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This commit is contained in:
trilene 2020-08-01 14:31:10 -04:00
parent f14d141cb5
commit e3e7595bab
13 changed files with 918 additions and 852 deletions
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123
src/ActiveCallBar.cpp
View File

@ -33,8 +33,7 @@ ActiveCallBar::ActiveCallBar(QWidget *parent)
layout_ = new QHBoxLayout(this);
layout_->setSpacing(widgetMargin);
layout_->setContentsMargins(
2 * widgetMargin, widgetMargin, 2 * widgetMargin, widgetMargin);
layout_->setContentsMargins(2 * widgetMargin, widgetMargin, 2 * widgetMargin, widgetMargin);
QFont labelFont;
labelFont.setPointSizeF(labelFont.pointSizeF() * 1.1);
@ -56,9 +55,9 @@ ActiveCallBar::ActiveCallBar(QWidget *parent)
setMuteIcon(false);
muteBtn_->setFixedSize(buttonSize_, buttonSize_);
muteBtn_->setCornerRadius(buttonSize_ / 2);
connect(muteBtn_, &FlatButton::clicked, this, [this](){
connect(muteBtn_, &FlatButton::clicked, this, [this]() {
if (WebRTCSession::instance().toggleMuteAudioSrc(muted_))
setMuteIcon(muted_);
setMuteIcon(muted_);
});
layout_->addWidget(avatar_, 0, Qt::AlignLeft);
@ -70,21 +69,21 @@ ActiveCallBar::ActiveCallBar(QWidget *parent)
layout_->addSpacing(18);
timer_ = new QTimer(this);
connect(timer_, &QTimer::timeout, this,
[this](){
auto seconds = QDateTime::currentSecsSinceEpoch() - callStartTime_;
int s = seconds % 60;
int m = (seconds / 60) % 60;
int h = seconds / 3600;
char buf[12];
if (h)
snprintf(buf, sizeof(buf), "%.2d:%.2d:%.2d", h, m, s);
else
snprintf(buf, sizeof(buf), "%.2d:%.2d", m, s);
durationLabel_->setText(buf);
connect(timer_, &QTimer::timeout, this, [this]() {
auto seconds = QDateTime::currentSecsSinceEpoch() - callStartTime_;
int s = seconds % 60;
int m = (seconds / 60) % 60;
int h = seconds / 3600;
char buf[12];
if (h)
snprintf(buf, sizeof(buf), "%.2d:%.2d:%.2d", h, m, s);
else
snprintf(buf, sizeof(buf), "%.2d:%.2d", m, s);
durationLabel_->setText(buf);
});
connect(&WebRTCSession::instance(), &WebRTCSession::stateChanged, this, &ActiveCallBar::update);
connect(
&WebRTCSession::instance(), &WebRTCSession::stateChanged, this, &ActiveCallBar::update);
}
void
@ -103,61 +102,59 @@ ActiveCallBar::setMuteIcon(bool muted)
}
void
ActiveCallBar::setCallParty(
const QString &userid,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl)
ActiveCallBar::setCallParty(const QString &userid,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl)
{
callPartyLabel_->setText(" " +
(displayName.isEmpty() ? userid : displayName) + " ");
callPartyLabel_->setText(" " + (displayName.isEmpty() ? userid : displayName) + " ");
if (!avatarUrl.isEmpty())
avatar_->setImage(avatarUrl);
avatar_->setImage(avatarUrl);
else
avatar_->setLetter(utils::firstChar(roomName));
avatar_->setLetter(utils::firstChar(roomName));
}
void
ActiveCallBar::update(WebRTCSession::State state)
{
switch (state) {
case WebRTCSession::State::INITIATING:
show();
stateLabel_->setText("Initiating call...");
break;
case WebRTCSession::State::INITIATED:
show();
stateLabel_->setText("Call initiated...");
break;
case WebRTCSession::State::OFFERSENT:
show();
stateLabel_->setText("Calling...");
break;
case WebRTCSession::State::CONNECTING:
show();
stateLabel_->setText("Connecting...");
break;
case WebRTCSession::State::CONNECTED:
show();
callStartTime_ = QDateTime::currentSecsSinceEpoch();
timer_->start(1000);
stateLabel_->setPixmap(QIcon(":/icons/icons/ui/place-call.png").
pixmap(QSize(buttonSize_, buttonSize_)));
durationLabel_->setText("00:00");
durationLabel_->show();
break;
case WebRTCSession::State::ICEFAILED:
case WebRTCSession::State::DISCONNECTED:
hide();
timer_->stop();
callPartyLabel_->setText(QString());
stateLabel_->setText(QString());
durationLabel_->setText(QString());
durationLabel_->hide();
setMuteIcon(false);
break;
default:
break;
case WebRTCSession::State::INITIATING:
show();
stateLabel_->setText("Initiating call...");
break;
case WebRTCSession::State::INITIATED:
show();
stateLabel_->setText("Call initiated...");
break;
case WebRTCSession::State::OFFERSENT:
show();
stateLabel_->setText("Calling...");
break;
case WebRTCSession::State::CONNECTING:
show();
stateLabel_->setText("Connecting...");
break;
case WebRTCSession::State::CONNECTED:
show();
callStartTime_ = QDateTime::currentSecsSinceEpoch();
timer_->start(1000);
stateLabel_->setPixmap(
QIcon(":/icons/icons/ui/place-call.png").pixmap(QSize(buttonSize_, buttonSize_)));
durationLabel_->setText("00:00");
durationLabel_->show();
break;
case WebRTCSession::State::ICEFAILED:
case WebRTCSession::State::DISCONNECTED:
hide();
timer_->stop();
callPartyLabel_->setText(QString());
stateLabel_->setText(QString());
durationLabel_->setText(QString());
durationLabel_->hide();
setMuteIcon(false);
break;
default:
break;
}
}

9
src/ActiveCallBar.h
View File

@ -19,11 +19,10 @@ public:
public slots:
void update(WebRTCSession::State);
void setCallParty(
const QString &userid,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl);
void setCallParty(const QString &userid,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl);
private:
QHBoxLayout *layout_ = nullptr;

569
src/CallManager.cpp
View File

@ -1,13 +1,13 @@
#include <algorithm>
#include <cctype>
#include <cstdint>
#include <chrono>
#include <cstdint>
#include <QMediaPlaylist>
#include <QUrl>
#include "CallManager.h"
#include "Cache.h"
#include "CallManager.h"
#include "ChatPage.h"
#include "Logging.h"
#include "MainWindow.h"
@ -34,389 +34,420 @@ getTurnURIs(const mtx::responses::TurnServer &turnServer);
}
CallManager::CallManager(QSharedPointer<UserSettings> userSettings)
: QObject(),
session_(WebRTCSession::instance()),
turnServerTimer_(this),
settings_(userSettings)
: QObject()
, session_(WebRTCSession::instance())
, turnServerTimer_(this)
, settings_(userSettings)
{
qRegisterMetaType<std::vector<mtx::events::msg::CallCandidates::Candidate>>();
qRegisterMetaType<mtx::events::msg::CallCandidates::Candidate>();
qRegisterMetaType<mtx::responses::TurnServer>();
qRegisterMetaType<std::vector<mtx::events::msg::CallCandidates::Candidate>>();
qRegisterMetaType<mtx::events::msg::CallCandidates::Candidate>();
qRegisterMetaType<mtx::responses::TurnServer>();
connect(&session_, &WebRTCSession::offerCreated, this,
[this](const std::string &sdp,
const std::vector<CallCandidates::Candidate> &candidates)
{
nhlog::ui()->debug("WebRTC: call id: {} - sending offer", callid_);
emit newMessage(roomid_, CallInvite{callid_, sdp, 0, timeoutms_});
emit newMessage(roomid_, CallCandidates{callid_, candidates, 0});
connect(
&session_,
&WebRTCSession::offerCreated,
this,
[this](const std::string &sdp, const std::vector<CallCandidates::Candidate> &candidates) {
nhlog::ui()->debug("WebRTC: call id: {} - sending offer", callid_);
emit newMessage(roomid_, CallInvite{callid_, sdp, 0, timeoutms_});
emit newMessage(roomid_, CallCandidates{callid_, candidates, 0});
QTimer::singleShot(timeoutms_, this, [this]() {
if (session_.state() == WebRTCSession::State::OFFERSENT) {
hangUp(CallHangUp::Reason::InviteTimeOut);
emit ChatPage::instance()->showNotification(
"The remote side failed to pick up.");
}
});
});
QTimer::singleShot(timeoutms_, this, [this](){
if (session_.state() == WebRTCSession::State::OFFERSENT) {
hangUp(CallHangUp::Reason::InviteTimeOut);
emit ChatPage::instance()->showNotification("The remote side failed to pick up.");
}
});
});
connect(
&session_,
&WebRTCSession::answerCreated,
this,
[this](const std::string &sdp, const std::vector<CallCandidates::Candidate> &candidates) {
nhlog::ui()->debug("WebRTC: call id: {} - sending answer", callid_);
emit newMessage(roomid_, CallAnswer{callid_, sdp, 0});
emit newMessage(roomid_, CallCandidates{callid_, candidates, 0});
});
connect(&session_, &WebRTCSession::answerCreated, this,
[this](const std::string &sdp,
const std::vector<CallCandidates::Candidate> &candidates)
{
nhlog::ui()->debug("WebRTC: call id: {} - sending answer", callid_);
emit newMessage(roomid_, CallAnswer{callid_, sdp, 0});
emit newMessage(roomid_, CallCandidates{callid_, candidates, 0});
});
connect(&session_,
&WebRTCSession::newICECandidate,
this,
[this](const CallCandidates::Candidate &candidate) {
nhlog::ui()->debug("WebRTC: call id: {} - sending ice candidate", callid_);
emit newMessage(roomid_, CallCandidates{callid_, {candidate}, 0});
});
connect(&session_, &WebRTCSession::newICECandidate, this,
[this](const CallCandidates::Candidate &candidate)
{
nhlog::ui()->debug("WebRTC: call id: {} - sending ice candidate", callid_);
emit newMessage(roomid_, CallCandidates{callid_, {candidate}, 0});
});
connect(&turnServerTimer_, &QTimer::timeout, this, &CallManager::retrieveTurnServer);
connect(&turnServerTimer_, &QTimer::timeout, this, &CallManager::retrieveTurnServer);
connect(this,
&CallManager::turnServerRetrieved,
this,
[this](const mtx::responses::TurnServer &res) {
nhlog::net()->info("TURN server(s) retrieved from homeserver:");
nhlog::net()->info("username: {}", res.username);
nhlog::net()->info("ttl: {} seconds", res.ttl);
for (const auto &u : res.uris)
nhlog::net()->info("uri: {}", u);
connect(this, &CallManager::turnServerRetrieved, this,
[this](const mtx::responses::TurnServer &res)
{
nhlog::net()->info("TURN server(s) retrieved from homeserver:");
nhlog::net()->info("username: {}", res.username);
nhlog::net()->info("ttl: {} seconds", res.ttl);
for (const auto &u : res.uris)
nhlog::net()->info("uri: {}", u);
// Request new credentials close to expiry
// See https://tools.ietf.org/html/draft-uberti-behave-turn-rest-00
turnURIs_ = getTurnURIs(res);
uint32_t ttl = std::max(res.ttl, UINT32_C(3600));
if (res.ttl < 3600)
nhlog::net()->warn("Setting ttl to 1 hour");
turnServerTimer_.setInterval(ttl * 1000 * 0.9);
});
// Request new credentials close to expiry
// See https://tools.ietf.org/html/draft-uberti-behave-turn-rest-00
turnURIs_ = getTurnURIs(res);
uint32_t ttl = std::max(res.ttl, UINT32_C(3600));
if (res.ttl < 3600)
nhlog::net()->warn("Setting ttl to 1 hour");
turnServerTimer_.setInterval(ttl * 1000 * 0.9);
});
connect(&session_, &WebRTCSession::stateChanged, this, [this](WebRTCSession::State state) {
switch (state) {
case WebRTCSession::State::DISCONNECTED:
playRingtone("qrc:/media/media/callend.ogg", false);
clear();
break;
case WebRTCSession::State::ICEFAILED: {
QString error("Call connection failed.");
if (turnURIs_.empty())
error += " Your homeserver has no configured TURN server.";
emit ChatPage::instance()->showNotification(error);
hangUp(CallHangUp::Reason::ICEFailed);
break;
}
default:
break;
}
});
connect(&session_, &WebRTCSession::stateChanged, this,
[this](WebRTCSession::State state) {
if (state == WebRTCSession::State::DISCONNECTED) {
playRingtone("qrc:/media/media/callend.ogg", false);
}
else if (state == WebRTCSession::State::ICEFAILED) {
QString error("Call connection failed.");
if (turnURIs_.empty())
error += " Your homeserver has no configured TURN server.";
emit ChatPage::instance()->showNotification(error);
hangUp(CallHangUp::Reason::ICEFailed);
}
});
connect(&player_, &QMediaPlayer::mediaStatusChanged, this,
[this](QMediaPlayer::MediaStatus status) {
if (status == QMediaPlayer::LoadedMedia)
player_.play();
});
connect(&player_,
&QMediaPlayer::mediaStatusChanged,
this,
[this](QMediaPlayer::MediaStatus status) {
if (status == QMediaPlayer::LoadedMedia)
player_.play();
});
}
void
CallManager::sendInvite(const QString &roomid)
{
if (onActiveCall())
return;
if (onActiveCall())
return;
auto roomInfo = cache::singleRoomInfo(roomid.toStdString());
if (roomInfo.member_count != 2) {
emit ChatPage::instance()->showNotification("Voice calls are limited to 1:1 rooms.");
return;
}
auto roomInfo = cache::singleRoomInfo(roomid.toStdString());
if (roomInfo.member_count != 2) {
emit ChatPage::instance()->showNotification(
"Voice calls are limited to 1:1 rooms.");
return;
}
std::string errorMessage;
if (!session_.init(&errorMessage)) {
emit ChatPage::instance()->showNotification(QString::fromStdString(errorMessage));
return;
}
std::string errorMessage;
if (!session_.init(&errorMessage)) {
emit ChatPage::instance()->showNotification(QString::fromStdString(errorMessage));
return;
}
roomid_ = roomid;
session_.setStunServer(settings_->useStunServer() ? STUN_SERVER : "");
session_.setTurnServers(turnURIs_);
roomid_ = roomid;
session_.setStunServer(settings_->useStunServer() ? STUN_SERVER : "");
session_.setTurnServers(turnURIs_);
generateCallID();
nhlog::ui()->debug("WebRTC: call id: {} - creating invite", callid_);
std::vector<RoomMember> members(cache::getMembers(roomid.toStdString()));
const RoomMember &callee = members.front().user_id == utils::localUser() ? members.back() : members.front();
emit newCallParty(callee.user_id, callee.display_name,
QString::fromStdString(roomInfo.name), QString::fromStdString(roomInfo.avatar_url));
playRingtone("qrc:/media/media/ringback.ogg", true);
if (!session_.createOffer()) {
emit ChatPage::instance()->showNotification("Problem setting up call.");
endCall();
}
generateCallID();
nhlog::ui()->debug("WebRTC: call id: {} - creating invite", callid_);
std::vector<RoomMember> members(cache::getMembers(roomid.toStdString()));
const RoomMember &callee =
members.front().user_id == utils::localUser() ? members.back() : members.front();
emit newCallParty(callee.user_id,
callee.display_name,
QString::fromStdString(roomInfo.name),
QString::fromStdString(roomInfo.avatar_url));
playRingtone("qrc:/media/media/ringback.ogg", true);
if (!session_.createOffer()) {
emit ChatPage::instance()->showNotification("Problem setting up call.");
endCall();
}
}
namespace {
std::string callHangUpReasonString(CallHangUp::Reason reason)
std::string
callHangUpReasonString(CallHangUp::Reason reason)
{
switch (reason) {
case CallHangUp::Reason::ICEFailed:
return "ICE failed";
case CallHangUp::Reason::InviteTimeOut:
return "Invite time out";
default:
return "User";
}
switch (reason) {
case CallHangUp::Reason::ICEFailed:
return "ICE failed";
case CallHangUp::Reason::InviteTimeOut:
return "Invite time out";
default:
return "User";
}
}
}
void
CallManager::hangUp(CallHangUp::Reason reason)
{
if (!callid_.empty()) {
nhlog::ui()->debug("WebRTC: call id: {} - hanging up ({})", callid_,
callHangUpReasonString(reason));
emit newMessage(roomid_, CallHangUp{callid_, 0, reason});
endCall();
}
if (!callid_.empty()) {
nhlog::ui()->debug(
"WebRTC: call id: {} - hanging up ({})", callid_, callHangUpReasonString(reason));
emit newMessage(roomid_, CallHangUp{callid_, 0, reason});
endCall();
}
}
bool
CallManager::onActiveCall()
{
return session_.state() != WebRTCSession::State::DISCONNECTED;
return session_.state() != WebRTCSession::State::DISCONNECTED;
}
void CallManager::syncEvent(const mtx::events::collections::TimelineEvents &event)
void
CallManager::syncEvent(const mtx::events::collections::TimelineEvents &event)
{
if (handleEvent_<CallInvite>(event) || handleEvent_<CallCandidates>(event)
|| handleEvent_<CallAnswer>(event) || handleEvent_<CallHangUp>(event))
return;
if (handleEvent_<CallInvite>(event) || handleEvent_<CallCandidates>(event) ||
handleEvent_<CallAnswer>(event) || handleEvent_<CallHangUp>(event))
return;
}
template<typename T>
bool
CallManager::handleEvent_(const mtx::events::collections::TimelineEvents &event)
{
if (std::holds_alternative<RoomEvent<T>>(event)) {
handleEvent(std::get<RoomEvent<T>>(event));
return true;
}
return false;
if (std::holds_alternative<RoomEvent<T>>(event)) {
handleEvent(std::get<RoomEvent<T>>(event));
return true;
}
return false;
}
void
CallManager::handleEvent(const RoomEvent<CallInvite> &callInviteEvent)
{
const char video[] = "m=video";
const std::string &sdp = callInviteEvent.content.sdp;
bool isVideo = std::search(sdp.cbegin(), sdp.cend(), std::cbegin(video), std::cend(video) - 1,
[](unsigned char c1, unsigned char c2) {return std::tolower(c1) == std::tolower(c2);})
!= sdp.cend();
const char video[] = "m=video";
const std::string &sdp = callInviteEvent.content.sdp;
bool isVideo = std::search(sdp.cbegin(),
sdp.cend(),
std::cbegin(video),
std::cend(video) - 1,
[](unsigned char c1, unsigned char c2) {
return std::tolower(c1) == std::tolower(c2);
}) != sdp.cend();
nhlog::ui()->debug(std::string("WebRTC: call id: {} - incoming ") + (isVideo ? "video" : "voice") +
" CallInvite from {}", callInviteEvent.content.call_id, callInviteEvent.sender);
nhlog::ui()->debug(std::string("WebRTC: call id: {} - incoming ") +
(isVideo ? "video" : "voice") + " CallInvite from {}",
callInviteEvent.content.call_id,
callInviteEvent.sender);
if (callInviteEvent.content.call_id.empty())
return;
if (callInviteEvent.content.call_id.empty())
return;
if (isVideo) {
emit newMessage(QString::fromStdString(callInviteEvent.room_id),
CallHangUp{callInviteEvent.content.call_id, 0, CallHangUp::Reason::InviteTimeOut});
return;
}
auto roomInfo = cache::singleRoomInfo(callInviteEvent.room_id);
if (onActiveCall() || roomInfo.member_count != 2 || isVideo) {
emit newMessage(QString::fromStdString(callInviteEvent.room_id),
CallHangUp{callInviteEvent.content.call_id,
0,
CallHangUp::Reason::InviteTimeOut});
return;
}
auto roomInfo = cache::singleRoomInfo(callInviteEvent.room_id);
if (onActiveCall() || roomInfo.member_count != 2) {
emit newMessage(QString::fromStdString(callInviteEvent.room_id),
CallHangUp{callInviteEvent.content.call_id, 0, CallHangUp::Reason::InviteTimeOut});
return;
}
playRingtone("qrc:/media/media/ring.ogg", true);
roomid_ = QString::fromStdString(callInviteEvent.room_id);
callid_ = callInviteEvent.content.call_id;
remoteICECandidates_.clear();
playRingtone("qrc:/media/media/ring.ogg", true);
roomid_ = QString::fromStdString(callInviteEvent.room_id);
callid_ = callInviteEvent.content.call_id;
remoteICECandidates_.clear();
std::vector<RoomMember> members(cache::getMembers(callInviteEvent.room_id));
const RoomMember &caller =
members.front().user_id == utils::localUser() ? members.back() : members.front();
emit newCallParty(caller.user_id,
caller.display_name,
QString::fromStdString(roomInfo.name),
QString::fromStdString(roomInfo.avatar_url));
std::vector<RoomMember> members(cache::getMembers(callInviteEvent.room_id));
const RoomMember &caller =
members.front().user_id == utils::localUser() ? members.back() : members.front();
emit newCallParty(caller.user_id, caller.display_name,
QString::fromStdString(roomInfo.name), QString::fromStdString(roomInfo.avatar_url));
auto dialog = new dialogs::AcceptCall(
caller.user_id,
caller.display_name,
QString::fromStdString(roomInfo.name),
QString::fromStdString(roomInfo.avatar_url),
MainWindow::instance());
connect(dialog, &dialogs::AcceptCall::accept, this,
[this, callInviteEvent](){
MainWindow::instance()->hideOverlay();
answerInvite(callInviteEvent.content);});
connect(dialog, &dialogs::AcceptCall::reject, this,
[this](){
MainWindow::instance()->hideOverlay();
hangUp();});
MainWindow::instance()->showSolidOverlayModal(dialog);
auto dialog = new dialogs::AcceptCall(caller.user_id,
caller.display_name,
QString::fromStdString(roomInfo.name),
QString::fromStdString(roomInfo.avatar_url),
MainWindow::instance());
connect(dialog, &dialogs::AcceptCall::accept, this, [this, callInviteEvent]() {
MainWindow::instance()->hideOverlay();
answerInvite(callInviteEvent.content);
});
connect(dialog, &dialogs::AcceptCall::reject, this, [this]() {
MainWindow::instance()->hideOverlay();
hangUp();
});
MainWindow::instance()->showSolidOverlayModal(dialog);
}
void
CallManager::answerInvite(const CallInvite &invite)
{
stopRingtone();
std::string errorMessage;
if (!session_.init(&errorMessage)) {
emit ChatPage::instance()->showNotification(QString::fromStdString(errorMessage));
hangUp();
return;
}
stopRingtone();
std::string errorMessage;
if (!session_.init(&errorMessage)) {
emit ChatPage::instance()->showNotification(QString::fromStdString(errorMessage));
hangUp();
return;
}
session_.setStunServer(settings_->useStunServer() ? STUN_SERVER : "");
session_.setTurnServers(turnURIs_);
session_.setStunServer(settings_->useStunServer() ? STUN_SERVER : "");
session_.setTurnServers(turnURIs_);
if (!session_.acceptOffer(invite.sdp)) {
emit ChatPage::instance()->showNotification("Problem setting up call.");
hangUp();
return;
}
session_.acceptICECandidates(remoteICECandidates_);
remoteICECandidates_.clear();
if (!session_.acceptOffer(invite.sdp)) {
emit ChatPage::instance()->showNotification("Problem setting up call.");
hangUp();
return;
}
session_.acceptICECandidates(remoteICECandidates_);
remoteICECandidates_.clear();
}
void
CallManager::handleEvent(const RoomEvent<CallCandidates> &callCandidatesEvent)
{
if (callCandidatesEvent.sender == utils::localUser().toStdString())
return;
if (callCandidatesEvent.sender == utils::localUser().toStdString())
return;
nhlog::ui()->debug("WebRTC: call id: {} - incoming CallCandidates from {}",
callCandidatesEvent.content.call_id, callCandidatesEvent.sender);
nhlog::ui()->debug("WebRTC: call id: {} - incoming CallCandidates from {}",
callCandidatesEvent.content.call_id,
callCandidatesEvent.sender);
if (callid_ == callCandidatesEvent.content.call_id) {
if (onActiveCall())
session_.acceptICECandidates(callCandidatesEvent.content.candidates);
else {
// CallInvite has been received and we're awaiting localUser to accept or reject the call
for (const auto &c : callCandidatesEvent.content.candidates)
remoteICECandidates_.push_back(c);
}
}
if (callid_ == callCandidatesEvent.content.call_id) {
if (onActiveCall())
session_.acceptICECandidates(callCandidatesEvent.content.candidates);
else {
// CallInvite has been received and we're awaiting localUser to accept or
// reject the call
for (const auto &c : callCandidatesEvent.content.candidates)
remoteICECandidates_.push_back(c);
}
}
}
void
CallManager::handleEvent(const RoomEvent<CallAnswer> &callAnswerEvent)
{
nhlog::ui()->debug("WebRTC: call id: {} - incoming CallAnswer from {}",
callAnswerEvent.content.call_id, callAnswerEvent.sender);
nhlog::ui()->debug("WebRTC: call id: {} - incoming CallAnswer from {}",
callAnswerEvent.content.call_id,
callAnswerEvent.sender);
if (!onActiveCall() && callAnswerEvent.sender == utils::localUser().toStdString() &&
callid_ == callAnswerEvent.content.call_id) {
emit ChatPage::instance()->showNotification("Call answered on another device.");
stopRingtone();
MainWindow::instance()->hideOverlay();
return;
}
if (!onActiveCall() && callAnswerEvent.sender == utils::localUser().toStdString() &&
callid_ == callAnswerEvent.content.call_id) {
emit ChatPage::instance()->showNotification("Call answered on another device.");
stopRingtone();
MainWindow::instance()->hideOverlay();
return;
}
if (onActiveCall() && callid_ == callAnswerEvent.content.call_id) {
stopRingtone();
if (!session_.acceptAnswer(callAnswerEvent.content.sdp)) {
emit ChatPage::instance()->showNotification("Problem setting up call.");
hangUp();
}
}
if (onActiveCall() && callid_ == callAnswerEvent.content.call_id) {
stopRingtone();
if (!session_.acceptAnswer(callAnswerEvent.content.sdp)) {
emit ChatPage::instance()->showNotification("Problem setting up call.");
hangUp();
}
}
}
void
CallManager::handleEvent(const RoomEvent<CallHangUp> &callHangUpEvent)
{
nhlog::ui()->debug("WebRTC: call id: {} - incoming CallHangUp ({}) from {}",
callHangUpEvent.content.call_id, callHangUpReasonString(callHangUpEvent.content.reason),
callHangUpEvent.sender);
nhlog::ui()->debug("WebRTC: call id: {} - incoming CallHangUp ({}) from {}",
callHangUpEvent.content.call_id,
callHangUpReasonString(callHangUpEvent.content.reason),
callHangUpEvent.sender);
if (callid_ == callHangUpEvent.content.call_id) {
MainWindow::instance()->hideOverlay();
endCall();
}
if (callid_ == callHangUpEvent.content.call_id) {
MainWindow::instance()->hideOverlay();
endCall();
}
}
void
CallManager::generateCallID()
{
using namespace std::chrono;
uint64_t ms = duration_cast<milliseconds>(system_clock::now().time_since_epoch()).count();
callid_ = "c" + std::to_string(ms);
using namespace std::chrono;
uint64_t ms = duration_cast<milliseconds>(system_clock::now().time_since_epoch()).count();
callid_ = "c" + std::to_string(ms);
}
void
CallManager::clear()
{
roomid_.clear();
callid_.clear();
remoteICECandidates_.clear();
}
void
CallManager::endCall()
{
stopRingtone();
session_.end();
roomid_.clear();
callid_.clear();
remoteICECandidates_.clear();
stopRingtone();
clear();
session_.end();
}
void
CallManager::refreshTurnServer()
{
turnURIs_.clear();
turnServerTimer_.start(2000);
turnURIs_.clear();
turnServerTimer_.start(2000);
}
void
CallManager::retrieveTurnServer()
{
http::client()->get_turn_server(
[this](const mtx::responses::TurnServer &res, mtx::http::RequestErr err) {
if (err) {
turnServerTimer_.setInterval(5000);
return;
}
emit turnServerRetrieved(res);
});
http::client()->get_turn_server(
[this](const mtx::responses::TurnServer &res, mtx::http::RequestErr err) {
if (err) {
turnServerTimer_.setInterval(5000);
return;
}
emit turnServerRetrieved(res);
});
}
void
CallManager::playRingtone(const QString &ringtone, bool repeat)
{
static QMediaPlaylist playlist;
playlist.clear();
playlist.setPlaybackMode(repeat ? QMediaPlaylist::CurrentItemInLoop : QMediaPlaylist::CurrentItemOnce);
playlist.addMedia(QUrl(ringtone));
player_.setVolume(100);
player_.setPlaylist(&playlist);
static QMediaPlaylist playlist;
playlist.clear();
playlist.setPlaybackMode(repeat ? QMediaPlaylist::CurrentItemInLoop
: QMediaPlaylist::CurrentItemOnce);
playlist.addMedia(QUrl(ringtone));
player_.setVolume(100);
player_.setPlaylist(&playlist);
}
void
CallManager::stopRingtone()
{
player_.setPlaylist(nullptr);
player_.setPlaylist(nullptr);
}
namespace {
std::vector<std::string>
getTurnURIs(const mtx::responses::TurnServer &turnServer)
{
// gstreamer expects: turn(s)://username:password@host:port?transport=udp(tcp)
// where username and password are percent-encoded
std::vector<std::string> ret;
for (const auto &uri : turnServer.uris) {
if (auto c = uri.find(':'); c == std::string::npos) {
nhlog::ui()->error("Invalid TURN server uri: {}", uri);
continue;
}
else {
std::string scheme = std::string(uri, 0, c);
if (scheme != "turn" && scheme != "turns") {
nhlog::ui()->error("Invalid TURN server uri: {}", uri);
continue;
}
// gstreamer expects: turn(s)://username:password@host:port?transport=udp(tcp)
// where username and password are percent-encoded
std::vector<std::string> ret;
for (const auto &uri : turnServer.uris) {
if (auto c = uri.find(':'); c == std::string::npos) {
nhlog::ui()->error("Invalid TURN server uri: {}", uri);
continue;
} else {
std::string scheme = std::string(uri, 0, c);
if (scheme != "turn" && scheme != "turns") {
nhlog::ui()->error("Invalid TURN server uri: {}", uri);
continue;
}
QString encodedUri = QString::fromStdString(scheme) + "://" +
QUrl::toPercentEncoding(QString::fromStdString(turnServer.username)) + ":" +
QUrl::toPercentEncoding(QString::fromStdString(turnServer.password)) + "@" +
QString::fromStdString(std::string(uri, ++c));
ret.push_back(encodedUri.toStdString());
}
}
return ret;
QString encodedUri =
QString::fromStdString(scheme) + "://" +
QUrl::toPercentEncoding(QString::fromStdString(turnServer.username)) +
":" +
QUrl::toPercentEncoding(QString::fromStdString(turnServer.password)) +
"@" + QString::fromStdString(std::string(uri, ++c));
ret.push_back(encodedUri.toStdString());
}
}
return ret;
}
}

37
src/CallManager.h
View File

@ -3,8 +3,8 @@
#include <string>
#include <vector>
#include <QObject>
#include <QMediaPlayer>
#include <QObject>
#include <QSharedPointer>
#include <QString>
#include <QTimer>
@ -27,7 +27,8 @@ public:
CallManager(QSharedPointer<UserSettings>);
void sendInvite(const QString &roomid);
void hangUp(mtx::events::msg::CallHangUp::Reason = mtx::events::msg::CallHangUp::Reason::User);
void hangUp(
mtx::events::msg::CallHangUp::Reason = mtx::events::msg::CallHangUp::Reason::User);
bool onActiveCall();
void refreshTurnServer();
@ -35,22 +36,21 @@ public slots:
void syncEvent(const mtx::events::collections::TimelineEvents &event);
signals:
void newMessage(const QString &roomid, const mtx::events::msg::CallInvite&);
void newMessage(const QString &roomid, const mtx::events::msg::CallCandidates&);
void newMessage(const QString &roomid, const mtx::events::msg::CallAnswer&);
void newMessage(const QString &roomid, const mtx::events::msg::CallHangUp&);
void turnServerRetrieved(const mtx::responses::TurnServer&);
void newCallParty(
const QString &userid,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl);
void newMessage(const QString &roomid, const mtx::events::msg::CallInvite &);
void newMessage(const QString &roomid, const mtx::events::msg::CallCandidates &);
void newMessage(const QString &roomid, const mtx::events::msg::CallAnswer &);
void newMessage(const QString &roomid, const mtx::events::msg::CallHangUp &);
void turnServerRetrieved(const mtx::responses::TurnServer &);
void newCallParty(const QString &userid,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl);
private slots:
void retrieveTurnServer();
private:
WebRTCSession& session_;
WebRTCSession &session_;
QString roomid_;
std::string callid_;
const uint32_t timeoutms_ = 120000;
@ -62,12 +62,13 @@ private:
template<typename T>
bool handleEvent_(const mtx::events::collections::TimelineEvents &event);
void handleEvent(const mtx::events::RoomEvent<mtx::events::msg::CallInvite>&);
void handleEvent(const mtx::events::RoomEvent<mtx::events::msg::CallCandidates>&);
void handleEvent(const mtx::events::RoomEvent<mtx::events::msg::CallAnswer>&);
void handleEvent(const mtx::events::RoomEvent<mtx::events::msg::CallHangUp>&);
void answerInvite(const mtx::events::msg::CallInvite&);
void handleEvent(const mtx::events::RoomEvent<mtx::events::msg::CallInvite> &);
void handleEvent(const mtx::events::RoomEvent<mtx::events::msg::CallCandidates> &);
void handleEvent(const mtx::events::RoomEvent<mtx::events::msg::CallAnswer> &);
void handleEvent(const mtx::events::RoomEvent<mtx::events::msg::CallHangUp> &);
void answerInvite(const mtx::events::msg::CallInvite &);
void generateCallID();
void clear();
void endCall();
void playRingtone(const QString &ringtone, bool repeat);
void stopRingtone();

15
src/ChatPage.cpp
View File

@ -460,9 +460,8 @@ ChatPage::ChatPage(QSharedPointer<UserSettings> userSettings, QWidget *parent)
if (callManager_.onActiveCall()) {
callManager_.hangUp();
} else {
if (auto roomInfo =
cache::singleRoomInfo(current_room_.toStdString());
roomInfo.member_count != 2) {
if (auto roomInfo = cache::singleRoomInfo(current_room_.toStdString());
roomInfo.member_count != 2) {
showNotification("Voice calls are limited to 1:1 rooms.");
} else {
std::vector<RoomMember> members(
@ -471,11 +470,11 @@ ChatPage::ChatPage(QSharedPointer<UserSettings> userSettings, QWidget *parent)
members.front().user_id == utils::localUser() ? members.back()
: members.front();
auto dialog = new dialogs::PlaceCall(
callee.user_id,
callee.display_name,
QString::fromStdString(roomInfo.name),
QString::fromStdString(roomInfo.avatar_url),
MainWindow::instance());
callee.user_id,
callee.display_name,
QString::fromStdString(roomInfo.name),
QString::fromStdString(roomInfo.avatar_url),
MainWindow::instance());
connect(dialog, &dialogs::PlaceCall::voice, this, [this]() {
callManager_.sendInvite(current_room_);
});

19
src/EventAccessors.cpp
View File

@ -72,12 +72,19 @@ struct CallType
template<class T>
std::string operator()(const T &e)
{
if constexpr (std::is_same_v<mtx::events::RoomEvent<mtx::events::msg::CallInvite>, T>) {
const char video[] = "m=video";
const std::string &sdp = e.content.sdp;
return std::search(sdp.cbegin(), sdp.cend(), std::cbegin(video), std::cend(video) - 1,
[](unsigned char c1, unsigned char c2) {return std::tolower(c1) == std::tolower(c2);})
!= sdp.cend() ? "video" : "voice";
if constexpr (std::is_same_v<mtx::events::RoomEvent<mtx::events::msg::CallInvite>,
T>) {
const char video[] = "m=video";
const std::string &sdp = e.content.sdp;
return std::search(sdp.cbegin(),
sdp.cend(),
std::cbegin(video),
std::cend(video) - 1,
[](unsigned char c1, unsigned char c2) {
return std::tolower(c1) == std::tolower(c2);
}) != sdp.cend()
? "video"
: "voice";
}
return std::string();
}

849
src/WebRTCSession.cpp
View File

@ -1,9 +1,10 @@
#include <cctype>
#include "WebRTCSession.h"
#include "Logging.h"
#include "WebRTCSession.h"
extern "C" {
extern "C"
{
#include "gst/gst.h"
#include "gst/sdp/sdp.h"
@ -13,180 +14,445 @@ extern "C" {
Q_DECLARE_METATYPE(WebRTCSession::State)
namespace {
bool isoffering_;
std::string localsdp_;
std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
gboolean newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data);
GstWebRTCSessionDescription* parseSDP(const std::string &sdp, GstWebRTCSDPType type);
void generateOffer(GstElement *webrtc);
void setLocalDescription(GstPromise *promise, gpointer webrtc);
void addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED);
gboolean onICEGatheringCompletion(gpointer timerid);
void iceConnectionStateChanged(GstElement *webrtcbin, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED);
void createAnswer(GstPromise *promise, gpointer webrtc);
void addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
void linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe);
std::string::const_iterator findName(const std::string &sdp, const std::string &name);
int getPayloadType(const std::string &sdp, const std::string &name);
}
WebRTCSession::WebRTCSession() : QObject()
WebRTCSession::WebRTCSession()
: QObject()
{
qRegisterMetaType<WebRTCSession::State>();
connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
qRegisterMetaType<WebRTCSession::State>();
connect(this, &WebRTCSession::stateChanged, this, &WebRTCSession::setState);
}
bool
WebRTCSession::init(std::string *errorMessage)
{
if (initialised_)
return true;
if (initialised_)
return true;
GError *error = nullptr;
if (!gst_init_check(nullptr, nullptr, &error)) {
std::string strError = std::string("WebRTC: failed to initialise GStreamer: ");
if (error) {
strError += error->message;
g_error_free(error);
}
nhlog::ui()->error(strError);
if (errorMessage)
*errorMessage = strError;
return false;
}
GError *error = nullptr;
if (!gst_init_check(nullptr, nullptr, &error)) {
std::string strError = std::string("WebRTC: failed to initialise GStreamer: ");
if (error) {
strError += error->message;
g_error_free(error);
}
nhlog::ui()->error(strError);
if (errorMessage)
*errorMessage = strError;
return false;
}
gchar *version = gst_version_string();
std::string gstVersion(version);
g_free(version);
nhlog::ui()->info("WebRTC: initialised " + gstVersion);
gchar *version = gst_version_string();
std::string gstVersion(version);
g_free(version);
nhlog::ui()->info("WebRTC: initialised " + gstVersion);
// GStreamer Plugins:
// Base: audioconvert, audioresample, opus, playback, volume
// Good: autodetect, rtpmanager
// Bad: dtls, srtp, webrtc
// libnice [GLib]: nice
initialised_ = true;
std::string strError = gstVersion + ": Missing plugins: ";
const gchar *needed[] = {"audioconvert", "audioresample", "autodetect", "dtls", "nice",
"opus", "playback", "rtpmanager", "srtp", "volume", "webrtc", nullptr};
GstRegistry *registry = gst_registry_get();
for (guint i = 0; i < g_strv_length((gchar**)needed); i++) {
GstPlugin *plugin = gst_registry_find_plugin(registry, needed[i]);
if (!plugin) {
strError += std::string(needed[i]) + " ";
initialised_ = false;
continue;
}
gst_object_unref(plugin);
}
// GStreamer Plugins:
// Base: audioconvert, audioresample, opus, playback, volume
// Good: autodetect, rtpmanager
// Bad: dtls, srtp, webrtc
// libnice [GLib]: nice
initialised_ = true;
std::string strError = gstVersion + ": Missing plugins: ";
const gchar *needed[] = {"audioconvert",
"audioresample",
"autodetect",
"dtls",
"nice",
"opus",
"playback",
"rtpmanager",
"srtp",
"volume",
"webrtc",
nullptr};
GstRegistry *registry = gst_registry_get();
for (guint i = 0; i < g_strv_length((gchar **)needed); i++) {
GstPlugin *plugin = gst_registry_find_plugin(registry, needed[i]);
if (!plugin) {
strError += std::string(needed[i]) + " ";
initialised_ = false;
continue;
}
gst_object_unref(plugin);
}
if (!initialised_) {
nhlog::ui()->error(strError);
if (errorMessage)
*errorMessage = strError;
}
return initialised_;
}
namespace {
bool isoffering_;
std::string localsdp_;
std::vector<mtx::events::msg::CallCandidates::Candidate> localcandidates_;
gboolean
newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
{
WebRTCSession *session = static_cast<WebRTCSession *>(user_data);
switch (GST_MESSAGE_TYPE(msg)) {
case GST_MESSAGE_EOS:
nhlog::ui()->error("WebRTC: end of stream");
session->end();
break;
case GST_MESSAGE_ERROR:
GError *error;
gchar *debug;
gst_message_parse_error(msg, &error, &debug);
nhlog::ui()->error(
"WebRTC: error from element {}: {}", GST_OBJECT_NAME(msg->src), error->message);
g_clear_error(&error);
g_free(debug);
session->end();
break;
default:
break;
}
return TRUE;
}
GstWebRTCSessionDescription *
parseSDP(const std::string &sdp, GstWebRTCSDPType type)
{
GstSDPMessage *msg;
gst_sdp_message_new(&msg);
if (gst_sdp_message_parse_buffer((guint8 *)sdp.c_str(), sdp.size(), msg) == GST_SDP_OK) {
return gst_webrtc_session_description_new(type, msg);
} else {
nhlog::ui()->error("WebRTC: failed to parse remote session description");
gst_object_unref(msg);
return nullptr;
}
}
void
setLocalDescription(GstPromise *promise, gpointer webrtc)
{
const GstStructure *reply = gst_promise_get_reply(promise);
gboolean isAnswer = gst_structure_id_has_field(reply, g_quark_from_string("answer"));
GstWebRTCSessionDescription *gstsdp = nullptr;
gst_structure_get(reply,
isAnswer ? "answer" : "offer",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
&gstsdp,
nullptr);
gst_promise_unref(promise);
g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr);
gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
localsdp_ = std::string(sdp);
g_free(sdp);
gst_webrtc_session_description_free(gstsdp);
nhlog::ui()->debug(
"WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_);
}
void
createOffer(GstElement *webrtc)
{
// create-offer first, then set-local-description
GstPromise *promise =
gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
g_signal_emit_by_name(webrtc, "create-offer", nullptr, promise);
}
void
createAnswer(GstPromise *promise, gpointer webrtc)
{
// create-answer first, then set-local-description
gst_promise_unref(promise);
promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
}
gboolean
onICEGatheringCompletion(gpointer timerid)
{
*(guint *)(timerid) = 0;
if (isoffering_) {
emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
} else {
emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ANSWERSENT);
}
return FALSE;
}
void
addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED,
guint mlineIndex,
gchar *candidate,
gpointer G_GNUC_UNUSED)
{
nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
emit WebRTCSession::instance().newICECandidate(
{"audio", (uint16_t)mlineIndex, candidate});
return;
}
localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers
// GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early. Fixed in v1.18. Use a 100ms timeout in
// the meantime
static guint timerid = 0;
if (timerid)
g_source_remove(timerid);
timerid = g_timeout_add(100, onICEGatheringCompletion, &timerid);
}
void
iceConnectionStateChanged(GstElement *webrtc,
GParamSpec *pspec G_GNUC_UNUSED,
gpointer user_data G_GNUC_UNUSED)
{
GstWebRTCICEConnectionState newState;
g_object_get(webrtc, "ice-connection-state", &newState, nullptr);
switch (newState) {
case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING:
nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking");
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
break;
case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ICEFAILED);
break;
default:
break;
}
}
void
linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
{
GstCaps *caps = gst_pad_get_current_caps(newpad);
if (!caps)
return;
const gchar *name = gst_structure_get_name(gst_caps_get_structure(caps, 0));
gst_caps_unref(caps);
GstPad *queuepad = nullptr;
if (g_str_has_prefix(name, "audio")) {
nhlog::ui()->debug("WebRTC: received incoming audio stream");
GstElement *queue = gst_element_factory_make("queue", nullptr);
GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
GstElement *resample = gst_element_factory_make("audioresample", nullptr);
GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
gst_bin_add_many(GST_BIN(pipe), queue, convert, resample, sink, nullptr);
gst_element_sync_state_with_parent(queue);
gst_element_sync_state_with_parent(convert);
gst_element_sync_state_with_parent(resample);
gst_element_sync_state_with_parent(sink);
gst_element_link_many(queue, convert, resample, sink, nullptr);
queuepad = gst_element_get_static_pad(queue, "sink");
}
if (queuepad) {
if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, queuepad)))
nhlog::ui()->error("WebRTC: unable to link new pad");
else {
emit WebRTCSession::instance().stateChanged(
WebRTCSession::State::CONNECTED);
}
gst_object_unref(queuepad);
}
}
void
addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
{
if (GST_PAD_DIRECTION(newpad) != GST_PAD_SRC)
return;
nhlog::ui()->debug("WebRTC: received incoming stream");
GstElement *decodebin = gst_element_factory_make("decodebin", nullptr);
g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
gst_bin_add(GST_BIN(pipe), decodebin);
gst_element_sync_state_with_parent(decodebin);
GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, sinkpad)))
nhlog::ui()->error("WebRTC: unable to link new pad");
gst_object_unref(sinkpad);
}
std::string::const_iterator
findName(const std::string &sdp, const std::string &name)
{
return std::search(
sdp.cbegin(),
sdp.cend(),
name.cbegin(),
name.cend(),
[](unsigned char c1, unsigned char c2) { return std::tolower(c1) == std::tolower(c2); });
}
int
getPayloadType(const std::string &sdp, const std::string &name)
{
// eg a=rtpmap:111 opus/48000/2
auto e = findName(sdp, name);
if (e == sdp.cend()) {
nhlog::ui()->error("WebRTC: remote offer - " + name + " attribute missing");
return -1;
}
if (auto s = sdp.rfind(':', e - sdp.cbegin()); s == std::string::npos) {
nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name +
" payload type");
return -1;
} else {
++s;
try {
return std::stoi(std::string(sdp, s, e - sdp.cbegin() - s));
} catch (...) {
nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name +
" payload type");
}
}
return -1;
}
if (!initialised_) {
nhlog::ui()->error(strError);
if (errorMessage)
*errorMessage = strError;
}
return initialised_;
}
bool
WebRTCSession::createOffer()
{
isoffering_ = true;
localsdp_.clear();
localcandidates_.clear();
return startPipeline(111); // a dynamic opus payload type
isoffering_ = true;
localsdp_.clear();
localcandidates_.clear();
return startPipeline(111); // a dynamic opus payload type
}
bool
WebRTCSession::acceptOffer(const std::string &sdp)
{
nhlog::ui()->debug("WebRTC: received offer:\n{}", sdp);
if (state_ != State::DISCONNECTED)
return false;
nhlog::ui()->debug("WebRTC: received offer:\n{}", sdp);
if (state_ != State::DISCONNECTED)
return false;
isoffering_ = false;
localsdp_.clear();
localcandidates_.clear();
isoffering_ = false;
localsdp_.clear();
localcandidates_.clear();
int opusPayloadType = getPayloadType(sdp, "opus");
if (opusPayloadType == -1)
return false;
int opusPayloadType = getPayloadType(sdp, "opus");
if (opusPayloadType == -1)
return false;
GstWebRTCSessionDescription *offer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_OFFER);
if (!offer)
return false;
GstWebRTCSessionDescription *offer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_OFFER);
if (!offer)
return false;
if (!startPipeline(opusPayloadType)) {
gst_webrtc_session_description_free(offer);
return false;
}
if (!startPipeline(opusPayloadType)) {
gst_webrtc_session_description_free(offer);
return false;
}
// set-remote-description first, then create-answer
GstPromise *promise = gst_promise_new_with_change_func(createAnswer, webrtc_, nullptr);
g_signal_emit_by_name(webrtc_, "set-remote-description", offer, promise);
gst_webrtc_session_description_free(offer);
return true;
// set-remote-description first, then create-answer
GstPromise *promise = gst_promise_new_with_change_func(createAnswer, webrtc_, nullptr);
g_signal_emit_by_name(webrtc_, "set-remote-description", offer, promise);
gst_webrtc_session_description_free(offer);
return true;
}
bool
WebRTCSession::acceptAnswer(const std::string &sdp)
{
nhlog::ui()->debug("WebRTC: received answer:\n{}", sdp);
if (state_ != State::OFFERSENT)
return false;
GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER);
if (!answer) {
end();
return false;
}
g_signal_emit_by_name(webrtc_, "set-remote-description", answer, nullptr);
gst_webrtc_session_description_free(answer);
return true;
}
void
WebRTCSession::acceptICECandidates(
const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates)
{
if (state_ >= State::INITIATED) {
for (const auto &c : candidates) {
nhlog::ui()->debug(
"WebRTC: remote candidate: (m-line:{}):{}", c.sdpMLineIndex, c.candidate);
g_signal_emit_by_name(
webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
}
}
}
bool
WebRTCSession::startPipeline(int opusPayloadType)
{
if (state_ != State::DISCONNECTED)
return false;
if (state_ != State::DISCONNECTED)
return false;
emit stateChanged(State::INITIATING);
emit stateChanged(State::INITIATING);
if (!createPipeline(opusPayloadType))
return false;
if (!createPipeline(opusPayloadType))
return false;
webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
webrtc_ = gst_bin_get_by_name(GST_BIN(pipe_), "webrtcbin");
if (!stunServer_.empty()) {
nhlog::ui()->info("WebRTC: setting STUN server: {}", stunServer_);
g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
}
if (!stunServer_.empty()) {
nhlog::ui()->info("WebRTC: setting STUN server: {}", stunServer_);
g_object_set(webrtc_, "stun-server", stunServer_.c_str(), nullptr);
}
for (const auto &uri : turnServers_) {
nhlog::ui()->info("WebRTC: setting TURN server: {}", uri);
gboolean udata;
g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata));
}
if (turnServers_.empty())
nhlog::ui()->warn("WebRTC: no TURN server provided");
for (const auto &uri : turnServers_) {
nhlog::ui()->info("WebRTC: setting TURN server: {}", uri);
gboolean udata;
g_signal_emit_by_name(webrtc_, "add-turn-server", uri.c_str(), (gpointer)(&udata));
}
if (turnServers_.empty())
nhlog::ui()->warn("WebRTC: no TURN server provided");
// generate the offer when the pipeline goes to PLAYING
if (isoffering_)
g_signal_connect(webrtc_, "on-negotiation-needed", G_CALLBACK(generateOffer), nullptr);
// generate the offer when the pipeline goes to PLAYING
if (isoffering_)
g_signal_connect(
webrtc_, "on-negotiation-needed", G_CALLBACK(::createOffer), nullptr);
// on-ice-candidate is emitted when a local ICE candidate has been gathered
g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr);
// on-ice-candidate is emitted when a local ICE candidate has been gathered
g_signal_connect(webrtc_, "on-ice-candidate", G_CALLBACK(addLocalICECandidate), nullptr);
// capture ICE failure
g_signal_connect(webrtc_, "notify::ice-connection-state",
G_CALLBACK(iceConnectionStateChanged), nullptr);
// capture ICE failure
g_signal_connect(
webrtc_, "notify::ice-connection-state", G_CALLBACK(iceConnectionStateChanged), nullptr);
// incoming streams trigger pad-added
gst_element_set_state(pipe_, GST_STATE_READY);
g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
// incoming streams trigger pad-added
gst_element_set_state(pipe_, GST_STATE_READY);
g_signal_connect(webrtc_, "pad-added", G_CALLBACK(addDecodeBin), pipe_);
// webrtcbin lifetime is the same as that of the pipeline
gst_object_unref(webrtc_);
// webrtcbin lifetime is the same as that of the pipeline
gst_object_unref(webrtc_);
// start the pipeline
GstStateChangeReturn ret = gst_element_set_state(pipe_, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
nhlog::ui()->error("WebRTC: unable to start pipeline");
end();
return false;
}
// start the pipeline
GstStateChangeReturn ret = gst_element_set_state(pipe_, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE) {
nhlog::ui()->error("WebRTC: unable to start pipeline");
end();
return false;
}
GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_));
gst_bus_add_watch(bus, newBusMessage, this);
gst_object_unref(bus);
emit stateChanged(State::INITIATED);
return true;
GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipe_));
gst_bus_add_watch(bus, newBusMessage, this);
gst_object_unref(bus);
emit stateChanged(State::INITIATED);
return true;
}
#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
@ -194,297 +460,52 @@ WebRTCSession::startPipeline(int opusPayloadType)
bool
WebRTCSession::createPipeline(int opusPayloadType)
{
std::string pipeline("webrtcbin bundle-policy=max-bundle name=webrtcbin "
"autoaudiosrc ! volume name=srclevel ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS + std::to_string(opusPayloadType) + " ! webrtcbin.");
std::string pipeline("webrtcbin bundle-policy=max-bundle name=webrtcbin "
"autoaudiosrc ! volume name=srclevel ! audioconvert ! "
"audioresample ! queue ! opusenc ! rtpopuspay ! "
"queue ! " RTP_CAPS_OPUS +
std::to_string(opusPayloadType) + " ! webrtcbin.");
webrtc_ = nullptr;
GError *error = nullptr;
pipe_ = gst_parse_launch(pipeline.c_str(), &error);
if (error) {
nhlog::ui()->error("WebRTC: failed to parse pipeline: {}", error->message);
g_error_free(error);
end();
return false;
}
return true;
}
bool
WebRTCSession::acceptAnswer(const std::string &sdp)
{
nhlog::ui()->debug("WebRTC: received answer:\n{}", sdp);
if (state_ != State::OFFERSENT)
return false;
GstWebRTCSessionDescription *answer = parseSDP(sdp, GST_WEBRTC_SDP_TYPE_ANSWER);
if (!answer) {
end();
return false;
}
g_signal_emit_by_name(webrtc_, "set-remote-description", answer, nullptr);
gst_webrtc_session_description_free(answer);
return true;
}
void
WebRTCSession::acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &candidates)
{
if (state_ >= State::INITIATED) {
for (const auto &c : candidates) {
nhlog::ui()->debug("WebRTC: remote candidate: (m-line:{}):{}", c.sdpMLineIndex, c.candidate);
g_signal_emit_by_name(webrtc_, "add-ice-candidate", c.sdpMLineIndex, c.candidate.c_str());
}
}
webrtc_ = nullptr;
GError *error = nullptr;
pipe_ = gst_parse_launch(pipeline.c_str(), &error);
if (error) {
nhlog::ui()->error("WebRTC: failed to parse pipeline: {}", error->message);
g_error_free(error);
end();
return false;
}
return true;
}
bool
WebRTCSession::toggleMuteAudioSrc(bool &isMuted)
{
if (state_ < State::INITIATED)
return false;
if (state_ < State::INITIATED)
return false;
GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
if (!srclevel)
return false;
GstElement *srclevel = gst_bin_get_by_name(GST_BIN(pipe_), "srclevel");
if (!srclevel)
return false;
gboolean muted;
g_object_get(srclevel, "mute", &muted, nullptr);
g_object_set(srclevel, "mute", !muted, nullptr);
gst_object_unref(srclevel);
isMuted = !muted;
return true;
gboolean muted;
g_object_get(srclevel, "mute", &muted, nullptr);
g_object_set(srclevel, "mute", !muted, nullptr);
gst_object_unref(srclevel);
isMuted = !muted;
return true;
}
void
WebRTCSession::end()
{
nhlog::ui()->debug("WebRTC: ending session");
if (pipe_) {
gst_element_set_state(pipe_, GST_STATE_NULL);
gst_object_unref(pipe_);
pipe_ = nullptr;
}
webrtc_ = nullptr;
if (state_ != State::DISCONNECTED)
emit stateChanged(State::DISCONNECTED);
}
namespace {
std::string::const_iterator findName(const std::string &sdp, const std::string &name)
{
return std::search(sdp.cbegin(), sdp.cend(), name.cbegin(), name.cend(),
[](unsigned char c1, unsigned char c2) {return std::tolower(c1) == std::tolower(c2);});
}
int getPayloadType(const std::string &sdp, const std::string &name)
{
// eg a=rtpmap:111 opus/48000/2
auto e = findName(sdp, name);
if (e == sdp.cend()) {
nhlog::ui()->error("WebRTC: remote offer - " + name + " attribute missing");
return -1;
}
if (auto s = sdp.rfind(':', e - sdp.cbegin()); s == std::string::npos) {
nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name + " payload type");
return -1;
}
else {
++s;
try {
return std::stoi(std::string(sdp, s, e - sdp.cbegin() - s));
}
catch(...) {
nhlog::ui()->error("WebRTC: remote offer - unable to determine " + name + " payload type");
}
}
return -1;
}
gboolean
newBusMessage(GstBus *bus G_GNUC_UNUSED, GstMessage *msg, gpointer user_data)
{
WebRTCSession *session = (WebRTCSession*)user_data;
switch (GST_MESSAGE_TYPE(msg)) {
case GST_MESSAGE_EOS:
nhlog::ui()->error("WebRTC: end of stream");
session->end();
break;
case GST_MESSAGE_ERROR:
GError *error;
gchar *debug;
gst_message_parse_error(msg, &error, &debug);
nhlog::ui()->error("WebRTC: error from element {}: {}", GST_OBJECT_NAME(msg->src), error->message);
g_clear_error(&error);
g_free(debug);
session->end();
break;
default:
break;
}
return TRUE;
}
GstWebRTCSessionDescription*
parseSDP(const std::string &sdp, GstWebRTCSDPType type)
{
GstSDPMessage *msg;
gst_sdp_message_new(&msg);
if (gst_sdp_message_parse_buffer((guint8*)sdp.c_str(), sdp.size(), msg) == GST_SDP_OK) {
return gst_webrtc_session_description_new(type, msg);
}
else {
nhlog::ui()->error("WebRTC: failed to parse remote session description");
gst_object_unref(msg);
return nullptr;
}
}
void
generateOffer(GstElement *webrtc)
{
// create-offer first, then set-local-description
GstPromise *promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
g_signal_emit_by_name(webrtc, "create-offer", nullptr, promise);
}
void
setLocalDescription(GstPromise *promise, gpointer webrtc)
{
const GstStructure *reply = gst_promise_get_reply(promise);
gboolean isAnswer = gst_structure_id_has_field(reply, g_quark_from_string("answer"));
GstWebRTCSessionDescription *gstsdp = nullptr;
gst_structure_get(reply, isAnswer ? "answer" : "offer", GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &gstsdp, nullptr);
gst_promise_unref(promise);
g_signal_emit_by_name(webrtc, "set-local-description", gstsdp, nullptr);
gchar *sdp = gst_sdp_message_as_text(gstsdp->sdp);
localsdp_ = std::string(sdp);
g_free(sdp);
gst_webrtc_session_description_free(gstsdp);
nhlog::ui()->debug("WebRTC: local description set ({}):\n{}", isAnswer ? "answer" : "offer", localsdp_);
}
void
addLocalICECandidate(GstElement *webrtc G_GNUC_UNUSED, guint mlineIndex, gchar *candidate, gpointer G_GNUC_UNUSED)
{
nhlog::ui()->debug("WebRTC: local candidate: (m-line:{}):{}", mlineIndex, candidate);
if (WebRTCSession::instance().state() >= WebRTCSession::State::OFFERSENT) {
emit WebRTCSession::instance().newICECandidate({"audio", (uint16_t)mlineIndex, candidate});
return;
}
localcandidates_.push_back({"audio", (uint16_t)mlineIndex, candidate});
// GStreamer v1.16: webrtcbin's notify::ice-gathering-state triggers GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE too early
// fixed in v1.18
// use a 100ms timeout in the meantime
static guint timerid = 0;
if (timerid)
g_source_remove(timerid);
timerid = g_timeout_add(100, onICEGatheringCompletion, &timerid);
}
gboolean
onICEGatheringCompletion(gpointer timerid)
{
*(guint*)(timerid) = 0;
if (isoffering_) {
emit WebRTCSession::instance().offerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::OFFERSENT);
}
else {
emit WebRTCSession::instance().answerCreated(localsdp_, localcandidates_);
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ANSWERSENT);
}
return FALSE;
}
void
iceConnectionStateChanged(GstElement *webrtc, GParamSpec *pspec G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED)
{
GstWebRTCICEConnectionState newState;
g_object_get(webrtc, "ice-connection-state", &newState, nullptr);
switch (newState) {
case GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING:
nhlog::ui()->debug("WebRTC: GstWebRTCICEConnectionState -> Checking");
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTING);
break;
case GST_WEBRTC_ICE_CONNECTION_STATE_FAILED:
nhlog::ui()->error("WebRTC: GstWebRTCICEConnectionState -> Failed");
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::ICEFAILED);
break;
default:
break;
}
}
void
createAnswer(GstPromise *promise, gpointer webrtc)
{
// create-answer first, then set-local-description
gst_promise_unref(promise);
promise = gst_promise_new_with_change_func(setLocalDescription, webrtc, nullptr);
g_signal_emit_by_name(webrtc, "create-answer", nullptr, promise);
}
void
addDecodeBin(GstElement *webrtc G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
{
if (GST_PAD_DIRECTION(newpad) != GST_PAD_SRC)
return;
nhlog::ui()->debug("WebRTC: received incoming stream");
GstElement *decodebin = gst_element_factory_make("decodebin", nullptr);
g_signal_connect(decodebin, "pad-added", G_CALLBACK(linkNewPad), pipe);
gst_bin_add(GST_BIN(pipe), decodebin);
gst_element_sync_state_with_parent(decodebin);
GstPad *sinkpad = gst_element_get_static_pad(decodebin, "sink");
if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, sinkpad)))
nhlog::ui()->error("WebRTC: unable to link new pad");
gst_object_unref(sinkpad);
}
void
linkNewPad(GstElement *decodebin G_GNUC_UNUSED, GstPad *newpad, GstElement *pipe)
{
GstCaps *caps = gst_pad_get_current_caps(newpad);
if (!caps)
return;
const gchar *name = gst_structure_get_name(gst_caps_get_structure(caps, 0));
gst_caps_unref(caps);
GstPad *queuepad = nullptr;
if (g_str_has_prefix(name, "audio")) {
nhlog::ui()->debug("WebRTC: received incoming audio stream");
GstElement *queue = gst_element_factory_make("queue", nullptr);
GstElement *convert = gst_element_factory_make("audioconvert", nullptr);
GstElement *resample = gst_element_factory_make("audioresample", nullptr);
GstElement *sink = gst_element_factory_make("autoaudiosink", nullptr);
gst_bin_add_many(GST_BIN(pipe), queue, convert, resample, sink, nullptr);
gst_element_sync_state_with_parent(queue);
gst_element_sync_state_with_parent(convert);
gst_element_sync_state_with_parent(resample);
gst_element_sync_state_with_parent(sink);
gst_element_link_many(queue, convert, resample, sink, nullptr);
queuepad = gst_element_get_static_pad(queue, "sink");
}
if (queuepad) {
if (GST_PAD_LINK_FAILED(gst_pad_link(newpad, queuepad)))
nhlog::ui()->error("WebRTC: unable to link new pad");
else {
emit WebRTCSession::instance().stateChanged(WebRTCSession::State::CONNECTED);
}
gst_object_unref(queuepad);
}
}
nhlog::ui()->debug("WebRTC: ending session");
if (pipe_) {
gst_element_set_state(pipe_, GST_STATE_NULL);
gst_object_unref(pipe_);
pipe_ = nullptr;
}
webrtc_ = nullptr;
if (state_ != State::DISCONNECTED)
emit stateChanged(State::DISCONNECTED);
}

53
src/WebRTCSession.h
View File

@ -14,52 +14,55 @@ class WebRTCSession : public QObject
Q_OBJECT
public:
enum class State {
ICEFAILED,
DISCONNECTED,
INITIATING,
INITIATED,
OFFERSENT,
ANSWERSENT,
CONNECTING,
CONNECTED
enum class State
{
DISCONNECTED,
ICEFAILED,
INITIATING,
INITIATED,
OFFERSENT,
ANSWERSENT,
CONNECTING,
CONNECTED
};
static WebRTCSession& instance()
static WebRTCSession &instance()
{
static WebRTCSession instance;
return instance;
static WebRTCSession instance;
return instance;
}
bool init(std::string *errorMessage = nullptr);
State state() const {return state_;}
State state() const { return state_; }
bool createOffer();
bool acceptOffer(const std::string &sdp);
bool acceptAnswer(const std::string &sdp);
void acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate>&);
void acceptICECandidates(const std::vector<mtx::events::msg::CallCandidates::Candidate> &);
bool toggleMuteAudioSrc(bool &isMuted);
void end();
void setStunServer(const std::string &stunServer) {stunServer_ = stunServer;}
void setTurnServers(const std::vector<std::string> &uris) {turnServers_ = uris;}
void setStunServer(const std::string &stunServer) { stunServer_ = stunServer; }
void setTurnServers(const std::vector<std::string> &uris) { turnServers_ = uris; }
signals:
void offerCreated(const std::string &sdp, const std::vector<mtx::events::msg::CallCandidates::Candidate>&);
void answerCreated(const std::string &sdp, const std::vector<mtx::events::msg::CallCandidates::Candidate>&);
void newICECandidate(const mtx::events::msg::CallCandidates::Candidate&);
void offerCreated(const std::string &sdp,
const std::vector<mtx::events::msg::CallCandidates::Candidate> &);
void answerCreated(const std::string &sdp,
const std::vector<mtx::events::msg::CallCandidates::Candidate> &);
void newICECandidate(const mtx::events::msg::CallCandidates::Candidate &);
void stateChanged(WebRTCSession::State); // explicit qualifier necessary for Qt
private slots:
void setState(State state) {state_ = state;}
void setState(State state) { state_ = state; }
private:
WebRTCSession();
bool initialised_ = false;
State state_ = State::DISCONNECTED;
GstElement *pipe_ = nullptr;
bool initialised_ = false;
State state_ = State::DISCONNECTED;
GstElement *pipe_ = nullptr;
GstElement *webrtc_ = nullptr;
std::string stunServer_;
std::vector<std::string> turnServers_;
@ -68,6 +71,6 @@ private:
bool createPipeline(int opusPayloadType);
public:
WebRTCSession(WebRTCSession const&) = delete;
void operator=(WebRTCSession const&) = delete;
WebRTCSession(WebRTCSession const &) = delete;
void operator=(WebRTCSession const &) = delete;
};

44
src/dialogs/AcceptCall.cpp
View File

@ -1,4 +1,5 @@
#include <QLabel>
#include <QPixmap>
#include <QPushButton>
#include <QString>
#include <QVBoxLayout>
@ -10,12 +11,12 @@
namespace dialogs {
AcceptCall::AcceptCall(
const QString &caller,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl,
QWidget *parent) : QWidget(parent)
AcceptCall::AcceptCall(const QString &caller,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl,
QWidget *parent)
: QWidget(parent)
{
setAutoFillBackground(true);
setWindowFlags(Qt::Tool | Qt::WindowStaysOnTopHint);
@ -39,8 +40,8 @@ AcceptCall::AcceptCall(
if (!displayName.isEmpty() && displayName != caller) {
displayNameLabel = new QLabel(displayName, this);
labelFont.setPointSizeF(f.pointSizeF() * 2);
displayNameLabel ->setFont(labelFont);
displayNameLabel ->setAlignment(Qt::AlignCenter);
displayNameLabel->setFont(labelFont);
displayNameLabel->setAlignment(Qt::AlignCenter);
}
QLabel *callerLabel = new QLabel(caller, this);
@ -48,19 +49,23 @@ AcceptCall::AcceptCall(
callerLabel->setFont(labelFont);
callerLabel->setAlignment(Qt::AlignCenter);
QLabel *voiceCallLabel = new QLabel("Voice Call", this);
labelFont.setPointSizeF(f.pointSizeF() * 1.1);
voiceCallLabel->setFont(labelFont);
voiceCallLabel->setAlignment(Qt::AlignCenter);
auto avatar = new Avatar(this, QFontMetrics(f).height() * 6);
if (!avatarUrl.isEmpty())
avatar->setImage(avatarUrl);
avatar->setImage(avatarUrl);
else
avatar->setLetter(utils::firstChar(roomName));
avatar->setLetter(utils::firstChar(roomName));
const int iconSize = 24;
auto buttonLayout = new QHBoxLayout();
const int iconSize = 24;
QLabel *callTypeIndicator = new QLabel(this);
QPixmap callIndicator(":/icons/icons/ui/place-call.png");
callTypeIndicator->setPixmap(callIndicator.scaled(iconSize * 2, iconSize * 2));
QLabel *callTypeLabel = new QLabel("Voice Call", this);
labelFont.setPointSizeF(f.pointSizeF() * 1.1);
callTypeLabel->setFont(labelFont);
callTypeLabel->setAlignment(Qt::AlignCenter);
auto buttonLayout = new QHBoxLayout;
buttonLayout->setSpacing(20);
acceptBtn_ = new QPushButton(tr("Accept"), this);
acceptBtn_->setDefault(true);
@ -74,10 +79,11 @@ AcceptCall::AcceptCall(
buttonLayout->addWidget(rejectBtn_);
if (displayNameLabel)
layout->addWidget(displayNameLabel, 0, Qt::AlignCenter);
layout->addWidget(displayNameLabel, 0, Qt::AlignCenter);
layout->addWidget(callerLabel, 0, Qt::AlignCenter);
layout->addWidget(voiceCallLabel, 0, Qt::AlignCenter);
layout->addWidget(avatar, 0, Qt::AlignCenter);
layout->addWidget(callTypeIndicator, 0, Qt::AlignCenter);
layout->addWidget(callTypeLabel, 0, Qt::AlignCenter);
layout->addLayout(buttonLayout);
connect(acceptBtn_, &QPushButton::clicked, this, [this]() {

11
src/dialogs/AcceptCall.h
View File

@ -12,12 +12,11 @@ class AcceptCall : public QWidget
Q_OBJECT
public:
AcceptCall(
const QString &caller,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl,
QWidget *parent = nullptr);
AcceptCall(const QString &caller,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl,
QWidget *parent = nullptr);
signals:
void accept();

24
src/dialogs/PlaceCall.cpp
View File

@ -10,12 +10,12 @@
namespace dialogs {
PlaceCall::PlaceCall(
const QString &callee,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl,
QWidget *parent) : QWidget(parent)
PlaceCall::PlaceCall(const QString &callee,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl,
QWidget *parent)
: QWidget(parent)
{
setAutoFillBackground(true);
setWindowFlags(Qt::Tool | Qt::WindowStaysOnTopHint);
@ -34,11 +34,13 @@ PlaceCall::PlaceCall(
f.setPointSizeF(f.pointSizeF());
auto avatar = new Avatar(this, QFontMetrics(f).height() * 3);
if (!avatarUrl.isEmpty())
avatar->setImage(avatarUrl);
avatar->setImage(avatarUrl);
else
avatar->setLetter(utils::firstChar(roomName));
voiceBtn_ = new QPushButton(tr("Voice Call"), this);
avatar->setLetter(utils::firstChar(roomName));
const int iconSize = 24;
voiceBtn_ = new QPushButton(tr("Voice"), this);
voiceBtn_->setIcon(QIcon(":/icons/icons/ui/place-call.png"));
voiceBtn_->setIconSize(QSize(iconSize, iconSize));
voiceBtn_->setDefault(true);
cancelBtn_ = new QPushButton(tr("Cancel"), this);
@ -47,7 +49,7 @@ PlaceCall::PlaceCall(
buttonLayout->addWidget(voiceBtn_);
buttonLayout->addWidget(cancelBtn_);
QString name = displayName.isEmpty() ? callee : displayName;
QString name = displayName.isEmpty() ? callee : displayName;
QLabel *label = new QLabel("Place a call to " + name + "?", this);
layout->addWidget(label);

11
src/dialogs/PlaceCall.h
View File

@ -12,12 +12,11 @@ class PlaceCall : public QWidget
Q_OBJECT
public:
PlaceCall(
const QString &callee,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl,
QWidget *parent = nullptr);
PlaceCall(const QString &callee,
const QString &displayName,
const QString &roomName,
const QString &avatarUrl,
QWidget *parent = nullptr);
signals:
void voice();

6
src/timeline/TimelineModel.cpp
View File

@ -796,9 +796,11 @@ TimelineModel::internalAddEvents(
} else if (std::holds_alternative<mtx::events::RoomEvent<
mtx::events::msg::CallCandidates>>(e_) ||
std::holds_alternative<
mtx::events::RoomEvent<mtx::events::msg::CallAnswer>>( e_) ||
mtx::events::RoomEvent<mtx::events::msg::CallAnswer>>(
e_) ||
std::holds_alternative<
mtx::events::RoomEvent<mtx::events::msg::CallHangUp>>( e_)) {
mtx::events::RoomEvent<mtx::events::msg::CallHangUp>>(
e_)) {
emit newCallEvent(e_);
}
}