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Code Issues Releases Wiki Activity

Fix build on Linux 64 bit.

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This commit is contained in:
John Preston
2020-08-11 13:59:48 +04:00
parent bd16708781
commit eda22b925f
9 changed files with 71 additions and 55 deletions
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28
Telegram/SourceFiles/calls/calls_call.cpp
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@@ -87,7 +87,7 @@ void AppendServer(
if (host.isEmpty()) {
return;
}
list.push_back({
list.push_back(tgcalls::RtcServer{
.host = host.toStdString(),
.port = port,
.isTurn = false
@@ -100,7 +100,7 @@ void AppendServer(
const auto password = qs(data.vpassword());
if (data.is_turn() && !username.isEmpty() && !password.isEmpty()) {
const auto pushTurn = [&](const QString &host) {
list.push_back({
list.push_back(tgcalls::RtcServer{
.host = host.toStdString(),
.port = port,
.login = username.toStdString(),
@@ -143,8 +143,8 @@ uint64 ComputeFingerprint(bytes::const_span authKey) {
return WrapVersions(tgcalls::Meta::Versions() | ranges::action::reverse);
}
[[nodiscard]] webrtc::VideoState StartVideoState(bool enabled) {
using State = webrtc::VideoState;
[[nodiscard]] Webrtc::VideoState StartVideoState(bool enabled) {
using State = Webrtc::VideoState;
return enabled ? State::Active : State::Inactive;
}
@@ -159,8 +159,8 @@ Call::Call(
, _user(user)
, _api(&_user->session().mtp())
, _type(type)
, _videoIncoming(std::make_unique<webrtc::VideoTrack>(StartVideoState(video)))
, _videoOutgoing(std::make_unique<webrtc::VideoTrack>(StartVideoState(video))) {
, _videoIncoming(std::make_unique<Webrtc::VideoTrack>(StartVideoState(video)))
, _videoOutgoing(std::make_unique<Webrtc::VideoTrack>(StartVideoState(video))) {
_discardByTimeoutTimer.setCallback([=] { hangup(); });
if (_type == Type::Outgoing) {
@@ -345,14 +345,14 @@ void Call::setMuted(bool mute) {
void Call::setupOutgoingVideo() {
const auto started = _videoOutgoing->state();
_videoOutgoing->stateValue(
) | rpl::start_with_next([=](webrtc::VideoState state) {
) | rpl::start_with_next([=](Webrtc::VideoState state) {
if (_state.current() != State::Established
&& state != started
&& !_videoCapture) {
_videoOutgoing->setState(started);
} else if (state != webrtc::VideoState::Inactive) {
} else if (state != Webrtc::VideoState::Inactive) {
// Paused not supported right now.
Assert(state == webrtc::VideoState::Active);
Assert(state == Webrtc::VideoState::Active);
if (!_videoCapture) {
_videoCapture = tgcalls::VideoCaptureInterface::Create();
_videoCapture->setOutput(_videoOutgoing->sink());
@@ -367,11 +367,11 @@ void Call::setupOutgoingVideo() {
}, _lifetime);
}
not_null<webrtc::VideoTrack*> Call::videoIncoming() const {
not_null<Webrtc::VideoTrack*> Call::videoIncoming() const {
return _videoIncoming.get();
}
not_null<webrtc::VideoTrack*> Call::videoOutgoing() const {
not_null<Webrtc::VideoTrack*> Call::videoOutgoing() const {
return _videoOutgoing.get();
}
@@ -589,7 +589,7 @@ void Call::updateRemoteMediaState(
}();
_videoIncoming->setState([&] {
using From = tgcalls::VideoState;
using To = webrtc::VideoState;
using To = Webrtc::VideoState;
switch (video) {
case From::Inactive: return To::Inactive;
case From::Paused: return To::Paused;
@@ -992,8 +992,8 @@ void Call::finish(FinishType type, const MTPPhoneCallDiscardReason &reason) {
auto duration = getDurationMs() / 1000;
auto connectionId = _instance ? _instance->getPreferredRelayId() : 0;
_finishByTimeoutTimer.call(kHangupTimeoutMs, [this, finalState] { setState(finalState); });
const auto flags = ((_videoIncoming->state() != webrtc::VideoState::Inactive)
|| (_videoOutgoing->state() != webrtc::VideoState::Inactive))
const auto flags = ((_videoIncoming->state() != Webrtc::VideoState::Inactive)
|| (_videoOutgoing->state() != Webrtc::VideoState::Inactive))
? MTPphone_DiscardCall::Flag::f_video
: MTPphone_DiscardCall::Flag(0);
_api.request(MTPphone_DiscardCall(