Checkpoint WebRTC Voice

master
Roxie Linden 2023-09-06 15:28:02 -07:00
parent 19b808316b
commit bf58173656
18 changed files with 9322 additions and 7 deletions

View File

@ -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>

View File

@ -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)

View File

@ -63,6 +63,7 @@ set(cmake_SOURCE_FILES
ViewerMiscLibs.cmake
VisualLeakDetector.cmake
LibVLCPlugin.cmake
WebRTC.cmake
XmlRpcEpi.cmake
xxHash.cmake
ZLIBNG.cmake

View File

@ -0,0 +1 @@
# -*- cmake -*-

43
indra/cmake/WebRTC.cmake Normal file
View File

@ -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}")

View File

@ -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)

486
indra/llwebrtc/llwebrtc.cpp Normal file
View File

@ -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

127
indra/llwebrtc/llwebrtc.h Normal file
View File

@ -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

View File

@ -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

View File

@ -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

View File

@ -15128,7 +15128,7 @@
<key>Type</key>
<string>String</string>
<key>Value</key>
<string>vivox</string>
<string>webrtc</string>
</map>
<key>WLSkyDetail</key>
<map>

View File

@ -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);
}
}
}

View File

@ -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");

View File

@ -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;

View File

@ -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"

File diff suppressed because it is too large Load Diff

File diff suppressed because it is too large Load Diff

View File

@ -1 +1 @@
e3bd0529a647d938ab6d48f26d21dd52c07ebc6e
dd15c52581b3fe99e072b26872deba2560893fc4