Version 0.1.0

.gitmodules

@@ -1,9 +1,6 @@
 [submodule "moonlight-common-c/moonlight-common-c"]
 	path = moonlight-common-c/moonlight-common-c
 	url = https://github.com/moonlight-stream/moonlight-common-c.git
-[submodule "opus/opus"]
-	path = opus/opus
-	url = https://github.com/xiph/opus.git
 [submodule "qmdnsengine/qmdnsengine"]
 	path = qmdnsengine/qmdnsengine
 	url = https://github.com/cgutman/qmdnsengine.git

.travis.yml

@@ -15,12 +15,18 @@ matrix:
         apt:
           sources:
             - sourceline: 'ppa:beineri/opt-qt596-xenial'
+            - sourceline: 'ppa:jonathonf/ffmpeg-4'
           packages:
             - qt59base
             - qt59quickcontrols2
             - qt59svg
             - libesd0-dev
             - libgl-dev
+            - libopus-dev
+            - libavcodec-dev
+            - libavutil-dev
+            - libva-dev
+            - libvdpau-dev
 
 install:
     - '[ "$TRAVIS_OS_NAME" != osx ] || brew install qt5'

README.md

@@ -4,7 +4,7 @@
 
 Moonlight also has mobile versions for [Android](https://github.com/moonlight-stream/moonlight-android) and  [iOS](https://github.com/moonlight-stream/moonlight-ios).
 
-This client is currently still in development (pre-alpha), but the streaming performance is already much better than the older Moonlight Chrome client on many machines. Test releases may be available for your OS on the [Releases page](https://github.com/moonlight-stream/moonlight-qt/releases).
+See the [releases page](https://github.com/moonlight-stream/moonlight-qt/releases) for download links for Windows and Mac. For Linux, we're on [Flathub](https://flathub.org/apps/details/com.moonlight_stream.Moonlight).
 
 You can follow development on our [Discord server](https://discord.gg/6ERtzFY).
 
@@ -13,7 +13,7 @@ You can follow development on our [Discord server](https://discord.gg/6ERtzFY).
 
 ## Features
  - Hardware accelerated video decoding on Windows, Mac, and Linux
- - Supports streaming at up to 90 FPS on high refresh rate monitors
+ - Supports streaming at up to 120 FPS on high refresh rate monitors
  - Supports streaming at 720p, 1080p, 1440p, or 4K
  - 5.1 surround sound audio
  - HEVC support for better image quality at reduced bandwidth

app/Info.plist

@@ -23,7 +23,7 @@
 	<key>NSSupportsAutomaticGraphicsSwitching</key>
 	<true/>
 	<key>CFBundleVersion</key>
-	<string>0.0.4</string>
+	<string>0.1.0</string>
 	<key>CFBundleDisplayName</key>
 	<string>Moonlight</string>
 </dict>

app/app.pro

@@ -43,7 +43,7 @@ macx {
 
 unix:!macx {
     CONFIG += link_pkgconfig
-    PKGCONFIG += openssl sdl2
+    PKGCONFIG += openssl sdl2 opus
     LIBS += -ldl
 
     packagesExist(libavcodec) {
@@ -53,14 +53,18 @@ unix:!macx {
         packagesExist(libva) {
             CONFIG += libva
         }
+
+        packagesExist(vdpau) {
+            CONFIG += libvdpau
+        }
     }
 }
 win32 {
-    LIBS += -llibssl -llibcrypto -lSDL2 -lavcodec -lavutil
+    LIBS += -llibssl -llibcrypto -lSDL2 -lavcodec -lavutil -lopus
     CONFIG += ffmpeg
 }
 macx {
-    LIBS += -lssl -lcrypto -lavcodec.58 -lavutil.56 -framework SDL2
+    LIBS += -lssl -lcrypto -lavcodec.58 -lavutil.56 -lopus -framework SDL2
     LIBS += -lobjc -framework VideoToolbox -framework AVFoundation -framework CoreVideo -framework CoreGraphics -framework CoreMedia -framework AppKit
     CONFIG += ffmpeg
 }
@@ -77,7 +81,8 @@ SOURCES += \
     streaming/session.cpp \
     streaming/audio.cpp \
     gui/computermodel.cpp \
-    gui/appmodel.cpp
+    gui/appmodel.cpp \
+    streaming/streamutils.cpp
 
 HEADERS += \
     utils.h \
@@ -91,7 +96,8 @@ HEADERS += \
     streaming/session.hpp \
     gui/computermodel.h \
     gui/appmodel.h \
-    streaming/video/decoder.h
+    streaming/video/decoder.h \
+    streaming/streamutils.h
 
 # Platform-specific renderers and decoders
 ffmpeg {
@@ -112,6 +118,13 @@ libva {
     SOURCES += streaming/video/ffmpeg-renderers/vaapi.cpp
     HEADERS += streaming/video/ffmpeg-renderers/vaapi.h
 }
+libvdpau {
+    message(VDPAU renderer selected)
+
+    DEFINES += HAVE_LIBVDPAU
+    SOURCES += streaming/video/ffmpeg-renderers/vdpau.cpp
+    HEADERS += streaming/video/ffmpeg-renderers/vdpau.h
+}
 config_SLVideo {
     message(SLVideo decoder/renderer selected)
 
@@ -150,13 +163,6 @@ else:unix: LIBS += -L$$OUT_PWD/../moonlight-common-c/ -lmoonlight-common-c
 INCLUDEPATH += $$PWD/../moonlight-common-c/moonlight-common-c/src
 DEPENDPATH += $$PWD/../moonlight-common-c/moonlight-common-c/src
 
-win32:CONFIG(release, debug|release): LIBS += -L$$OUT_PWD/../opus/release/ -lopus
-else:win32:CONFIG(debug, debug|release): LIBS += -L$$OUT_PWD/../opus/debug/ -lopus
-else:unix: LIBS += -L$$OUT_PWD/../opus/ -lopus
-
-INCLUDEPATH += $$PWD/../opus/opus/include
-DEPENDPATH += $$PWD/../opus/opus/include
-
 win32:CONFIG(release, debug|release): LIBS += -L$$OUT_PWD/../qmdnsengine/release/ -lqmdnsengine
 else:win32:CONFIG(debug, debug|release): LIBS += -L$$OUT_PWD/../qmdnsengine/debug/ -lqmdnsengine
 else:unix: LIBS += -L$$OUT_PWD/../qmdnsengine/ -lqmdnsengine
@@ -196,4 +202,4 @@ macx {
     QMAKE_BUNDLE_DATA += APP_BUNDLE_RESOURCES
 }
 
-VERSION = 0.0.4
+VERSION = 0.1.0

app/backend/computermanager.cpp

@@ -51,6 +51,8 @@ NvComputer::NvComputer(QSettings& settings)
     this->appVersion = nullptr;
     this->maxLumaPixelsHEVC = 0;
     this->serverCodecModeSupport = 0;
+    this->pendingQuit = false;
+    this->gpuModel = nullptr;
 }
 
 void
@@ -133,8 +135,10 @@ NvComputer::NvComputer(QString address, QString serverInfo)
     this->currentGameId = NvHTTP::getCurrentGame(serverInfo);
     this->appVersion = NvHTTP::getXmlString(serverInfo, "appversion");
     this->gfeVersion = NvHTTP::getXmlString(serverInfo, "GfeVersion");
+    this->gpuModel = NvHTTP::getXmlString(serverInfo, "gputype");
     this->activeAddress = address;
     this->state = NvComputer::CS_ONLINE;
+    this->pendingQuit = false;
 }
 
 bool NvComputer::wake()
@@ -264,6 +268,7 @@ bool NvComputer::update(NvComputer& that)
     ASSIGN_IF_CHANGED(gfeVersion);
     ASSIGN_IF_CHANGED(appVersion);
     ASSIGN_IF_CHANGED(maxLumaPixelsHEVC);
+    ASSIGN_IF_CHANGED(gpuModel);
     ASSIGN_IF_CHANGED_AND_NONEMPTY(appList);
     ASSIGN_IF_CHANGED_AND_NONEMPTY(displayModes);
     return changed;
@@ -446,6 +451,15 @@ void ComputerManager::pairHost(NvComputer* computer, QString pin)
     QThreadPool::globalInstance()->start(pairing);
 }
 
+void ComputerManager::quitRunningApp(NvComputer* computer)
+{
+    QWriteLocker lock(&computer->lock);
+    computer->pendingQuit = true;
+
+    PendingQuitTask* quit = new PendingQuitTask(this, computer);
+    QThreadPool::globalInstance()->start(quit);
+}
+
 void ComputerManager::stopPollingAsync()
 {
     QWriteLocker lock(&m_Lock);
@@ -497,6 +511,11 @@ ComputerManager::handleComputerStateChanged(NvComputer* computer)
 {
     emit computerStateChanged(computer);
 
+    if (computer->pendingQuit && computer->currentGameId == 0) {
+        computer->pendingQuit = false;
+        emit quitAppCompleted(QVariant());
+    }
+
     // Save updated hosts to QSettings
     saveHosts();
 }

app/backend/computermanager.h

@@ -16,10 +16,14 @@
 class NvComputer
 {
     friend class PcMonitorThread;
+    friend class ComputerManager;
+    friend class PendingQuitTask;
 
 private:
     void sortAppList();
 
+    bool pendingQuit;
+
 public:
     explicit NvComputer(QString address, QString serverInfo);
 
@@ -61,6 +65,7 @@ public:
     QVector<NvDisplayMode> displayModes;
     int maxLumaPixelsHEVC;
     int serverCodecModeSupport;
+    QString gpuModel;
 
     // Persisted traits
     QString localAddress;
@@ -254,6 +259,8 @@ public:
 
     void pairHost(NvComputer* computer, QString pin);
 
+    void quitRunningApp(NvComputer* computer);
+
     QVector<NvComputer*> getComputers();
 
     // computer is deleted inside this call
@@ -266,6 +273,8 @@ signals:
 
     void computerAddCompleted(QVariant success);
 
+    void quitAppCompleted(QVariant error);
+
 private slots:
     void handleComputerStateChanged(NvComputer* computer);
 
@@ -333,6 +342,49 @@ private:
     QString m_Pin;
 };
 
+class PendingQuitTask : public QObject, public QRunnable
+{
+    Q_OBJECT
+
+public:
+    PendingQuitTask(ComputerManager* computerManager, NvComputer* computer)
+        : m_Computer(computer)
+    {
+        connect(this, &PendingQuitTask::quitAppFailed,
+                computerManager, &ComputerManager::quitAppCompleted);
+    }
+
+signals:
+    void quitAppFailed(QString error);
+
+private:
+    void run()
+    {
+        NvHTTP http(m_Computer->activeAddress);
+
+        try {
+            if (m_Computer->currentGameId != 0) {
+                http.quitApp();
+            }
+        } catch (const GfeHttpResponseException& e) {
+            {
+                QWriteLocker lock(&m_Computer->lock);
+                m_Computer->pendingQuit = false;
+            }
+            if (e.getStatusCode() == 599) {
+                // 599 is a special code we make a custom message for
+                emit quitAppFailed("The running game wasn't started by this PC. "
+                                   "You must quit the game on the host PC manually or use the device that originally started the game.");
+            }
+            else {
+                emit quitAppFailed(e.toQString());
+            }
+        }
+    }
+
+    NvComputer* m_Computer;
+};
+
 class PendingAddTask : public QObject, public QRunnable
 {
     Q_OBJECT

app/backend/nvhttp.cpp

@@ -67,6 +67,7 @@ NvHTTP::getServerInfo()
         serverInfo = openConnectionToString(m_BaseUrlHttps,
                                             "serverinfo",
                                             nullptr,
+                                            true,
                                             true);
         // Throws if the request failed
         verifyResponseStatus(serverInfo);
@@ -79,6 +80,7 @@ NvHTTP::getServerInfo()
             serverInfo = openConnectionToString(m_BaseUrlHttp,
                                                 "serverinfo",
                                                 nullptr,
+                                                true,
                                                 true);
             verifyResponseStatus(serverInfo);
         }
@@ -225,6 +227,7 @@ NvHTTP::getAppList()
     QString appxml = openConnectionToString(m_BaseUrlHttps,
                                             "applist",
                                             nullptr,
+                                            true,
                                             true);
     verifyResponseStatus(appxml);
 
@@ -338,9 +341,10 @@ QString
 NvHTTP::openConnectionToString(QUrl baseUrl,
                                QString command,
                                QString arguments,
-                               bool enableTimeout)
+                               bool enableTimeout,
+                               bool suppressLogging)
 {
-    QNetworkReply* reply = openConnection(baseUrl, command, arguments, enableTimeout);
+    QNetworkReply* reply = openConnection(baseUrl, command, arguments, enableTimeout, suppressLogging);
     QString ret;
 
     QTextStream stream(reply);
@@ -355,7 +359,8 @@ QNetworkReply*
 NvHTTP::openConnection(QUrl baseUrl,
                        QString command,
                        QString arguments,
-                       bool enableTimeout)
+                       bool enableTimeout,
+                       bool suppressLogging)
 {
     // Build a URL for the request
     QUrl url(baseUrl);
@@ -381,13 +386,17 @@ NvHTTP::openConnection(QUrl baseUrl,
     {
         QTimer::singleShot(REQUEST_TIMEOUT_MS, &loop, SLOT(quit()));
     }
-    qInfo() << "Executing request:" << url.toString();
+    if (!suppressLogging) {
+        qInfo() << "Executing request:" << url.toString();
+    }
     loop.exec(QEventLoop::ExcludeUserInputEvents);
 
     // Abort the request if it timed out
     if (!reply->isFinished())
     {
-        qWarning() << "Aborting timed out request for" << url.toString();
+        if (!suppressLogging) {
+            qWarning() << "Aborting timed out request for" << url.toString();
+        }
         reply->abort();
     }
 
@@ -398,7 +407,9 @@ NvHTTP::openConnection(QUrl baseUrl,
     // Handle error
     if (reply->error() != QNetworkReply::NoError)
     {
-        qWarning() << command << " request failed with error " << reply->error();
+        if (!suppressLogging) {
+            qWarning() << command << " request failed with error " << reply->error();
+        }
         GfeHttpResponseException exception(reply->error(), reply->errorString());
         delete reply;
         throw exception;

app/backend/nvhttp.h

@@ -107,7 +107,8 @@ public:
     openConnectionToString(QUrl baseUrl,
                            QString command,
                            QString arguments,
-                           bool enableTimeout);
+                           bool enableTimeout,
+                           bool suppressLogging = false);
 
     static
     QVector<int>
@@ -143,7 +144,8 @@ private:
     openConnection(QUrl baseUrl,
                    QString command,
                    QString arguments,
-                   bool enableTimeout);
+                   bool enableTimeout,
+                   bool suppressLogging = false);
 
     QString m_Address;
     QNetworkAccessManager m_Nam;

app/deploy/linux/com.moonlight_stream.Moonlight.appdata.xml

@@ -1,5 +1,5 @@
- <?xml version="1.0" encoding="UTF-8"?>
- <component type="desktop">
+<?xml version="1.0" encoding="UTF-8"?>
+<component type="desktop">
 	<id>com.moonlight_stream.Moonlight</id>
 	<metadata_license>CC0-1.0</metadata_license>
 	<project_license>GPL-3.0+</project_license>
@@ -7,8 +7,16 @@
 	<summary>Stream games from your NVIDIA GameStream-enabled PC</summary>
 
 	<description>
-		<p>Moonlight can stream games or other applications from a PC with an NVIDIA GeForce GTX 600-series or higher GPU and GeForce Experience installed.</p>
-		<p>GameStream utilizes the NVENC hardware encoder present on NVIDIA GPUs for low-latency streaming at up 4K 60 FPS. Moonlight can utilize VAAPI hardware acceleration for high-performance H.264 and HEVC video decoding with low CPU usage.</p>
+		<p>Moonlight can stream games and other applications from a PC with an NVIDIA GeForce GTX 600-series or higher GPU and GeForce Experience installed.</p>
+		<p>Features include:</p>
+		<ul>
+			<li>Streaming at up to 4K resolution</li>
+			<li>Support for up to 120 FPS streaming on high refresh rate monitors</li>
+			<li>Hardware accelerated video decoding with VAAPI and VDPAU support</li>
+			<li>5.1 surround sound audio support</li>
+			<li>HEVC support for better video compression efficiency</li>
+			<li>Gamepad support with SDL mapping compatibility</li>
+		</ul>
 	</description>
 
 	<url type="homepage">https://moonlight-stream.com</url>
@@ -25,7 +33,37 @@
 	</screenshots>
 
 	<releases>
-		<release version="0.0.4" date="2018-07-29">
+		<release version="0.1.0" date="2018-08-05">
+			<description>
+				<p>New features:</p>
+				<ul>
+					<li>Allow streaming at the client's native resolution</li>
+					<li>VDPAU hardware decoding support for NVIDIA GPUs</li>
+					<li>Updated UI to replace status text with icons</li>
+					<li>Stream window is now resizable</li>
+					<li>Display warnings during game launch for configuration issues</li>
+				</ul>
+				<p>Bugfixes:</p>
+				<ul>
+					<li>Gamepad input now works even if the app doesn't have focus</li>
+					<li>Fixed streaming black screen if the host supported HEVC but the client didn't</li>
+					<li>Fixed broken video scaling causing extra stretching and off-center video</li>
+					<li>Fixed gamepad input issue causing some Y axis inputs to be reversed</li>
+				</ul>
+			</description>
+		</release>
+		<release version="0.0.5" date="2018-08-01">
+			<description>
+				<p>Changes from 0.0.4:</p>
+				<ul>
+					<li>Fixed a bug causing increased frame drops and input lag</li>
+					<li>Added support for quitting running games</li>
+					<li>Added support for waking sleeping PCs with Wake-on-LAN</li>
+					<li>Increased the FPS limit to 120 FPS</li>
+				</ul>
+			</description>
+		</release>
+		<release version="0.0.4a" date="2018-07-30">
 			<description>
 				<p>Initial pre-alpha release for Linux. Please file bugs!</p>
 			</description>

app/gui/AppView.qml

@@ -1,4 +1,5 @@
 import QtQuick 2.9
+import QtQuick.Dialogs 1.3
 import QtQuick.Controls 2.2
 
 import AppModel 1.0
@@ -50,12 +51,6 @@ GridView {
     delegate: Item {
         width: 200; height: 300;
 
-        Component.onCompleted: {
-            if (model.running) {
-                appNameText.text = "<font color=\"green\">Running</font><font color=\"white\"> - </font>" + appNameText.text
-            }
-        }
-
         Image {
             id: appIcon
             anchors.horizontalCenter: parent.horizontalCenter;
@@ -66,11 +61,21 @@ GridView {
             }
         }
 
+        Image {
+            id: runningIcon
+            anchors.centerIn: appIcon
+            visible: model.running
+            source: "qrc:/res/baseline-play_circle_filled_white-48px.svg"
+            sourceSize {
+                width: 75
+                height: 75
+            }
+        }
+
         Text {
             id: appNameText
             text: model.name
             color: "white"
-
             width: parent.width
             height: 125
             anchors.top: appIcon.bottom
@@ -80,16 +85,86 @@ GridView {
             elide: Text.ElideRight
         }
 
+        function launchOrResumeSelectedApp()
+        {
+            var runningIndex = appModel.getRunningAppIndex()
+            if (runningIndex >= 0 && runningIndex !== index) {
+                quitAppDialog.appName = appModel.getRunningAppName()
+                quitAppDialog.segueToStream = true
+                quitAppDialog.open()
+                return
+            }
+
+            var component = Qt.createComponent("StreamSegue.qml")
+            var segue = component.createObject(stackView)
+            segue.appName = model.name
+            segue.session = appModel.createSessionForApp(index)
+            stackView.push(segue)
+        }
+
+        MessageDialog {
+            id: quitAppDialog
+            modality:Qt.WindowModal
+            property string appName : "";
+            property bool segueToStream : false
+            text:"Are you sure you want to quit " + appName +"? Any unsaved progress will be lost."
+            standardButtons: StandardButton.Yes | StandardButton.No
+            onYes: {
+                var component = Qt.createComponent("QuitSegue.qml")
+                var segue = component.createObject(stackView)
+                segue.appName = appName
+                if (segueToStream) {
+                    // Store the session and app name if we're going to stream after
+                    // successfully quitting the old app.
+                    segue.nextAppName = model.name
+                    segue.nextSession = appModel.createSessionForApp(index)
+                }
+                else {
+                    segue.nextAppName = null
+                    segue.nextSession = null
+                }
+
+                stackView.push(segue)
+
+                // Trigger the quit after pushing the quit segue on screen
+                appModel.quitRunningApp()
+            }
+        }
+
         MouseArea {
             anchors.fill: parent
+            acceptedButtons: Qt.LeftButton | Qt.RightButton
             onClicked: {
-                // TODO: Check if a different game is running
+                if (mouse.button === Qt.LeftButton) {
+                    // Nothing is running or this app is running
+                    launchOrResumeSelectedApp()
+                }
+                else {
+                    // Right click
+                    appContextMenu.open()
+                }
+            }
+        }
 
-                var component = Qt.createComponent("StreamSegue.qml")
-                var segue = component.createObject(stackView)
-                segue.appname = model.name
-                segue.session = appModel.createSessionForApp(index)
-                stackView.push(segue)
+        Menu {
+            id: appContextMenu
+            MenuItem {
+                text: model.running ? "Resume Game" : "Launch Game"
+                onTriggered: {
+                    appContextMenu.close()
+                    launchOrResumeSelectedApp()
+                }
+                height: visible ? implicitHeight : 0
+            }
+            MenuItem {
+                text: "Quit Game"
+                onTriggered: {
+                    quitAppDialog.appName = appModel.getRunningAppName()
+                    quitAppDialog.segueToStream = false
+                    quitAppDialog.open()
+                }
+                visible: model.running
+                height: visible ? implicitHeight : 0
             }
         }
     }

app/gui/PcView.qml

@@ -81,14 +81,14 @@ GridView {
         return model
     }
 
-    model: createModel()
+    model: computerModel
 
     delegate: Item {
         width: 300; height: 300;
 
         Image {
             id: pcIcon
-            anchors.horizontalCenter: parent.horizontalCenter;
+            anchors.horizontalCenter: parent.horizontalCenter
             source: {
                 model.addPc ? "qrc:/res/ic_add_to_queue_white_48px.svg" : "qrc:/res/ic_tv_white_48px.svg"
             }
@@ -98,6 +98,30 @@ GridView {
             }
         }
 
+        Image {
+            // TODO: Tooltip
+            id: stateIcon
+            anchors.horizontalCenter: pcIcon.horizontalCenter
+            anchors.verticalCenter: pcIcon.verticalCenter
+            anchors.verticalCenterOffset: -10
+            visible: !model.addPc && !model.statusUnknown && (!model.online || !model.paired)
+            source: !model.online ? "qrc:/res/baseline-warning-24px.svg" : "qrc:/res/baseline-lock-24px.svg"
+            sourceSize {
+                width: 75
+                height: 75
+            }
+        }
+
+        BusyIndicator {
+            id: statusUnknownSpinner
+            anchors.horizontalCenter: pcIcon.horizontalCenter
+            anchors.verticalCenter: pcIcon.verticalCenter
+            anchors.verticalCenterOffset: -10
+            width: 75
+            height: 75
+            visible: !model.addPc && model.statusUnknown
+        }
+
         Text {
             id: pcNameText
             text: model.name
@@ -110,43 +134,13 @@ GridView {
             wrapMode: Text.Wrap
         }
 
-        Text {
-            function getStatusText(model)
-            {
-                if (model.online) {
-                    var text = "<font color=\"green\">Online</font>"
-                    text += "<font color=\"white\"> - </font>"
-                    if (model.paired) {
-                        text += "<font color=\"skyblue\">Paired</font>"
-                    }
-                    else if (model.busy) {
-                        text += "<font color=\"red\">Busy</font>"
-                    }
-                    else {
-                        text += "<font color=\"red\">Not Paired</font>"
-                    }
-                    return text
-                }
-                else {
-                    return "<font color=\"red\">Offline</font>";
-                }
-            }
-
-            id: pcPairedText
-            text: getStatusText(model)
-            visible: !model.addPc
-
-            width: parent.width
-            anchors.top: pcNameText.bottom
-            font.pointSize: 24
-            horizontalAlignment: Text.AlignHCenter
-        }
-
         Menu {
             id: pcContextMenu
             MenuItem {
                 text: "Wake PC"
                 onTriggered: computerModel.wakeComputer(index)
+                visible: !model.addPc && !model.online && model.wakeable
+                height: visible ? implicitHeight : 0
             }
             MenuItem {
                 text: "Delete PC"
@@ -247,17 +241,14 @@ GridView {
         // don't allow edits to the rest of the window while open
         modality:Qt.WindowModal
         property int pcIndex : -1;
-        text:"Are you sure you want to unpair from this PC?"
-        standardButtons: StandardButton.Yes |StandardButton.No
+        text:"Are you sure you want to remove this PC?"
+        standardButtons: StandardButton.Yes | StandardButton.No
         onYes: {
             console.log("deleting PC pairing for PC at index: " + pcIndex)
             computerModel.deleteComputer(pcIndex);
-            // hack to remove the child from the gridview
-            model = createModel()
         }
     }
 
-
     Dialog {
         id: addPcDialog
         property string label: "Enter the IP address of your GameStream PC"

app/gui/QuitSegue.qml

@@ -0,0 +1,86 @@
+import QtQuick 2.0
+import QtQuick.Controls 2.2
+import QtQuick.Dialogs 1.3
+
+import ComputerManager 1.0
+import Session 1.0
+
+Item {
+    property string appName
+    property Session nextSession : null
+    property string nextAppName : ""
+
+    property string stageText : "Quitting " + appName + "..."
+
+    anchors.fill: parent
+
+    // The StackView will trigger a visibility change when
+    // we're pushed onto it, causing our onVisibleChanged
+    // routine to run, but only if we start as invisible
+    visible: false
+
+    function quitAppCompleted(error)
+    {
+        // Display a failed dialog if we got an error
+        if (error !== undefined) {
+            errorDialog.text = error
+            errorDialog.open()
+        }
+
+        // Exit this view
+        stackView.pop()
+
+        // If we're supposed to launch another game after this, do so now
+        if (error === undefined && nextSession !== null) {
+            var component = Qt.createComponent("StreamSegue.qml")
+            var segue = component.createObject(stackView)
+            segue.appName = nextAppName
+            segue.session = nextSession
+            stackView.push(segue)
+        }
+    }
+
+    onVisibleChanged: {
+        if (visible) {
+            // Connect the quit completion signal
+            ComputerManager.quitAppCompleted.connect(quitAppCompleted)
+
+            // We must be polling or we won't get the quitAppCompleted() callback
+            ComputerManager.startPolling()
+        }
+        else {
+            // Disconnect the signal
+            ComputerManager.quitAppCompleted.disconnect(quitAppCompleted)
+
+            // Stop polling when this view is not on top
+            ComputerManager.stopPollingAsync()
+        }
+    }
+
+    Row {
+        anchors.centerIn: parent
+        spacing: 5
+
+        BusyIndicator {
+            id: stageSpinner
+        }
+
+        Label {
+            id: stageLabel
+            height: stageSpinner.height
+            text: stageText
+            font.pointSize: 20
+            verticalAlignment: Text.AlignVCenter
+
+            wrapMode: Text.Wrap
+            color: "white"
+        }
+    }
+
+    MessageDialog {
+        id: errorDialog
+        modality:Qt.WindowModal
+        icon: StandardIcon.Critical
+        standardButtons: StandardButton.Ok
+    }
+}

app/gui/SettingsView.qml

@@ -43,7 +43,7 @@ ScrollView {
                 Label {
                     width: parent.width
                     id: resFPSdesc
-                    text: qsTr("Setting values too high for your PC may cause lag, stuttering, or errors")
+                    text: qsTr("Setting values too high for your PC may cause lag, stuttering, or errors.")
                     font.pointSize: 9
                     wrapMode: Text.Wrap
                     color: "white"
@@ -55,6 +55,41 @@ ScrollView {
                     ComboBox {
                         // ignore setting the index at first, and actually set it when the component is loaded
                         Component.onCompleted: {
+                            // Add native resolutions for all attached displays
+                            for (var displayIndex = 0;; displayIndex++) {
+                                var screenRect = prefs.getDisplayResolution(displayIndex)
+                                if (screenRect.width === 0) {
+                                    // Exceeded max count of displays
+                                    break
+                                }
+
+                                var indexToAdd = 0
+                                for (var j = 0; j < resolutionComboBox.count; j++) {
+                                    var existing_width = parseInt(resolutionListModel.get(j).video_width);
+                                    var existing_height = parseInt(resolutionListModel.get(j).video_height);
+
+                                    if (screenRect.width === existing_width && screenRect.height === existing_height) {
+                                        // Duplicate entry, skip
+                                        indexToAdd = -1
+                                        break
+                                    }
+                                    else if (screenRect.width * screenRect.height > existing_width * existing_height) {
+                                        // Candidate entrypoint after this entry
+                                        indexToAdd = j + 1
+                                    }
+                                }
+
+                                // Insert this display's resolution if it's not a duplicate
+                                if (indexToAdd >= 0) {
+                                    resolutionListModel.insert(indexToAdd,
+                                                               {
+                                                                   "text": "Native ("+screenRect.width+"x"+screenRect.height+")",
+                                                                   "video_width": ""+screenRect.width,
+                                                                   "video_height": ""+screenRect.height
+                                                               })
+                                }
+                            }
+
                             // load the saved width/height, and iterate through the ComboBox until a match is found
                             // and set it to that index.
                             var saved_width = prefs.width
@@ -63,17 +98,21 @@ ScrollView {
                             for (var i = 0; i < resolutionComboBox.count; i++) {
                                 var el_width = parseInt(resolutionListModel.get(i).video_width);
                                 var el_height = parseInt(resolutionListModel.get(i).video_height);
-                                if (saved_width === el_width && saved_height === el_height) {
+
+                                // Pick the highest value lesser or equal to the saved resolution
+                                if (saved_width * saved_height >= el_width * el_height) {
                                     currentIndex = i
                                 }
                             }
                         }
 
                         id: resolutionComboBox
-                        font.pointSize: 9
+                        width: 200
                         textRole: "text"
                         model: ListModel {
                             id: resolutionListModel
+                            // Other elements may be added at runtime
+                            // based on attached display resolution
                             ListElement {
                                 text: "720p"
                                 video_width: "1280"
@@ -121,14 +160,15 @@ ScrollView {
                             currentIndex = 0
                             for (var i = 0; i < fpsComboBox.count; i++) {
                                 var el_fps = parseInt(fpsListModel.get(i).video_fps);
-                                if (el_fps === saved_fps) {
+
+                                // Pick the highest value lesser or equal to the saved FPS
+                                if (saved_fps >= el_fps) {
                                     currentIndex = i
                                 }
                             }
                         }
 
                         id: fpsComboBox
-                        font.pointSize: 9
                         textRole: "text"
                         model: ListModel {
                             id: fpsListModel
@@ -238,7 +278,6 @@ ScrollView {
 
                     id: audioComboBox
                     width: Math.min(bitrateDesc.implicitWidth, parent.width)
-                    font.pointSize: 9
                     textRole: "text"
                     model: ListModel {
                         id: audioListModel
@@ -362,7 +401,6 @@ ScrollView {
 
                     id: decoderComboBox
                     width: Math.min(bitrateDesc.implicitWidth, parent.width)
-                    font.pointSize: 9
                     textRole: "text"
                     model: ListModel {
                         id: decoderListModel
@@ -409,7 +447,6 @@ ScrollView {
 
                     id: codecComboBox
                     width: Math.min(bitrateDesc.implicitWidth, parent.width)
-                    font.pointSize: 9
                     textRole: "text"
                     model: ListModel {
                         id: codecListModel

app/gui/StreamSegue.qml

@@ -6,8 +6,8 @@ import Session 1.0
 
 Item {
     property Session session
-    property string appname
-    property string stageText : "Starting " + appname + "..."
+    property string appName
+    property string stageText : "Starting " + appName + "..."
 
     anchors.fill: parent
 
@@ -47,7 +47,13 @@ Item {
 
     function displayLaunchWarning(text)
     {
-        // TODO: toast
+        // This toast appears for 3 seconds, just shorter than how long
+        // Session will wait for it to be displayed. This gives it time
+        // to transition to invisible before continuing.
+        var toast = Qt.createQmlObject('import QtQuick.Controls 2.3; ToolTip {}', parent, '')
+        toast.text = text
+        toast.timeout = 3000
+        toast.visible = true
     }
 
     onVisibleChanged: {

app/gui/appmodel.cpp

@@ -19,6 +19,32 @@ void AppModel::initialize(ComputerManager* computerManager, int computerIndex)
     m_CurrentGameId = m_Computer->currentGameId;
 }
 
+int AppModel::getRunningAppIndex()
+{
+    if (m_CurrentGameId != 0) {
+        for (int i = 0; i < m_Apps.count(); i++) {
+            if (m_Apps[i].id == m_CurrentGameId) {
+                return i;
+            }
+        }
+    }
+
+    return -1;
+}
+
+QString AppModel::getRunningAppName()
+{
+    if (m_CurrentGameId != 0) {
+        for (int i = 0; i < m_Apps.count(); i++) {
+            if (m_Apps[i].id == m_CurrentGameId) {
+                return m_Apps[i].name;
+            }
+        }
+    }
+
+    return nullptr;
+}
+
 Session* AppModel::createSessionForApp(int appIndex)
 {
     Q_ASSERT(appIndex < m_Apps.count());
@@ -70,6 +96,11 @@ QHash<int, QByteArray> AppModel::roleNames() const
     return names;
 }
 
+void AppModel::quitRunningApp()
+{
+    m_ComputerManager->quitRunningApp(m_Computer);
+}
+
 void AppModel::handleComputerStateChanged(NvComputer* computer)
 {
     // Ignore updates for computers that aren't ours

app/gui/appmodel.h

@@ -25,6 +25,12 @@ public:
 
     Q_INVOKABLE Session* createSessionForApp(int appIndex);
 
+    Q_INVOKABLE int getRunningAppIndex();
+
+    Q_INVOKABLE QString getRunningAppName();
+
+    Q_INVOKABLE void quitRunningApp();
+
     QVariant data(const QModelIndex &index, int role) const override;
 
     int rowCount(const QModelIndex &parent) const override;

app/gui/computermodel.cpp

@@ -46,8 +46,12 @@ QVariant ComputerModel::data(const QModelIndex& index, int role) const
         return computer->pairState == NvComputer::PS_PAIRED;
     case BusyRole:
         return computer->currentGameId != 0;
+    case WakeableRole:
+        return !computer->macAddress.isEmpty();
     case AddPcRole:
         return false;
+    case StatusUnknownRole:
+        return computer->state == NvComputer::CS_UNKNOWN;
     default:
         return QVariant();
     }
@@ -74,6 +78,8 @@ QHash<int, QByteArray> ComputerModel::roleNames() const
     names[PairedRole] = "paired";
     names[BusyRole] = "busy";
     names[AddPcRole] = "addPc";
+    names[WakeableRole] = "wakeable";
+    names[StatusUnknownRole] = "statusUnknown";
 
     return names;
 }

app/gui/computermodel.h

@@ -13,6 +13,8 @@ class ComputerModel : public QAbstractListModel
         OnlineRole,
         PairedRole,
         BusyRole,
+        WakeableRole,
+        StatusUnknownRole,
         AddPcRole
     };
 

app/qml.qrc

@@ -6,5 +6,6 @@
         <file>gui/SettingsView.qml</file>
         <file>gui/StreamSegue.qml</file>
         <file>gui/GamepadMapper.qml</file>
+        <file>gui/QuitSegue.qml</file>
     </qresource>
 </RCC>

app/res/baseline-lock-24px.svg

@@ -0,0 +1,4 @@
+<svg xmlns="http://www.w3.org/2000/svg" fill="#FFFFFF" width="24" height="24" viewBox="0 0 24 24">
+    <path d="M0 0h24v24H0z" fill="none"/>
+    <path d="M18 8h-1V6c0-2.76-2.24-5-5-5S7 3.24 7 6v2H6c-1.1 0-2 .9-2 2v10c0 1.1.9 2 2 2h12c1.1 0 2-.9 2-2V10c0-1.1-.9-2-2-2zm-6 9c-1.1 0-2-.9-2-2s.9-2 2-2 2 .9 2 2-.9 2-2 2zm3.1-9H8.9V6c0-1.71 1.39-3.1 3.1-3.1 1.71 0 3.1 1.39 3.1 3.1v2z"/>
+</svg>

app/res/baseline-play_circle_filled_white-48px.svg

@@ -0,0 +1,4 @@
+<svg xmlns="http://www.w3.org/2000/svg" width="48" height="48" viewBox="0 0 48 48" fill="#fff">
+    <path d="M0 0h48v48H0z" fill="none"/>
+    <path d="M24 4C12.95 4 4 12.95 4 24s8.95 20 20 20 20-8.95 20-20S35.05 4 24 4zm-4 29V15l12 9-12 9z"/>
+</svg>

app/res/baseline-warning-24px.svg

@@ -0,0 +1,4 @@
+<svg xmlns="http://www.w3.org/2000/svg" fill="#FFFFFF" width="24" height="24" viewBox="0 0 24 24">
+    <path d="M0 0h24v24H0z" fill="none"/>
+    <path d="M1 21h22L12 2 1 21zm12-3h-2v-2h2v2zm0-4h-2v-4h2v4z"/>
+</svg>

app/resources.qrc

@@ -5,6 +5,9 @@
         <file>res/ic_videogame_asset_white_48px.svg</file>
         <file>res/ic_tv_white_48px.svg</file>
         <file>res/ic_add_to_queue_white_48px.svg</file>
+        <file>res/baseline-lock-24px.svg</file>
+        <file>res/baseline-play_circle_filled_white-48px.svg</file>
+        <file>res/baseline-warning-24px.svg</file>
         <file>res/no_app_image.png</file>
         <file>res/settings.svg</file>
         <file>res/arrow_left.svg</file>

app/settings/streamingpreferences.cpp

@@ -82,9 +82,9 @@ int StreamingPreferences::getMaximumStreamingFrameRate()
     for (int i = 0; i < SDL_GetNumVideoDisplays(); i++) {
         SDL_DisplayMode mode;
         if (SDL_GetCurrentDisplayMode(i, &mode) == 0) {
-            // Cap the frame rate at 90 FPS, since I can't seem to get
-            // much more out of GFE even with the game rendering at > 300 FPS.
-            maxFrameRate = qMax(maxFrameRate, qMin(90, mode.refresh_rate));
+            // Cap the frame rate at 120 FPS. Past this, the encoders start
+            // to max out and drop frames.
+            maxFrameRate = qMax(maxFrameRate, qMin(120, mode.refresh_rate));
         }
     }
 
@@ -93,15 +93,50 @@ int StreamingPreferences::getMaximumStreamingFrameRate()
     return maxFrameRate;
 }
 
+QRect StreamingPreferences::getDisplayResolution(int displayIndex)
+{
+    if (SDL_InitSubSystem(SDL_INIT_VIDEO) != 0) {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "SDL_InitSubSystem(SDL_INIT_VIDEO) failed: %s",
+                     SDL_GetError());
+        return QRect();
+    }
+
+    if (displayIndex >= SDL_GetNumVideoDisplays()) {
+        SDL_QuitSubSystem(SDL_INIT_VIDEO);
+        return QRect();
+    }
+
+    SDL_DisplayMode mode;
+    int err = SDL_GetCurrentDisplayMode(displayIndex, &mode);
+    SDL_QuitSubSystem(SDL_INIT_VIDEO);
+
+    if (err == 0) {
+        return QRect(0, 0, mode.w, mode.h);
+    }
+    else {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "SDL_GetCurrentDisplayMode() failed: %s",
+                     SDL_GetError());
+        return QRect();
+    }
+}
+
 int StreamingPreferences::getDefaultBitrate(int width, int height, int fps)
 {
-    if (width * height <= 1280 * 720) {
+    // This table prefers 16:10 resolutions because they are
+    // only slightly more pixels than the 16:9 equivalents, so
+    // we don't want to bump those 16:10 resolutions up to the
+    // next 16:9 slot.
+
+    // This covers 1280x720 and 1280x800 too
+    if (width * height <= 1366 * 768) {
         return static_cast<int>(5000 * (fps / 30.0));
     }
-    else if (width * height <= 1920 * 1080) {
+    else if (width * height <= 1920 * 1200) {
         return static_cast<int>(10000 * (fps / 30.0));
     }
-    else if (width * height <= 2560 * 1440) {
+    else if (width * height <= 2560 * 1600) {
         return static_cast<int>(20000 * (fps / 30.0));
     }
     else /* if (width * height <= 3840 * 2160) */ {

app/settings/streamingpreferences.h

@@ -1,6 +1,7 @@
 #pragma once
 
 #include <QObject>
+#include <QRect>
 
 class StreamingPreferences : public QObject
 {
@@ -18,6 +19,8 @@ public:
 
     Q_INVOKABLE static int getMaximumStreamingFrameRate();
 
+    Q_INVOKABLE QRect getDisplayResolution(int displayIndex);
+
     void reload();
 
     enum AudioConfig

app/streaming/input.cpp

@@ -2,6 +2,8 @@
 #include <SDL.h>
 #include "streaming/session.hpp"
 
+#include <QtGlobal>
+
 #define VK_0 0x30
 #define VK_A 0x41
 
@@ -23,6 +25,9 @@ const int SdlInputHandler::k_ButtonMap[] = {
 SdlInputHandler::SdlInputHandler(bool multiController)
     : m_MultiController(multiController)
 {
+    // Allow gamepad input when the app doesn't have focus
+    SDL_SetHint(SDL_HINT_JOYSTICK_ALLOW_BACKGROUND_EVENTS, "1");
+
     // We need to reinit this each time, since you only get
     // an initial set of gamepad arrival events once per init.
     SDL_assert(!SDL_WasInit(SDL_INIT_GAMECONTROLLER));
@@ -438,13 +443,18 @@ void SdlInputHandler::handleControllerAxisEvent(SDL_ControllerAxisEvent* event)
             state->lsX = event->value;
             break;
         case SDL_CONTROLLER_AXIS_LEFTY:
-            state->lsY = -event->value;
+            // Signed values have one more negative value than
+            // positive value, so inverting the sign on -32768
+            // could actually cause the value to overflow and
+            // wrap around to be negative again. Avoid that by
+            // capping the value at 32767.
+            state->lsY = -qMax(event->value, (short)-32767);
             break;
         case SDL_CONTROLLER_AXIS_RIGHTX:
             state->rsX = event->value;
             break;
         case SDL_CONTROLLER_AXIS_RIGHTY:
-            state->rsY = -event->value;
+            state->rsY = -qMax(event->value, (short)-32767);
             break;
         case SDL_CONTROLLER_AXIS_TRIGGERLEFT:
             state->lt = (unsigned char)(event->value * 255UL / 32767);

app/streaming/session.cpp

@@ -13,10 +13,10 @@
 #include "video/sl.h"
 #endif
 
-#ifdef Q_OS_WIN32
-// Using full-screen desktop allows us to avoid needing to enable V-sync
+#if defined(Q_OS_WIN32) || defined(Q_OS_DARWIN)
+// Using full-screen desktop allows us to avoid needing to enable V-sync on Windows
 // and it also avoids some strange flickering issues on my Win7 test machine
-// with Intel HD 5500 graphics.
+// with Intel HD 5500 graphics. It also allows Spaces to work on macOS.
 #define SDL_OS_FULLSCREEN_FLAG SDL_WINDOW_FULLSCREEN_DESKTOP
 #else
 #define SDL_OS_FULLSCREEN_FLAG SDL_WINDOW_FULLSCREEN
@@ -138,6 +138,10 @@ bool Session::chooseDecoder(StreamingPreferences::VideoDecoderSelection vds,
     }
 #endif
 
+#if !defined(HAVE_FFMPEG) && !defined(HAVE_SLVIDEO)
+#error No video decoding libraries available!
+#endif
+
     // If we reach this, we didn't initialize any decoders successfully
     return false;
 }
@@ -232,6 +236,8 @@ Session::Session(NvComputer* computer, NvApp& app)
       m_DecoderLock(0),
       m_NeedsIdr(false)
 {
+    qDebug() << "Server GPU:" << m_Computer->gpuModel;
+
     LiInitializeVideoCallbacks(&m_VideoCallbacks);
     m_VideoCallbacks.setup = drSetup;
     m_VideoCallbacks.submitDecodeUnit = drSubmitDecodeUnit;
@@ -240,7 +246,11 @@ Session::Session(NvComputer* computer, NvApp& app)
     m_VideoCallbacks.capabilities |= CAPABILITY_DIRECT_SUBMIT;
 
     // Slice up to 4 times for parallel decode, once slice per core
-    m_VideoCallbacks.capabilities |= CAPABILITY_SLICES_PER_FRAME(qMin(MAX_SLICES, SDL_GetCPUCount()));
+    int slices = qMin(MAX_SLICES, SDL_GetCPUCount());
+    m_VideoCallbacks.capabilities |= CAPABILITY_SLICES_PER_FRAME(slices);
+    SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
+                "Encoder configured for %d slices per frame",
+                slices);
 
     LiInitializeStreamConfiguration(&m_StreamConfig);
     m_StreamConfig.width = m_Preferences.width;
@@ -248,6 +258,11 @@ Session::Session(NvComputer* computer, NvApp& app)
     m_StreamConfig.fps = m_Preferences.fps;
     m_StreamConfig.bitrate = m_Preferences.bitrateKbps;
     m_StreamConfig.hevcBitratePercentageMultiplier = 75;
+
+    SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
+                "Video bitrate: %d kbps",
+                m_StreamConfig.bitrate);
+
     RAND_bytes(reinterpret_cast<unsigned char*>(m_StreamConfig.remoteInputAesKey),
                sizeof(m_StreamConfig.remoteInputAesKey));
 
@@ -313,10 +328,30 @@ Session::Session(NvComputer* computer, NvApp& app)
     }
 }
 
+void Session::emitLaunchWarning(QString text)
+{
+    // Emit the warning to the UI
+    emit displayLaunchWarning(text);
+
+    // Wait a little bit so the user can actually read what we just said.
+    // This wait is a little longer than the actual toast timeout (3 seconds)
+    // to allow it to transition off the screen before continuing.
+    uint32_t start = SDL_GetTicks();
+    while (!SDL_TICKS_PASSED(SDL_GetTicks(), start + 3500)) {
+        // Pump the UI loop while we wait
+        SDL_Delay(5);
+        QCoreApplication::processEvents(QEventLoop::ExcludeUserInputEvents);
+    }
+}
+
 bool Session::validateLaunch()
 {
     QStringList warningList;
 
+    if (m_Preferences.videoDecoderSelection == StreamingPreferences::VDS_FORCE_SOFTWARE) {
+        emitLaunchWarning("Your settings selection to force software decoding may cause poor streaming performance.");
+    }
+
     if (m_StreamConfig.supportsHevc) {
         bool hevcForced = m_Preferences.videoCodecConfig == StreamingPreferences::VCC_FORCE_HEVC ||
                 m_Preferences.videoCodecConfig == StreamingPreferences::VCC_FORCE_HEVC_HDR;
@@ -325,22 +360,21 @@ bool Session::validateLaunch()
                                        VIDEO_FORMAT_H265,
                                        m_StreamConfig.width,
                                        m_StreamConfig.height,
-                                       m_StreamConfig.fps)) {
-            // NOTE: HEVC currently uses only 1 slice regardless of what
-            // we provide in CAPABILITY_SLICES_PER_FRAME(), so we should
-            // never use it for software decoding (unless common-c starts
-            // respecting it for HEVC).
-            m_StreamConfig.supportsHevc = false;
-
+                                       m_StreamConfig.fps) &&
+                m_Preferences.videoDecoderSelection != StreamingPreferences::VDS_FORCE_SOFTWARE) {
             if (hevcForced) {
-                emit displayLaunchWarning("This PC's GPU doesn't support HEVC decoding.");
+                emitLaunchWarning("Using software decoding due to your selection to force HEVC. This may cause poor streaming performance.");
+            }
+            else {
+                emitLaunchWarning("This PC's GPU doesn't support HEVC decoding.");
+                m_StreamConfig.supportsHevc = false;
             }
         }
 
         if (hevcForced) {
             if (m_Computer->maxLumaPixelsHEVC == 0) {
-                emit displayLaunchWarning("Your host PC GPU doesn't support HEVC. "
-                                          "A GeForce GTX 900-series (Maxwell) or later GPU is required for HEVC streaming.");
+                emitLaunchWarning("Your host PC GPU doesn't support HEVC. "
+                                  "A GeForce GTX 900-series (Maxwell) or later GPU is required for HEVC streaming.");
             }
         }
     }
@@ -351,19 +385,19 @@ bool Session::validateLaunch()
 
         // Check that the app supports HDR
         if (!m_App.hdrSupported) {
-            emit displayLaunchWarning(m_App.name + " doesn't support HDR10.");
+            emitLaunchWarning(m_App.name + " doesn't support HDR10.");
         }
         // Check that the server GPU supports HDR
         else if (!(m_Computer->serverCodecModeSupport & 0x200)) {
-            emit displayLaunchWarning("Your host PC GPU doesn't support HDR streaming. "
-                                      "A GeForce GTX 1000-series (Pascal) or later GPU is required for HDR streaming.");
+            emitLaunchWarning("Your host PC GPU doesn't support HDR streaming. "
+                              "A GeForce GTX 1000-series (Pascal) or later GPU is required for HDR streaming.");
         }
         else if (!isHardwareDecodeAvailable(m_Preferences.videoDecoderSelection,
                                             VIDEO_FORMAT_H265_MAIN10,
                                             m_StreamConfig.width,
                                             m_StreamConfig.height,
                                             m_StreamConfig.fps)) {
-            emit displayLaunchWarning("This PC's GPU doesn't support HEVC Main10 decoding for HDR streaming.");
+            emitLaunchWarning("This PC's GPU doesn't support HEVC Main10 decoding for HDR streaming.");
         }
         else {
             // TODO: Also validate display capabilites
@@ -376,7 +410,7 @@ bool Session::validateLaunch()
     if (m_StreamConfig.width >= 3840) {
         // Only allow 4K on GFE 3.x+
         if (m_Computer->gfeVersion.isNull() || m_Computer->gfeVersion.startsWith("2.")) {
-            emit displayLaunchWarning("GeForce Experience 3.0 or higher is required for 4K streaming.");
+            emitLaunchWarning("GeForce Experience 3.0 or higher is required for 4K streaming.");
 
             m_StreamConfig.width = 1920;
             m_StreamConfig.height = 1080;
@@ -384,8 +418,8 @@ bool Session::validateLaunch()
         // This list is sorted from least to greatest
         else if (m_Computer->displayModes.last().width < 3840 ||
                  (m_Computer->displayModes.last().refreshRate < 60 && m_StreamConfig.fps >= 60)) {
-            emit displayLaunchWarning("Your host PC GPU doesn't support 4K streaming. "
-                                      "A GeForce GTX 900-series (Maxwell) or later GPU is required for 4K streaming.");
+            emitLaunchWarning("Your host PC GPU doesn't support 4K streaming. "
+                              "A GeForce GTX 900-series (Maxwell) or later GPU is required for 4K streaming.");
 
             m_StreamConfig.width = 1920;
             m_StreamConfig.height = 1080;
@@ -416,6 +450,24 @@ void Session::getWindowDimensions(bool fullScreen,
 {
     int displayIndex = 0;
 
+    // If there's a display matching this exact resolution, pick that
+    // one (for native full-screen streaming). Otherwise, assume
+    // display 0 for now. TODO: Default to the screen that the Qt window is on
+    if (fullScreen) {
+        for (int i = 0; i < SDL_GetNumVideoDisplays(); i++) {
+            SDL_DisplayMode mode;
+            if (SDL_GetCurrentDisplayMode(i, &mode) == 0 &&
+                    m_ActiveVideoWidth == mode.w &&
+                    m_ActiveVideoHeight == mode.h) {
+                SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
+                            "Found exact resolution match on display: %d",
+                            i);
+                displayIndex = i;
+                break;
+            }
+        }
+    }
+
     if (m_Window != nullptr) {
         displayIndex = SDL_GetWindowDisplayIndex(m_Window);
         if (displayIndex < 0) {
@@ -471,6 +523,13 @@ void Session::getWindowDimensions(bool fullScreen,
 
                 width -= left + right;
                 height -= top + bottom;
+
+                // If the stream window can fit within the usable drawing area with 1:1
+                // scaling, do that rather than filling the screen.
+                if (m_StreamConfig.width < width && m_StreamConfig.height < height) {
+                    width = m_StreamConfig.width;
+                    height = m_StreamConfig.height;
+                }
             }
         }
         else {
@@ -495,6 +554,7 @@ void Session::toggleFullscreen()
     // that way we have accurate display size metrics (not the size our stream changed it to).
     if (!fullScreen) {
         SDL_SetWindowFullscreen(m_Window, 0);
+        SDL_SetWindowResizable(m_Window, SDL_TRUE);
     }
 
     getWindowDimensions(fullScreen, x, y, width, height);
@@ -503,6 +563,7 @@ void Session::toggleFullscreen()
     SDL_SetWindowSize(m_Window, width, height);
 
     if (fullScreen) {
+        SDL_SetWindowResizable(m_Window, SDL_FALSE);
         SDL_SetWindowFullscreen(m_Window, SDL_OS_FULLSCREEN_FLAG);
     }
 }
@@ -533,6 +594,18 @@ void Session::exec()
     Q_ASSERT(m_Computer->currentGameId == 0 ||
              m_Computer->currentGameId == m_App.id);
 
+    bool enableGameOptimizations = false;
+    for (const NvDisplayMode &mode : m_Computer->displayModes) {
+        if (mode.width == m_StreamConfig.width &&
+                mode.height == m_StreamConfig.height) {
+            SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
+                        "Found host supported resolution: %dx%d",
+                        mode.width, mode.height);
+            enableGameOptimizations = prefs.gameOptimizations;
+            break;
+        }
+    }
+
     try {
         NvHTTP http(m_Computer->activeAddress);
         if (m_Computer->currentGameId != 0) {
@@ -540,7 +613,7 @@ void Session::exec()
         }
         else {
             http.launchApp(m_App.id, &m_StreamConfig,
-                           prefs.gameOptimizations,
+                           enableGameOptimizations,
                            prefs.playAudioOnHost,
                            inputHandler.getAttachedGamepadMask());
         }
@@ -619,6 +692,10 @@ void Session::exec()
 
         SDL_SetWindowPosition(m_Window, x, y);
         SDL_SetWindowSize(m_Window, width, height);
+
+        // Passing SDL_WINDOW_RESIZABLE to set this during window
+        // creation causes our window to be full screen for some reason
+        SDL_SetWindowResizable(m_Window, SDL_TRUE);
     }
 
     QSvgRenderer svgIconRenderer(QString(":/res/moonlight.svg"));
@@ -647,6 +724,12 @@ void Session::exec()
     SDL_SetRelativeMouseMode(SDL_TRUE);
 #endif
 
+    // Stop text input. SDL enables it by default
+    // when we initialize the video subsystem, but this
+    // causes an IME popup when certain keys are held down
+    // on macOS.
+    SDL_StopTextInput();
+
     // Disable the screen saver
     SDL_DisableScreenSaver();
 
@@ -661,7 +744,15 @@ void Session::exec()
     // Hijack this thread to be the SDL main thread. We have to do this
     // because we want to suspend all Qt processing until the stream is over.
     SDL_Event event;
-    while (SDL_WaitEvent(&event)) {
+    for (;;) {
+        // We explicitly use SDL_PollEvent() and SDL_Delay() because
+        // SDL_WaitEvent() has an internal SDL_Delay(10) inside which
+        // blocks this thread too long for high polling rate mice and high
+        // refresh rate displays.
+        if (!SDL_PollEvent(&event)) {
+            SDL_Delay(1);
+            continue;
+        }
         switch (event.type) {
         case SDL_QUIT:
             SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,

app/streaming/session.hpp

@@ -39,6 +39,8 @@ signals:
 private:
     bool validateLaunch();
 
+    void emitLaunchWarning(QString text);
+
     int getDecoderCapabilities();
 
     int sdlDetermineAudioConfiguration();

app/streaming/streamutils.cpp

@@ -0,0 +1,20 @@
+#include "streamutils.h"
+
+void StreamUtils::scaleSourceToDestinationSurface(SDL_Rect* src, SDL_Rect* dst)
+{
+    int dstH = dst->w * src->h / src->w;
+    int dstW = dst->h * src->w / src->h;
+
+    if (dstH > dst->h) {
+        dst->y = 0;
+        dst->x = (dst->w - dstW) / 2;
+        dst->w = dstW;
+        SDL_assert(dst->w * src->h / src->w <= dst->h);
+    }
+    else {
+        dst->x = 0;
+        dst->y = (dst->h - dstH) / 2;
+        dst->h = dstH;
+        SDL_assert(dst->h * src->w / src->h <= dst->w);
+    }
+}

app/streaming/streamutils.h

@@ -0,0 +1,10 @@
+#pragma once
+
+#include <SDL.h>
+
+class StreamUtils
+{
+public:
+    static
+    void scaleSourceToDestinationSurface(SDL_Rect* src, SDL_Rect* dst);
+};

app/streaming/video/ffmpeg-renderers/dxva2.cpp

@@ -1,5 +1,7 @@
 #include <initguid.h>
 #include "dxva2.h"
+#include "../ffmpeg.h"
+#include <streaming/streamutils.h>
 
 #include <Limelight.h>
 
@@ -75,11 +77,8 @@ bool DXVA2Renderer::prepareDecoderContext(AVCodecContext* context)
 
     context->hwaccel_context = &m_DXVAContext;
 
-    context->get_format = ffGetFormat;
     context->get_buffer2 = ffGetBuffer2;
 
-    context->opaque = this;
-
     m_Pool = av_buffer_pool_init2(ARRAYSIZE(m_DecSurfaces), this, ffPoolAlloc, nullptr);
     if (!m_Pool) {
         SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
@@ -95,7 +94,7 @@ bool DXVA2Renderer::prepareDecoderContext(AVCodecContext* context)
 
 int DXVA2Renderer::ffGetBuffer2(AVCodecContext* context, AVFrame* frame, int)
 {
-    DXVA2Renderer* me = reinterpret_cast<DXVA2Renderer*>(context->opaque);
+    DXVA2Renderer* me = (DXVA2Renderer*)((FFmpegVideoDecoder*)context->opaque)->getRenderer();
 
     frame->buf[0] = av_buffer_pool_get(me->m_Pool);
     if (!frame->buf[0]) {
@@ -110,25 +109,14 @@ int DXVA2Renderer::ffGetBuffer2(AVCodecContext* context, AVFrame* frame, int)
     return 0;
 }
 
-enum AVPixelFormat DXVA2Renderer::ffGetFormat(AVCodecContext*,
-                                              const enum AVPixelFormat* pixFmts)
-{
-    const enum AVPixelFormat *p;
-
-    for (p = pixFmts; *p != -1; p++) {
-        if (*p == AV_PIX_FMT_DXVA2_VLD)
-            return *p;
-    }
-
-    SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
-                 "Failed to find DXVA2 HW surface format");
-    return AV_PIX_FMT_NONE;
-}
-
 bool DXVA2Renderer::initializeDecoder()
 {
     HRESULT hr;
 
+    if (isDecoderBlacklisted()) {
+        return false;
+    }
+
     hr = DXVA2CreateVideoService(m_Device, IID_IDirectXVideoDecoderService,
                                  reinterpret_cast<void**>(&m_DecService));
     if (FAILED(hr)) {
@@ -211,18 +199,11 @@ bool DXVA2Renderer::initializeDecoder()
         return false;
     }
 
-    int alignment;
-
-    // HEVC using DXVA requires 128B alignment
-    if (m_VideoFormat & VIDEO_FORMAT_MASK_H265) {
-        alignment = 128;
-    }
-    else {
-        alignment = 16;
-    }
-
-    hr = m_DecService->CreateSurface(FFALIGN(m_VideoWidth, alignment),
-                                     FFALIGN(m_VideoHeight, alignment),
+    // Alignment was already taken care of
+    SDL_assert(m_Desc.SampleWidth % 16 == 0);
+    SDL_assert(m_Desc.SampleHeight % 16 == 0);
+    hr = m_DecService->CreateSurface(m_Desc.SampleWidth,
+                                     m_Desc.SampleHeight,
                                      ARRAYSIZE(m_DecSurfaces) - 1,
                                      m_Desc.Format,
                                      D3DPOOL_DEFAULT,
@@ -352,6 +333,103 @@ bool DXVA2Renderer::initializeRenderer()
     return true;
 }
 
+bool DXVA2Renderer::isDecoderBlacklisted()
+{
+    IDirect3D9* d3d9;
+    HRESULT hr;
+    bool result = false;
+
+    // TODO: Update for HEVC Main10
+    SDL_assert(m_VideoFormat != VIDEO_FORMAT_H265_MAIN10);
+
+    hr = m_Device->GetDirect3D(&d3d9);
+    if (SUCCEEDED(hr)) {
+        D3DCAPS9 caps;
+
+        hr = m_Device->GetDeviceCaps(&caps);
+        if (SUCCEEDED(hr)) {
+            D3DADAPTER_IDENTIFIER9 id;
+
+            hr = d3d9->GetAdapterIdentifier(caps.AdapterOrdinal, 0, &id);
+            if (SUCCEEDED(hr)) {
+                SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
+                            "Detected GPU: %s (%x:%x)",
+                            id.Description,
+                            id.VendorId,
+                            id.DeviceId);
+
+                if (id.VendorId == 0x8086) {
+                    // Intel seems to encode the series in the high byte of
+                    // the device ID. We want to avoid the "Partial" acceleration
+                    // support explicitly. Those will claim to have HW acceleration
+                    // but perform badly.
+                    // https://en.wikipedia.org/wiki/Intel_Graphics_Technology#Capabilities_(GPU_video_acceleration)
+                    // https://raw.githubusercontent.com/GameTechDev/gpudetect/master/IntelGfx.cfg
+                    switch (id.DeviceId & 0xFF00) {
+                    case 0x0400: // Haswell
+                    case 0x0A00: // Haswell
+                    case 0x0D00: // Haswell
+                    case 0x1600: // Broadwell
+                    case 0x2200: // Cherry Trail and Braswell
+                        // Blacklist these for HEVC to avoid hybrid decode
+                        result = (m_VideoFormat & VIDEO_FORMAT_MASK_H265) != 0;
+                        break;
+                    default:
+                        // Everything else is fine with whatever it says it supports
+                        result = false;
+                        break;
+                    }
+                }
+                else if (id.VendorId == 0x10DE) {
+                    // For NVIDIA, we wait to avoid those GPUs with Feature Set E
+                    // for HEVC decoding, since that's hybrid.
+                    // https://en.wikipedia.org/wiki/Nvidia_PureVideo
+                    // http://envytools.readthedocs.io/en/latest/hw/pciid.html (missing GM200)
+                    if ((id.DeviceId >= 0x1340 && id.DeviceId <= 0x137F) || // GM108
+                            (id.DeviceId >= 0x1380 && id.DeviceId <= 0x13BF) || // GM107
+                            (id.DeviceId >= 0x13C0 && id.DeviceId <= 0x13FF) || // GM204
+                            (id.DeviceId >= 0x1617 && id.DeviceId <= 0x161A) || // GM204
+                            (id.DeviceId == 0x1667) || // GM204
+                            (id.DeviceId >= 0x17C0 && id.DeviceId <= 0x17FF)) { // GM200
+                        // Avoid HEVC on Feature Set E GPUs
+                        result = (m_VideoFormat & VIDEO_FORMAT_MASK_H265) != 0;
+                    }
+                    else {
+                        result = false;
+                    }
+                }
+                else if (id.VendorId == 0x1002) {
+                    // AMD doesn't seem to do hybrid acceleration?
+                    result = false;
+                }
+                else {
+                    SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
+                                "Unrecognized vendor ID: %x",
+                                id.VendorId);
+                }
+            }
+        }
+        else {
+            SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                         "GetDeviceCaps() failed: %x", hr);
+        }
+
+        d3d9->Release();
+    }
+    else {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "GetDirect3D() failed: %x", hr);
+    }
+
+    if (result) {
+        SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
+                    "GPU blacklisted for format %x",
+                    m_VideoFormat);
+    }
+
+    return result;
+}
+
 bool DXVA2Renderer::initialize(SDL_Window* window, int videoFormat, int width, int height)
 {
     m_VideoFormat = videoFormat;
@@ -381,8 +459,19 @@ bool DXVA2Renderer::initialize(SDL_Window* window, int videoFormat, int width, i
     SDL_RenderPresent(m_SdlRenderer);
 
     RtlZeroMemory(&m_Desc, sizeof(m_Desc));
-    m_Desc.SampleWidth = m_VideoWidth;
-    m_Desc.SampleHeight = m_VideoHeight;
+
+    int alignment;
+
+    // HEVC using DXVA requires 128B alignment
+    if (m_VideoFormat & VIDEO_FORMAT_MASK_H265) {
+        alignment = 128;
+    }
+    else {
+        alignment = 16;
+    }
+
+    m_Desc.SampleWidth = FFALIGN(m_VideoWidth, alignment);
+    m_Desc.SampleHeight = FFALIGN(m_VideoHeight, alignment);
     m_Desc.SampleFormat.VideoChromaSubsampling = DXVA2_VideoChromaSubsampling_Unknown;
     m_Desc.SampleFormat.NominalRange = DXVA2_NominalRange_Unknown;
     m_Desc.SampleFormat.VideoTransferMatrix = DXVA2_VideoTransferMatrix_Unknown;
@@ -503,28 +592,26 @@ void DXVA2Renderer::renderFrame(AVFrame* frame)
     sample.Start = m_FrameIndex;
     sample.End = m_FrameIndex + 1;
     sample.SrcSurface = surface;
-    sample.SrcRect.right = m_Desc.SampleWidth;
-    sample.SrcRect.bottom = m_Desc.SampleHeight;
+    sample.SrcRect.right = m_VideoWidth;
+    sample.SrcRect.bottom = m_VideoHeight;
     sample.SampleFormat = m_Desc.SampleFormat;
     sample.PlanarAlpha = DXVA2_Fixed32OpaqueAlpha();
 
     // Center in frame and preserve aspect ratio
-    double srcAspectRatio = (double)m_Desc.SampleWidth / (double)m_Desc.SampleHeight;
-    double dstAspectRatio = (double)m_DisplayWidth / (double)m_DisplayHeight;
-    if (dstAspectRatio < srcAspectRatio) {
-        // Greater height per width
-        int drawHeight = (int)(m_DisplayWidth / srcAspectRatio);
-        sample.DstRect.top = (m_DisplayHeight - drawHeight) / 2;
-        sample.DstRect.bottom = sample.DstRect.bottom + drawHeight;
-        sample.DstRect.right = m_DisplayWidth;
-    }
-    else {
-        // Greater width per height
-        int drawWidth = (int)(m_DisplayHeight * srcAspectRatio);
-        sample.DstRect.bottom = m_DisplayHeight;
-        sample.DstRect.left = (m_DisplayWidth - drawWidth) / 2;
-        sample.DstRect.right = sample.DstRect.left + drawWidth;
-    }
+    SDL_Rect src, dst;
+    src.x = src.y = 0;
+    src.w = m_VideoWidth;
+    src.h = m_VideoHeight;
+    dst.x = dst.y = 0;
+    dst.w = m_DisplayWidth;
+    dst.h = m_DisplayHeight;
+
+    StreamUtils::scaleSourceToDestinationSurface(&src, &dst);
+
+    sample.DstRect.left = dst.x;
+    sample.DstRect.right = dst.x + dst.w;
+    sample.DstRect.top = dst.y;
+    sample.DstRect.bottom = dst.y + dst.h;
 
     DXVA2_VideoProcessBltParams bltParams = {};
 

app/streaming/video/ffmpeg-renderers/dxva2.h

@@ -24,6 +24,7 @@ public:
 private:
     bool initializeDecoder();
     bool initializeRenderer();
+    bool isDecoderBlacklisted();
 
     static
     AVBufferRef* ffPoolAlloc(void* opaque, int size);
@@ -34,10 +35,6 @@ private:
     static
     int ffGetBuffer2(AVCodecContext* context, AVFrame* frame, int flags);
 
-    static
-    enum AVPixelFormat ffGetFormat(AVCodecContext*,
-                                   const enum AVPixelFormat* pixFmts);
-
     int m_VideoFormat;
     int m_VideoWidth;
     int m_VideoHeight;

app/streaming/video/ffmpeg-renderers/sdl.cpp

@@ -31,18 +31,10 @@ bool SdlRenderer::prepareDecoderContext(AVCodecContext*)
 }
 
 bool SdlRenderer::initialize(SDL_Window* window,
-                             int videoFormat,
+                             int,
                              int width,
                              int height)
 {
-    // NOTE: HEVC currently uses only 1 slice regardless of what
-    // we provide in CAPABILITY_SLICES_PER_FRAME(), so we should
-    // never use it for software decoding (unless common-c starts
-    // respecting it for HEVC).
-    if (videoFormat != VIDEO_FORMAT_H264) {
-        return false;
-    }
-
     m_Renderer = SDL_CreateRenderer(window, -1, SDL_RENDERER_ACCELERATED);
     if (!m_Renderer) {
         SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,

app/streaming/video/ffmpeg-renderers/vaapi.cpp

@@ -1,4 +1,5 @@
 #include "vaapi.h"
+#include <streaming/streamutils.h>
 
 #include <dlfcn.h>
 
@@ -70,7 +71,6 @@ VAAPIRenderer::initialize(SDL_Window* window, int, int width, int height)
         return false;
     }
 
-
     err = av_hwdevice_ctx_create(&m_HwContext,
                                  AV_HWDEVICE_TYPE_VAAPI,
                                  nullptr, nullptr, 0);
@@ -81,6 +81,14 @@ VAAPIRenderer::initialize(SDL_Window* window, int, int width, int height)
         return false;
     }
 
+    // This quirk is set for the VDPAU wrapper which doesn't work with our VAAPI renderer
+    if (((AVVAAPIDeviceContext*)((AVHWDeviceContext*)(m_HwContext->data))->hwctx)->driver_quirks & AV_VAAPI_DRIVER_QUIRK_SURFACE_ATTRIBUTES) {
+        // Fail and let our VDPAU renderer pick this up
+        SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
+                    "Avoiding VDPAU wrapper for VAAPI decoding");
+        return false;
+    }
+
     return true;
 }
 
@@ -102,34 +110,23 @@ VAAPIRenderer::renderFrame(AVFrame* frame)
     AVHWDeviceContext* deviceContext = (AVHWDeviceContext*)m_HwContext->data;
     AVVAAPIDeviceContext* vaDeviceContext = (AVVAAPIDeviceContext*)deviceContext->hwctx;
 
-    // Center in frame and preserve aspect ratio
-    int x, y, width, height;
-    double srcAspectRatio = (double)m_VideoWidth / (double)m_VideoHeight;
-    double dstAspectRatio = (double)m_DisplayWidth / (double)m_DisplayHeight;
-    if (dstAspectRatio < srcAspectRatio) {
-        // Greater height per width
-        int drawHeight = (int)(m_DisplayWidth / srcAspectRatio);
-        y = (m_DisplayHeight - drawHeight) / 2;
-        height = drawHeight;
-        x = 0;
-        width = m_DisplayWidth;
-    }
-    else {
-        // Greater width per height
-        int drawWidth = (int)(m_DisplayHeight * srcAspectRatio);
-        y = 0;
-        height = m_DisplayHeight;
-        x = (m_DisplayWidth - drawWidth) / 2;
-        width = drawWidth;
-    }
+    SDL_Rect src, dst;
+    src.x = src.y = 0;
+    src.w = m_VideoWidth;
+    src.h = m_VideoHeight;
+    dst.x = dst.y = 0;
+    dst.w = m_DisplayWidth;
+    dst.h = m_DisplayHeight;
+
+    StreamUtils::scaleSourceToDestinationSurface(&src, &dst);
 
     m_vaPutSurface(vaDeviceContext->display,
                    surface,
                    m_XWindow,
                    0, 0,
                    m_VideoWidth, m_VideoHeight,
-                   x, y,
-                   width, height,
+                   dst.x, dst.y,
+                   dst.w, dst.h,
                    NULL, 0, 0);
 
     av_frame_free(&frame);

app/streaming/video/ffmpeg-renderers/vaapi.h

@@ -47,4 +47,3 @@ private:
     int m_DisplayWidth;
     int m_DisplayHeight;
 };
-

app/streaming/video/ffmpeg-renderers/vdpau.cpp

@@ -0,0 +1,332 @@
+#include "vdpau.h"
+#include <streaming/streamutils.h>
+
+#include <SDL_syswm.h>
+
+#define BAIL_ON_FAIL(status, something) if ((status) != VDP_STATUS_OK) { \
+                                            SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION, \
+                                                        #something " failed: %d", (status)); \
+                                            return false; \
+                                        }
+
+#define GET_PROC_ADDRESS(id, func) status = vdpauCtx->get_proc_address(m_Device, id, (void**)func); \
+                                   BAIL_ON_FAIL(status, id)
+
+const VdpRGBAFormat VDPAURenderer::k_OutputFormats[] = {
+    VDP_RGBA_FORMAT_B8G8R8A8,
+    VDP_RGBA_FORMAT_R8G8B8A8
+};
+
+VDPAURenderer::VDPAURenderer()
+    : m_HwContext(nullptr),
+      m_PresentationQueueTarget(0),
+      m_PresentationQueue(0),
+      m_VideoMixer(0),
+      m_NextSurfaceIndex(0)
+{
+    SDL_zero(m_OutputSurface);
+}
+
+VDPAURenderer::~VDPAURenderer()
+{
+    if (m_PresentationQueue != 0) {
+        m_VdpPresentationQueueDestroy(m_PresentationQueue);
+    }
+
+    if (m_VideoMixer != 0) {
+        m_VdpVideoMixerDestroy(m_VideoMixer);
+    }
+
+    if (m_PresentationQueueTarget != 0) {
+        m_VdpPresentationQueueTargetDestroy(m_PresentationQueueTarget);
+    }
+
+    for (int i = 0; i < OUTPUT_SURFACE_COUNT; i++) {
+        if (m_OutputSurface[i] != 0) {
+            m_VdpOutputSurfaceDestroy(m_OutputSurface[i]);
+        }
+    }
+
+    // This must be done last as it frees VDPAU context required to call
+    // the functions above.
+    if (m_HwContext != nullptr) {
+        av_buffer_unref(&m_HwContext);
+    }
+}
+
+bool VDPAURenderer::initialize(SDL_Window* window, int, int width, int height)
+{
+    int err;
+    VdpStatus status;
+    SDL_SysWMinfo info;
+
+    m_VideoWidth = width;
+    m_VideoHeight = height;
+
+    err = av_hwdevice_ctx_create(&m_HwContext,
+                                 AV_HWDEVICE_TYPE_VDPAU,
+                                 nullptr, nullptr, 0);
+    if (err < 0) {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "Failed to create VDPAU context: %d",
+                     err);
+        return false;
+    }
+
+    AVHWDeviceContext* devCtx = (AVHWDeviceContext*)m_HwContext->data;
+    AVVDPAUDeviceContext* vdpauCtx = (AVVDPAUDeviceContext*)devCtx->hwctx;
+    m_Device = vdpauCtx->device;
+
+    GET_PROC_ADDRESS(VDP_FUNC_ID_GET_ERROR_STRING, &m_VdpGetErrorString);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_PRESENTATION_QUEUE_TARGET_DESTROY, &m_VdpPresentationQueueTargetDestroy);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_VIDEO_MIXER_CREATE, &m_VdpVideoMixerCreate);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_VIDEO_MIXER_DESTROY, &m_VdpVideoMixerDestroy);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_VIDEO_MIXER_RENDER, &m_VdpVideoMixerRender);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_PRESENTATION_QUEUE_CREATE, &m_VdpPresentationQueueCreate);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_PRESENTATION_QUEUE_DESTROY, &m_VdpPresentationQueueDestroy);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_PRESENTATION_QUEUE_DISPLAY, &m_VdpPresentationQueueDisplay);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_PRESENTATION_QUEUE_SET_BACKGROUND_COLOR, &m_VdpPresentationQueueSetBackgroundColor);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_PRESENTATION_QUEUE_BLOCK_UNTIL_SURFACE_IDLE, &m_VdpPresentationQueueBlockUntilSurfaceIdle);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_OUTPUT_SURFACE_CREATE, &m_VdpOutputSurfaceCreate);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_OUTPUT_SURFACE_DESTROY, &m_VdpOutputSurfaceDestroy);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_OUTPUT_SURFACE_QUERY_CAPABILITIES, &m_VdpOutputSurfaceQueryCapabilities);
+    GET_PROC_ADDRESS(VDP_FUNC_ID_VIDEO_SURFACE_GET_PARAMETERS, &m_VdpVideoSurfaceGetParameters);
+
+    SDL_GetWindowSize(window, (int*)&m_DisplayWidth, (int*)&m_DisplayHeight);
+
+    SDL_VERSION(&info.version);
+
+    if (!SDL_GetWindowWMInfo(window, &info)) {
+        SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
+                    "SDL_GetWindowWMInfo() failed: %s",
+                    SDL_GetError());
+        return false;
+    }
+
+    SDL_assert(info.subsystem == SDL_SYSWM_X11);
+
+    if (info.subsystem == SDL_SYSWM_X11) {
+        GET_PROC_ADDRESS(VDP_FUNC_ID_PRESENTATION_QUEUE_TARGET_CREATE_X11,
+                         &m_VdpPresentationQueueTargetCreateX11);
+        status = m_VdpPresentationQueueTargetCreateX11(m_Device,
+                                                       info.info.x11.window,
+                                                       &m_PresentationQueueTarget);
+        if (status != VDP_STATUS_OK) {
+            SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                         "VdpPresentationQueueTargetCreateX11() failed: %s",
+                         m_VdpGetErrorString(status));
+            return false;
+        }
+    }
+    else if (info.subsystem == SDL_SYSWM_WAYLAND) {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "VDPAU backend does not currently support Wayland");
+        return false;
+    }
+    else {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "Unsupported VDPAU rendering subsystem: %d",
+                     info.subsystem);
+        return false;
+    }
+
+    // Try our available output formats to find something the GPU supports
+    bool foundFormat = false;
+    for (int i = 0; i < OUTPUT_SURFACE_FORMAT_COUNT; i++) {
+        VdpBool supported;
+        uint32_t maxWidth, maxHeight;
+        status = m_VdpOutputSurfaceQueryCapabilities(m_Device, k_OutputFormats[i],
+                                                     &supported, &maxWidth, &maxHeight);
+        if (status != VDP_STATUS_OK) {
+            SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                         "VdpOutputSurfaceQueryCapabilities() failed: %s",
+                         m_VdpGetErrorString(status));
+            return false;
+        }
+
+        if (supported) {
+            if (m_DisplayWidth <= maxWidth && m_DisplayHeight <= maxHeight) {
+                m_OutputSurfaceFormat = k_OutputFormats[i];
+                foundFormat = true;
+                break;
+            }
+            else  {
+                SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
+                            "Display size not within capabilities %dx%d vs %dx%d",
+                            m_DisplayWidth, m_DisplayWidth,
+                            maxWidth, maxHeight);
+            }
+        }
+    }
+
+    if (!foundFormat) {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "No compatible output surface format found!");
+        return false;
+    }
+
+    // Create the output surfaces
+    for (int i = 0; i < OUTPUT_SURFACE_COUNT; i++) {
+        // It seems there's some lazy freeing going on or something in VDPAU
+        // because we can get VDP_STATUS_RESOURCES, then wait a bit and it'll
+        // complete without a problem.
+        int tries = 1;
+        do {
+            status = m_VdpOutputSurfaceCreate(m_Device, m_OutputSurfaceFormat,
+                                              m_DisplayWidth, m_DisplayHeight,
+                                              &m_OutputSurface[i]);
+            if (status != VDP_STATUS_OK) {
+                SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
+                            "VdpOutputSurfaceCreate() try #%d: %s",
+                            tries,
+                            m_VdpGetErrorString(status));
+                SDL_Delay(250);
+            }
+        } while (status == VDP_STATUS_RESOURCES && ++tries <= 10);
+
+        if (status != VDP_STATUS_OK) {
+            SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                         "VdpOutputSurfaceCreate() failed: %s",
+                         m_VdpGetErrorString(status));
+            return false;
+        }
+    }
+
+    status = m_VdpPresentationQueueCreate(m_Device, m_PresentationQueueTarget,
+                                          &m_PresentationQueue);
+    if (status != VDP_STATUS_OK) {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "VdpPresentationQueueCreate() failed: %s",
+                     m_VdpGetErrorString(status));
+        return false;
+    }
+
+    // Set the background to opaque black
+    VdpColor color = {0.0, 0.0, 0.0, 1.0};
+    m_VdpPresentationQueueSetBackgroundColor(m_PresentationQueue, &color);
+
+    return true;
+}
+
+bool VDPAURenderer::prepareDecoderContext(AVCodecContext* context)
+{
+    context->hw_device_ctx = av_buffer_ref(m_HwContext);
+
+    // Allow HEVC usage on VDPAU. This was disabled by FFmpeg due to
+    // GL interop issues, but we use VDPAU for rendering so it's no issue.
+    // https://github.com/FFmpeg/FFmpeg/commit/64ecb78b7179cab2dbdf835463104679dbb7c895
+    context->hwaccel_flags |= AV_HWACCEL_FLAG_ALLOW_PROFILE_MISMATCH;
+
+    SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
+                "Using VDPAU accelerated renderer");
+
+    return true;
+}
+
+void VDPAURenderer::renderFrame(AVFrame* frame)
+{
+    VdpStatus status;
+    VdpVideoSurface videoSurface = (VdpVideoSurface)(uintptr_t)frame->data[3];
+
+    // This is safe without locking because this is always called on the main thread
+    VdpOutputSurface chosenSurface = m_OutputSurface[m_NextSurfaceIndex];
+    m_NextSurfaceIndex = (m_NextSurfaceIndex + 1) % OUTPUT_SURFACE_COUNT;
+
+    // We need to create the mixer on the fly, because we don't know the dimensions
+    // of our video surfaces in advance of decoding
+    if (m_VideoMixer == 0) {
+        VdpChromaType chroma;
+        uint32_t videoSurfaceWidth, videoSurfaceHeight;
+        status = m_VdpVideoSurfaceGetParameters(videoSurface, &chroma, &videoSurfaceWidth, &videoSurfaceHeight);
+        if (status != VDP_STATUS_OK) {
+            SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                         "VdpVideoSurfaceGetParameters() failed: %s",
+                         m_VdpGetErrorString(status));
+            av_frame_free(&frame);
+            return;
+        }
+
+        SDL_LogInfo(SDL_LOG_CATEGORY_APPLICATION,
+                    "VDPAU surface size: %dx%d",
+                    videoSurfaceWidth, videoSurfaceHeight);
+
+    #define PARAM_COUNT 2
+        const VdpVideoMixerParameter params[PARAM_COUNT] = {
+            VDP_VIDEO_MIXER_PARAMETER_VIDEO_SURFACE_WIDTH,
+            VDP_VIDEO_MIXER_PARAMETER_VIDEO_SURFACE_HEIGHT,
+        };
+        const void* const paramValues[PARAM_COUNT] = {
+            &videoSurfaceWidth,
+            &videoSurfaceHeight,
+        };
+
+        status = m_VdpVideoMixerCreate(m_Device, 0, nullptr,
+                                       PARAM_COUNT, params, paramValues,
+                                       &m_VideoMixer);
+        if (status != VDP_STATUS_OK) {
+            SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                         "VdpVideoMixerCreate() failed: %s",
+                         m_VdpGetErrorString(status));
+            av_frame_free(&frame);
+            return;
+        }
+    }
+
+    // Wait for this frame to be off the screen
+    VdpTime pts;
+    m_VdpPresentationQueueBlockUntilSurfaceIdle(m_PresentationQueue, chosenSurface, &pts);
+
+    VdpRect sourceRect, outputRect;
+
+    SDL_Rect src, dst;
+    src.x = src.y = 0;
+    src.w = m_VideoWidth;
+    src.h = m_VideoHeight;
+    dst.x = dst.y = 0;
+    dst.w = m_DisplayWidth;
+    dst.h = m_DisplayHeight;
+
+    StreamUtils::scaleSourceToDestinationSurface(&src, &dst);
+
+    outputRect.x0 = dst.x;
+    outputRect.x1 = dst.x + dst.w;
+    outputRect.y0 = dst.y;
+    outputRect.y1 = dst.y + dst.h;
+
+    sourceRect.x0 = sourceRect.y0 = 0;
+    sourceRect.x1 = m_VideoWidth;
+    sourceRect.y1 = m_VideoHeight;
+
+    // Render the next frame into the output surface
+    status = m_VdpVideoMixerRender(m_VideoMixer,
+                                   VDP_INVALID_HANDLE, nullptr,
+                                   VDP_VIDEO_MIXER_PICTURE_STRUCTURE_FRAME,
+                                   0, nullptr,
+                                   videoSurface,
+                                   0, nullptr,
+                                   &sourceRect,
+                                   chosenSurface,
+                                   &outputRect,
+                                   nullptr,
+                                   0,
+                                   nullptr);
+
+    // The decoder can have this surface back now
+    av_frame_free(&frame);
+
+    if (status != VDP_STATUS_OK) {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "VdpVideoMixerRender() failed: %s",
+                     m_VdpGetErrorString(status));
+        return;
+    }
+
+    // Queue the frame for display immediately
+    status = m_VdpPresentationQueueDisplay(m_PresentationQueue, chosenSurface, 0, 0, 0);
+    if (status != VDP_STATUS_OK) {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "VdpPresentationQueueDisplay() failed: %s",
+                     m_VdpGetErrorString(status));
+        return;
+    }
+}

app/streaming/video/ffmpeg-renderers/vdpau.h

@@ -0,0 +1,59 @@
+#pragma once
+
+#include "renderer.h"
+
+extern "C" {
+#include <vdpau/vdpau.h>
+#include <vdpau/vdpau_x11.h>
+#include <libavutil/hwcontext_vdpau.h>
+}
+
+class VDPAURenderer : public IFFmpegRenderer
+{
+public:
+    VDPAURenderer();
+    virtual ~VDPAURenderer();
+    virtual bool initialize(SDL_Window* window,
+                            int videoFormat,
+                            int width,
+                            int height);
+    virtual bool prepareDecoderContext(AVCodecContext* context);
+    virtual void renderFrame(AVFrame* frame);
+
+private:
+    uint32_t m_VideoWidth, m_VideoHeight;
+    uint32_t m_DisplayWidth, m_DisplayHeight;
+    AVBufferRef* m_HwContext;
+    VdpPresentationQueueTarget m_PresentationQueueTarget;
+    VdpPresentationQueue m_PresentationQueue;
+    VdpVideoMixer m_VideoMixer;
+    VdpRGBAFormat m_OutputSurfaceFormat;
+    VdpDevice m_Device;
+
+#define OUTPUT_SURFACE_COUNT 3
+    VdpOutputSurface m_OutputSurface[OUTPUT_SURFACE_COUNT];
+    int m_NextSurfaceIndex;
+
+#define OUTPUT_SURFACE_FORMAT_COUNT 2
+    static const VdpRGBAFormat k_OutputFormats[OUTPUT_SURFACE_FORMAT_COUNT];
+
+    VdpGetErrorString* m_VdpGetErrorString;
+    VdpPresentationQueueTargetDestroy* m_VdpPresentationQueueTargetDestroy;
+    VdpVideoMixerCreate* m_VdpVideoMixerCreate;
+    VdpVideoMixerDestroy* m_VdpVideoMixerDestroy;
+    VdpVideoMixerRender* m_VdpVideoMixerRender;
+    VdpPresentationQueueCreate* m_VdpPresentationQueueCreate;
+    VdpPresentationQueueDestroy* m_VdpPresentationQueueDestroy;
+    VdpPresentationQueueDisplay* m_VdpPresentationQueueDisplay;
+    VdpPresentationQueueSetBackgroundColor* m_VdpPresentationQueueSetBackgroundColor;
+    VdpPresentationQueueBlockUntilSurfaceIdle* m_VdpPresentationQueueBlockUntilSurfaceIdle;
+    VdpOutputSurfaceCreate* m_VdpOutputSurfaceCreate;
+    VdpOutputSurfaceDestroy* m_VdpOutputSurfaceDestroy;
+    VdpOutputSurfaceQueryCapabilities* m_VdpOutputSurfaceQueryCapabilities;
+    VdpVideoSurfaceGetParameters* m_VdpVideoSurfaceGetParameters;
+
+    // X11 stuff
+    VdpPresentationQueueTargetCreateX11* m_VdpPresentationQueueTargetCreateX11;
+};
+
+

app/streaming/video/ffmpeg.cpp

@@ -13,93 +13,39 @@
 #include "ffmpeg-renderers/vaapi.h"
 #endif
 
-bool FFmpegVideoDecoder::chooseDecoder(
-        StreamingPreferences::VideoDecoderSelection vds,
-        SDL_Window* window,
-        int videoFormat,
-        int width, int height,
-        AVCodec*& chosenDecoder,
-        const AVCodecHWConfig*& chosenHwConfig,
-        IFFmpegRenderer*& newRenderer)
-{
-    if (videoFormat & VIDEO_FORMAT_MASK_H264) {
-        chosenDecoder = avcodec_find_decoder(AV_CODEC_ID_H264);
-    }
-    else if (videoFormat & VIDEO_FORMAT_MASK_H265) {
-        chosenDecoder = avcodec_find_decoder(AV_CODEC_ID_HEVC);
-    }
-    else {
-        Q_ASSERT(false);
-        chosenDecoder = nullptr;
-    }
-
-    if (!chosenDecoder) {
-        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
-                     "Unable to find decoder for format: %x",
-                     videoFormat);
-        return false;
-    }
-
-    for (int i = 0;; i++) {
-        const AVCodecHWConfig *config = avcodec_get_hw_config(chosenDecoder, i);
-        if (!config || vds == StreamingPreferences::VDS_FORCE_SOFTWARE) {
-            // No matching hardware acceleration support.
-            // This is not an error.
-            chosenHwConfig = nullptr;
-            newRenderer = new SdlRenderer();
-            if (vds != StreamingPreferences::VDS_FORCE_HARDWARE &&
-                    newRenderer->initialize(window, videoFormat, width, height)) {
-                return true;
-            }
-            else {
-                delete newRenderer;
-                newRenderer = nullptr;
-                return false;
-            }
-        }
-
-        if (!(config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX)) {
-            continue;
-        }
-
-        // Look for acceleration types we support
-        switch (config->device_type) {
-#ifdef Q_OS_WIN32
-        case AV_HWDEVICE_TYPE_DXVA2:
-            newRenderer = new DXVA2Renderer();
-            break;
+#ifdef HAVE_LIBVDPAU
+#include "ffmpeg-renderers/vdpau.h"
 #endif
-#ifdef Q_OS_DARWIN
-        case AV_HWDEVICE_TYPE_VIDEOTOOLBOX:
-            newRenderer = VTRendererFactory::createRenderer();
-            break;
-#endif
-#ifdef HAVE_LIBVA
-        case AV_HWDEVICE_TYPE_VAAPI:
-            newRenderer = new VAAPIRenderer();
-            break;
-#endif
-        default:
-            continue;
-        }
 
-        if (newRenderer->initialize(window, videoFormat, width, height)) {
-            chosenHwConfig = config;
-            return true;
-        }
-        else {
-            // Failed to initialize
-            delete newRenderer;
-            newRenderer = nullptr;
-        }
-    }
-}
+// This is gross but it allows us to use sizeof()
+#include "ffmpeg_videosamples.cpp"
 
 bool FFmpegVideoDecoder::isHardwareAccelerated()
 {
     return m_HwDecodeCfg != nullptr;
 }
 
+enum AVPixelFormat FFmpegVideoDecoder::ffGetFormat(AVCodecContext* context,
+                                                   const enum AVPixelFormat* pixFmts)
+{
+    FFmpegVideoDecoder* decoder = (FFmpegVideoDecoder*)context->opaque;
+    const enum AVPixelFormat *p;
+
+    for (p = pixFmts; *p != -1; p++) {
+        // Only match our hardware decoding codec or SW pixel
+        // format (if not using hardware decoding). It's crucial
+        // to override the default get_format() which will try
+        // to gracefully fall back to software decode and break us.
+        if (*p == (decoder->m_HwDecodeCfg ?
+                   decoder->m_HwDecodeCfg->pix_fmt :
+                   context->pix_fmt)) {
+            return *p;
+        }
+    }
+
+    return AV_PIX_FMT_NONE;
+}
+
 FFmpegVideoDecoder::FFmpegVideoDecoder()
     : m_VideoDecoderCtx(nullptr),
       m_DecodeBuffer(1024 * 1024, 0),
@@ -114,6 +60,16 @@ FFmpegVideoDecoder::FFmpegVideoDecoder()
 }
 
 FFmpegVideoDecoder::~FFmpegVideoDecoder()
+{
+    reset();
+}
+
+IFFmpegRenderer* FFmpegVideoDecoder::getRenderer()
+{
+    return m_Renderer;
+}
+
+void FFmpegVideoDecoder::reset()
 {
     // Drop any frames still queued to ensure
     // they are properly freed.
@@ -140,24 +96,11 @@ FFmpegVideoDecoder::~FFmpegVideoDecoder()
     avcodec_free_context(&m_VideoDecoderCtx);
 
     delete m_Renderer;
+    m_Renderer = nullptr;
 }
 
-bool FFmpegVideoDecoder::initialize(
-        StreamingPreferences::VideoDecoderSelection vds,
-        SDL_Window* window,
-        int videoFormat,
-        int width,
-        int height,
-        int)
+bool FFmpegVideoDecoder::completeInitialization(AVCodec* decoder, int videoFormat, int width, int height, bool testOnly)
 {
-    AVCodec* decoder;
-
-    if (!chooseDecoder(vds, window, videoFormat, width, height,
-                       decoder, m_HwDecodeCfg, m_Renderer)) {
-        // Error logged in chooseDecoder()
-        return false;
-    }
-
     m_VideoDecoderCtx = avcodec_alloc_context3(decoder);
     if (!m_VideoDecoderCtx) {
         SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
@@ -182,12 +125,20 @@ bool FFmpegVideoDecoder::initialize(
     m_VideoDecoderCtx->width = width;
     m_VideoDecoderCtx->height = height;
     m_VideoDecoderCtx->pix_fmt = AV_PIX_FMT_YUV420P; // FIXME: HDR
+    m_VideoDecoderCtx->get_format = ffGetFormat;
 
     // Allow the renderer to attach data to this decoder
     if (!m_Renderer->prepareDecoderContext(m_VideoDecoderCtx)) {
         return false;
     }
 
+    // Nobody must override our ffGetFormat
+    SDL_assert(m_VideoDecoderCtx->get_format == ffGetFormat);
+
+    // Stash a pointer to this object in the context
+    SDL_assert(m_VideoDecoderCtx->opaque == nullptr);
+    m_VideoDecoderCtx->opaque = this;
+
     int err = avcodec_open2(m_VideoDecoderCtx, decoder, nullptr);
     if (err < 0) {
         SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
@@ -196,9 +147,148 @@ bool FFmpegVideoDecoder::initialize(
         return false;
     }
 
+    // FFMpeg doesn't completely initialize the codec until the codec
+    // config data comes in. This would be too late for us to change
+    // our minds on the selected video codec, so we'll do a trial run
+    // now to see if things will actually work when the video stream
+    // comes in.
+    if (testOnly) {
+        if (videoFormat & VIDEO_FORMAT_MASK_H264) {
+            m_Pkt.data = (uint8_t*)k_H264TestFrame;
+            m_Pkt.size = sizeof(k_H264TestFrame);
+        }
+        else {
+            m_Pkt.data = (uint8_t*)k_HEVCTestFrame;
+            m_Pkt.size = sizeof(k_HEVCTestFrame);
+        }
+
+        err = avcodec_send_packet(m_VideoDecoderCtx, &m_Pkt);
+        if (err < 0) {
+            char errorstring[512];
+            av_strerror(err, errorstring, sizeof(errorstring));
+            SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
+                        "Test decode failed: %s", errorstring);
+            return false;
+        }
+    }
+
+#ifdef QT_DEBUG
+    // Restore default log level before streaming
+    av_log_set_level(AV_LOG_INFO);
+#endif
+
     return true;
 }
 
+IFFmpegRenderer* FFmpegVideoDecoder::createAcceleratedRenderer(const AVCodecHWConfig* hwDecodeCfg)
+{
+    if (!(hwDecodeCfg->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX)) {
+        return nullptr;
+    }
+
+    // Look for acceleration types we support
+    switch (hwDecodeCfg->device_type) {
+#ifdef Q_OS_WIN32
+    case AV_HWDEVICE_TYPE_DXVA2:
+        return new DXVA2Renderer();
+#endif
+#ifdef Q_OS_DARWIN
+    case AV_HWDEVICE_TYPE_VIDEOTOOLBOX:
+        return VTRendererFactory::createRenderer();
+#endif
+#ifdef HAVE_LIBVA
+    case AV_HWDEVICE_TYPE_VAAPI:
+        return new VAAPIRenderer();
+#endif
+#ifdef HAVE_LIBVDPAU
+    case AV_HWDEVICE_TYPE_VDPAU:
+        return new VDPAURenderer();
+#endif
+    default:
+        return nullptr;
+    }
+}
+
+bool FFmpegVideoDecoder::initialize(
+        StreamingPreferences::VideoDecoderSelection vds,
+        SDL_Window* window,
+        int videoFormat,
+        int width,
+        int height,
+        int)
+{
+    AVCodec* decoder;
+
+#ifdef QT_DEBUG
+    // Increase log level during initialization
+    av_log_set_level(AV_LOG_DEBUG);
+#endif
+
+    if (videoFormat & VIDEO_FORMAT_MASK_H264) {
+        decoder = avcodec_find_decoder(AV_CODEC_ID_H264);
+    }
+    else if (videoFormat & VIDEO_FORMAT_MASK_H265) {
+        decoder = avcodec_find_decoder(AV_CODEC_ID_HEVC);
+    }
+    else {
+        Q_ASSERT(false);
+        decoder = nullptr;
+    }
+
+    if (!decoder) {
+        SDL_LogError(SDL_LOG_CATEGORY_APPLICATION,
+                     "Unable to find decoder for format: %x",
+                     videoFormat);
+        return false;
+    }
+
+    for (int i = 0;; i++) {
+        const AVCodecHWConfig *config = avcodec_get_hw_config(decoder, i);
+        if (!config || vds == StreamingPreferences::VDS_FORCE_SOFTWARE) {
+            // No matching hardware acceleration support.
+            // This is not an error.
+            m_HwDecodeCfg = nullptr;
+            m_Renderer = new SdlRenderer();
+            if (vds != StreamingPreferences::VDS_FORCE_HARDWARE &&
+                    m_Renderer->initialize(window, videoFormat, width, height) &&
+                    completeInitialization(decoder, videoFormat, width, height, false)) {
+                return true;
+            }
+            else {
+                reset();
+                return false;
+            }
+        }
+
+        m_Renderer = createAcceleratedRenderer(config);
+        if (!m_Renderer) {
+            continue;
+        }
+
+        m_HwDecodeCfg = config;
+        // Submit test frame to ensure this codec really works
+        if (m_Renderer->initialize(window, videoFormat, width, height) &&
+                completeInitialization(decoder, videoFormat, width, height, true)) {
+            // OK, it worked, so now let's initialize it for real
+            reset();
+            if ((m_Renderer = createAcceleratedRenderer(config)) != nullptr &&
+                    m_Renderer->initialize(window, videoFormat, width, height) &&
+                    completeInitialization(decoder, videoFormat, width, height, false)) {
+                return true;
+            }
+            else {
+                SDL_LogCritical(SDL_LOG_CATEGORY_APPLICATION,
+                                "Decoder failed to initialize after successful test");
+                reset();
+            }
+        }
+        else {
+            // Failed to initialize or test frame failed, so keep looking
+            reset();
+        }
+    }
+}
+
 int FFmpegVideoDecoder::submitDecodeUnit(PDECODE_UNIT du)
 {
     PLENTRY entry = du->bufferList;
@@ -224,8 +314,6 @@ int FFmpegVideoDecoder::submitDecodeUnit(PDECODE_UNIT du)
 
     err = avcodec_send_packet(m_VideoDecoderCtx, &m_Pkt);
     if (err < 0) {
-        SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,
-                    "Decoding failed: %d", err);
         char errorstring[512];
         av_strerror(err, errorstring, sizeof(errorstring));
         SDL_LogWarn(SDL_LOG_CATEGORY_APPLICATION,

app/streaming/video/ffmpeg.h

@@ -22,16 +22,19 @@ public:
     virtual void renderFrame(SDL_UserEvent* event) override;
     virtual void dropFrame(SDL_UserEvent* event) override;
 
+    virtual IFFmpegRenderer* getRenderer();
+
 private:
-    bool
-    chooseDecoder(
-        StreamingPreferences::VideoDecoderSelection vds,
-        SDL_Window* window,
-        int videoFormat,
-        int width, int height,
-        AVCodec*& chosenDecoder,
-        const AVCodecHWConfig*& chosenHwConfig,
-        IFFmpegRenderer*& newRenderer);
+    bool completeInitialization(AVCodec* decoder, int videoFormat,
+                                int width, int height, bool testOnly);
+
+    IFFmpegRenderer* createAcceleratedRenderer(const AVCodecHWConfig* hwDecodeCfg);
+
+    void reset();
+
+    static
+    enum AVPixelFormat ffGetFormat(AVCodecContext* context,
+                                   const enum AVPixelFormat* pixFmts);
 
     AVPacket m_Pkt;
     AVCodecContext* m_VideoDecoderCtx;
@@ -39,4 +42,7 @@ private:
     const AVCodecHWConfig* m_HwDecodeCfg;
     IFFmpegRenderer* m_Renderer;
     SDL_atomic_t m_QueuedFrames;
+
+    static const uint8_t k_H264TestFrame[];
+    static const uint8_t k_HEVCTestFrame[];
 };

app/streaming/video/ffmpeg_videosamples.cpp

@@ -0,0 +1,19 @@
+#include "ffmpeg.h"
+
+// 720p 60 FPS H.264 with 1 reference frame
+const uint8_t FFmpegVideoDecoder::k_H264TestFrame[] = {
+    0x00, 0x00, 0x00, 0x01, 0x67, 0x64, 0x00, 0x20, 0xac, 0x2b, 0x40, 0x28, 0x02, 0xdd, 0x80, 0xb5, 0x06, 0x06, 0x06, 0xa5, 0x00, 0x00, 0x03, 0x03, 0xe8, 0x00, 0x01, 0xd4, 0xc0, 0x8f, 0x4a, 0xa0,
+    0x00, 0x00, 0x00, 0x01, 0x68, 0xee, 0x3c, 0xb0,
+    0x00, 0x00, 0x00, 0x01, 0x65, 0xb8, 0x02, 0x01, 0x67, 0x25, 0x1b, 0xf4, 0xfa, 0x7d, 0x40, 0x1a, 0x78, 0xe5, 0x10, 0x52, 0xc2, 0xee, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0xc6, 0xef, 0xbb, 0x81, 0x85, 0x2d, 0x47, 0xda, 0xca, 0x4c, 0x00, 0x00, 0x03, 0x00, 0x02, 0x7b, 0xcf, 0x80, 0x00, 0x45, 0x40, 0x01, 0x8d, 0xa6, 0x00, 0x01, 0x64, 0x00, 0x0e, 0x03, 0xc8, 0x00, 0x0e, 0x10, 0x00, 0xbd, 0xc5, 0x00, 0x01, 0x11, 0x00, 0x0e, 0xa3, 0x80, 0x00, 0x38, 0xa0,
+    0x00, 0x00, 0x01, 0x65, 0x00, 0x6e, 0x2e, 0x00, 0x83, 0x7f, 0xb4, 0x8e, 0x79, 0xa5, 0xff, 0x84, 0x3f, 0x7f, 0x34, 0x3f, 0x97, 0x1b, 0xaf, 0x31, 0x5f, 0x6e, 0xaa, 0xb6, 0xac, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x78, 0x36, 0x9d, 0xa4, 0x65, 0xf9, 0x1e, 0x5b, 0x3a, 0xb0, 0x40, 0x00, 0x00, 0x03, 0x00, 0x00, 0x41, 0x80, 0x00, 0xc5, 0xc8, 0x00, 0x00, 0xfa, 0x00, 0x03, 0x24, 0x00, 0x0e, 0x20, 0x00, 0x3f, 0x80, 0x01, 0x32, 0x00, 0x08, 0x68, 0x00, 0x3e, 0xc0, 0x03, 0x8a,
+    0x00, 0x00, 0x01, 0x65, 0x00, 0x37, 0x0b, 0x80, 0x21, 0x7f, 0xe3, 0x85, 0x1c, 0xd9, 0xff, 0xe1, 0x1b, 0x01, 0xfa, 0xc0, 0x3e, 0x11, 0x7e, 0xfe, 0x45, 0x5c, 0x43, 0x69, 0x31, 0x4b, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x02, 0x24, 0xae, 0x1a, 0xbb, 0xae, 0x75, 0x9e, 0x35, 0xae, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x03, 0x64, 0x00, 0x09, 0x98, 0x00, 0x1e, 0xc0, 0x00, 0x64, 0x80, 0x01, 0xc4, 0x00, 0x07, 0xf0, 0x00, 0x42, 0xf0, 0x00, 0x00, 0xe1, 0x98, 0x00, 0x05, 0x4b, 0x28, 0x00, 0x06, 0x04,
+    0x00, 0x00, 0x01, 0x65, 0x00, 0x14, 0xa2, 0xe0, 0x08, 0x5f, 0xe3, 0x85, 0x1c, 0xd9, 0xff, 0xe1, 0x1b, 0x01, 0xfa, 0xc0, 0x3e, 0x11, 0x7e, 0xfe, 0x45, 0x5c, 0x43, 0x69, 0x31, 0x4b, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x02, 0x24, 0xb9, 0x7d, 0xb4, 0x70, 0x4d, 0x15, 0xc5, 0x0a, 0x4e, 0x60, 0x00, 0x00, 0x03, 0x00, 0x00, 0x26, 0xa8, 0xb0, 0x00, 0x13, 0xcd, 0xcc, 0x00, 0x07, 0x48, 0x88, 0x00, 0x06, 0x29, 0x69, 0x00, 0x01, 0x16, 0xac, 0x80, 0x00, 0x9e, 0x4e, 0x80, 0x00, 0x3f, 0x84, 0x20, 0x00, 0x6f, 0x41, 0xa0, 0x00, 0x20, 0x00, 0x02, 0x16, 0xb8, 0x00, 0x08, 0x08
+};
+
+// 720p 60 FPS HEVC with 1 reference frame
+const uint8_t FFmpegVideoDecoder::k_HEVCTestFrame[] = {
+    0x00, 0x00, 0x00, 0x01, 0x40, 0x01, 0x0c, 0x01, 0xff, 0xff, 0x01, 0x40, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x78, 0xac, 0x09,
+    0x00, 0x00, 0x00, 0x01, 0x42, 0x01, 0x01, 0x01, 0x40, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x78, 0xa0, 0x02, 0x80, 0x80, 0x2e, 0x1f, 0x13, 0x96, 0xb4, 0xa4, 0x25, 0x92, 0xe3, 0x01, 0x6a, 0x0c, 0x0c, 0x0d, 0x48, 0x20, 0x00, 0x00, 0x03, 0x00, 0x20, 0x00, 0x00, 0x07, 0x85, 0xf1, 0xa2, 0xd0,
+    0x00, 0x00, 0x00, 0x01, 0x44, 0x01, 0xc0, 0xf7, 0xc0, 0xcc, 0x90,
+    0x00, 0x00, 0x00, 0x01, 0x2a, 0x01, 0xac, 0x0c, 0xf0, 0xe6, 0xf6, 0x5e, 0xff, 0xd9, 0x97, 0xeb, 0x44, 0x7d, 0x1f, 0xa4, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x03, 0x52, 0x7c, 0x4e, 0x5b, 0xfe, 0xbe, 0xc2, 0x87, 0x20, 0x00, 0x00, 0x03, 0x00, 0x00, 0x03, 0x00, 0x2b, 0x60,
+};

libs/mac/include/opus.h

@@ -0,0 +1,981 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus.h
+ * @brief Opus reference implementation API
+ */
+
+#ifndef OPUS_H
+#define OPUS_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * @mainpage Opus
+ *
+ * The Opus codec is designed for interactive speech and audio transmission over the Internet.
+ * It is designed by the IETF Codec Working Group and incorporates technology from
+ * Skype's SILK codec and Xiph.Org's CELT codec.
+ *
+ * The Opus codec is designed to handle a wide range of interactive audio applications,
+ * including Voice over IP, videoconferencing, in-game chat, and even remote live music
+ * performances. It can scale from low bit-rate narrowband speech to very high quality
+ * stereo music. Its main features are:
+
+ * @li Sampling rates from 8 to 48 kHz
+ * @li Bit-rates from 6 kb/s to 510 kb/s
+ * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
+ * @li Audio bandwidth from narrowband to full-band
+ * @li Support for speech and music
+ * @li Support for mono and stereo
+ * @li Support for multichannel (up to 255 channels)
+ * @li Frame sizes from 2.5 ms to 60 ms
+ * @li Good loss robustness and packet loss concealment (PLC)
+ * @li Floating point and fixed-point implementation
+ *
+ * Documentation sections:
+ * @li @ref opus_encoder
+ * @li @ref opus_decoder
+ * @li @ref opus_repacketizer
+ * @li @ref opus_multistream
+ * @li @ref opus_libinfo
+ * @li @ref opus_custom
+ */
+
+/** @defgroup opus_encoder Opus Encoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to encode Opus.
+  *
+  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
+  * state. This can be done with:
+  *
+  * @code
+  * int          error;
+  * OpusEncoder *enc;
+  * enc = opus_encoder_create(Fs, channels, application, &error);
+  * @endcode
+  *
+  * From this point, @c enc can be used for encoding an audio stream. An encoder state
+  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
+  * state @b must @b not be re-initialized for each frame.
+  *
+  * While opus_encoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  *
+  * @code
+  * int          size;
+  * int          error;
+  * OpusEncoder *enc;
+  * size = opus_encoder_get_size(channels);
+  * enc = malloc(size);
+  * error = opus_encoder_init(enc, Fs, channels, application);
+  * @endcode
+  *
+  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The encoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
+  * interface. All these settings already default to the recommended value, so they should
+  * only be changed when necessary. The most common settings one may want to change are:
+  *
+  * @code
+  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
+  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
+  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
+  * @endcode
+  *
+  * where
+  *
+  * @arg bitrate is in bits per second (b/s)
+  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
+  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
+  *
+  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
+  *
+  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
+  * @code
+  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
+  * @endcode
+  *
+  * where
+  * <ul>
+  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
+  * <li>frame_size is the duration of the frame in samples (per channel)</li>
+  * <li>packet is the byte array to which the compressed data is written</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
+  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
+  * </ul>
+  *
+  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
+  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
+  * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
+  *
+  * Once the encoder state if no longer needed, it can be destroyed with
+  *
+  * @code
+  * opus_encoder_destroy(enc);
+  * @endcode
+  *
+  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
+  * then no action is required aside from potentially freeing the memory that was manually
+  * allocated for it (calling free(enc) for the example above)
+  *
+  */
+
+/** Opus encoder state.
+  * This contains the complete state of an Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_encoder_create,opus_encoder_init
+  */
+typedef struct OpusEncoder OpusEncoder;
+
+/** Gets the size of an <code>OpusEncoder</code> structure.
+  * @param[in] channels <tt>int</tt>: Number of channels.
+  *                                   This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
+
+/**
+ */
+
+/** Allocates and initializes an encoder state.
+ * There are three coding modes:
+ *
+ * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
+ *    signals. It enhances the  input signal by high-pass filtering and
+ *    emphasizing formants and harmonics. Optionally  it includes in-band
+ *    forward error correction to protect against packet loss. Use this
+ *    mode for typical VoIP applications. Because of the enhancement,
+ *    even at high bitrates the output may sound different from the input.
+ *
+ * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
+ *    non-voice signals like music. Use this mode for music and mixed
+ *    (music/voice) content, broadcast, and applications requiring less
+ *    than 15 ms of coding delay.
+ *
+ * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
+ *    disables the speech-optimized mode in exchange for slightly reduced delay.
+ *    This mode can only be set on an newly initialized or freshly reset encoder
+ *    because it changes the codec delay.
+ *
+ * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
+ * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                     This must be one of 8000, 12000, 16000,
+ *                                     24000, or 48000.
+ * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+ * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
+ * @note Regardless of the sampling rate and number channels selected, the Opus encoder
+ * can switch to a lower audio bandwidth or number of channels if the bitrate
+ * selected is too low. This also means that it is safe to always use 48 kHz stereo input
+ * and let the encoder optimize the encoding.
+ */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int application,
+    int *error
+);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_encoder_create(),opus_encoder_get_size()
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                      This must be one of 8000, 12000, 16000,
+ *                                      24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_encoder_init(
+    OpusEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int application
+) OPUS_ARG_NONNULL(1);
+
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
+    OpusEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes an Opus frame from floating point input.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range.
+  *          length is frame_size*channels*sizeof(float)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
+    OpusEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
+  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
+
+/** Perform a CTL function on an Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusEncoder*</tt>: Encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_encoderctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  */
+OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+/**@}*/
+
+/** @defgroup opus_decoder Opus Decoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to decode Opus.
+  *
+  * The decoding process also starts with creating a decoder
+  * state. This can be done with:
+  * @code
+  * int          error;
+  * OpusDecoder *dec;
+  * dec = opus_decoder_create(Fs, channels, &error);
+  * @endcode
+  * where
+  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
+  * @li channels is the number of channels (1 or 2)
+  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
+  * @li the return value is a newly created decoder state to be used for decoding
+  *
+  * While opus_decoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  * @code
+  * int          size;
+  * int          error;
+  * OpusDecoder *dec;
+  * size = opus_decoder_get_size(channels);
+  * dec = malloc(size);
+  * error = opus_decoder_init(dec, Fs, channels);
+  * @endcode
+  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The decoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
+  * @code
+  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
+  * @endcode
+  * where
+  *
+  * @li packet is the byte array containing the compressed data
+  * @li len is the exact number of bytes contained in the packet
+  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
+  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
+  *
+  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
+  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
+  * buffer is too small to hold the decoded audio.
+  *
+  * Opus is a stateful codec with overlapping blocks and as a result Opus
+  * packets are not coded independently of each other. Packets must be
+  * passed into the decoder serially and in the correct order for a correct
+  * decode. Lost packets can be replaced with loss concealment by calling
+  * the decoder with a null pointer and zero length for the missing packet.
+  *
+  * A single codec state may only be accessed from a single thread at
+  * a time and any required locking must be performed by the caller. Separate
+  * streams must be decoded with separate decoder states and can be decoded
+  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
+  * defined.
+  *
+  */
+
+/** Opus decoder state.
+  * This contains the complete state of an Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_decoder_create,opus_decoder_init
+  */
+typedef struct OpusDecoder OpusDecoder;
+
+/** Gets the size of an <code>OpusDecoder</code> structure.
+  * @param [in] channels <tt>int</tt>: Number of channels.
+  *                                    This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
+
+/** Allocates and initializes a decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
+  *
+  * Internally Opus stores data at 48000 Hz, so that should be the default
+  * value for Fs. However, the decoder can efficiently decode to buffers
+  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
+  * data at the full sample rate, or knows the compressed data doesn't
+  * use the full frequency range, it can request decoding at a reduced
+  * rate. Likewise, the decoder is capable of filling in either mono or
+  * interleaved stereo pcm buffers, at the caller's request.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int *error
+);
+
+/** Initializes a previously allocated decoder state.
+  * The state must be at least the size returned by opus_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_init(
+    OpusDecoder *st,
+    opus_int32 Fs,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an Opus packet with floating point output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_decoderctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  */
+OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
+  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
+
+/** Parse an opus packet into one or more frames.
+  * Opus_decode will perform this operation internally so most applications do
+  * not need to use this function.
+  * This function does not copy the frames, the returned pointers are pointers into
+  * the input packet.
+  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
+  * @param [in] len <tt>opus_int32</tt>: size of data
+  * @param [out] out_toc <tt>char*</tt>: TOC pointer
+  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
+  * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
+  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
+  * @returns number of frames
+  */
+OPUS_EXPORT int opus_packet_parse(
+   const unsigned char *data,
+   opus_int32 len,
+   unsigned char *out_toc,
+   const unsigned char *frames[48],
+   opus_int16 size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Gets the bandwidth of an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
+  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
+  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
+  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
+  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples per frame from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet.
+  *                                  This must contain at least one byte of
+  *                                  data.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples per frame.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of channels from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @returns Number of channels
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of frames in an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of frames
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
+/**@}*/
+
+/** @defgroup opus_repacketizer Repacketizer
+  * @{
+  *
+  * The repacketizer can be used to merge multiple Opus packets into a single
+  * packet or alternatively to split Opus packets that have previously been
+  * merged. Splitting valid Opus packets is always guaranteed to succeed,
+  * whereas merging valid packets only succeeds if all frames have the same
+  * mode, bandwidth, and frame size, and when the total duration of the merged
+  * packet is no more than 120 ms. The 120 ms limit comes from the
+  * specification and limits decoder memory requirements at a point where
+  * framing overhead becomes negligible.
+  *
+  * The repacketizer currently only operates on elementary Opus
+  * streams. It will not manipualte multistream packets successfully, except in
+  * the degenerate case where they consist of data from a single stream.
+  *
+  * The repacketizing process starts with creating a repacketizer state, either
+  * by calling opus_repacketizer_create() or by allocating the memory yourself,
+  * e.g.,
+  * @code
+  * OpusRepacketizer *rp;
+  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
+  * if (rp != NULL)
+  *     opus_repacketizer_init(rp);
+  * @endcode
+  *
+  * Then the application should submit packets with opus_repacketizer_cat(),
+  * extract new packets with opus_repacketizer_out() or
+  * opus_repacketizer_out_range(), and then reset the state for the next set of
+  * input packets via opus_repacketizer_init().
+  *
+  * For example, to split a sequence of packets into individual frames:
+  * @code
+  * unsigned char *data;
+  * int len;
+  * while (get_next_packet(&data, &len))
+  * {
+  *   unsigned char out[1276];
+  *   opus_int32 out_len;
+  *   int nb_frames;
+  *   int err;
+  *   int i;
+  *   err = opus_repacketizer_cat(rp, data, len);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packet(data);
+  *     return err;
+  *   }
+  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
+  *   for (i = 0; i < nb_frames; i++)
+  *   {
+  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packet(data);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *   }
+  *   opus_repacketizer_init(rp);
+  *   release_packet(data);
+  * }
+  * @endcode
+  *
+  * Alternatively, to combine a sequence of frames into packets that each
+  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
+  * @code
+  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
+  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
+  * // packets.
+  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
+  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
+  * int nb_packets;
+  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
+  * opus_int32 out_len;
+  * int prev_toc;
+  * nb_packets = 0;
+  * while (get_next_packet(data+nb_packets, len+nb_packets))
+  * {
+  *   int nb_frames;
+  *   int err;
+  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
+  *   if (nb_frames < 1)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return nb_frames;
+  *   }
+  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
+  *   // If adding the next packet would exceed our target, or it has an
+  *   // incompatible TOC sequence, output the packets we already have before
+  *   // submitting it.
+  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
+  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
+  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
+  *   // output an (invalid) empty packet. It also ensures that prev_toc has
+  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
+  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
+  *   // reference to data[nb_packets][0] should be valid.
+  *   if (nb_packets > 0 && (
+  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
+  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
+  *       TARGET_DURATION_MS*48))
+  *   {
+  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packets(data, nb_packets+1);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *     opus_repacketizer_init(rp);
+  *     release_packets(data, nb_packets);
+  *     data[0] = data[nb_packets];
+  *     len[0] = len[nb_packets];
+  *     nb_packets = 0;
+  *   }
+  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return err;
+  *   }
+  *   prev_toc = data[nb_packets][0];
+  *   nb_packets++;
+  * }
+  * // Output the final, partial packet.
+  * if (nb_packets > 0)
+  * {
+  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *   release_packets(data, nb_packets);
+  *   if (out_len < 0)
+  *     return (int)out_len;
+  *   output_next_packet(out, out_len);
+  * }
+  * @endcode
+  *
+  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
+  * unconditionally until it fails. At that point, the merged packet can be
+  * obtained with opus_repacketizer_out() and the input packet for which
+  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
+  * repacketizer state.
+  */
+
+typedef struct OpusRepacketizer OpusRepacketizer;
+
+/** Gets the size of an <code>OpusRepacketizer</code> structure.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
+
+/** (Re)initializes a previously allocated repacketizer state.
+  * The state must be at least the size returned by opus_repacketizer_get_size().
+  * This can be used for applications which use their own allocator instead of
+  * malloc().
+  * It must also be called to reset the queue of packets waiting to be
+  * repacketized, which is necessary if the maximum packet duration of 120 ms
+  * is reached or if you wish to submit packets with a different Opus
+  * configuration (coding mode, audio bandwidth, frame size, or channel count).
+  * Failure to do so will prevent a new packet from being added with
+  * opus_repacketizer_cat().
+  * @see opus_repacketizer_create
+  * @see opus_repacketizer_get_size
+  * @see opus_repacketizer_cat
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
+  *                                       (re)initialize.
+  * @returns A pointer to the same repacketizer state that was passed in.
+  */
+OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Allocates memory and initializes the new repacketizer with
+ * opus_repacketizer_init().
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
+
+/** Frees an <code>OpusRepacketizer</code> allocated by
+  * opus_repacketizer_create().
+  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
+
+/** Add a packet to the current repacketizer state.
+  * This packet must match the configuration of any packets already submitted
+  * for repacketization since the last call to opus_repacketizer_init().
+  * This means that it must have the same coding mode, audio bandwidth, frame
+  * size, and channel count.
+  * This can be checked in advance by examining the top 6 bits of the first
+  * byte of the packet, and ensuring they match the top 6 bits of the first
+  * byte of any previously submitted packet.
+  * The total duration of audio in the repacketizer state also must not exceed
+  * 120 ms, the maximum duration of a single packet, after adding this packet.
+  *
+  * The contents of the current repacketizer state can be extracted into new
+  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
+  *
+  * In order to add a packet with a different configuration or to add more
+  * audio beyond 120 ms, you must clear the repacketizer state by calling
+  * opus_repacketizer_init().
+  * If a packet is too large to add to the current repacketizer state, no part
+  * of it is added, even if it contains multiple frames, some of which might
+  * fit.
+  * If you wish to be able to add parts of such packets, you should first use
+  * another repacketizer to split the packet into pieces and add them
+  * individually.
+  * @see opus_repacketizer_out_range
+  * @see opus_repacketizer_out
+  * @see opus_repacketizer_init
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
+  *                                       add the packet.
+  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
+  *                                                The application must ensure
+  *                                                this pointer remains valid
+  *                                                until the next call to
+  *                                                opus_repacketizer_init() or
+  *                                                opus_repacketizer_destroy().
+  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
+  * @returns An error code indicating whether or not the operation succeeded.
+  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
+  *                  state.
+  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
+  *                              the packet's TOC sequence was not compatible
+  *                              with previously submitted packets (because
+  *                              the coding mode, audio bandwidth, frame size,
+  *                              or channel count did not match), or adding
+  *                              this packet would increase the total amount of
+  *                              audio stored in the repacketizer state to more
+  *                              than 120 ms.
+  */
+OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param begin <tt>int</tt>: The index of the first frame in the current
+  *                            repacketizer state to include in the output.
+  * @param end <tt>int</tt>: One past the index of the last frame in the
+  *                          current repacketizer state to include in the
+  *                          output.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1276</code> for a single frame,
+  *                                    or for multiple frames,
+  *                                    <code>1277*(end-begin)</code>.
+  *                                    However, <code>1*(end-begin)</code> plus
+  *                                    the size of all packet data submitted to
+  *                                    the repacketizer since the last call to
+  *                                    opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
+  *                       frames (begin < 0, begin >= end, or end >
+  *                       opus_repacketizer_get_nb_frames()).
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Return the total number of frames contained in packet data submitted to
+  * the repacketizer state so far via opus_repacketizer_cat() since the last
+  * call to opus_repacketizer_init() or opus_repacketizer_create().
+  * This defines the valid range of packets that can be extracted with
+  * opus_repacketizer_out_range() or opus_repacketizer_out().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
+  *                                       frames.
+  * @returns The total number of frames contained in the packet data submitted
+  *          to the repacketizer state.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * This is a convenience routine that returns all the data submitted so far
+  * in a single packet.
+  * It is equivalent to calling
+  * @code
+  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
+  *                             data, maxlen)
+  * @endcode
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
+  *                                    However,
+  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
+  *                                    plus the size of all packet data
+  *                                    submitted to the repacketizer since the
+  *                                    last call to opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
+
+/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
+
+/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
+
+/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */

libs/mac/include/opus_custom.h

@@ -0,0 +1,342 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008-2012 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+  @file opus_custom.h
+  @brief Opus-Custom reference implementation API
+ */
+
+#ifndef OPUS_CUSTOM_H
+#define OPUS_CUSTOM_H
+
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef CUSTOM_MODES
+# define OPUS_CUSTOM_EXPORT OPUS_EXPORT
+# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT
+#else
+# define OPUS_CUSTOM_EXPORT
+# ifdef OPUS_BUILD
+#  define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE
+# else
+#  define OPUS_CUSTOM_EXPORT_STATIC
+# endif
+#endif
+
+/** @defgroup opus_custom Opus Custom
+  * @{
+  *  Opus Custom is an optional part of the Opus specification and
+  * reference implementation which uses a distinct API from the regular
+  * API and supports frame sizes that are not normally supported.\ Use
+  * of Opus Custom is discouraged for all but very special applications
+  * for which a frame size different from 2.5, 5, 10, or 20 ms is needed
+  * (for either complexity or latency reasons) and where interoperability
+  * is less important.
+  *
+  * In addition to the interoperability limitations the use of Opus custom
+  * disables a substantial chunk of the codec and generally lowers the
+  * quality available at a given bitrate. Normally when an application needs
+  * a different frame size from the codec it should buffer to match the
+  * sizes but this adds a small amount of delay which may be important
+  * in some very low latency applications. Some transports (especially
+  * constant rate RF transports) may also work best with frames of
+  * particular durations.
+  *
+  * Libopus only supports custom modes if they are enabled at compile time.
+  *
+  * The Opus Custom API is similar to the regular API but the
+  * @ref opus_encoder_create and @ref opus_decoder_create calls take
+  * an additional mode parameter which is a structure produced by
+  * a call to @ref opus_custom_mode_create. Both the encoder and decoder
+  * must create a mode using the same sample rate (fs) and frame size
+  * (frame size) so these parameters must either be signaled out of band
+  * or fixed in a particular implementation.
+  *
+  * Similar to regular Opus the custom modes support on the fly frame size
+  * switching, but the sizes available depend on the particular frame size in
+  * use. For some initial frame sizes on a single on the fly size is available.
+  */
+
+/** Contains the state of an encoder. One encoder state is needed
+    for each stream. It is initialized once at the beginning of the
+    stream. Do *not* re-initialize the state for every frame.
+   @brief Encoder state
+ */
+typedef struct OpusCustomEncoder OpusCustomEncoder;
+
+/** State of the decoder. One decoder state is needed for each stream.
+    It is initialized once at the beginning of the stream. Do *not*
+    re-initialize the state for every frame.
+   @brief Decoder state
+ */
+typedef struct OpusCustomDecoder OpusCustomDecoder;
+
+/** The mode contains all the information necessary to create an
+    encoder. Both the encoder and decoder need to be initialized
+    with exactly the same mode, otherwise the output will be
+    corrupted.
+   @brief Mode configuration
+ */
+typedef struct OpusCustomMode OpusCustomMode;
+
+/** Creates a new mode struct. This will be passed to an encoder or
+  * decoder. The mode MUST NOT BE DESTROYED until the encoders and
+  * decoders that use it are destroyed as well.
+  * @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz)
+  * @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each
+  *        packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes)
+  * @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned)
+  * @return A newly created mode
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error);
+
+/** Destroys a mode struct. Only call this after all encoders and
+  * decoders using this mode are destroyed as well.
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C)
+
+/* Encoder */
+/** Gets the size of an OpusCustomEncoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+# ifdef CUSTOM_MODES
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_encoder_create(),opus_custom_encoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_init(
+    OpusCustomEncoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+# endif
+#endif
+
+
+/** Creates a new encoder state. Each stream needs its own encoder
+  * state (can't be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created encoder state.
+*/
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+
+/** Destroys a an encoder state.
+  * @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range. There must be exactly
+  *          frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float(
+    OpusCustomEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian).
+  *          There must be exactly frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+ */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
+    OpusCustomEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_encoderctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C)
+/* Decoder */
+
+/** Gets the size of an OpusCustomDecoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Initializes a previously allocated decoder state
+  * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_decoder_create(),opus_custom_decoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init(
+    OpusCustomDecoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+#endif
+
+
+/** Creates a new decoder state. Each stream needs its own decoder state (can't
+  * be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the
+  *          stream (must be the same characteristics as used for the encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created decoder state.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+/** Destroys a an decoder state.
+  * @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st);
+
+/** Decode an opus custom frame with floating point output
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    float *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an opus custom frame
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    opus_int16 *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_genericctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_CUSTOM_H */

libs/mac/include/opus_defines.h

@@ -0,0 +1,785 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_defines.h
+ * @brief Opus reference implementation constants
+ */
+
+#ifndef OPUS_DEFINES_H
+#define OPUS_DEFINES_H
+
+#include "opus_types.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @defgroup opus_errorcodes Error codes
+ * @{
+ */
+/** No error @hideinitializer*/
+#define OPUS_OK                0
+/** One or more invalid/out of range arguments @hideinitializer*/
+#define OPUS_BAD_ARG          -1
+/** Not enough bytes allocated in the buffer @hideinitializer*/
+#define OPUS_BUFFER_TOO_SMALL -2
+/** An internal error was detected @hideinitializer*/
+#define OPUS_INTERNAL_ERROR   -3
+/** The compressed data passed is corrupted @hideinitializer*/
+#define OPUS_INVALID_PACKET   -4
+/** Invalid/unsupported request number @hideinitializer*/
+#define OPUS_UNIMPLEMENTED    -5
+/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
+#define OPUS_INVALID_STATE    -6
+/** Memory allocation has failed @hideinitializer*/
+#define OPUS_ALLOC_FAIL       -7
+/**@}*/
+
+/** @cond OPUS_INTERNAL_DOC */
+/**Export control for opus functions */
+
+#ifndef OPUS_EXPORT
+# if defined(WIN32)
+#  if defined(OPUS_BUILD) && defined(DLL_EXPORT)
+#   define OPUS_EXPORT __declspec(dllexport)
+#  else
+#   define OPUS_EXPORT
+#  endif
+# elif defined(__GNUC__) && defined(OPUS_BUILD)
+#  define OPUS_EXPORT __attribute__ ((visibility ("default")))
+# else
+#  define OPUS_EXPORT
+# endif
+#endif
+
+# if !defined(OPUS_GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define OPUS_GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define OPUS_GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(3,0)
+#  define OPUS_RESTRICT __restrict__
+# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
+#  define OPUS_RESTRICT __restrict
+# else
+#  define OPUS_RESTRICT
+# endif
+#else
+# define OPUS_RESTRICT restrict
+#endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(2,7)
+#  define OPUS_INLINE __inline__
+# elif (defined(_MSC_VER))
+#  define OPUS_INLINE __inline
+# else
+#  define OPUS_INLINE
+# endif
+#else
+# define OPUS_INLINE inline
+#endif
+
+/**Warning attributes for opus functions
+  * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
+  * some paranoid null checks. */
+#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
+#else
+# define OPUS_WARN_UNUSED_RESULT
+#endif
+#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_ARG_NONNULL(_x)  __attribute__ ((__nonnull__(_x)))
+#else
+# define OPUS_ARG_NONNULL(_x)
+#endif
+
+/** These are the actual Encoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+#define OPUS_SET_APPLICATION_REQUEST         4000
+#define OPUS_GET_APPLICATION_REQUEST         4001
+#define OPUS_SET_BITRATE_REQUEST             4002
+#define OPUS_GET_BITRATE_REQUEST             4003
+#define OPUS_SET_MAX_BANDWIDTH_REQUEST       4004
+#define OPUS_GET_MAX_BANDWIDTH_REQUEST       4005
+#define OPUS_SET_VBR_REQUEST                 4006
+#define OPUS_GET_VBR_REQUEST                 4007
+#define OPUS_SET_BANDWIDTH_REQUEST           4008
+#define OPUS_GET_BANDWIDTH_REQUEST           4009
+#define OPUS_SET_COMPLEXITY_REQUEST          4010
+#define OPUS_GET_COMPLEXITY_REQUEST          4011
+#define OPUS_SET_INBAND_FEC_REQUEST          4012
+#define OPUS_GET_INBAND_FEC_REQUEST          4013
+#define OPUS_SET_PACKET_LOSS_PERC_REQUEST    4014
+#define OPUS_GET_PACKET_LOSS_PERC_REQUEST    4015
+#define OPUS_SET_DTX_REQUEST                 4016
+#define OPUS_GET_DTX_REQUEST                 4017
+#define OPUS_SET_VBR_CONSTRAINT_REQUEST      4020
+#define OPUS_GET_VBR_CONSTRAINT_REQUEST      4021
+#define OPUS_SET_FORCE_CHANNELS_REQUEST      4022
+#define OPUS_GET_FORCE_CHANNELS_REQUEST      4023
+#define OPUS_SET_SIGNAL_REQUEST              4024
+#define OPUS_GET_SIGNAL_REQUEST              4025
+#define OPUS_GET_LOOKAHEAD_REQUEST           4027
+/* #define OPUS_RESET_STATE 4028 */
+#define OPUS_GET_SAMPLE_RATE_REQUEST         4029
+#define OPUS_GET_FINAL_RANGE_REQUEST         4031
+#define OPUS_GET_PITCH_REQUEST               4033
+#define OPUS_SET_GAIN_REQUEST                4034
+#define OPUS_GET_GAIN_REQUEST                4045 /* Should have been 4035 */
+#define OPUS_SET_LSB_DEPTH_REQUEST           4036
+#define OPUS_GET_LSB_DEPTH_REQUEST           4037
+#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
+#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
+#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
+#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
+#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
+/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
+#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
+#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
+
+/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
+#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
+#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
+#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
+#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
+/** @endcond */
+
+/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
+  * @see opus_genericctls, opus_encoderctls
+  * @{
+  */
+/* Values for the various encoder CTLs */
+#define OPUS_AUTO                           -1000 /**<Auto/default setting @hideinitializer*/
+#define OPUS_BITRATE_MAX                       -1 /**<Maximum bitrate @hideinitializer*/
+
+/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
+ * @hideinitializer */
+#define OPUS_APPLICATION_VOIP                2048
+/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
+ * @hideinitializer */
+#define OPUS_APPLICATION_AUDIO               2049
+/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
+ * @hideinitializer */
+#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
+
+#define OPUS_SIGNAL_VOICE                    3001 /**< Signal being encoded is voice */
+#define OPUS_SIGNAL_MUSIC                    3002 /**< Signal being encoded is music */
+#define OPUS_BANDWIDTH_NARROWBAND            1101 /**< 4 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_MEDIUMBAND            1102 /**< 6 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_WIDEBAND              1103 /**< 8 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_SUPERWIDEBAND         1104 /**<12 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_FULLBAND              1105 /**<20 kHz bandpass @hideinitializer*/
+
+#define OPUS_FRAMESIZE_ARG                   5000 /**< Select frame size from the argument (default) */
+#define OPUS_FRAMESIZE_2_5_MS                5001 /**< Use 2.5 ms frames */
+#define OPUS_FRAMESIZE_5_MS                  5002 /**< Use 5 ms frames */
+#define OPUS_FRAMESIZE_10_MS                 5003 /**< Use 10 ms frames */
+#define OPUS_FRAMESIZE_20_MS                 5004 /**< Use 20 ms frames */
+#define OPUS_FRAMESIZE_40_MS                 5005 /**< Use 40 ms frames */
+#define OPUS_FRAMESIZE_60_MS                 5006 /**< Use 60 ms frames */
+#define OPUS_FRAMESIZE_80_MS                 5007 /**< Use 80 ms frames */
+#define OPUS_FRAMESIZE_100_MS                5008 /**< Use 100 ms frames */
+#define OPUS_FRAMESIZE_120_MS                5009 /**< Use 120 ms frames */
+
+/**@}*/
+
+
+/** @defgroup opus_encoderctls Encoder related CTLs
+  *
+  * These are convenience macros for use with the \c opus_encode_ctl
+  * interface. They are used to generate the appropriate series of
+  * arguments for that call, passing the correct type, size and so
+  * on as expected for each particular request.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
+  * if (ret != OPUS_OK) return ret;
+  *
+  * opus_int32 rate;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * @endcode
+  *
+  * @see opus_genericctls, opus_encoder
+  * @{
+  */
+
+/** Configures the encoder's computational complexity.
+  * The supported range is 0-10 inclusive with 10 representing the highest complexity.
+  * @see OPUS_GET_COMPLEXITY
+  * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
+  *
+  * @hideinitializer */
+#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
+/** Gets the encoder's complexity configuration.
+  * @see OPUS_SET_COMPLEXITY
+  * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
+  *                                      inclusive.
+  * @hideinitializer */
+#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the bitrate in the encoder.
+  * Rates from 500 to 512000 bits per second are meaningful, as well as the
+  * special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
+  * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
+  * rate as it can, which is useful for controlling the rate by adjusting the
+  * output buffer size.
+  * @see OPUS_GET_BITRATE
+  * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
+  *                                   is determined based on the number of
+  *                                   channels and the input sampling rate.
+  * @hideinitializer */
+#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
+/** Gets the encoder's bitrate configuration.
+  * @see OPUS_SET_BITRATE
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
+  *                                      The default is determined based on the
+  *                                      number of channels and the input
+  *                                      sampling rate.
+  * @hideinitializer */
+#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables variable bitrate (VBR) in the encoder.
+  * The configured bitrate may not be met exactly because frames must
+  * be an integer number of bytes in length.
+  * @see OPUS_GET_VBR
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
+  *               cause noticeable quality degradation.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
+  *               #OPUS_SET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
+/** Determine if variable bitrate (VBR) is enabled in the encoder.
+  * @see OPUS_SET_VBR
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
+  *               #OPUS_GET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables constrained VBR in the encoder.
+  * This setting is ignored when the encoder is in CBR mode.
+  * @warning Only the MDCT mode of Opus currently heeds the constraint.
+  *  Speech mode ignores it completely, hybrid mode may fail to obey it
+  *  if the LPC layer uses more bitrate than the constraint would have
+  *  permitted.
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @see OPUS_SET_VBR
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
+  *               frame of buffering delay assuming a transport with a
+  *               serialization speed of the nominal bitrate.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
+/** Determine if constrained VBR is enabled in the encoder.
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @see OPUS_GET_VBR
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default).</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures mono/stereo forcing in the encoder.
+  * This can force the encoder to produce packets encoded as either mono or
+  * stereo, regardless of the format of the input audio. This is useful when
+  * the caller knows that the input signal is currently a mono source embedded
+  * in a stereo stream.
+  * @see OPUS_GET_FORCE_CHANNELS
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
+/** Gets the encoder's forced channel configuration.
+  * @see OPUS_SET_FORCE_CHANNELS
+  * @param[out] x <tt>opus_int32 *</tt>:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the maximum bandpass that the encoder will select automatically.
+  * Applications should normally use this instead of #OPUS_SET_BANDWIDTH
+  * (leaving that set to the default, #OPUS_AUTO). This allows the
+  * application to set an upper bound based on the type of input it is
+  * providing, but still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_MAX_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Gets the encoder's configured maximum allowed bandpass.
+  * @see OPUS_SET_MAX_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Sets the encoder's bandpass to a specific value.
+  * This prevents the encoder from automatically selecting the bandpass based
+  * on the available bitrate. If an application knows the bandpass of the input
+  * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
+  * instead, which still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Configures the type of signal being encoded.
+  * This is a hint which helps the encoder's mode selection.
+  * @see OPUS_GET_SIGNAL
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal type.
+  * @see OPUS_SET_SIGNAL
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Configures the encoder's intended application.
+  * The initial value is a mandatory argument to the encoder_create function.
+  * @see OPUS_GET_APPLICATION
+  * @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured application.
+  * @see OPUS_SET_APPLICATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the total samples of delay added by the entire codec.
+  * This can be queried by the encoder and then the provided number of samples can be
+  * skipped on from the start of the decoder's output to provide time aligned input
+  * and output. From the perspective of a decoding application the real data begins this many
+  * samples late.
+  *
+  * The decoder contribution to this delay is identical for all decoders, but the
+  * encoder portion of the delay may vary from implementation to implementation,
+  * version to version, or even depend on the encoder's initial configuration.
+  * Applications needing delay compensation should call this CTL rather than
+  * hard-coding a value.
+  * @param[out] x <tt>opus_int32 *</tt>:   Number of lookahead samples
+  * @hideinitializer */
+#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of inband forward error correction (FEC).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_INBAND_FEC
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable inband FEC (default).</dd>
+  * <dt>1</dt><dd>Enable inband FEC.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of inband forward error correction.
+  * @see OPUS_SET_INBAND_FEC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Inband FEC disabled (default).</dd>
+  * <dt>1</dt><dd>Inband FEC enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's expected packet loss percentage.
+  * Higher values trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the absence of packet loss, but
+  * greater quality under loss.
+  * @see OPUS_GET_PACKET_LOSS_PERC
+  * @param[in] x <tt>opus_int32</tt>:   Loss percentage in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured packet loss percentage.
+  * @see OPUS_SET_PACKET_LOSS_PERC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
+  *                                      in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of discontinuous transmission (DTX).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_DTX
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable DTX (default).</dd>
+  * <dt>1</dt><dd>Enabled DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of discontinuous transmission.
+  * @see OPUS_SET_DTX
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>DTX disabled (default).</dd>
+  * <dt>1</dt><dd>DTX enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
+/** Configures the depth of signal being encoded.
+  *
+  * This is a hint which helps the encoder identify silence and near-silence.
+  * It represents the number of significant bits of linear intensity below
+  * which the signal contains ignorable quantization or other noise.
+  *
+  * For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
+  * for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
+  * for 16-bit linear pcm input with opus_encode_float().
+  *
+  * When using opus_encode() instead of opus_encode_float(), or when libopus
+  * is compiled for fixed-point, the encoder uses the minimum of the value
+  * set here and the value 16.
+  *
+  * @see OPUS_GET_LSB_DEPTH
+  * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
+  *                                   (default: 24).
+  * @hideinitializer */
+#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal depth.
+  * @see OPUS_SET_LSB_DEPTH
+  * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
+  *                                      24 (default: 24).
+  * @hideinitializer */
+#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of variable duration frames.
+  * When variable duration is enabled, the encoder is free to use a shorter frame
+  * size than the one requested in the opus_encode*() call.
+  * It is then the user's responsibility
+  * to verify how much audio was encoded by checking the ToC byte of the encoded
+  * packet. The part of the audio that was not encoded needs to be resent to the
+  * encoder for the next call. Do not use this option unless you <b>really</b>
+  * know what you are doing.
+  * @see OPUS_GET_EXPERT_FRAME_DURATION
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured use of variable duration frames.
+  * @see OPUS_SET_EXPERT_FRAME_DURATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables almost all use of prediction, making frames almost
+  * completely independent. This reduces quality.
+  * @see OPUS_GET_PREDICTION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable prediction (default).</dd>
+  * <dt>1</dt><dd>Disable prediction.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured prediction status.
+  * @see OPUS_SET_PREDICTION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Prediction enabled (default).</dd>
+  * <dt>1</dt><dd>Prediction disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_genericctls Generic CTLs
+  *
+  * These macros are used with the \c opus_decoder_ctl and
+  * \c opus_encoder_ctl calls to generate a particular
+  * request.
+  *
+  * When called on an \c OpusDecoder they apply to that
+  * particular decoder instance. When called on an
+  * \c OpusEncoder they apply to the corresponding setting
+  * on that encoder instance, if present.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * opus_int32 pitch;
+  * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
+  * if (ret == OPUS_OK) return ret;
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
+  *
+  * opus_int32 enc_bw, dec_bw;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
+  * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
+  * if (enc_bw != dec_bw) {
+  *   printf("packet bandwidth mismatch!\n");
+  * }
+  * @endcode
+  *
+  * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
+  * @{
+  */
+
+/** Resets the codec state to be equivalent to a freshly initialized state.
+  * This should be called when switching streams in order to prevent
+  * the back to back decoding from giving different results from
+  * one at a time decoding.
+  * @hideinitializer */
+#define OPUS_RESET_STATE 4028
+
+/** Gets the final state of the codec's entropy coder.
+  * This is used for testing purposes,
+  * The encoder and decoder state should be identical after coding a payload
+  * (assuming no data corruption or software bugs)
+  *
+  * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
+  *
+  * @hideinitializer */
+#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
+
+/** Gets the encoder's configured bandpass or the decoder's last bandpass.
+  * @see OPUS_SET_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the sampling rate the encoder or decoder was initialized with.
+  * This simply returns the <code>Fs</code> value passed to opus_encoder_init()
+  * or opus_decoder_init().
+  * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
+  * @hideinitializer
+  */
+#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables the use of phase inversion for intensity stereo,
+  * improving the quality of mono downmixes, but slightly reducing normal
+  * stereo quality. Disabling phase inversion in the decoder does not comply
+  * with RFC 6716, although it does not cause any interoperability issue and
+  * is expected to become part of the Opus standard once RFC 6716 is updated
+  * by draft-ietf-codec-opus-update.
+  * @see OPUS_GET_PHASE_INVERSION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable phase inversion (default).</dd>
+  * <dt>1</dt><dd>Disable phase inversion.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured phase inversion status.
+  * @see OPUS_SET_PHASE_INVERSION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
+  * <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_decoderctls Decoder related CTLs
+  * @see opus_genericctls, opus_encoderctls, opus_decoder
+  * @{
+  */
+
+/** Configures decoder gain adjustment.
+  * Scales the decoded output by a factor specified in Q8 dB units.
+  * This has a maximum range of -32768 to 32767 inclusive, and returns
+  * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
+  * This setting survives decoder reset.
+  *
+  * gain = pow(10, x/(20.0*256))
+  *
+  * @param[in] x <tt>opus_int32</tt>:   Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
+/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
+  * @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
+  * @hideinitializer */
+#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the pitch of the last decoded frame, if available.
+  * This can be used for any post-processing algorithm requiring the use of pitch,
+  * e.g. time stretching/shortening. If the last frame was not voiced, or if the
+  * pitch was not coded in the frame, then zero is returned.
+  *
+  * This CTL is only implemented for decoder instances.
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
+  *
+  * @hideinitializer */
+#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_libinfo Opus library information functions
+  * @{
+  */
+
+/** Converts an opus error code into a human readable string.
+  *
+  * @param[in] error <tt>int</tt>: Error number
+  * @returns Error string
+  */
+OPUS_EXPORT const char *opus_strerror(int error);
+
+/** Gets the libopus version string.
+  *
+  * Applications may look for the substring "-fixed" in the version string to
+  * determine whether they have a fixed-point or floating-point build at
+  * runtime.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */

libs/mac/include/opus_multistream.h

@@ -0,0 +1,660 @@
+/* Copyright (c) 2011 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_multistream.h
+ * @brief Opus reference implementation multistream API
+ */
+
+#ifndef OPUS_MULTISTREAM_H
+#define OPUS_MULTISTREAM_H
+
+#include "opus.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** Macros to trigger compilation errors when the wrong types are provided to a
+  * CTL. */
+/**@{*/
+#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
+#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
+/**@}*/
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
+#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
+/**@}*/
+
+/** @endcond */
+
+/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
+  * @ref opus_decoderctls may be applied to a multistream encoder or decoder as
+  * well.
+  * In addition, you may retrieve the encoder or decoder state for an specific
+  * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
+  * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
+  */
+/**@{*/
+
+/** Gets the encoder state for an individual stream of a multistream encoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the encoder.
+  * @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
+  *                                       encoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
+
+/** Gets the decoder state for an individual stream of a multistream decoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the decoder.
+  * @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
+  *                                       decoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
+
+/**@}*/
+
+/** @defgroup opus_multistream Opus Multistream API
+  * @{
+  *
+  * The multistream API allows individual Opus streams to be combined into a
+  * single packet, enabling support for up to 255 channels. Unlike an
+  * elementary Opus stream, the encoder and decoder must negotiate the channel
+  * configuration before the decoder can successfully interpret the data in the
+  * packets produced by the encoder. Some basic information, such as packet
+  * duration, can be computed without any special negotiation.
+  *
+  * The format for multistream Opus packets is defined in
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
+  * and is based on the self-delimited Opus framing described in Appendix B of
+  * <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * Normal Opus packets are just a degenerate case of multistream Opus packets,
+  * and can be encoded or decoded with the multistream API by setting
+  * <code>streams</code> to <code>1</code> when initializing the encoder or
+  * decoder.
+  *
+  * Multistream Opus streams can contain up to 255 elementary Opus streams.
+  * These may be either "uncoupled" or "coupled", indicating that the decoder
+  * is configured to decode them to either 1 or 2 channels, respectively.
+  * The streams are ordered so that all coupled streams appear at the
+  * beginning.
+  *
+  * A <code>mapping</code> table defines which decoded channel <code>i</code>
+  * should be used for each input/output (I/O) channel <code>j</code>. This table is
+  * typically provided as an unsigned char array.
+  * Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
+  * If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
+  * encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
+  * is even, or  as the right channel of stream <code>(i/2)</code> if
+  * <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
+  * mono in stream <code>(i - coupled_streams)</code>, unless it has the special
+  * value 255, in which case it is omitted from the encoding entirely (the
+  * decoder will reproduce it as silence). Each value <code>i</code> must either
+  * be the special value 255 or be less than <code>streams + coupled_streams</code>.
+  *
+  * The output channels specified by the encoder
+  * should use the
+  * <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
+  * channel ordering</a>. A decoder may wish to apply an additional permutation
+  * to the mapping the encoder used to achieve a different output channel
+  * order (e.g. for outputing in WAV order).
+  *
+  * Each multistream packet contains an Opus packet for each stream, and all of
+  * the Opus packets in a single multistream packet must have the same
+  * duration. Therefore the duration of a multistream packet can be extracted
+  * from the TOC sequence of the first stream, which is located at the
+  * beginning of the packet, just like an elementary Opus stream:
+  *
+  * @code
+  * int nb_samples;
+  * int nb_frames;
+  * nb_frames = opus_packet_get_nb_frames(data, len);
+  * if (nb_frames < 1)
+  *   return nb_frames;
+  * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
+  * @endcode
+  *
+  * The general encoding and decoding process proceeds exactly the same as in
+  * the normal @ref opus_encoder and @ref opus_decoder APIs.
+  * See their documentation for an overview of how to use the corresponding
+  * multistream functions.
+  */
+
+/** Opus multistream encoder state.
+  * This contains the complete state of a multistream Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_init
+  */
+typedef struct OpusMSEncoder OpusMSEncoder;
+
+/** Opus multistream decoder state.
+  * This contains the complete state of a multistream Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_decoder_init
+  */
+typedef struct OpusMSDecoder OpusMSDecoder;
+
+/**\name Multistream encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusMSEncoder structure.
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
+      int channels,
+      int mapping_family
+);
+
+
+/** Allocates and initializes a multistream encoder state.
+  * Call opus_multistream_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+/** Initialize a previously allocated multistream encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+OPUS_EXPORT int opus_multistream_surround_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7);
+
+/** Encodes a multistream Opus frame.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
+    OpusMSEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a multistream Opus frame from floating point input.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
+      OpusMSEncoder *st,
+      const float *pcm,
+      int frame_size,
+      unsigned char *data,
+      opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusMSEncoder</code> allocated by
+  * opus_multistream_encoder_create().
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
+
+/** Perform a CTL function on a multistream Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+/**\name Multistream decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusMSDecoder</code> structure.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+/** Allocates and initializes a multistream decoder state.
+  * Call opus_multistream_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Intialize a previously allocated decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_deocder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_decoder_init(
+      OpusMSDecoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Decode a multistream Opus packet.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode a multistream Opus packet with floating point output.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on a multistream Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusMSDecoder</code> allocated by
+  * opus_multistream_decoder_create().
+  * @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_MULTISTREAM_H */

libs/mac/include/opus_types.h

@@ -0,0 +1,159 @@
+/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
+/* Modified by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+/* opus_types.h based on ogg_types.h from libogg */
+
+/**
+   @file opus_types.h
+   @brief Opus reference implementation types
+*/
+#ifndef OPUS_TYPES_H
+#define OPUS_TYPES_H
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+#elif defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <_G_config.h>
+     typedef _G_int32_t opus_int32;
+     typedef _G_uint32_t opus_uint32;
+     typedef _G_int16 opus_int16;
+     typedef _G_uint16 opus_uint16;
+#  elif defined(__MINGW32__)
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+#  elif defined(__MWERKS__)
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+#  else
+     /* MSVC/Borland */
+     typedef __int32 opus_int32;
+     typedef unsigned __int32 opus_uint32;
+     typedef __int16 opus_int16;
+     typedef unsigned __int16 opus_uint16;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 opus_int16;
+   typedef UInt16 opus_uint16;
+   typedef SInt32 opus_int32;
+   typedef UInt32 opus_uint32;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <sys/types.h>
+   typedef int16_t opus_int16;
+   typedef u_int16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16 opus_int16;
+   typedef u_int16 opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef int opus_int32;
+   typedef unsigned opus_uint32;
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef signed int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef long opus_int32;
+   typedef unsigned long opus_uint32;
+
+#elif defined(CONFIG_TI_C6X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#else
+
+   /* Give up, take a reasonable guess */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#endif
+
+#define opus_int         int                     /* used for counters etc; at least 16 bits */
+#define opus_int64       long long
+#define opus_int8        signed char
+
+#define opus_uint        unsigned int            /* used for counters etc; at least 16 bits */
+#define opus_uint64      unsigned long long
+#define opus_uint8       unsigned char
+
+#endif  /* OPUS_TYPES_H */

libs/windows/include/opus.h

@@ -0,0 +1,981 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus.h
+ * @brief Opus reference implementation API
+ */
+
+#ifndef OPUS_H
+#define OPUS_H
+
+#include "opus_types.h"
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * @mainpage Opus
+ *
+ * The Opus codec is designed for interactive speech and audio transmission over the Internet.
+ * It is designed by the IETF Codec Working Group and incorporates technology from
+ * Skype's SILK codec and Xiph.Org's CELT codec.
+ *
+ * The Opus codec is designed to handle a wide range of interactive audio applications,
+ * including Voice over IP, videoconferencing, in-game chat, and even remote live music
+ * performances. It can scale from low bit-rate narrowband speech to very high quality
+ * stereo music. Its main features are:
+
+ * @li Sampling rates from 8 to 48 kHz
+ * @li Bit-rates from 6 kb/s to 510 kb/s
+ * @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
+ * @li Audio bandwidth from narrowband to full-band
+ * @li Support for speech and music
+ * @li Support for mono and stereo
+ * @li Support for multichannel (up to 255 channels)
+ * @li Frame sizes from 2.5 ms to 60 ms
+ * @li Good loss robustness and packet loss concealment (PLC)
+ * @li Floating point and fixed-point implementation
+ *
+ * Documentation sections:
+ * @li @ref opus_encoder
+ * @li @ref opus_decoder
+ * @li @ref opus_repacketizer
+ * @li @ref opus_multistream
+ * @li @ref opus_libinfo
+ * @li @ref opus_custom
+ */
+
+/** @defgroup opus_encoder Opus Encoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to encode Opus.
+  *
+  * Since Opus is a stateful codec, the encoding process starts with creating an encoder
+  * state. This can be done with:
+  *
+  * @code
+  * int          error;
+  * OpusEncoder *enc;
+  * enc = opus_encoder_create(Fs, channels, application, &error);
+  * @endcode
+  *
+  * From this point, @c enc can be used for encoding an audio stream. An encoder state
+  * @b must @b not be used for more than one stream at the same time. Similarly, the encoder
+  * state @b must @b not be re-initialized for each frame.
+  *
+  * While opus_encoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  *
+  * @code
+  * int          size;
+  * int          error;
+  * OpusEncoder *enc;
+  * size = opus_encoder_get_size(channels);
+  * enc = malloc(size);
+  * error = opus_encoder_init(enc, Fs, channels, application);
+  * @endcode
+  *
+  * where opus_encoder_get_size() returns the required size for the encoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The encoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * It is possible to change some of the encoder's settings using the opus_encoder_ctl()
+  * interface. All these settings already default to the recommended value, so they should
+  * only be changed when necessary. The most common settings one may want to change are:
+  *
+  * @code
+  * opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
+  * opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
+  * opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
+  * @endcode
+  *
+  * where
+  *
+  * @arg bitrate is in bits per second (b/s)
+  * @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
+  * @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
+  *
+  * See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
+  *
+  * To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
+  * @code
+  * len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
+  * @endcode
+  *
+  * where
+  * <ul>
+  * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
+  * <li>frame_size is the duration of the frame in samples (per channel)</li>
+  * <li>packet is the byte array to which the compressed data is written</li>
+  * <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
+  *     Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
+  * </ul>
+  *
+  * opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
+  * The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
+  * is 2 bytes or less, then the packet does not need to be transmitted (DTX).
+  *
+  * Once the encoder state if no longer needed, it can be destroyed with
+  *
+  * @code
+  * opus_encoder_destroy(enc);
+  * @endcode
+  *
+  * If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
+  * then no action is required aside from potentially freeing the memory that was manually
+  * allocated for it (calling free(enc) for the example above)
+  *
+  */
+
+/** Opus encoder state.
+  * This contains the complete state of an Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_encoder_create,opus_encoder_init
+  */
+typedef struct OpusEncoder OpusEncoder;
+
+/** Gets the size of an <code>OpusEncoder</code> structure.
+  * @param[in] channels <tt>int</tt>: Number of channels.
+  *                                   This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
+
+/**
+ */
+
+/** Allocates and initializes an encoder state.
+ * There are three coding modes:
+ *
+ * @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
+ *    signals. It enhances the  input signal by high-pass filtering and
+ *    emphasizing formants and harmonics. Optionally  it includes in-band
+ *    forward error correction to protect against packet loss. Use this
+ *    mode for typical VoIP applications. Because of the enhancement,
+ *    even at high bitrates the output may sound different from the input.
+ *
+ * @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
+ *    non-voice signals like music. Use this mode for music and mixed
+ *    (music/voice) content, broadcast, and applications requiring less
+ *    than 15 ms of coding delay.
+ *
+ * @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
+ *    disables the speech-optimized mode in exchange for slightly reduced delay.
+ *    This mode can only be set on an newly initialized or freshly reset encoder
+ *    because it changes the codec delay.
+ *
+ * This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
+ * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                     This must be one of 8000, 12000, 16000,
+ *                                     24000, or 48000.
+ * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+ * @param [in] application <tt>int</tt>: Coding mode (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+ * @param [out] error <tt>int*</tt>: @ref opus_errorcodes
+ * @note Regardless of the sampling rate and number channels selected, the Opus encoder
+ * can switch to a lower audio bandwidth or number of channels if the bitrate
+ * selected is too low. This also means that it is safe to always use 48 kHz stereo input
+ * and let the encoder optimize the encoding.
+ */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
+    opus_int32 Fs,
+    int channels,
+    int application,
+    int *error
+);
+
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_encoder_create(),opus_encoder_get_size()
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
+ *                                      This must be one of 8000, 12000, 16000,
+ *                                      24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
+  * @param [in] application <tt>int</tt>: Coding mode (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_encoder_init(
+    OpusEncoder *st,
+    opus_int32 Fs,
+    int channels,
+    int application
+) OPUS_ARG_NONNULL(1);
+
+/** Encodes an Opus frame.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
+    OpusEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes an Opus frame from floating point input.
+  * @param [in] st <tt>OpusEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range.
+  *          length is frame_size*channels*sizeof(float)
+  * @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
+  *                                      input signal.
+  *                                      This must be an Opus frame size for
+  *                                      the encoder's sampling rate.
+  *                                      For example, at 48 kHz the permitted
+  *                                      values are 120, 240, 480, 960, 1920,
+  *                                      and 2880.
+  *                                      Passing in a duration of less than
+  *                                      10 ms (480 samples at 48 kHz) will
+  *                                      prevent the encoder from using the LPC
+  *                                      or hybrid modes.
+  * @param [out] data <tt>unsigned char*</tt>: Output payload.
+  *                                            This must contain storage for at
+  *                                            least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
+    OpusEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
+  * @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
+
+/** Perform a CTL function on an Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusEncoder*</tt>: Encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_encoderctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  */
+OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+/**@}*/
+
+/** @defgroup opus_decoder Opus Decoder
+  * @{
+  *
+  * @brief This page describes the process and functions used to decode Opus.
+  *
+  * The decoding process also starts with creating a decoder
+  * state. This can be done with:
+  * @code
+  * int          error;
+  * OpusDecoder *dec;
+  * dec = opus_decoder_create(Fs, channels, &error);
+  * @endcode
+  * where
+  * @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
+  * @li channels is the number of channels (1 or 2)
+  * @li error will hold the error code in case of failure (or #OPUS_OK on success)
+  * @li the return value is a newly created decoder state to be used for decoding
+  *
+  * While opus_decoder_create() allocates memory for the state, it's also possible
+  * to initialize pre-allocated memory:
+  * @code
+  * int          size;
+  * int          error;
+  * OpusDecoder *dec;
+  * size = opus_decoder_get_size(channels);
+  * dec = malloc(size);
+  * error = opus_decoder_init(dec, Fs, channels);
+  * @endcode
+  * where opus_decoder_get_size() returns the required size for the decoder state. Note that
+  * future versions of this code may change the size, so no assuptions should be made about it.
+  *
+  * The decoder state is always continuous in memory and only a shallow copy is sufficient
+  * to copy it (e.g. memcpy())
+  *
+  * To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
+  * @code
+  * frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
+  * @endcode
+  * where
+  *
+  * @li packet is the byte array containing the compressed data
+  * @li len is the exact number of bytes contained in the packet
+  * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
+  * @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
+  *
+  * opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
+  * If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
+  * buffer is too small to hold the decoded audio.
+  *
+  * Opus is a stateful codec with overlapping blocks and as a result Opus
+  * packets are not coded independently of each other. Packets must be
+  * passed into the decoder serially and in the correct order for a correct
+  * decode. Lost packets can be replaced with loss concealment by calling
+  * the decoder with a null pointer and zero length for the missing packet.
+  *
+  * A single codec state may only be accessed from a single thread at
+  * a time and any required locking must be performed by the caller. Separate
+  * streams must be decoded with separate decoder states and can be decoded
+  * in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
+  * defined.
+  *
+  */
+
+/** Opus decoder state.
+  * This contains the complete state of an Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_decoder_create,opus_decoder_init
+  */
+typedef struct OpusDecoder OpusDecoder;
+
+/** Gets the size of an <code>OpusDecoder</code> structure.
+  * @param [in] channels <tt>int</tt>: Number of channels.
+  *                                    This must be 1 or 2.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
+
+/** Allocates and initializes a decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
+  *
+  * Internally Opus stores data at 48000 Hz, so that should be the default
+  * value for Fs. However, the decoder can efficiently decode to buffers
+  * at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
+  * data at the full sample rate, or knows the compressed data doesn't
+  * use the full frequency range, it can request decoding at a reduced
+  * rate. Likewise, the decoder is capable of filling in either mono or
+  * interleaved stereo pcm buffers, at the caller's request.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
+    opus_int32 Fs,
+    int channels,
+    int *error
+);
+
+/** Initializes a previously allocated decoder state.
+  * The state must be at least the size returned by opus_decoder_get_size().
+  * This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
+  *                                     This must be one of 8000, 12000, 16000,
+  *                                     24000, or 48000.
+  * @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
+  * @retval #OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_EXPORT int opus_decoder_init(
+    OpusDecoder *st,
+    opus_int32 Fs,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Decode an Opus packet.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available, the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an Opus packet with floating point output.
+  * @param [in] st <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in \a pcm.
+  *  If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
+  *  not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
+  *  then frame_size needs to be exactly the duration of audio that is missing, otherwise the
+  *  decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
+  *  FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
+  * @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
+  *  decoded. If no such data is available the frame is decoded as if it were lost.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
+    OpusDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @param st <tt>OpusDecoder*</tt>: Decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls or
+  *                @ref opus_decoderctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  */
+OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
+  * @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
+
+/** Parse an opus packet into one or more frames.
+  * Opus_decode will perform this operation internally so most applications do
+  * not need to use this function.
+  * This function does not copy the frames, the returned pointers are pointers into
+  * the input packet.
+  * @param [in] data <tt>char*</tt>: Opus packet to be parsed
+  * @param [in] len <tt>opus_int32</tt>: size of data
+  * @param [out] out_toc <tt>char*</tt>: TOC pointer
+  * @param [out] frames <tt>char*[48]</tt> encapsulated frames
+  * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
+  * @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
+  * @returns number of frames
+  */
+OPUS_EXPORT int opus_packet_parse(
+   const unsigned char *data,
+   opus_int32 len,
+   unsigned char *out_toc,
+   const unsigned char *frames[48],
+   opus_int16 size[48],
+   int *payload_offset
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Gets the bandwidth of an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
+  * @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
+  * @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
+  * @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
+  * @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples per frame from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet.
+  *                                  This must contain at least one byte of
+  *                                  data.
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples per frame.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of channels from an Opus packet.
+  * @param [in] data <tt>char*</tt>: Opus packet
+  * @returns Number of channels
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of frames in an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of frames
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
+  *                                     This must be a multiple of 400, or
+  *                                     inaccurate results will be returned.
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
+
+/** Gets the number of samples of an Opus packet.
+  * @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
+  * @param [in] packet <tt>char*</tt>: Opus packet
+  * @param [in] len <tt>opus_int32</tt>: Length of packet
+  * @returns Number of samples
+  * @retval OPUS_BAD_ARG Insufficient data was passed to the function
+  * @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
+  * the signal is already in that range, nothing is done. If there are values
+  * outside of [-1,1], then the signal is clipped as smoothly as possible to
+  * both fit in the range and avoid creating excessive distortion in the
+  * process.
+  * @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
+  * @param [in] frame_size <tt>int</tt> Number of samples per channel to process
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
+  */
+OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
+
+
+/**@}*/
+
+/** @defgroup opus_repacketizer Repacketizer
+  * @{
+  *
+  * The repacketizer can be used to merge multiple Opus packets into a single
+  * packet or alternatively to split Opus packets that have previously been
+  * merged. Splitting valid Opus packets is always guaranteed to succeed,
+  * whereas merging valid packets only succeeds if all frames have the same
+  * mode, bandwidth, and frame size, and when the total duration of the merged
+  * packet is no more than 120 ms. The 120 ms limit comes from the
+  * specification and limits decoder memory requirements at a point where
+  * framing overhead becomes negligible.
+  *
+  * The repacketizer currently only operates on elementary Opus
+  * streams. It will not manipualte multistream packets successfully, except in
+  * the degenerate case where they consist of data from a single stream.
+  *
+  * The repacketizing process starts with creating a repacketizer state, either
+  * by calling opus_repacketizer_create() or by allocating the memory yourself,
+  * e.g.,
+  * @code
+  * OpusRepacketizer *rp;
+  * rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
+  * if (rp != NULL)
+  *     opus_repacketizer_init(rp);
+  * @endcode
+  *
+  * Then the application should submit packets with opus_repacketizer_cat(),
+  * extract new packets with opus_repacketizer_out() or
+  * opus_repacketizer_out_range(), and then reset the state for the next set of
+  * input packets via opus_repacketizer_init().
+  *
+  * For example, to split a sequence of packets into individual frames:
+  * @code
+  * unsigned char *data;
+  * int len;
+  * while (get_next_packet(&data, &len))
+  * {
+  *   unsigned char out[1276];
+  *   opus_int32 out_len;
+  *   int nb_frames;
+  *   int err;
+  *   int i;
+  *   err = opus_repacketizer_cat(rp, data, len);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packet(data);
+  *     return err;
+  *   }
+  *   nb_frames = opus_repacketizer_get_nb_frames(rp);
+  *   for (i = 0; i < nb_frames; i++)
+  *   {
+  *     out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packet(data);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *   }
+  *   opus_repacketizer_init(rp);
+  *   release_packet(data);
+  * }
+  * @endcode
+  *
+  * Alternatively, to combine a sequence of frames into packets that each
+  * contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
+  * @code
+  * // The maximum number of packets with duration TARGET_DURATION_MS occurs
+  * // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
+  * // packets.
+  * unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
+  * opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
+  * int nb_packets;
+  * unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
+  * opus_int32 out_len;
+  * int prev_toc;
+  * nb_packets = 0;
+  * while (get_next_packet(data+nb_packets, len+nb_packets))
+  * {
+  *   int nb_frames;
+  *   int err;
+  *   nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
+  *   if (nb_frames < 1)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return nb_frames;
+  *   }
+  *   nb_frames += opus_repacketizer_get_nb_frames(rp);
+  *   // If adding the next packet would exceed our target, or it has an
+  *   // incompatible TOC sequence, output the packets we already have before
+  *   // submitting it.
+  *   // N.B., The nb_packets > 0 check ensures we've submitted at least one
+  *   // packet since the last call to opus_repacketizer_init(). Otherwise a
+  *   // single packet longer than TARGET_DURATION_MS would cause us to try to
+  *   // output an (invalid) empty packet. It also ensures that prev_toc has
+  *   // been set to a valid value. Additionally, len[nb_packets] > 0 is
+  *   // guaranteed by the call to opus_packet_get_nb_frames() above, so the
+  *   // reference to data[nb_packets][0] should be valid.
+  *   if (nb_packets > 0 && (
+  *       ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
+  *       opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
+  *       TARGET_DURATION_MS*48))
+  *   {
+  *     out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *     if (out_len < 0)
+  *     {
+  *        release_packets(data, nb_packets+1);
+  *        return (int)out_len;
+  *     }
+  *     output_next_packet(out, out_len);
+  *     opus_repacketizer_init(rp);
+  *     release_packets(data, nb_packets);
+  *     data[0] = data[nb_packets];
+  *     len[0] = len[nb_packets];
+  *     nb_packets = 0;
+  *   }
+  *   err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
+  *   if (err != OPUS_OK)
+  *   {
+  *     release_packets(data, nb_packets+1);
+  *     return err;
+  *   }
+  *   prev_toc = data[nb_packets][0];
+  *   nb_packets++;
+  * }
+  * // Output the final, partial packet.
+  * if (nb_packets > 0)
+  * {
+  *   out_len = opus_repacketizer_out(rp, out, sizeof(out));
+  *   release_packets(data, nb_packets);
+  *   if (out_len < 0)
+  *     return (int)out_len;
+  *   output_next_packet(out, out_len);
+  * }
+  * @endcode
+  *
+  * An alternate way of merging packets is to simply call opus_repacketizer_cat()
+  * unconditionally until it fails. At that point, the merged packet can be
+  * obtained with opus_repacketizer_out() and the input packet for which
+  * opus_repacketizer_cat() needs to be re-added to a newly reinitialized
+  * repacketizer state.
+  */
+
+typedef struct OpusRepacketizer OpusRepacketizer;
+
+/** Gets the size of an <code>OpusRepacketizer</code> structure.
+  * @returns The size in bytes.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
+
+/** (Re)initializes a previously allocated repacketizer state.
+  * The state must be at least the size returned by opus_repacketizer_get_size().
+  * This can be used for applications which use their own allocator instead of
+  * malloc().
+  * It must also be called to reset the queue of packets waiting to be
+  * repacketized, which is necessary if the maximum packet duration of 120 ms
+  * is reached or if you wish to submit packets with a different Opus
+  * configuration (coding mode, audio bandwidth, frame size, or channel count).
+  * Failure to do so will prevent a new packet from being added with
+  * opus_repacketizer_cat().
+  * @see opus_repacketizer_create
+  * @see opus_repacketizer_get_size
+  * @see opus_repacketizer_cat
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
+  *                                       (re)initialize.
+  * @returns A pointer to the same repacketizer state that was passed in.
+  */
+OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Allocates memory and initializes the new repacketizer with
+ * opus_repacketizer_init().
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
+
+/** Frees an <code>OpusRepacketizer</code> allocated by
+  * opus_repacketizer_create().
+  * @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
+  */
+OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
+
+/** Add a packet to the current repacketizer state.
+  * This packet must match the configuration of any packets already submitted
+  * for repacketization since the last call to opus_repacketizer_init().
+  * This means that it must have the same coding mode, audio bandwidth, frame
+  * size, and channel count.
+  * This can be checked in advance by examining the top 6 bits of the first
+  * byte of the packet, and ensuring they match the top 6 bits of the first
+  * byte of any previously submitted packet.
+  * The total duration of audio in the repacketizer state also must not exceed
+  * 120 ms, the maximum duration of a single packet, after adding this packet.
+  *
+  * The contents of the current repacketizer state can be extracted into new
+  * packets using opus_repacketizer_out() or opus_repacketizer_out_range().
+  *
+  * In order to add a packet with a different configuration or to add more
+  * audio beyond 120 ms, you must clear the repacketizer state by calling
+  * opus_repacketizer_init().
+  * If a packet is too large to add to the current repacketizer state, no part
+  * of it is added, even if it contains multiple frames, some of which might
+  * fit.
+  * If you wish to be able to add parts of such packets, you should first use
+  * another repacketizer to split the packet into pieces and add them
+  * individually.
+  * @see opus_repacketizer_out_range
+  * @see opus_repacketizer_out
+  * @see opus_repacketizer_init
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
+  *                                       add the packet.
+  * @param[in] data <tt>const unsigned char*</tt>: The packet data.
+  *                                                The application must ensure
+  *                                                this pointer remains valid
+  *                                                until the next call to
+  *                                                opus_repacketizer_init() or
+  *                                                opus_repacketizer_destroy().
+  * @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
+  * @returns An error code indicating whether or not the operation succeeded.
+  * @retval #OPUS_OK The packet's contents have been added to the repacketizer
+  *                  state.
+  * @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
+  *                              the packet's TOC sequence was not compatible
+  *                              with previously submitted packets (because
+  *                              the coding mode, audio bandwidth, frame size,
+  *                              or channel count did not match), or adding
+  *                              this packet would increase the total amount of
+  *                              audio stored in the repacketizer state to more
+  *                              than 120 ms.
+  */
+OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param begin <tt>int</tt>: The index of the first frame in the current
+  *                            repacketizer state to include in the output.
+  * @param end <tt>int</tt>: One past the index of the last frame in the
+  *                          current repacketizer state to include in the
+  *                          output.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1276</code> for a single frame,
+  *                                    or for multiple frames,
+  *                                    <code>1277*(end-begin)</code>.
+  *                                    However, <code>1*(end-begin)</code> plus
+  *                                    the size of all packet data submitted to
+  *                                    the repacketizer since the last call to
+  *                                    opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
+  *                       frames (begin < 0, begin >= end, or end >
+  *                       opus_repacketizer_get_nb_frames()).
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Return the total number of frames contained in packet data submitted to
+  * the repacketizer state so far via opus_repacketizer_cat() since the last
+  * call to opus_repacketizer_init() or opus_repacketizer_create().
+  * This defines the valid range of packets that can be extracted with
+  * opus_repacketizer_out_range() or opus_repacketizer_out().
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
+  *                                       frames.
+  * @returns The total number of frames contained in the packet data submitted
+  *          to the repacketizer state.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
+
+/** Construct a new packet from data previously submitted to the repacketizer
+  * state via opus_repacketizer_cat().
+  * This is a convenience routine that returns all the data submitted so far
+  * in a single packet.
+  * It is equivalent to calling
+  * @code
+  * opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
+  *                             data, maxlen)
+  * @endcode
+  * @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
+  *                                       construct the new packet.
+  * @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
+  *                                                 store the output packet.
+  * @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
+  *                                    the output buffer. In order to guarantee
+  *                                    success, this should be at least
+  *                                    <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
+  *                                    However,
+  *                                    <code>1*opus_repacketizer_get_nb_frames(rp)</code>
+  *                                    plus the size of all packet data
+  *                                    submitted to the repacketizer since the
+  *                                    last call to opus_repacketizer_init() or
+  *                                    opus_repacketizer_create() is also
+  *                                    sufficient, and possibly much smaller.
+  * @returns The total size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
+  *                                complete output packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
+
+/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
+
+/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
+
+/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to pad.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least as large as len.
+  * @returns an error code
+  * @retval #OPUS_OK \a on success.
+  * @retval #OPUS_BAD_ARG \a len was less than 1.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
+
+/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
+  * minimize space usage.
+  * @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
+  *                                                   packet to strip.
+  * @param len <tt>opus_int32</tt>: The size of the packet.
+  *                                 This must be at least 1.
+  * @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
+  *                                 This must be at least 1.
+  * @returns The new size of the output packet on success, or an error code
+  *          on failure.
+  * @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
+  * @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */

libs/windows/include/opus_custom.h

@@ -0,0 +1,342 @@
+/* Copyright (c) 2007-2008 CSIRO
+   Copyright (c) 2007-2009 Xiph.Org Foundation
+   Copyright (c) 2008-2012 Gregory Maxwell
+   Written by Jean-Marc Valin and Gregory Maxwell */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+  @file opus_custom.h
+  @brief Opus-Custom reference implementation API
+ */
+
+#ifndef OPUS_CUSTOM_H
+#define OPUS_CUSTOM_H
+
+#include "opus_defines.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef CUSTOM_MODES
+# define OPUS_CUSTOM_EXPORT OPUS_EXPORT
+# define OPUS_CUSTOM_EXPORT_STATIC OPUS_EXPORT
+#else
+# define OPUS_CUSTOM_EXPORT
+# ifdef OPUS_BUILD
+#  define OPUS_CUSTOM_EXPORT_STATIC static OPUS_INLINE
+# else
+#  define OPUS_CUSTOM_EXPORT_STATIC
+# endif
+#endif
+
+/** @defgroup opus_custom Opus Custom
+  * @{
+  *  Opus Custom is an optional part of the Opus specification and
+  * reference implementation which uses a distinct API from the regular
+  * API and supports frame sizes that are not normally supported.\ Use
+  * of Opus Custom is discouraged for all but very special applications
+  * for which a frame size different from 2.5, 5, 10, or 20 ms is needed
+  * (for either complexity or latency reasons) and where interoperability
+  * is less important.
+  *
+  * In addition to the interoperability limitations the use of Opus custom
+  * disables a substantial chunk of the codec and generally lowers the
+  * quality available at a given bitrate. Normally when an application needs
+  * a different frame size from the codec it should buffer to match the
+  * sizes but this adds a small amount of delay which may be important
+  * in some very low latency applications. Some transports (especially
+  * constant rate RF transports) may also work best with frames of
+  * particular durations.
+  *
+  * Libopus only supports custom modes if they are enabled at compile time.
+  *
+  * The Opus Custom API is similar to the regular API but the
+  * @ref opus_encoder_create and @ref opus_decoder_create calls take
+  * an additional mode parameter which is a structure produced by
+  * a call to @ref opus_custom_mode_create. Both the encoder and decoder
+  * must create a mode using the same sample rate (fs) and frame size
+  * (frame size) so these parameters must either be signaled out of band
+  * or fixed in a particular implementation.
+  *
+  * Similar to regular Opus the custom modes support on the fly frame size
+  * switching, but the sizes available depend on the particular frame size in
+  * use. For some initial frame sizes on a single on the fly size is available.
+  */
+
+/** Contains the state of an encoder. One encoder state is needed
+    for each stream. It is initialized once at the beginning of the
+    stream. Do *not* re-initialize the state for every frame.
+   @brief Encoder state
+ */
+typedef struct OpusCustomEncoder OpusCustomEncoder;
+
+/** State of the decoder. One decoder state is needed for each stream.
+    It is initialized once at the beginning of the stream. Do *not*
+    re-initialize the state for every frame.
+   @brief Decoder state
+ */
+typedef struct OpusCustomDecoder OpusCustomDecoder;
+
+/** The mode contains all the information necessary to create an
+    encoder. Both the encoder and decoder need to be initialized
+    with exactly the same mode, otherwise the output will be
+    corrupted.
+   @brief Mode configuration
+ */
+typedef struct OpusCustomMode OpusCustomMode;
+
+/** Creates a new mode struct. This will be passed to an encoder or
+  * decoder. The mode MUST NOT BE DESTROYED until the encoders and
+  * decoders that use it are destroyed as well.
+  * @param [in] Fs <tt>int</tt>: Sampling rate (8000 to 96000 Hz)
+  * @param [in] frame_size <tt>int</tt>: Number of samples (per channel) to encode in each
+  *        packet (64 - 1024, prime factorization must contain zero or more 2s, 3s, or 5s and no other primes)
+  * @param [out] error <tt>int*</tt>: Returned error code (if NULL, no error will be returned)
+  * @return A newly created mode
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomMode *opus_custom_mode_create(opus_int32 Fs, int frame_size, int *error);
+
+/** Destroys a mode struct. Only call this after all encoders and
+  * decoders using this mode are destroyed as well.
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Mode to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_mode_destroy(OpusCustomMode *mode);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_ENCODER_C)
+
+/* Encoder */
+/** Gets the size of an OpusCustomEncoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_encoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+# ifdef CUSTOM_MODES
+/** Initializes a previously allocated encoder state
+  * The memory pointed to by st must be the size returned by opus_custom_encoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_encoder_create(),opus_custom_encoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_init(
+    OpusCustomEncoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+# endif
+#endif
+
+
+/** Creates a new encoder state. Each stream needs its own encoder
+  * state (can't be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode*</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  decoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created encoder state.
+*/
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomEncoder *opus_custom_encoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+
+/** Destroys a an encoder state.
+  * @param[in] st <tt>OpusCustomEncoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_encoder_destroy(OpusCustomEncoder *st);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>float*</tt>: PCM audio in float format, with a normal range of +/-1.0.
+  *          Samples with a range beyond +/-1.0 are supported but will
+  *          be clipped by decoders using the integer API and should
+  *          only be used if it is known that the far end supports
+  *          extended dynamic range. There must be exactly
+  *          frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode_float(
+    OpusCustomEncoder *st,
+    const float *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a frame of audio.
+  * @param [in] st <tt>OpusCustomEncoder*</tt>: Encoder state
+  * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian).
+  *          There must be exactly frame_size samples per channel.
+  * @param [in] frame_size <tt>int</tt>: Number of samples per frame of input signal
+  * @param [out] compressed <tt>char *</tt>: The compressed data is written here. This may not alias pcm and must be at least maxCompressedBytes long.
+  * @param [in] maxCompressedBytes <tt>int</tt>: Maximum number of bytes to use for compressing the frame
+  *          (can change from one frame to another)
+  * @return Number of bytes written to "compressed".
+  *       If negative, an error has occurred (see error codes). It is IMPORTANT that
+  *       the length returned be somehow transmitted to the decoder. Otherwise, no
+  *       decoding is possible.
+ */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_encode(
+    OpusCustomEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *compressed,
+    int maxCompressedBytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom encoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_encoderctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_encoder_ctl(OpusCustomEncoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+
+#if !defined(OPUS_BUILD) || defined(CELT_DECODER_C)
+/* Decoder */
+
+/** Gets the size of an OpusCustomDecoder structure.
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Mode configuration
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @returns size
+  */
+OPUS_CUSTOM_EXPORT_STATIC OPUS_WARN_UNUSED_RESULT int opus_custom_decoder_get_size(
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1);
+
+/** Initializes a previously allocated decoder state
+  * The memory pointed to by st must be the size returned by opus_custom_decoder_get_size.
+  * This is intended for applications which use their own allocator instead of malloc.
+  * @see opus_custom_decoder_create(),opus_custom_decoder_get_size()
+  * To reset a previously initialized state use the OPUS_RESET_STATE CTL.
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] mode <tt>OpusCustomMode *</tt>: Contains all the information about the characteristics of
+  *  the stream (must be the same characteristics as used for the
+  *  encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @return OPUS_OK Success or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT_STATIC int opus_custom_decoder_init(
+    OpusCustomDecoder *st,
+    const OpusCustomMode *mode,
+    int channels
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
+
+#endif
+
+
+/** Creates a new decoder state. Each stream needs its own decoder state (can't
+  * be shared across simultaneous streams).
+  * @param [in] mode <tt>OpusCustomMode</tt>: Contains all the information about the characteristics of the
+  *          stream (must be the same characteristics as used for the encoder)
+  * @param [in] channels <tt>int</tt>: Number of channels
+  * @param [out] error <tt>int*</tt>: Returns an error code
+  * @return Newly created decoder state.
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT OpusCustomDecoder *opus_custom_decoder_create(
+    const OpusCustomMode *mode,
+    int channels,
+    int *error
+) OPUS_ARG_NONNULL(1);
+
+/** Destroys a an decoder state.
+  * @param[in] st <tt>OpusCustomDecoder*</tt>: State to be freed.
+  */
+OPUS_CUSTOM_EXPORT void opus_custom_decoder_destroy(OpusCustomDecoder *st);
+
+/** Decode an opus custom frame with floating point output
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(float)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode_float(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    float *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode an opus custom frame
+  * @param [in] st <tt>OpusCustomDecoder*</tt>: Decoder state
+  * @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
+  * @param [in] len <tt>int</tt>: Number of bytes in payload
+  * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
+  *  is frame_size*channels*sizeof(opus_int16)
+  * @param [in] frame_size Number of samples per channel of available space in *pcm.
+  * @returns Number of decoded samples or @ref opus_errorcodes
+  */
+OPUS_CUSTOM_EXPORT OPUS_WARN_UNUSED_RESULT int opus_custom_decode(
+    OpusCustomDecoder *st,
+    const unsigned char *data,
+    int len,
+    opus_int16 *pcm,
+    int frame_size
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on an Opus custom decoder.
+  *
+  * Generally the request and subsequent arguments are generated
+  * by a convenience macro.
+  * @see opus_genericctls
+  */
+OPUS_CUSTOM_EXPORT int opus_custom_decoder_ctl(OpusCustomDecoder * OPUS_RESTRICT st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_CUSTOM_H */

libs/windows/include/opus_defines.h

@@ -0,0 +1,785 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_defines.h
+ * @brief Opus reference implementation constants
+ */
+
+#ifndef OPUS_DEFINES_H
+#define OPUS_DEFINES_H
+
+#include "opus_types.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @defgroup opus_errorcodes Error codes
+ * @{
+ */
+/** No error @hideinitializer*/
+#define OPUS_OK                0
+/** One or more invalid/out of range arguments @hideinitializer*/
+#define OPUS_BAD_ARG          -1
+/** Not enough bytes allocated in the buffer @hideinitializer*/
+#define OPUS_BUFFER_TOO_SMALL -2
+/** An internal error was detected @hideinitializer*/
+#define OPUS_INTERNAL_ERROR   -3
+/** The compressed data passed is corrupted @hideinitializer*/
+#define OPUS_INVALID_PACKET   -4
+/** Invalid/unsupported request number @hideinitializer*/
+#define OPUS_UNIMPLEMENTED    -5
+/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
+#define OPUS_INVALID_STATE    -6
+/** Memory allocation has failed @hideinitializer*/
+#define OPUS_ALLOC_FAIL       -7
+/**@}*/
+
+/** @cond OPUS_INTERNAL_DOC */
+/**Export control for opus functions */
+
+#ifndef OPUS_EXPORT
+# if defined(WIN32)
+#  if defined(OPUS_BUILD) && defined(DLL_EXPORT)
+#   define OPUS_EXPORT __declspec(dllexport)
+#  else
+#   define OPUS_EXPORT
+#  endif
+# elif defined(__GNUC__) && defined(OPUS_BUILD)
+#  define OPUS_EXPORT __attribute__ ((visibility ("default")))
+# else
+#  define OPUS_EXPORT
+# endif
+#endif
+
+# if !defined(OPUS_GNUC_PREREQ)
+#  if defined(__GNUC__)&&defined(__GNUC_MINOR__)
+#   define OPUS_GNUC_PREREQ(_maj,_min) \
+ ((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
+#  else
+#   define OPUS_GNUC_PREREQ(_maj,_min) 0
+#  endif
+# endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(3,0)
+#  define OPUS_RESTRICT __restrict__
+# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
+#  define OPUS_RESTRICT __restrict
+# else
+#  define OPUS_RESTRICT
+# endif
+#else
+# define OPUS_RESTRICT restrict
+#endif
+
+#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
+# if OPUS_GNUC_PREREQ(2,7)
+#  define OPUS_INLINE __inline__
+# elif (defined(_MSC_VER))
+#  define OPUS_INLINE __inline
+# else
+#  define OPUS_INLINE
+# endif
+#else
+# define OPUS_INLINE inline
+#endif
+
+/**Warning attributes for opus functions
+  * NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
+  * some paranoid null checks. */
+#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
+#else
+# define OPUS_WARN_UNUSED_RESULT
+#endif
+#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
+# define OPUS_ARG_NONNULL(_x)  __attribute__ ((__nonnull__(_x)))
+#else
+# define OPUS_ARG_NONNULL(_x)
+#endif
+
+/** These are the actual Encoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+#define OPUS_SET_APPLICATION_REQUEST         4000
+#define OPUS_GET_APPLICATION_REQUEST         4001
+#define OPUS_SET_BITRATE_REQUEST             4002
+#define OPUS_GET_BITRATE_REQUEST             4003
+#define OPUS_SET_MAX_BANDWIDTH_REQUEST       4004
+#define OPUS_GET_MAX_BANDWIDTH_REQUEST       4005
+#define OPUS_SET_VBR_REQUEST                 4006
+#define OPUS_GET_VBR_REQUEST                 4007
+#define OPUS_SET_BANDWIDTH_REQUEST           4008
+#define OPUS_GET_BANDWIDTH_REQUEST           4009
+#define OPUS_SET_COMPLEXITY_REQUEST          4010
+#define OPUS_GET_COMPLEXITY_REQUEST          4011
+#define OPUS_SET_INBAND_FEC_REQUEST          4012
+#define OPUS_GET_INBAND_FEC_REQUEST          4013
+#define OPUS_SET_PACKET_LOSS_PERC_REQUEST    4014
+#define OPUS_GET_PACKET_LOSS_PERC_REQUEST    4015
+#define OPUS_SET_DTX_REQUEST                 4016
+#define OPUS_GET_DTX_REQUEST                 4017
+#define OPUS_SET_VBR_CONSTRAINT_REQUEST      4020
+#define OPUS_GET_VBR_CONSTRAINT_REQUEST      4021
+#define OPUS_SET_FORCE_CHANNELS_REQUEST      4022
+#define OPUS_GET_FORCE_CHANNELS_REQUEST      4023
+#define OPUS_SET_SIGNAL_REQUEST              4024
+#define OPUS_GET_SIGNAL_REQUEST              4025
+#define OPUS_GET_LOOKAHEAD_REQUEST           4027
+/* #define OPUS_RESET_STATE 4028 */
+#define OPUS_GET_SAMPLE_RATE_REQUEST         4029
+#define OPUS_GET_FINAL_RANGE_REQUEST         4031
+#define OPUS_GET_PITCH_REQUEST               4033
+#define OPUS_SET_GAIN_REQUEST                4034
+#define OPUS_GET_GAIN_REQUEST                4045 /* Should have been 4035 */
+#define OPUS_SET_LSB_DEPTH_REQUEST           4036
+#define OPUS_GET_LSB_DEPTH_REQUEST           4037
+#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
+#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
+#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
+#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
+#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
+/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
+#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
+#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
+
+/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
+#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
+#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
+#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
+#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
+/** @endcond */
+
+/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
+  * @see opus_genericctls, opus_encoderctls
+  * @{
+  */
+/* Values for the various encoder CTLs */
+#define OPUS_AUTO                           -1000 /**<Auto/default setting @hideinitializer*/
+#define OPUS_BITRATE_MAX                       -1 /**<Maximum bitrate @hideinitializer*/
+
+/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
+ * @hideinitializer */
+#define OPUS_APPLICATION_VOIP                2048
+/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
+ * @hideinitializer */
+#define OPUS_APPLICATION_AUDIO               2049
+/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
+ * @hideinitializer */
+#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
+
+#define OPUS_SIGNAL_VOICE                    3001 /**< Signal being encoded is voice */
+#define OPUS_SIGNAL_MUSIC                    3002 /**< Signal being encoded is music */
+#define OPUS_BANDWIDTH_NARROWBAND            1101 /**< 4 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_MEDIUMBAND            1102 /**< 6 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_WIDEBAND              1103 /**< 8 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_SUPERWIDEBAND         1104 /**<12 kHz bandpass @hideinitializer*/
+#define OPUS_BANDWIDTH_FULLBAND              1105 /**<20 kHz bandpass @hideinitializer*/
+
+#define OPUS_FRAMESIZE_ARG                   5000 /**< Select frame size from the argument (default) */
+#define OPUS_FRAMESIZE_2_5_MS                5001 /**< Use 2.5 ms frames */
+#define OPUS_FRAMESIZE_5_MS                  5002 /**< Use 5 ms frames */
+#define OPUS_FRAMESIZE_10_MS                 5003 /**< Use 10 ms frames */
+#define OPUS_FRAMESIZE_20_MS                 5004 /**< Use 20 ms frames */
+#define OPUS_FRAMESIZE_40_MS                 5005 /**< Use 40 ms frames */
+#define OPUS_FRAMESIZE_60_MS                 5006 /**< Use 60 ms frames */
+#define OPUS_FRAMESIZE_80_MS                 5007 /**< Use 80 ms frames */
+#define OPUS_FRAMESIZE_100_MS                5008 /**< Use 100 ms frames */
+#define OPUS_FRAMESIZE_120_MS                5009 /**< Use 120 ms frames */
+
+/**@}*/
+
+
+/** @defgroup opus_encoderctls Encoder related CTLs
+  *
+  * These are convenience macros for use with the \c opus_encode_ctl
+  * interface. They are used to generate the appropriate series of
+  * arguments for that call, passing the correct type, size and so
+  * on as expected for each particular request.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
+  * if (ret != OPUS_OK) return ret;
+  *
+  * opus_int32 rate;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * @endcode
+  *
+  * @see opus_genericctls, opus_encoder
+  * @{
+  */
+
+/** Configures the encoder's computational complexity.
+  * The supported range is 0-10 inclusive with 10 representing the highest complexity.
+  * @see OPUS_GET_COMPLEXITY
+  * @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
+  *
+  * @hideinitializer */
+#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
+/** Gets the encoder's complexity configuration.
+  * @see OPUS_SET_COMPLEXITY
+  * @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
+  *                                      inclusive.
+  * @hideinitializer */
+#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the bitrate in the encoder.
+  * Rates from 500 to 512000 bits per second are meaningful, as well as the
+  * special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
+  * The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
+  * rate as it can, which is useful for controlling the rate by adjusting the
+  * output buffer size.
+  * @see OPUS_GET_BITRATE
+  * @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
+  *                                   is determined based on the number of
+  *                                   channels and the input sampling rate.
+  * @hideinitializer */
+#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
+/** Gets the encoder's bitrate configuration.
+  * @see OPUS_SET_BITRATE
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
+  *                                      The default is determined based on the
+  *                                      number of channels and the input
+  *                                      sampling rate.
+  * @hideinitializer */
+#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables variable bitrate (VBR) in the encoder.
+  * The configured bitrate may not be met exactly because frames must
+  * be an integer number of bytes in length.
+  * @see OPUS_GET_VBR
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
+  *               cause noticeable quality degradation.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
+  *               #OPUS_SET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
+/** Determine if variable bitrate (VBR) is enabled in the encoder.
+  * @see OPUS_SET_VBR
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Hard CBR.</dd>
+  * <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
+  *               #OPUS_GET_VBR_CONSTRAINT.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
+
+/** Enables or disables constrained VBR in the encoder.
+  * This setting is ignored when the encoder is in CBR mode.
+  * @warning Only the MDCT mode of Opus currently heeds the constraint.
+  *  Speech mode ignores it completely, hybrid mode may fail to obey it
+  *  if the LPC layer uses more bitrate than the constraint would have
+  *  permitted.
+  * @see OPUS_GET_VBR_CONSTRAINT
+  * @see OPUS_SET_VBR
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
+  *               frame of buffering delay assuming a transport with a
+  *               serialization speed of the nominal bitrate.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
+/** Determine if constrained VBR is enabled in the encoder.
+  * @see OPUS_SET_VBR_CONSTRAINT
+  * @see OPUS_GET_VBR
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Unconstrained VBR.</dd>
+  * <dt>1</dt><dd>Constrained VBR (default).</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures mono/stereo forcing in the encoder.
+  * This can force the encoder to produce packets encoded as either mono or
+  * stereo, regardless of the format of the input audio. This is useful when
+  * the caller knows that the input signal is currently a mono source embedded
+  * in a stereo stream.
+  * @see OPUS_GET_FORCE_CHANNELS
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
+/** Gets the encoder's forced channel configuration.
+  * @see OPUS_SET_FORCE_CHANNELS
+  * @param[out] x <tt>opus_int32 *</tt>:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
+  * <dt>1</dt>         <dd>Forced mono</dd>
+  * <dt>2</dt>         <dd>Forced stereo</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the maximum bandpass that the encoder will select automatically.
+  * Applications should normally use this instead of #OPUS_SET_BANDWIDTH
+  * (leaving that set to the default, #OPUS_AUTO). This allows the
+  * application to set an upper bound based on the type of input it is
+  * providing, but still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_MAX_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Gets the encoder's configured maximum allowed bandpass.
+  * @see OPUS_SET_MAX_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband (default)</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Sets the encoder's bandpass to a specific value.
+  * This prevents the encoder from automatically selecting the bandpass based
+  * on the available bitrate. If an application knows the bandpass of the input
+  * audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
+  * instead, which still gives the encoder the freedom to reduce the bandpass
+  * when the bitrate becomes too low, for better overall quality.
+  * @see OPUS_GET_BANDWIDTH
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
+
+/** Configures the type of signal being encoded.
+  * This is a hint which helps the encoder's mode selection.
+  * @see OPUS_GET_SIGNAL
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal type.
+  * @see OPUS_SET_SIGNAL
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>        <dd>(default)</dd>
+  * <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
+  * <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
+
+
+/** Configures the encoder's intended application.
+  * The initial value is a mandatory argument to the encoder_create function.
+  * @see OPUS_GET_APPLICATION
+  * @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured application.
+  * @see OPUS_SET_APPLICATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the total samples of delay added by the entire codec.
+  * This can be queried by the encoder and then the provided number of samples can be
+  * skipped on from the start of the decoder's output to provide time aligned input
+  * and output. From the perspective of a decoding application the real data begins this many
+  * samples late.
+  *
+  * The decoder contribution to this delay is identical for all decoders, but the
+  * encoder portion of the delay may vary from implementation to implementation,
+  * version to version, or even depend on the encoder's initial configuration.
+  * Applications needing delay compensation should call this CTL rather than
+  * hard-coding a value.
+  * @param[out] x <tt>opus_int32 *</tt>:   Number of lookahead samples
+  * @hideinitializer */
+#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of inband forward error correction (FEC).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_INBAND_FEC
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable inband FEC (default).</dd>
+  * <dt>1</dt><dd>Enable inband FEC.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of inband forward error correction.
+  * @see OPUS_SET_INBAND_FEC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Inband FEC disabled (default).</dd>
+  * <dt>1</dt><dd>Inband FEC enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's expected packet loss percentage.
+  * Higher values trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the absence of packet loss, but
+  * greater quality under loss.
+  * @see OPUS_GET_PACKET_LOSS_PERC
+  * @param[in] x <tt>opus_int32</tt>:   Loss percentage in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured packet loss percentage.
+  * @see OPUS_SET_PACKET_LOSS_PERC
+  * @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
+  *                                      in the range 0-100, inclusive (default: 0).
+  * @hideinitializer */
+#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of discontinuous transmission (DTX).
+  * @note This is only applicable to the LPC layer
+  * @see OPUS_GET_DTX
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Disable DTX (default).</dd>
+  * <dt>1</dt><dd>Enabled DTX.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
+/** Gets encoder's configured use of discontinuous transmission.
+  * @see OPUS_SET_DTX
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>DTX disabled (default).</dd>
+  * <dt>1</dt><dd>DTX enabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
+/** Configures the depth of signal being encoded.
+  *
+  * This is a hint which helps the encoder identify silence and near-silence.
+  * It represents the number of significant bits of linear intensity below
+  * which the signal contains ignorable quantization or other noise.
+  *
+  * For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
+  * for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
+  * for 16-bit linear pcm input with opus_encode_float().
+  *
+  * When using opus_encode() instead of opus_encode_float(), or when libopus
+  * is compiled for fixed-point, the encoder uses the minimum of the value
+  * set here and the value 16.
+  *
+  * @see OPUS_GET_LSB_DEPTH
+  * @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
+  *                                   (default: 24).
+  * @hideinitializer */
+#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured signal depth.
+  * @see OPUS_SET_LSB_DEPTH
+  * @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
+  *                                      24 (default: 24).
+  * @hideinitializer */
+#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Configures the encoder's use of variable duration frames.
+  * When variable duration is enabled, the encoder is free to use a shorter frame
+  * size than the one requested in the opus_encode*() call.
+  * It is then the user's responsibility
+  * to verify how much audio was encoded by checking the ToC byte of the encoded
+  * packet. The part of the audio that was not encoded needs to be resent to the
+  * encoder for the next call. Do not use this option unless you <b>really</b>
+  * know what you are doing.
+  * @see OPUS_GET_EXPERT_FRAME_DURATION
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured use of variable duration frames.
+  * @see OPUS_SET_EXPERT_FRAME_DURATION
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
+  * <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
+  * <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables almost all use of prediction, making frames almost
+  * completely independent. This reduces quality.
+  * @see OPUS_GET_PREDICTION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable prediction (default).</dd>
+  * <dt>1</dt><dd>Disable prediction.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured prediction status.
+  * @see OPUS_SET_PREDICTION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Prediction enabled (default).</dd>
+  * <dt>1</dt><dd>Prediction disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_genericctls Generic CTLs
+  *
+  * These macros are used with the \c opus_decoder_ctl and
+  * \c opus_encoder_ctl calls to generate a particular
+  * request.
+  *
+  * When called on an \c OpusDecoder they apply to that
+  * particular decoder instance. When called on an
+  * \c OpusEncoder they apply to the corresponding setting
+  * on that encoder instance, if present.
+  *
+  * Some usage examples:
+  *
+  * @code
+  * int ret;
+  * opus_int32 pitch;
+  * ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
+  * if (ret == OPUS_OK) return ret;
+  *
+  * opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
+  * opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
+  *
+  * opus_int32 enc_bw, dec_bw;
+  * opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
+  * opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
+  * if (enc_bw != dec_bw) {
+  *   printf("packet bandwidth mismatch!\n");
+  * }
+  * @endcode
+  *
+  * @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
+  * @{
+  */
+
+/** Resets the codec state to be equivalent to a freshly initialized state.
+  * This should be called when switching streams in order to prevent
+  * the back to back decoding from giving different results from
+  * one at a time decoding.
+  * @hideinitializer */
+#define OPUS_RESET_STATE 4028
+
+/** Gets the final state of the codec's entropy coder.
+  * This is used for testing purposes,
+  * The encoder and decoder state should be identical after coding a payload
+  * (assuming no data corruption or software bugs)
+  *
+  * @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
+  *
+  * @hideinitializer */
+#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
+
+/** Gets the encoder's configured bandpass or the decoder's last bandpass.
+  * @see OPUS_SET_BANDWIDTH
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>#OPUS_AUTO</dt>                    <dd>(default)</dd>
+  * <dt>#OPUS_BANDWIDTH_NARROWBAND</dt>    <dd>4 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt>    <dd>6 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_WIDEBAND</dt>      <dd>8 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
+  * <dt>#OPUS_BANDWIDTH_FULLBAND</dt>     <dd>20 kHz passband</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the sampling rate the encoder or decoder was initialized with.
+  * This simply returns the <code>Fs</code> value passed to opus_encoder_init()
+  * or opus_decoder_init().
+  * @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
+  * @hideinitializer
+  */
+#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
+
+/** If set to 1, disables the use of phase inversion for intensity stereo,
+  * improving the quality of mono downmixes, but slightly reducing normal
+  * stereo quality. Disabling phase inversion in the decoder does not comply
+  * with RFC 6716, although it does not cause any interoperability issue and
+  * is expected to become part of the Opus standard once RFC 6716 is updated
+  * by draft-ietf-codec-opus-update.
+  * @see OPUS_GET_PHASE_INVERSION_DISABLED
+  * @param[in] x <tt>opus_int32</tt>: Allowed values:
+  * <dl>
+  * <dt>0</dt><dd>Enable phase inversion (default).</dd>
+  * <dt>1</dt><dd>Disable phase inversion.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x)
+/** Gets the encoder's configured phase inversion status.
+  * @see OPUS_SET_PHASE_INVERSION_DISABLED
+  * @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
+  * <dl>
+  * <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
+  * <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
+  * </dl>
+  * @hideinitializer */
+#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_decoderctls Decoder related CTLs
+  * @see opus_genericctls, opus_encoderctls, opus_decoder
+  * @{
+  */
+
+/** Configures decoder gain adjustment.
+  * Scales the decoded output by a factor specified in Q8 dB units.
+  * This has a maximum range of -32768 to 32767 inclusive, and returns
+  * OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
+  * This setting survives decoder reset.
+  *
+  * gain = pow(10, x/(20.0*256))
+  *
+  * @param[in] x <tt>opus_int32</tt>:   Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
+/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
+  * @hideinitializer */
+#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
+  * @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
+  * @hideinitializer */
+#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
+
+/** Gets the pitch of the last decoded frame, if available.
+  * This can be used for any post-processing algorithm requiring the use of pitch,
+  * e.g. time stretching/shortening. If the last frame was not voiced, or if the
+  * pitch was not coded in the frame, then zero is returned.
+  *
+  * This CTL is only implemented for decoder instances.
+  *
+  * @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
+  *
+  * @hideinitializer */
+#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
+
+/**@}*/
+
+/** @defgroup opus_libinfo Opus library information functions
+  * @{
+  */
+
+/** Converts an opus error code into a human readable string.
+  *
+  * @param[in] error <tt>int</tt>: Error number
+  * @returns Error string
+  */
+OPUS_EXPORT const char *opus_strerror(int error);
+
+/** Gets the libopus version string.
+  *
+  * Applications may look for the substring "-fixed" in the version string to
+  * determine whether they have a fixed-point or floating-point build at
+  * runtime.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */

libs/windows/include/opus_multistream.h

@@ -0,0 +1,660 @@
+/* Copyright (c) 2011 Xiph.Org Foundation
+   Written by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+/**
+ * @file opus_multistream.h
+ * @brief Opus reference implementation multistream API
+ */
+
+#ifndef OPUS_MULTISTREAM_H
+#define OPUS_MULTISTREAM_H
+
+#include "opus.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** @cond OPUS_INTERNAL_DOC */
+
+/** Macros to trigger compilation errors when the wrong types are provided to a
+  * CTL. */
+/**@{*/
+#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
+#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
+/**@}*/
+
+/** These are the actual encoder and decoder CTL ID numbers.
+  * They should not be used directly by applications.
+  * In general, SETs should be even and GETs should be odd.*/
+/**@{*/
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
+#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
+/**@}*/
+
+/** @endcond */
+
+/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
+  *
+  * These are convenience macros that are specific to the
+  * opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
+  * interface.
+  * The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
+  * @ref opus_decoderctls may be applied to a multistream encoder or decoder as
+  * well.
+  * In addition, you may retrieve the encoder or decoder state for an specific
+  * stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
+  * #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
+  */
+/**@{*/
+
+/** Gets the encoder state for an individual stream of a multistream encoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the encoder.
+  * @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
+  *                                       encoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
+
+/** Gets the decoder state for an individual stream of a multistream decoder.
+  * @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
+  *                                   wish to retrieve.
+  *                                   This must be non-negative and less than
+  *                                   the <code>streams</code> parameter used
+  *                                   to initialize the decoder.
+  * @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
+  *                                       decoder state.
+  * @retval OPUS_BAD_ARG The index of the requested stream was out of range.
+  * @hideinitializer
+  */
+#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
+
+/**@}*/
+
+/** @defgroup opus_multistream Opus Multistream API
+  * @{
+  *
+  * The multistream API allows individual Opus streams to be combined into a
+  * single packet, enabling support for up to 255 channels. Unlike an
+  * elementary Opus stream, the encoder and decoder must negotiate the channel
+  * configuration before the decoder can successfully interpret the data in the
+  * packets produced by the encoder. Some basic information, such as packet
+  * duration, can be computed without any special negotiation.
+  *
+  * The format for multistream Opus packets is defined in
+  * <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
+  * and is based on the self-delimited Opus framing described in Appendix B of
+  * <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
+  * Normal Opus packets are just a degenerate case of multistream Opus packets,
+  * and can be encoded or decoded with the multistream API by setting
+  * <code>streams</code> to <code>1</code> when initializing the encoder or
+  * decoder.
+  *
+  * Multistream Opus streams can contain up to 255 elementary Opus streams.
+  * These may be either "uncoupled" or "coupled", indicating that the decoder
+  * is configured to decode them to either 1 or 2 channels, respectively.
+  * The streams are ordered so that all coupled streams appear at the
+  * beginning.
+  *
+  * A <code>mapping</code> table defines which decoded channel <code>i</code>
+  * should be used for each input/output (I/O) channel <code>j</code>. This table is
+  * typically provided as an unsigned char array.
+  * Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
+  * If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
+  * encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
+  * is even, or  as the right channel of stream <code>(i/2)</code> if
+  * <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
+  * mono in stream <code>(i - coupled_streams)</code>, unless it has the special
+  * value 255, in which case it is omitted from the encoding entirely (the
+  * decoder will reproduce it as silence). Each value <code>i</code> must either
+  * be the special value 255 or be less than <code>streams + coupled_streams</code>.
+  *
+  * The output channels specified by the encoder
+  * should use the
+  * <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
+  * channel ordering</a>. A decoder may wish to apply an additional permutation
+  * to the mapping the encoder used to achieve a different output channel
+  * order (e.g. for outputing in WAV order).
+  *
+  * Each multistream packet contains an Opus packet for each stream, and all of
+  * the Opus packets in a single multistream packet must have the same
+  * duration. Therefore the duration of a multistream packet can be extracted
+  * from the TOC sequence of the first stream, which is located at the
+  * beginning of the packet, just like an elementary Opus stream:
+  *
+  * @code
+  * int nb_samples;
+  * int nb_frames;
+  * nb_frames = opus_packet_get_nb_frames(data, len);
+  * if (nb_frames < 1)
+  *   return nb_frames;
+  * nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
+  * @endcode
+  *
+  * The general encoding and decoding process proceeds exactly the same as in
+  * the normal @ref opus_encoder and @ref opus_decoder APIs.
+  * See their documentation for an overview of how to use the corresponding
+  * multistream functions.
+  */
+
+/** Opus multistream encoder state.
+  * This contains the complete state of a multistream Opus encoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_init
+  */
+typedef struct OpusMSEncoder OpusMSEncoder;
+
+/** Opus multistream decoder state.
+  * This contains the complete state of a multistream Opus decoder.
+  * It is position independent and can be freely copied.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_decoder_init
+  */
+typedef struct OpusMSDecoder OpusMSDecoder;
+
+/**\name Multistream encoder functions */
+/**@{*/
+
+/** Gets the size of an OpusMSEncoder structure.
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
+      int channels,
+      int mapping_family
+);
+
+
+/** Allocates and initializes a multistream encoder state.
+  * Call opus_multistream_encoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application,
+      int *error
+) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
+
+/** Initialize a previously allocated multistream encoder state.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_encoder_create
+  * @see opus_multistream_encoder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels in the input signal.
+  *                               This must be at most 255.
+  *                               It may be greater than the number of
+  *                               coded channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams to encode from the
+  *                              input.
+  *                              This must be no more than the number of channels.
+  * @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
+  *                                      to encode.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      encoded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than the number of input channels.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    encoded channels to input channels, as described in
+  *                    @ref opus_multistream. As an extra constraint, the
+  *                    multistream encoder does not allow encoding coupled
+  *                    streams for which one channel is unused since this
+  *                    is never a good idea.
+  * @param application <tt>int</tt>: The target encoder application.
+  *                                  This must be one of the following:
+  * <dl>
+  * <dt>#OPUS_APPLICATION_VOIP</dt>
+  * <dd>Process signal for improved speech intelligibility.</dd>
+  * <dt>#OPUS_APPLICATION_AUDIO</dt>
+  * <dd>Favor faithfulness to the original input.</dd>
+  * <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
+  * <dd>Configure the minimum possible coding delay by disabling certain modes
+  * of operation.</dd>
+  * </dl>
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+OPUS_EXPORT int opus_multistream_surround_encoder_init(
+      OpusMSEncoder *st,
+      opus_int32 Fs,
+      int channels,
+      int mapping_family,
+      int *streams,
+      int *coupled_streams,
+      unsigned char *mapping,
+      int application
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7);
+
+/** Encodes a multistream Opus frame.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
+  *                                            samples.
+  *                                            This must contain
+  *                                            <code>frame_size*channels</code>
+  *                                            samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
+    OpusMSEncoder *st,
+    const opus_int16 *pcm,
+    int frame_size,
+    unsigned char *data,
+    opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Encodes a multistream Opus frame from floating point input.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
+  *                                       samples with a normal range of
+  *                                       +/-1.0.
+  *                                       Samples with a range beyond +/-1.0
+  *                                       are supported but will be clipped by
+  *                                       decoders using the integer API and
+  *                                       should only be used if it is known
+  *                                       that the far end supports extended
+  *                                       dynamic range.
+  *                                       This must contain
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: Number of samples per channel in the input
+  *                                 signal.
+  *                                 This must be an Opus frame size for the
+  *                                 encoder's sampling rate.
+  *                                 For example, at 48 kHz the permitted values
+  *                                 are 120, 240, 480, 960, 1920, and 2880.
+  *                                 Passing in a duration of less than 10 ms
+  *                                 (480 samples at 48 kHz) will prevent the
+  *                                 encoder from using the LPC or hybrid modes.
+  * @param[out] data <tt>unsigned char*</tt>: Output payload.
+  *                                           This must contain storage for at
+  *                                           least \a max_data_bytes.
+  * @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
+  *                                                 memory for the output
+  *                                                 payload. This may be
+  *                                                 used to impose an upper limit on
+  *                                                 the instant bitrate, but should
+  *                                                 not be used as the only bitrate
+  *                                                 control. Use #OPUS_SET_BITRATE to
+  *                                                 control the bitrate.
+  * @returns The length of the encoded packet (in bytes) on success or a
+  *          negative error code (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
+      OpusMSEncoder *st,
+      const float *pcm,
+      int frame_size,
+      unsigned char *data,
+      opus_int32 max_data_bytes
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
+
+/** Frees an <code>OpusMSEncoder</code> allocated by
+  * opus_multistream_encoder_create().
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
+
+/** Perform a CTL function on a multistream Opus encoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_encoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_encoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/**@}*/
+
+/**\name Multistream decoder functions */
+/**@{*/
+
+/** Gets the size of an <code>OpusMSDecoder</code> structure.
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @returns The size in bytes on success, or a negative error code
+  *          (see @ref opus_errorcodes) on error.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
+      int streams,
+      int coupled_streams
+);
+
+/** Allocates and initializes a multistream decoder state.
+  * Call opus_multistream_decoder_destroy() to release
+  * this object when finished.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
+  *                                   code (see @ref opus_errorcodes) on
+  *                                   failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping,
+      int *error
+) OPUS_ARG_NONNULL(5);
+
+/** Intialize a previously allocated decoder state object.
+  * The memory pointed to by \a st must be at least the size returned by
+  * opus_multistream_encoder_get_size().
+  * This is intended for applications which use their own allocator instead of
+  * malloc.
+  * To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
+  * @see opus_multistream_decoder_create
+  * @see opus_multistream_deocder_get_size
+  * @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
+  * @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
+  *                                This must be one of 8000, 12000, 16000,
+  *                                24000, or 48000.
+  * @param channels <tt>int</tt>: Number of channels to output.
+  *                               This must be at most 255.
+  *                               It may be different from the number of coded
+  *                               channels (<code>streams +
+  *                               coupled_streams</code>).
+  * @param streams <tt>int</tt>: The total number of streams coded in the
+  *                              input.
+  *                              This must be no more than 255.
+  * @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
+  *                                      (2 channel) streams.
+  *                                      This must be no larger than the total
+  *                                      number of streams.
+  *                                      Additionally, The total number of
+  *                                      coded channels (<code>streams +
+  *                                      coupled_streams</code>) must be no
+  *                                      more than 255.
+  * @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
+  *                    coded channels to output channels, as described in
+  *                    @ref opus_multistream.
+  * @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
+  *          on failure.
+  */
+OPUS_EXPORT int opus_multistream_decoder_init(
+      OpusMSDecoder *st,
+      opus_int32 Fs,
+      int channels,
+      int streams,
+      int coupled_streams,
+      const unsigned char *mapping
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
+
+/** Decode a multistream Opus packet.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    opus_int16 *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Decode a multistream Opus packet with floating point output.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param[in] data <tt>const unsigned char*</tt>: Input payload.
+  *                                                Use a <code>NULL</code>
+  *                                                pointer to indicate packet
+  *                                                loss.
+  * @param len <tt>opus_int32</tt>: Number of bytes in payload.
+  * @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
+  *                                       samples.
+  *                                       This must contain room for
+  *                                       <code>frame_size*channels</code>
+  *                                       samples.
+  * @param frame_size <tt>int</tt>: The number of samples per channel of
+  *                                 available space in \a pcm.
+  *                                 If this is less than the maximum packet duration
+  *                                 (120 ms; 5760 for 48kHz), this function will not be capable
+  *                                 of decoding some packets. In the case of PLC (data==NULL)
+  *                                 or FEC (decode_fec=1), then frame_size needs to be exactly
+  *                                 the duration of audio that is missing, otherwise the
+  *                                 decoder will not be in the optimal state to decode the
+  *                                 next incoming packet. For the PLC and FEC cases, frame_size
+  *                                 <b>must</b> be a multiple of 2.5 ms.
+  * @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
+  *                                 forward error correction data be decoded.
+  *                                 If no such data is available, the frame is
+  *                                 decoded as if it were lost.
+  * @returns Number of samples decoded on success or a negative error code
+  *          (see @ref opus_errorcodes) on failure.
+  */
+OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
+    OpusMSDecoder *st,
+    const unsigned char *data,
+    opus_int32 len,
+    float *pcm,
+    int frame_size,
+    int decode_fec
+) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
+
+/** Perform a CTL function on a multistream Opus decoder.
+  *
+  * Generally the request and subsequent arguments are generated by a
+  * convenience macro.
+  * @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
+  * @param request This and all remaining parameters should be replaced by one
+  *                of the convenience macros in @ref opus_genericctls,
+  *                @ref opus_decoderctls, or @ref opus_multistream_ctls.
+  * @see opus_genericctls
+  * @see opus_decoderctls
+  * @see opus_multistream_ctls
+  */
+OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
+
+/** Frees an <code>OpusMSDecoder</code> allocated by
+  * opus_multistream_decoder_create().
+  * @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
+  */
+OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
+
+/**@}*/
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_MULTISTREAM_H */

libs/windows/include/opus_types.h

@@ -0,0 +1,159 @@
+/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
+/* Modified by Jean-Marc Valin */
+/*
+   Redistribution and use in source and binary forms, with or without
+   modification, are permitted provided that the following conditions
+   are met:
+
+   - Redistributions of source code must retain the above copyright
+   notice, this list of conditions and the following disclaimer.
+
+   - Redistributions in binary form must reproduce the above copyright
+   notice, this list of conditions and the following disclaimer in the
+   documentation and/or other materials provided with the distribution.
+
+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+   A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
+   OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+/* opus_types.h based on ogg_types.h from libogg */
+
+/**
+   @file opus_types.h
+   @brief Opus reference implementation types
+*/
+#ifndef OPUS_TYPES_H
+#define OPUS_TYPES_H
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+#elif defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <_G_config.h>
+     typedef _G_int32_t opus_int32;
+     typedef _G_uint32_t opus_uint32;
+     typedef _G_int16 opus_int16;
+     typedef _G_uint16 opus_uint16;
+#  elif defined(__MINGW32__)
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+#  elif defined(__MWERKS__)
+     typedef int opus_int32;
+     typedef unsigned int opus_uint32;
+     typedef short opus_int16;
+     typedef unsigned short opus_uint16;
+#  else
+     /* MSVC/Borland */
+     typedef __int32 opus_int32;
+     typedef unsigned __int32 opus_uint32;
+     typedef __int16 opus_int16;
+     typedef unsigned __int16 opus_uint16;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 opus_int16;
+   typedef UInt16 opus_uint16;
+   typedef SInt32 opus_int32;
+   typedef UInt32 opus_uint32;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <sys/types.h>
+   typedef int16_t opus_int16;
+   typedef u_int16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16 opus_int16;
+   typedef u_int16 opus_uint16;
+   typedef int32_t opus_int32;
+   typedef u_int32_t opus_uint32;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef int opus_int32;
+   typedef unsigned opus_uint32;
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef signed int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef long opus_int32;
+   typedef unsigned long opus_uint32;
+
+#elif defined(CONFIG_TI_C6X)
+
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#else
+
+   /* Give up, take a reasonable guess */
+   typedef short opus_int16;
+   typedef unsigned short opus_uint16;
+   typedef int opus_int32;
+   typedef unsigned int opus_uint32;
+
+#endif
+
+#define opus_int         int                     /* used for counters etc; at least 16 bits */
+#define opus_int64       long long
+#define opus_int8        signed char
+
+#define opus_uint        unsigned int            /* used for counters etc; at least 16 bits */
+#define opus_uint64      unsigned long long
+#define opus_uint8       unsigned char
+
+#endif  /* OPUS_TYPES_H */

moonlight-qt.pro

@@ -1,7 +1,6 @@
 TEMPLATE = subdirs
 SUBDIRS = \
     moonlight-common-c \
-    opus \
     qmdnsengine \
     app
 

opus/opus.pro

@@ -1,272 +0,0 @@
-QT -= core gui
-
-TARGET = opus
-TEMPLATE = lib
-
-# Support debug and release builds from command line for CI
-CONFIG += debug_and_release
-
-OPUS_DIR = $$PWD/opus
-DEFINES += \
-    USE_ALLOCA=1 \
-    HAVE_LRINT=1 \
-    HAVE_LRINTF=1 \
-    OPUS_HAVE_RTCD=1 \
-    OPUS_BUILD=1
-
-contains(QT_ARCH, i386) {
-    CONFIG += x86
-    DEFINES += \
-        OPUS_X86_MAY_HAVE_SSE=1 \
-        OPUS_X86_MAY_HAVE_SSE2=1 \
-        OPUS_X86_MAY_HAVE_SSE4_1=1 \
-        OPUS_X86_MAY_HAVE_AVX=1
-}
-
-# AMD64 implies SSE2 support
-contains(QT_ARCH, x86_64) {
-    CONFIG += x86
-    DEFINES += \
-        OPUS_X86_MAY_HAVE_SSE=1 \
-        OPUS_X86_MAY_HAVE_SSE2=1 \
-        OPUS_X86_MAY_HAVE_SSE4_1=1 \
-        OPUS_X86_MAY_HAVE_AVX=1 \
-        OPUS_X86_PRESUME_SSE=1 \
-        OPUS_X86_PRESUME_SSE2=1
-}
-
-contains(QT_ARCH, arm) {
-    CONFIG += arm
-    DEFINES += \
-        OPUS_ARM_MAY_HAVE_NEON_INTR=1
-}
-
-QMAKE_CFLAGS += -O2
-
-x86 {
-    SOURCES += \
-        $$OPUS_DIR/celt/x86/x86_celt_map.c \
-        $$OPUS_DIR/silk/x86/x86_silk_map.c \
-        $$OPUS_DIR/celt/x86/x86cpu.c
-
-    SOURCES_SSE = \
-        $$OPUS_DIR/celt/x86/pitch_sse.c
-
-    SOURCES_SSE2 = \
-        $$OPUS_DIR/celt/x86/pitch_sse2.c \
-        $$OPUS_DIR/celt/x86/vq_sse2.c
-
-    SOURCES_SSE41 = \
-        $$OPUS_DIR/celt/x86/pitch_sse4_1.c \
-        $$OPUS_DIR/celt/x86/celt_lpc_sse4_1.c \
-        $$OPUS_DIR/silk/x86/NSQ_sse4_1.c \
-        $$OPUS_DIR/silk/x86/VAD_sse4_1.c \
-        $$OPUS_DIR/silk/x86/NSQ_del_dec_sse4_1.c \
-        $$OPUS_DIR/silk/x86/VQ_WMat_EC_sse4_1.c
-}
-
-arm {
-    SOURCES += \
-        $$OPUS_DIR/celt/arm/arm_celt_map.c \
-        $$OPUS_DIR/celt/arm/armcpu.c
-
-    SOURCES_NEON = \
-        $$OPUS_DIR/celt/arm/celt_neon_intr.c \
-        $$OPUS_DIR/celt/arm/pitch_neon_intr.c \
-        $$OPUS_DIR/silk/arm/arm_silk_map.c \
-        $$OPUS_DIR/silk/arm/biquad_alt_neon_intr.c \
-        $$OPUS_DIR/silk/arm/LPC_inv_pred_gain_neon_intr.c \
-        $$OPUS_DIR/silk/arm/NSQ_del_dec_neon_intr.c \
-        $$OPUS_DIR/silk/arm/NSQ_neon.c
-}
-
-SOURCES += \
-    $$OPUS_DIR/celt/quant_bands.c \
-    $$OPUS_DIR/celt/laplace.c \
-    $$OPUS_DIR/celt/rate.c \
-    $$OPUS_DIR/celt/celt_decoder.c \
-    $$OPUS_DIR/celt/celt_encoder.c \
-    $$OPUS_DIR/celt/modes.c \
-    $$OPUS_DIR/celt/entenc.c \
-    $$OPUS_DIR/celt/entcode.c \
-    $$OPUS_DIR/celt/cwrs.c \
-    $$OPUS_DIR/celt/pitch.c \
-    $$OPUS_DIR/celt/entdec.c \
-    $$OPUS_DIR/celt/mathops.c \
-    $$OPUS_DIR/celt/vq.c \
-    $$OPUS_DIR/celt/bands.c \
-    $$OPUS_DIR/celt/celt.c \
-    $$OPUS_DIR/celt/celt_lpc.c \
-    $$OPUS_DIR/celt/kiss_fft.c \
-    $$OPUS_DIR/celt/mdct.c \
-    $$OPUS_DIR/src/opus_decoder.c \
-    $$OPUS_DIR/src/opus_compare.c \
-    $$OPUS_DIR/src/opus_multistream_encoder.c \
-    $$OPUS_DIR/src/opus.c \
-    $$OPUS_DIR/src/mapping_matrix.c \
-    $$OPUS_DIR/src/mlp.c \
-    $$OPUS_DIR/src/opus_multistream_decoder.c \
-    $$OPUS_DIR/src/analysis.c \
-    $$OPUS_DIR/src/opus_projection_encoder.c \
-    $$OPUS_DIR/src/repacketizer.c \
-    $$OPUS_DIR/src/mlp_data.c \
-    $$OPUS_DIR/src/opus_projection_decoder.c \
-    $$OPUS_DIR/src/opus_multistream.c \
-    $$OPUS_DIR/src/opus_encoder.c \
-    $$OPUS_DIR/silk/NSQ_del_dec.c \
-    $$OPUS_DIR/silk/ana_filt_bank_1.c \
-    $$OPUS_DIR/silk/stereo_find_predictor.c \
-    $$OPUS_DIR/silk/resampler.c \
-    $$OPUS_DIR/silk/HP_variable_cutoff.c \
-    $$OPUS_DIR/silk/shell_coder.c \
-    $$OPUS_DIR/silk/inner_prod_aligned.c \
-    $$OPUS_DIR/silk/tables_gain.c \
-    $$OPUS_DIR/silk/bwexpander_32.c \
-    $$OPUS_DIR/silk/stereo_quant_pred.c \
-    $$OPUS_DIR/silk/gain_quant.c \
-    $$OPUS_DIR/silk/debug.c \
-    $$OPUS_DIR/silk/decoder_set_fs.c \
-    $$OPUS_DIR/silk/LPC_inv_pred_gain.c \
-    $$OPUS_DIR/silk/tables_pulses_per_block.c \
-    $$OPUS_DIR/silk/NLSF_VQ.c \
-    $$OPUS_DIR/silk/control_audio_bandwidth.c \
-    $$OPUS_DIR/silk/tables_LTP.c \
-    $$OPUS_DIR/silk/dec_API.c \
-    $$OPUS_DIR/silk/tables_NLSF_CB_WB.c \
-    $$OPUS_DIR/silk/bwexpander.c \
-    $$OPUS_DIR/silk/tables_NLSF_CB_NB_MB.c \
-    $$OPUS_DIR/silk/LP_variable_cutoff.c \
-    $$OPUS_DIR/silk/float/LPC_inv_pred_gain_FLP.c \
-    $$OPUS_DIR/silk/float/schur_FLP.c \
-    $$OPUS_DIR/silk/float/wrappers_FLP.c \
-    $$OPUS_DIR/silk/float/autocorrelation_FLP.c \
-    $$OPUS_DIR/silk/float/regularize_correlations_FLP.c \
-    $$OPUS_DIR/silk/float/warped_autocorrelation_FLP.c \
-    $$OPUS_DIR/silk/float/k2a_FLP.c \
-    $$OPUS_DIR/silk/float/find_LPC_FLP.c \
-    $$OPUS_DIR/silk/float/noise_shape_analysis_FLP.c \
-    $$OPUS_DIR/silk/float/scale_vector_FLP.c \
-    $$OPUS_DIR/silk/float/apply_sine_window_FLP.c \
-    $$OPUS_DIR/silk/float/bwexpander_FLP.c \
-    $$OPUS_DIR/silk/float/inner_product_FLP.c \
-    $$OPUS_DIR/silk/float/scale_copy_vector_FLP.c \
-    $$OPUS_DIR/silk/float/energy_FLP.c \
-    $$OPUS_DIR/silk/float/pitch_analysis_core_FLP.c \
-    $$OPUS_DIR/silk/float/process_gains_FLP.c \
-    $$OPUS_DIR/silk/float/residual_energy_FLP.c \
-    $$OPUS_DIR/silk/float/find_pred_coefs_FLP.c \
-    $$OPUS_DIR/silk/float/encode_frame_FLP.c \
-    $$OPUS_DIR/silk/float/find_LTP_FLP.c \
-    $$OPUS_DIR/silk/float/find_pitch_lags_FLP.c \
-    $$OPUS_DIR/silk/float/burg_modified_FLP.c \
-    $$OPUS_DIR/silk/float/corrMatrix_FLP.c \
-    $$OPUS_DIR/silk/float/LTP_scale_ctrl_FLP.c \
-    $$OPUS_DIR/silk/float/sort_FLP.c \
-    $$OPUS_DIR/silk/float/LPC_analysis_filter_FLP.c \
-    $$OPUS_DIR/silk/float/LTP_analysis_filter_FLP.c \
-    $$OPUS_DIR/silk/stereo_MS_to_LR.c \
-    $$OPUS_DIR/silk/log2lin.c \
-    $$OPUS_DIR/silk/resampler_private_up2_HQ.c \
-    $$OPUS_DIR/silk/resampler_rom.c \
-    $$OPUS_DIR/silk/A2NLSF.c \
-    $$OPUS_DIR/silk/NLSF_decode.c \
-    $$OPUS_DIR/silk/resampler_down2.c \
-    $$OPUS_DIR/silk/NLSF2A.c \
-    $$OPUS_DIR/silk/table_LSF_cos.c \
-    $$OPUS_DIR/silk/control_codec.c \
-    $$OPUS_DIR/silk/pitch_est_tables.c \
-    $$OPUS_DIR/silk/NLSF_unpack.c \
-    $$OPUS_DIR/silk/resampler_private_down_FIR.c \
-    $$OPUS_DIR/silk/decode_frame.c \
-    $$OPUS_DIR/silk/PLC.c \
-    $$OPUS_DIR/silk/check_control_input.c \
-    $$OPUS_DIR/silk/decode_indices.c \
-    $$OPUS_DIR/silk/encode_pulses.c \
-    $$OPUS_DIR/silk/LPC_fit.c \
-    $$OPUS_DIR/silk/enc_API.c \
-    $$OPUS_DIR/silk/init_decoder.c \
-    $$OPUS_DIR/silk/biquad_alt.c \
-    $$OPUS_DIR/silk/resampler_private_IIR_FIR.c \
-    $$OPUS_DIR/silk/resampler_private_AR2.c \
-    $$OPUS_DIR/silk/interpolate.c \
-    $$OPUS_DIR/silk/stereo_encode_pred.c \
-    $$OPUS_DIR/silk/sort.c \
-    $$OPUS_DIR/silk/process_NLSFs.c \
-    $$OPUS_DIR/silk/tables_other.c \
-    $$OPUS_DIR/silk/encode_indices.c \
-    $$OPUS_DIR/silk/NLSF_encode.c \
-    $$OPUS_DIR/silk/VQ_WMat_EC.c \
-    $$OPUS_DIR/silk/NSQ.c \
-    $$OPUS_DIR/silk/NLSF_stabilize.c \
-    $$OPUS_DIR/silk/CNG.c \
-    $$OPUS_DIR/silk/sum_sqr_shift.c \
-    $$OPUS_DIR/silk/tables_pitch_lag.c \
-    $$OPUS_DIR/silk/NLSF_del_dec_quant.c \
-    $$OPUS_DIR/silk/init_encoder.c \
-    $$OPUS_DIR/silk/decode_core.c \
-    $$OPUS_DIR/silk/decode_pulses.c \
-    $$OPUS_DIR/silk/stereo_decode_pred.c \
-    $$OPUS_DIR/silk/VAD.c \
-    $$OPUS_DIR/silk/decode_pitch.c \
-    $$OPUS_DIR/silk/NLSF_VQ_weights_laroia.c \
-    $$OPUS_DIR/silk/control_SNR.c \
-    $$OPUS_DIR/silk/code_signs.c \
-    $$OPUS_DIR/silk/quant_LTP_gains.c \
-    $$OPUS_DIR/silk/sigm_Q15.c \
-    $$OPUS_DIR/silk/lin2log.c \
-    $$OPUS_DIR/silk/decode_parameters.c \
-    $$OPUS_DIR/silk/resampler_down2_3.c \
-    $$OPUS_DIR/silk/stereo_LR_to_MS.c \
-    $$OPUS_DIR/silk/LPC_analysis_filter.c
-
-INCLUDEPATH += \
-    $$OPUS_DIR            \
-    $$OPUS_DIR/include    \
-    $$OPUS_DIR/src        \
-    $$OPUS_DIR/silk       \
-    $$OPUS_DIR/celt       \
-    $$OPUS_DIR/silk/fixed \
-    $$OPUS_DIR/silk/float
-
-CONFIG += warn_off staticlib
-
-win32-msvc* {
-    # No flags required to build with SSE or NEON on MSVC
-    x86 {
-        SOURCES += $$SOURCES_SSE $$SOURCES_SSE2 $$SOURCES_SSE41
-    }
-    arm {
-        SOURCES += $$SOURCES_NEON
-    }
-}
-else {
-    sse.name = sse
-    sse.input = SOURCES_SSE
-    sse.dependency_type = TYPE_C
-    sse.variable_out = OBJECTS
-    sse.output = ${QMAKE_VAR_OBJECTS_DIR}${QMAKE_FILE_IN_BASE}$${first(QMAKE_EXT_OBJ)}
-    sse.commands = $${QMAKE_CC} -c $(CFLAGS) -msse $(INCPATH) -o ${QMAKE_FILE_OUT} ${QMAKE_FILE_IN}
-
-    sse2.name = sse2
-    sse2.input = SOURCES_SSE2
-    sse2.dependency_type = TYPE_C
-    sse2.variable_out = OBJECTS
-    sse2.output = ${QMAKE_VAR_OBJECTS_DIR}${QMAKE_FILE_IN_BASE}$${first(QMAKE_EXT_OBJ)}
-    sse2.commands = $${QMAKE_CC} -c $(CFLAGS) -msse2 $(INCPATH) -o ${QMAKE_FILE_OUT} ${QMAKE_FILE_IN}
-
-    sse41.name = sse41
-    sse41.input = SOURCES_SSE41
-    sse41.dependency_type = TYPE_C
-    sse41.variable_out = OBJECTS
-    sse41.output = ${QMAKE_VAR_OBJECTS_DIR}${QMAKE_FILE_IN_BASE}$${first(QMAKE_EXT_OBJ)}
-    sse41.commands = $${QMAKE_CC} -c $(CFLAGS) -msse4.1 $(INCPATH) -o ${QMAKE_FILE_OUT} ${QMAKE_FILE_IN}
-
-    neon.name = neon
-    neon.input = SOURCES_NEON
-    neon.dependency_type = TYPE_C
-    neon.variable_out = OBJECTS
-    neon.output = ${QMAKE_VAR_OBJECTS_DIR}${QMAKE_FILE_IN_BASE}$${first(QMAKE_EXT_OBJ)}
-    neon.commands = $${QMAKE_CC} -c $(CFLAGS) $(INCPATH) -o ${QMAKE_FILE_OUT} ${QMAKE_FILE_IN}
-
-    QMAKE_EXTRA_COMPILERS += sse sse2 sse41 neon
-}

scripts/build-ffmpeg-mac.sh

@@ -0,0 +1,3 @@
+./configure --extra-cflags="-mmacosx-version-min=10.10" --enable-pic --enable-shared --disable-static --disable-all --enable-avcodec --enable-decoder=h264 --enable-decoder=hevc --enable-hwaccel=h264_videotoolbox --enable-hwaccel=hevc_videotoolbox
+make clean
+make -j4

scripts/build-ffmpeg-win32.sh

@@ -0,0 +1,3 @@
+./configure --toolchain=msvc --enable-cross-compile --enable-pic --enable-shared --disable-static --disable-all --enable-avcodec --enable-decoder=h264 --enable-decoder=hevc --enable-hwaccel=h264_dxva2 --enable-hwaccel=hevc_dxva2
+make clean
+make -j4

scripts/build-ffmpeg-win64.sh

@@ -0,0 +1,3 @@
+./configure --toolchain=msvc --enable-pic --enable-shared --disable-static --disable-all --enable-avcodec --enable-decoder=h264 --enable-decoder=hevc --enable-hwaccel=h264_dxva2 --enable-hwaccel=hevc_dxva2
+make clean
+make -j4

scripts/generate-installers.bat

@@ -13,8 +13,13 @@ if /I "%BUILD_CONFIG%"=="debug" (
     if /I "%BUILD_CONFIG%"=="release" (
         set BUILD_CONFIG=release
     ) else (
-        echo Invalid build configuration
-        exit /b 1
+        if /I "%BUILD_CONFIG%"=="signed-release" (
+            set BUILD_CONFIG=release
+            set SIGN=1
+        ) else (
+            echo Invalid build configuration
+            exit /b 1
+        )
     )
 )
 
@@ -26,6 +31,7 @@ if /I "%ARCH%" NEQ "x86" (
 )
 
 set VS_PATH=%ProgramFiles(x86)%\Microsoft Visual Studio\2017\Community
+set SIGNTOOL_PARAMS=sign /tr http://timestamp.digicert.com /td sha256 /fd sha256 /sha1 1B3C676E831A94EC0327C3347EB32C68C26B3A67 /v
 
 call "%VS_PATH%\VC\Auxiliary\Build\vcvarsall.bat" %ARCH%
 if !ERRORLEVEL! NEQ 0 goto Error
@@ -61,27 +67,47 @@ echo Copying DLL dependencies
 copy %SOURCE_ROOT%\libs\windows\lib\%ARCH%\*.dll %DEPLOY_FOLDER%
 if !ERRORLEVEL! NEQ 0 goto Error
 
-echo Deploying Qt dependencies 
-windeployqt.exe --dir %DEPLOY_FOLDER% --%BUILD_CONFIG% --qmldir %SOURCE_ROOT%\app\gui --no-angle --no-opengl-sw --no-compiler-runtime %BUILD_FOLDER%\app\%BUILD_CONFIG%\Moonlight.exe
+echo Deploying Qt dependencies
+windeployqt.exe --dir %DEPLOY_FOLDER% --%BUILD_CONFIG% --qmldir %SOURCE_ROOT%\app\gui --no-opengl-sw --no-compiler-runtime %BUILD_FOLDER%\app\%BUILD_CONFIG%\Moonlight.exe
 if !ERRORLEVEL! NEQ 0 goto Error
 
 echo Harvesting files for WiX
-heat dir %DEPLOY_FOLDER% -srd -sfrag -ag -sw5150 -cg MoonlightDependencies -var var.SourceDir -dr INSTALLFOLDER -out %BUILD_FOLDER%\Dependencies.wxs
+"%WIX%\bin\heat" dir %DEPLOY_FOLDER% -srd -sfrag -ag -sw5150 -cg MoonlightDependencies -var var.SourceDir -dr INSTALLFOLDER -out %BUILD_FOLDER%\Dependencies.wxs
 if !ERRORLEVEL! NEQ 0 goto Error
 
 echo Copying application binary to deployment directory
 copy %BUILD_FOLDER%\app\%BUILD_CONFIG%\Moonlight.exe %DEPLOY_FOLDER%
 if !ERRORLEVEL! NEQ 0 goto Error
 
+if "%SIGN%"=="1" (
+    echo Signing deployed binaries
+    for /r "%DEPLOY_FOLDER%" %%f in (*.dll *.exe) do (
+        signtool %SIGNTOOL_PARAMS% %%f
+        if !ERRORLEVEL! NEQ 0 goto Error
+    )
+)
+
 echo Building installer
 set VCREDIST_INSTALLER=%VS_PATH%\VC\Redist\MSVC\14.14.26405\vcredist_%ARCH%.exe
 msbuild %SOURCE_ROOT%\wix\Moonlight.sln /p:Configuration=%BUILD_CONFIG% /p:Platform=%ARCH%
 if !ERRORLEVEL! NEQ 0 goto Error
 
+if "%SIGN%"=="1" (
+    echo Signing installer binary
+    "%WIX%\bin\insignia" -ib %INSTALLER_FOLDER%\MoonlightSetup.exe -o %BUILD_FOLDER%\engine.exe
+    if !ERRORLEVEL! NEQ 0 goto Error
+    signtool %SIGNTOOL_PARAMS% %BUILD_FOLDER%\engine.exe
+    if !ERRORLEVEL! NEQ 0 goto Error
+    "%WIX%\bin\insignia" -ab %BUILD_FOLDER%\engine.exe %INSTALLER_FOLDER%\MoonlightSetup.exe -o %INSTALLER_FOLDER%\MoonlightSetup.exe
+    if !ERRORLEVEL! NEQ 0 goto Error
+    signtool %SIGNTOOL_PARAMS% %INSTALLER_FOLDER%\MoonlightSetup.exe
+    if !ERRORLEVEL! NEQ 0 goto Error
+)
+
 if /i "%APPVEYOR%"=="true" (
     echo Pushing artifacts
-    appveyor PushArtifact %DEPLOY_FOLDER%
     appveyor PushArtifact %INSTALLER_FOLDER%\MoonlightSetup.exe -FileName MoonlightSetup-%ARCH%-%BUILD_CONFIG%.exe
+    if !ERRORLEVEL! NEQ 0 goto Error
 )
 
 echo Build successful!