningSpeakerVolume;
+ bool mTuningSpeakerVolumeDirty;
+ bool mDevicesListUpdated; // set to true when the device list has been updated
+ // and false when the panelvoicedevicesettings has queried for an update status.
+
+ std::string mSpatialSessionURI;
+ std::string mSpatialSessionCredentials;
+
+ std::string mMainSessionGroupHandle; // handle of the "main" session group.
+
+ std::string mChannelName; // Name of the channel to be looked up
+ bool mAreaVoiceDisabled;
+ sessionStatePtr_t mAudioSession; // Session state for the current audio session
+ bool mAudioSessionChanged; // set to true when the above pointer gets changed, so observers can be notified.
+
+ sessionStatePtr_t mNextAudioSession; // Session state for the audio session we're trying to join
+
+ S32 mCurrentParcelLocalID; // Used to detect parcel boundary crossings
+ std::string mCurrentRegionName; // Used to detect parcel boundary crossings
+
+ bool mConnectorEstablished; // set by "Create Connector" response
+ bool mAccountLoggedIn; // set by login message
+ int mNumberOfAliases;
+ U32 mCommandCookie;
+
+ int mLoginRetryCount;
+
+ sessionMap mSessionsByHandle; // Active sessions, indexed by session handle. Sessions which are being initiated may not be in this map.
+#if 0
+ sessionSet mSessions; // All sessions, not indexed. This is the canonical session list.
+#endif
+
+ bool mBuddyListMapPopulated;
+ bool mBlockRulesListReceived;
+ bool mAutoAcceptRulesListReceived;
+ buddyListMap mBuddyListMap;
+
+ llwebrtc::LLWebRTCDeviceInterface *mWebRTCDeviceInterface;
+ llwebrtc::LLWebRTCSignalInterface *mWebRTCSignalingInterface;
+ llwebrtc::LLWebRTCAudioInterface *mWebRTCAudioInterface;
+
+ LLVoiceDeviceList mCaptureDevices;
+ LLVoiceDeviceList mRenderDevices;
+ std::vector<llwebrtc::LLWebRTCIceCandidate> mIceCandidates;
+ bool mIceCompleted;
+
+ bool mIsInitialized;
+ bool mShutdownComplete;
+
+ bool checkParcelChanged(bool update = false);
+ bool switchChannel(std::string uri = std::string(), bool spatial = true, bool no_reconnect = false, bool is_p2p = false, std::string hash = "");
+ void joinSession(const sessionStatePtr_t &session);
+
+ std::string nameFromAvatar(LLVOAvatar *avatar);
+ std::string nameFromID(const LLUUID &id);
+ bool IDFromName(const std::string name, LLUUID &uuid);
+ std::string displayNameFromAvatar(LLVOAvatar *avatar);
+
+
+ bool inSpatialChannel(void);
+ std::string getAudioSessionURI();
+ std::string getAudioSessionHandle();
+
+ void setHidden(bool hidden); //virtual
+ void sendPositionAndVolumeUpdate(void);
+
+ void sendFriendsListUpdates();
+
+#if 0
+ // start a text IM session with the specified user
+ // This will be asynchronous, the session may be established at a future time.
+ sessionStatePtr_t startUserIMSession(const LLUUID& uuid);
+#endif
+
+ void enforceTether(void);
+
+ bool mSpatialCoordsDirty;
+
+ LLVector3d mCameraPosition;
+ LLVector3d mCameraRequestedPosition;
+ LLVector3 mCameraVelocity;
+ LLMatrix3 mCameraRot;
+
+ LLVector3d mAvatarPosition;
+ LLVector3 mAvatarVelocity;
+ LLQuaternion mAvatarRot;
+
+ bool mMuteMic;
+ bool mMuteMicDirty;
+ bool mHidden; //Set to true during teleport to hide the agent's position.
+
+ // Set to true when the friends list is known to have changed.
+ bool mFriendsListDirty;
+
+ enum
+ {
+ earLocCamera = 0, // ear at camera
+ earLocAvatar, // ear at avatar
+ earLocMixed // ear at avatar location/camera direction
+ };
+
+ S32 mEarLocation;
+
+ bool mSpeakerVolumeDirty;
+ bool mSpeakerMuteDirty;
+ int mSpeakerVolume;
+
+ int mMicVolume;
+ bool mMicVolumeDirty;
+
+ bool mVoiceEnabled;
+ bool mWriteInProgress;
+ std::string mWriteString;
+ size_t mWriteOffset;
+
+ BOOL mLipSyncEnabled;
+
+ typedef std::set<LLVoiceClientParticipantObserver*> observer_set_t;
+ observer_set_t mParticipantObservers;
+
+ void notifyParticipantObservers();
+
+ typedef std::set<LLVoiceClientStatusObserver*> status_observer_set_t;
+ status_observer_set_t mStatusObservers;
+
+ void notifyStatusObservers(LLVoiceClientStatusObserver::EStatusType status);
+
+ typedef std::set<LLFriendObserver*> friend_observer_set_t;
+ friend_observer_set_t mFriendObservers;
+ void notifyFriendObservers();
+
+ // Voice Fonts
+
+ void expireVoiceFonts();
+ void deleteVoiceFont(const LLUUID& id);
+ void deleteAllVoiceFonts();
+ void deleteVoiceFontTemplates();
+
+ S32 getVoiceFontIndex(const LLUUID& id) const;
+ S32 getVoiceFontTemplateIndex(const LLUUID& id) const;
+
+ void accountGetSessionFontsSendMessage();
+ void accountGetTemplateFontsSendMessage();
+ void sessionSetVoiceFontSendMessage(const sessionStatePtr_t &session);
+
+ void updateVoiceMorphingMenu();
+ void notifyVoiceFontObservers();
+
+ typedef enum e_voice_font_type
+ {
+ VOICE_FONT_TYPE_NONE = 0,
+ VOICE_FONT_TYPE_ROOT = 1,
+ VOICE_FONT_TYPE_USER = 2,
+ VOICE_FONT_TYPE_UNKNOWN
+ } EVoiceFontType;
+
+ typedef enum e_voice_font_status
+ {
+ VOICE_FONT_STATUS_NONE = 0,
+ VOICE_FONT_STATUS_FREE = 1,
+ VOICE_FONT_STATUS_NOT_FREE = 2,
+ VOICE_FONT_STATUS_UNKNOWN
+ } EVoiceFontStatus;
+
+ struct voiceFontEntry
+ {
+ voiceFontEntry(LLUUID& id);
+ ~voiceFontEntry();
+
+ LLUUID mID;
+ S32 mFontIndex;
+ std::string mName;
+ LLDate mExpirationDate;
+ S32 mFontType;
+ S32 mFontStatus;
+ bool mIsNew;
+
+ LLFrameTimer mExpiryTimer;
+ LLFrameTimer mExpiryWarningTimer;
+ };
+
+ bool mVoiceFontsReceived;
+ bool mVoiceFontsNew;
+ bool mVoiceFontListDirty;
+ voice_effect_list_t mVoiceFontList;
+ voice_effect_list_t mVoiceFontTemplateList;
+
+ typedef std::map<const LLUUID, voiceFontEntry*> voice_font_map_t;
+ voice_font_map_t mVoiceFontMap;
+ voice_font_map_t mVoiceFontTemplateMap;
+
+ typedef std::set<LLVoiceEffectObserver*> voice_font_observer_set_t;
+ voice_font_observer_set_t mVoiceFontObservers;
+
+ LLFrameTimer mVoiceFontExpiryTimer;
+
+
+ // Audio capture buffer
+
+ void captureBufferRecordStartSendMessage();
+ void captureBufferRecordStopSendMessage();
+ void captureBufferPlayStartSendMessage(const LLUUID& voice_font_id = LLUUID::null);
+ void captureBufferPlayStopSendMessage();
+
+ bool mCaptureBufferMode; // Disconnected from voice channels while using the capture buffer.
+ bool mCaptureBufferRecording; // A voice sample is being captured.
+ bool mCaptureBufferRecorded; // A voice sample is captured in the buffer ready to play.
+ bool mCaptureBufferPlaying; // A voice sample is being played.
+
+ LLTimer mCaptureTimer;
+ LLUUID mPreviewVoiceFont;
+ LLUUID mPreviewVoiceFontLast;
+ S32 mPlayRequestCount;
+ bool mIsInTuningMode;
+ bool mIsInChannel;
+ bool mIsJoiningSession;
+ bool mIsWaitingForFonts;
+ bool mIsLoggingIn;
+ bool mIsLoggedIn;
+ bool mIsProcessingChannels;
+ bool mIsCoroutineActive;
+
+ // This variables can last longer than WebRTC in coroutines so we need them as static
+ static bool sShuttingDown;
+ static bool sConnected;
+ static LLPumpIO* sPump;
+
+ LLEventMailDrop mWebRTCPump;
+};
+
+
+/**
+ * @class LLWebRTCProtocolParser
+ * @brief This class helps construct new LLIOPipe specializations
+ * @see LLIOPipe
+ *
+ * THOROUGH_DESCRIPTION
+ */
+class LLWebRTCProtocolParser : public LLIOPipe
+{
+ LOG_CLASS(LLWebRTCProtocolParser);
+public:
+ LLWebRTCProtocolParser();
+ virtual ~LLWebRTCProtocolParser();
+
+protected:
+ /* @name LLIOPipe virtual implementations
+ */
+ //@{
+ /**
+ * @brief Process the data in buffer
+ */
+ virtual EStatus process_impl(
+ const LLChannelDescriptors& channels,
+ buffer_ptr_t& buffer,
+ bool& eos,
+ LLSD& context,
+ LLPumpIO* pump);
+ //@}
+
+ std::string mInput;
+
+ // Expat control members
+ XML_Parser parser;
+ int responseDepth;
+ bool ignoringTags;
+ bool isEvent;
+ int ignoreDepth;
+
+ // Members for processing responses. The values are transient and only valid within a call to processResponse().
+ int returnCode;
+ int statusCode;
+ std::string statusString;
+ std::string requestId;
+ std::string actionString;
+ std::string connectorHandle;
+ std::string versionID;
+ std::string mBuildID;
+ std::string accountHandle;
+ std::string sessionHandle;
+ std::string sessionGroupHandle;
+ std::string alias;
+ std::string applicationString;
+
+ // Members for processing events. The values are transient and only valid within a call to processResponse().
+ std::string eventTypeString;
+ int state;
+ std::string uriString;
+ bool isChannel;
+ bool incoming;
+ bool enabled;
+ std::string nameString;
+ std::string audioMediaString;
+ std::string deviceString;
+ std::string displayNameString;
+ int participantType;
+ bool isLocallyMuted;
+ bool isModeratorMuted;
+ bool isSpeaking;
+ int volume;
+ F32 energy;
+ std::string messageHeader;
+ std::string messageBody;
+ std::string notificationType;
+ bool hasText;
+ bool hasAudio;
+ bool hasVideo;
+ bool terminated;
+ std::string blockMask;
+ std::string presenceOnly;
+ std::string autoAcceptMask;
+ std::string autoAddAsBuddy;
+ int numberOfAliases;
+ std::string subscriptionHandle;
+ std::string subscriptionType;
+ S32 id;
+ std::string descriptionString;
+ LLDate expirationDate;
+ bool hasExpired;
+ S32 fontType;
+ S32 fontStatus;
+ std::string mediaCompletionType;
+
+ // Members for processing text between tags
+ std::string textBuffer;
+ bool accumulateText;
+
+ void reset();
+
+ void processResponse(std::string tag);
+
+ static void XMLCALL ExpatStartTag(void *data, const char *el, const char **attr);
+ static void XMLCALL ExpatEndTag(void *data, const char *el);
+ static void XMLCALL ExpatCharHandler(void *data, const XML_Char *s, int len);
+
+ void StartTag(const char *tag, const char **attr);
+ void EndTag(const char *tag);
+ void CharData(const char *buffer, int length);
+ LLDate expiryTimeStampToLLDate(const std::string& WebRTC_ts);
+
+};
+
+class LLWebRTCSecurity : public LLSingleton<LLWebRTCSecurity>
+{
+ LLSINGLETON(LLWebRTCSecurity);
+ virtual ~LLWebRTCSecurity();
+
+ public:
+ std::string connectorHandle() { return mConnectorHandle; };
+ std::string accountHandle() { return mAccountHandle; };
+
+ private:
+ std::string mConnectorHandle;
+ std::string mAccountHandle;
+};
+
+class LLVoiceWebRTCStats : public LLSingleton<LLVoiceWebRTCStats>
+{
+ LLSINGLETON(LLVoiceWebRTCStats);
+ LOG_CLASS(LLVoiceWebRTCStats);
+ virtual ~LLVoiceWebRTCStats();
+
+ private:
+ F64SecondsImplicit mStartTime;
+
+ U32 mConnectCycles;
+
+ F64 mConnectTime;
+ U32 mConnectAttempts;
+
+ F64 mProvisionTime;
+ U32 mProvisionAttempts;
+
+ F64 mEstablishTime;
+ U32 mEstablishAttempts;
+
+ public:
+
+ void reset();
+ void connectionAttemptStart();
+ void connectionAttemptEnd(bool success);
+ void provisionAttemptStart();
+ void provisionAttemptEnd(bool success);
+ void establishAttemptStart();
+ void establishAttemptEnd(bool success);
+ LLSD read();
+};
+
+#endif //LL_WebRTC_VOICE_CLIENT_H
+
diff --git a/scripts/messages/message_template.msg.sha1 b/scripts/messages/message_template.msg.sha1
index 5ad85458e9..47291dcd66 100755
--- a/scripts/messages/message_template.msg.sha1
+++ b/scripts/messages/message_template.msg.sha1
@@ -1 +1 @@
-e3bd0529a647d938ab6d48f26d21dd52c07ebc6e \ No newline at end of file
+dd15c52581b3fe99e072b26872deba2560893fc4
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-22 06:05:29 +0000