Merge pull request #1469 from vircadia/webapp

Enable WebRTC connections to be made to the Domain, i.e., add Web client support.

Author: digisomni

CODING_STANDARD.md

@@ -948,7 +948,7 @@ In an international environment English is the preferred language.
 
 #### [4.3.2] Use // for all comments, including multi-line comments.
 
-An exception to this rule applies for jsdoc or Doxygen comments.
+An exception to this rule applies to JSDoc and Doxygen comments.
 
 ```cpp
 // Comment spanning
@@ -1008,3 +1008,11 @@ These types of comments are explicitly not allowed. If you need to break up sect
 //--------------------------------------------------------------------------------
 ```
 
+#### [4.3.6] Doxygen comments should use "///"
+
+Use the `///` style of [Doxygen](https://www.doxygen.nl/index.html) comments when documenting public interfaces.
+
+Some editors can automatically create a Doxygen documentation stub if you type `///` in the line above the item to be
+documented.
+
+**Visual Studio:** To configure Visual Studio's Doxygen commenting behavior, search for "Doxygen" in Tools > Options.

assignment-client/src/AssignmentClient.cpp

@@ -14,6 +14,7 @@
 
 #include <assert.h>
 
+#include <QJsonDocument>
 #include <QProcess>
 #include <QSharedMemory>
 #include <QThread>
@@ -84,7 +85,7 @@ AssignmentClient::AssignmentClient(Assignment::Type requestAssignmentType, QStri
         _assignmentServerHostname = assignmentServerHostname;
     }
 
-    _assignmentServerSocket = SockAddr(_assignmentServerHostname, assignmentServerPort, true);
+    _assignmentServerSocket = SockAddr(SocketType::UDP, _assignmentServerHostname, assignmentServerPort, true);
     if (_assignmentServerSocket.isNull()) {
         qCCritical(assignment_client) << "PAGE: Couldn't resolve domain server address" << _assignmentServerHostname;
     }
@@ -119,7 +120,8 @@ AssignmentClient::AssignmentClient(Assignment::Type requestAssignmentType, QStri
 
     // did we get an assignment-client monitor port?
     if (assignmentMonitorPort > 0) {
-        _assignmentClientMonitorSocket = SockAddr(DEFAULT_ASSIGNMENT_CLIENT_MONITOR_HOSTNAME, assignmentMonitorPort);
+        _assignmentClientMonitorSocket = SockAddr(SocketType::UDP, DEFAULT_ASSIGNMENT_CLIENT_MONITOR_HOSTNAME, 
+            assignmentMonitorPort);
         _assignmentClientMonitorSocket.setObjectName("AssignmentClientMonitor");
 
         qCDebug(assignment_client) << "Assignment-client monitor socket is" << _assignmentClientMonitorSocket;
@@ -132,6 +134,18 @@ AssignmentClient::AssignmentClient(Assignment::Type requestAssignmentType, QStri
         PacketReceiver::makeUnsourcedListenerReference<AssignmentClient>(this, &AssignmentClient::handleCreateAssignmentPacket));
     packetReceiver.registerListener(PacketType::StopNode,
         PacketReceiver::makeUnsourcedListenerReference<AssignmentClient>(this, &AssignmentClient::handleStopNodePacket));
+
+#if defined(WEBRTC_DATA_CHANNELS)
+    auto webrtcSocket = nodeList->getWebRTCSocket();
+
+    // Route inbound WebRTC signaling messages from the Domain Server.
+    packetReceiver.registerListener(PacketType::WebRTCSignaling,
+        PacketReceiver::makeUnsourcedListenerReference<AssignmentClient>(this, &AssignmentClient::handleWebRTCSignalingPacket));
+    connect(this, &AssignmentClient::webrtcSignalingMessageFromUserClient, webrtcSocket, &WebRTCSocket::onSignalingMessage);
+
+    // Route outbound WebRTC signaling messages via the Domain Server to the user client.
+    connect(webrtcSocket, &WebRTCSocket::sendSignalingMessage, this, &AssignmentClient::sendSignalingMessageToUserClient);
+#endif
 }
 
 void AssignmentClient::stopAssignmentClient() {
@@ -340,3 +354,45 @@ void AssignmentClient::assignmentCompleted() {
 
     _isAssigned = false;
 }
+
+#if defined(WEBRTC_DATA_CHANNELS)
+
+void AssignmentClient::handleWebRTCSignalingPacket(QSharedPointer<ReceivedMessage> message) {
+    auto messageString = message->readString();
+    auto json = QJsonDocument::fromJson(messageString.toUtf8()).object();
+    if (json.keys().contains("echo")) {
+        // Echo message back to sender.
+
+        if (!json.keys().contains("to") || !json.keys().contains("from")) {
+            qCDebug(assignment_client) << "Invalid WebRTC signaling echo message received.";
+            return;
+        }
+
+        // Swap to/from.
+        auto to = json.value("to");
+        json.insert("to", json.value("from"));
+        json.insert("from", to);
+
+        // Send back to sender via the Domain Server.
+        auto packetList = NLPacketList::create(PacketType::WebRTCSignaling, QByteArray(), true, true);
+        packetList->writeString(QJsonDocument(json).toJson(QJsonDocument::Compact));
+        auto nodeList = DependencyManager::get<NodeList>();
+        auto domainServerAddress = nodeList->getDomainHandler().getSockAddr();
+        nodeList->sendPacketList(std::move(packetList), domainServerAddress);
+
+    } else {
+        // WebRTC signaling message.
+        emit webrtcSignalingMessageFromUserClient(json);
+    }
+}
+
+// Sends a signaling message from the assignment client to the user client via the Domain Server.
+void AssignmentClient::sendSignalingMessageToUserClient(const QJsonObject& json) {
+    auto packetList = NLPacketList::create(PacketType::WebRTCSignaling, QByteArray(), true, true);
+    packetList->writeString(QJsonDocument(json).toJson(QJsonDocument::Compact));
+    auto nodeList = DependencyManager::get<NodeList>();
+    auto domainServerAddress = nodeList->getDomainHandler().getSockAddr();
+    nodeList->sendPacketList(std::move(packetList), domainServerAddress);
+}
+
+#endif

assignment-client/src/AssignmentClient.h

@@ -17,6 +17,8 @@
 #include <QtCore/QPointer>
 #include <QtCore/QSharedPointer>
 
+#include <shared/WebRTC.h>
+
 #include "ThreadedAssignment.h"
 
 class QSharedMemory;
@@ -30,19 +32,26 @@ class AssignmentClient : public QObject {
                      bool disableDomainPortAutoDiscovery);
     ~AssignmentClient();
 
+public slots:
+    void aboutToQuit();
+
 private slots:
     void sendAssignmentRequest();
     void assignmentCompleted();
     void handleAuthenticationRequest();
     void sendStatusPacketToACM();
     void stopAssignmentClient();
-
-public slots:
-    void aboutToQuit();
-
-private slots:
     void handleCreateAssignmentPacket(QSharedPointer<ReceivedMessage> message);
     void handleStopNodePacket(QSharedPointer<ReceivedMessage> message);
+#if defined(WEBRTC_DATA_CHANNELS)
+    void handleWebRTCSignalingPacket(QSharedPointer<ReceivedMessage> message);
+    void sendSignalingMessageToUserClient(const QJsonObject& json);
+#endif
+
+signals:
+#if defined(WEBRTC_DATA_CHANNELS)
+    void webrtcSignalingMessageFromUserClient(const QJsonObject& json);
+#endif
 
 private:
     void setUpStatusToMonitor();

cmake/ports/webrtc/CONTROL

@@ -1,3 +1,3 @@
 Source: webrtc
-Version: 20190626
+Version: 20210105
 Description: WebRTC

cmake/ports/webrtc/README.md

@@ -0,0 +1,215 @@
+# WebRTC
+
+WebRTC Information:
+- https://webrtc.org/
+- https://webrtc.googlesource.com/src
+- https://webrtc.googlesource.com/src/+/refs/heads/master/docs/native-code/index.md
+  - https://webrtc.googlesource.com/src/+/refs/heads/master/docs/native-code/development/prerequisite-sw/index.md
+  - https://webrtc.googlesource.com/src/+/refs/heads/master/docs/native-code/development/index.md
+- https://www.chromium.org/developers/calendar
+- https://github.com/microsoft/winrtc
+- https://docs.microsoft.com/en-us/winrtc/getting-started
+- https://groups.google.com/g/discuss-webrtc \
+  See "PSA" posts for release information.
+- https://bugs.chromium.org/p/webrtc/issues/list
+- https://stackoverflow.com/questions/27809193/webrtc-not-building-for-windows
+- https://github.com/aisouard/libwebrtc/issues/57
+
+## Windows - M84
+
+WebRTC's M84 release is currently used because it corresponded to Microsoft's latest WinRTC release at the time of development,
+and WinRTC is a source of potentially useful patches.
+
+The following notes document how the M84-based Windows VCPKG was created, using Visual Studio 2019.
+
+### Set Up depot_tools
+
+Install Google's depot_tools.
+- Download depot_tools.zip.
+- Extract somewhere.
+- Add the extracted directory to the start of the system `PATH` environment variable.
+
+Configure depot_tools.
+- Set an environment variable `DEPOT_TOOLS_WIN_TOOLCHAIN=0`
+- Set an environment variable `GYP_MSVS_VERSION=2019`
+
+Initialize depot_tools.
+- VS2019 developer command prompt in the directory where the source tree will be created.  
+- `gclient`
+
+### Get the Code
+
+Fetch the code into a *\src* subdirectory. This may take some time!
+- `fetch --nohooks webrtc`
+
+Switch to the M84 branch.
+- `cd src`
+- `git checkout branch-heads/4147`
+
+Fetch all the subrepositories.  
+- `gclient sync -D -r branch-heads/4147`
+
+### Patch the Code
+
+#### Modify compiler switches
+- Edit *build\config\win\BUILD.gn*:
+  - Change all `/MT` to `/MD`, and `/MTd` to `/MDd`.
+  - Change all `cflags = [ "/MDd" ]` to `[ "/MDd", "-D_ITERATOR_DEBUG_LEVEL=2", "-D_HAS_ITERATOR_DEBUGGING=1" ]`.
+- Edit *build\config\compiler\BUILD.gn*:\
+  Change:
+  ```
+    if (is_win) {
+      if (is_clang) {
+        cflags = [ "/Z7" ]  # Debug information in the .obj files.
+      } else {
+        cflags = [ "/Zi" ]  # Produce PDB file, no edit and continue.
+      }
+  ```
+  to:
+  ```
+    if (is_win) {
+      if (is_clang) {
+        cflags = [ "/Z7", "/std:c++17", "/Zc:__cplusplus" ]  # Debug information in the .obj files.
+      } else {
+        cflags = [ "/Zi", "/std:c++17", "/Zc:__cplusplus" ]  # Produce PDB file, no edit and continue.
+      }
+  ```
+
+#### H265 Codec Fixes  
+https://bugs.webrtc.org/9213#c13
+- Edit the following files:
+  - *modules\video_coding\codecs\h264\h264_color_space.h*
+  - *modules\video_coding\codecs\h264\h264_decoder_impl.h*
+  - *modules\video_coding\codecs\h264\h264_encoder_impl.h*
+  In each, comment out the following lines:
+  ```
+  #if defined(WEBRTC_WIN) && !defined(__clang__)
+  #error "See: bugs.webrtc.org/9213#c13."
+  #endif
+  ```
+- Edit *third_party\ffmpeg\libavcodec\fft_template.c*:\
+  Comment out all of `ff_fft_init` except the fail clause at the end.
+- Edit *third_party\ffmpeg\libavcodec\pcm.c*:\
+  Comment out last line, containing `PCM Archimedes VIDC`.
+- Edit *third_party\ffmpeg\libavutil\x86\imgutils_init.c*:\
+  Add the following method to the end of the file:
+  ```
+  void avpriv_emms_asm(void) {}  // Fix missing symbol in FFMPEG.
+  ```
+
+#### Exclude BoringSSL
+A separate OpenSSL VCPKG is used for building Vircadia.
+The following patches are needed even though SSL is excluded in the `gn gen` build commands.
+- Rename *third_party\boringssl* to *third_party\boringssl-NO*.
+- Edit *third_party\libsrtp\BUILD.gn:\
+  Change:
+  ```
+  public_deps = [
+    "//third_party/boringssl:boringssl",
+  ]
+  ```
+  To:
+  ```
+  public_deps = [
+    # "//third_party/boringssl:boringssl",
+  ]
+  ```
+
+- Edit *third_party\usrsctp\BUILD.gn*:\
+  Change:
+  ```
+  deps = [ "//third_party/boringssl" ]
+  ```
+  To:
+  ```
+  deps = [
+    # "//third_party/boringssl"
+  ]
+  ```
+- Edit *base\BUILD.gn*:\
+  In the code under:
+  ```
+  # Use the base implementation of hash functions when building for
+  # NaCl. Otherwise, use boringssl.
+  ```
+  Change:
+  ```
+  if (is_nacl) {
+  ```
+  To:
+  ```
+  # if (is_nacl) {
+  if (true) {
+  ```
+- Edit *rtc_base\BUILD.gn*:\
+  Change:  
+  ```
+  if (rtc_build_ssl) {
+    deps += [ "//third_party/boringssl" ]
+  } else {
+  ```
+  To:  
+  ```
+  if (rtc_build_ssl) {
+    # deps += [ "//third_party/boringssl" ]
+  } else {
+  ```
+
+### Set Up OpenSSL
+
+Do one of the following to provide OpenSSL for building against:
+a. If you have built Vircadia, find the **HIFI_VCPKG_BASE** subdirectory used in your build and make note of the path to and 
+including the *installed\x64-windows\include* directory (which includes an *openssl* directory).
+a. Follow https://github.com/vircadia/vcpkg to install *vcpkg* and then *openssl*. Make note of the path to and including the 
+*packages\openssl-windows_x64-windows\include* directory (which includes an *openssl* directory).
+
+Copy the *\<path\>\openssl* directory to the following locations (i.e., add as *openssl* subdirectories):
+- *third_party\libsrtp\crypto\include*
+- *third_party\usrsctp\usrsctplib\usrsctplib*
+
+Also use the path in the `gn gen` commands, below, making sure to escape `\`s as `\\`s.
+
+### Build and Package
+
+Use a VS2019 developer command prompt in the *src* directory.
+
+Create a release build of the WebRTC library:
+- `gn gen --ide=vs2019 out\Release --filters=//:webrtc "--args=is_debug=false is_clang=false use_custom_libcxx=false libcxx_is_shared=true symbol_level=2 use_lld=false rtc_include_tests=false rtc_build_tools=false rtc_build_examples=false proprietary_codecs=true rtc_use_h264=true enable_libaom=false rtc_enable_protobuf=false rtc_build_ssl=false rtc_ssl_root=\"<path>\""`
+- `ninja -C out\Release`
+
+Create a debug build of the WebRTC library:
+- `gn gen --ide=vs2019 out\Debug --filters=//:webrtc "--args=is_debug=true is_clang=false use_custom_libcxx=false libcxx_is_shared=true enable_iterator_debugging=true use_lld=false rtc_include_tests=false rtc_build_tools=false rtc_build_examples=false proprietary_codecs=true rtc_use_h264=true enable_libaom=false rtc_enable_protobuf=false rtc_build_ssl=false rtc_ssl_root=\"<path>\""`
+- `ninja -C out\Debug`
+
+Create VCPKG file:
+- Assemble files in VCPKG directory structure: Use the *copy-VCPKG-files-win.cmd* batch file per instructions in it.  
+`cd ..`\
+`copy-VCPKG-files-win`
+- Zip up the VCPKG *webrtc* directory created by the batch file.  
+`cd vcpkg`\
+`7z a -tzip webrtc-m84-yyyymmdd-windows.zip webrtc`
+- Calculate the SHA512 of the zip file. E.g., using a Windows PowerShell command window:\
+  `Get-FileHash <filename> -Algorithm SHA512 | Format-List`
+- Convert the SHA512 to lower case. E.g., using Microsoft Word, select the SHA512 text and use Shift-F3 to change the case.
+- Host the zip file on the Web.
+- Update *CONTROL* and *portfile.cmake* with version number (revision date), zip file Web URL, and SHA512.
+
+### Tidying up
+
+Disable the depot_tools:
+- Rename the *depot_tools* directory so that these tools don't interfere with the rest of your development environment for 
+  other work.
+
+
+## Linux - M81
+
+The original, High Fidelity-provided WebRTC VCPKG library is used for AEC (audio echo cancellation) only.
+
+**TODO:** Update to M84 and include WebRTC components per Windows WebRTC.
+
+
+## MacOS - M78
+
+The original, High Fidelity-provided WebRTC VCPKG library is used for AEC (audio echo cancellation) only.
+
+**TODO:** Update to M84 and include WebRTC components per Windows WebRTC.