1122 lines
		
	
	
	
		
			32 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			1122 lines
		
	
	
	
		
			32 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
| /*
 | |
| This file is part of Telegram Desktop,
 | |
| the official desktop application for the Telegram messaging service.
 | |
| 
 | |
| For license and copyright information please follow this link:
 | |
| https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
 | |
| */
 | |
| #include "calls/calls_call.h"
 | |
| 
 | |
| #include "main/main_session.h"
 | |
| #include "main/main_account.h"
 | |
| #include "main/main_app_config.h"
 | |
| #include "apiwrap.h"
 | |
| #include "lang/lang_keys.h"
 | |
| #include "boxes/confirm_box.h"
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| #include "boxes/rate_call_box.h"
 | |
| #include "calls/calls_instance.h"
 | |
| #include "base/openssl_help.h"
 | |
| #include "mtproto/mtproto_dh_utils.h"
 | |
| #include "mtproto/mtproto_config.h"
 | |
| #include "core/application.h"
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| #include "core/core_settings.h"
 | |
| #include "media/audio/media_audio_track.h"
 | |
| #include "base/platform/base_platform_info.h"
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| #include "calls/calls_panel.h"
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| #include "webrtc/webrtc_video_track.h"
 | |
| #include "webrtc/webrtc_media_devices.h"
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| #include "webrtc/webrtc_create_adm.h"
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| #include "data/data_user.h"
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| #include "data/data_session.h"
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| 
 | |
| #include <tgcalls/Instance.h>
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| #include <tgcalls/VideoCaptureInterface.h>
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| #include <tgcalls/StaticThreads.h>
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| 
 | |
| namespace tgcalls {
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| class InstanceImpl;
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| class InstanceImplLegacy;
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| class InstanceImplReference;
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| void SetLegacyGlobalServerConfig(const std::string &serverConfig);
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| } // namespace tgcalls
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| 
 | |
| namespace Calls {
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| namespace {
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| 
 | |
| constexpr auto kMinLayer = 65;
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| constexpr auto kHangupTimeoutMs = 5000;
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| constexpr auto kSha256Size = 32;
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| const auto kDefaultVersion = "2.4.4"_q;
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| 
 | |
| const auto RegisterTag = tgcalls::Register<tgcalls::InstanceImpl>();
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| const auto RegisterTagLegacy = tgcalls::Register<tgcalls::InstanceImplLegacy>();
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| 
 | |
| void AppendEndpoint(
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| 		std::vector<tgcalls::Endpoint> &list,
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| 		const MTPPhoneConnection &connection) {
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| 	connection.match([&](const MTPDphoneConnection &data) {
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| 		if (data.vpeer_tag().v.length() != 16) {
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| 			return;
 | |
| 		}
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| 		tgcalls::Endpoint endpoint = {
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| 			.endpointId = (int64_t)data.vid().v,
 | |
| 			.host = tgcalls::EndpointHost{
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| 				.ipv4 = data.vip().v.toStdString(),
 | |
| 				.ipv6 = data.vipv6().v.toStdString() },
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| 			.port = (uint16_t)data.vport().v,
 | |
| 			.type = tgcalls::EndpointType::UdpRelay,
 | |
| 		};
 | |
| 		const auto tag = data.vpeer_tag().v;
 | |
| 		if (tag.size() >= 16) {
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| 			memcpy(endpoint.peerTag, tag.data(), 16);
 | |
| 		}
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| 		list.push_back(std::move(endpoint));
 | |
| 	}, [&](const MTPDphoneConnectionWebrtc &data) {
 | |
| 	});
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| }
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| 
 | |
| void AppendServer(
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| 		std::vector<tgcalls::RtcServer> &list,
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| 		const MTPPhoneConnection &connection) {
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| 	connection.match([&](const MTPDphoneConnection &data) {
 | |
| 	}, [&](const MTPDphoneConnectionWebrtc &data) {
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| 		const auto host = qs(data.vip());
 | |
| 		const auto hostv6 = qs(data.vipv6());
 | |
| 		const auto port = uint16_t(data.vport().v);
 | |
| 		if (data.is_stun()) {
 | |
| 			const auto pushStun = [&](const QString &host) {
 | |
| 				if (host.isEmpty()) {
 | |
| 					return;
 | |
| 				}
 | |
| 				list.push_back(tgcalls::RtcServer{
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| 					.host = host.toStdString(),
 | |
| 					.port = port,
 | |
| 					.isTurn = false
 | |
| 				});
 | |
| 			};
 | |
| 			pushStun(host);
 | |
| 			pushStun(hostv6);
 | |
| 		}
 | |
| 		const auto username = qs(data.vusername());
 | |
| 		const auto password = qs(data.vpassword());
 | |
| 		if (data.is_turn() && !username.isEmpty() && !password.isEmpty()) {
 | |
| 			const auto pushTurn = [&](const QString &host) {
 | |
| 				list.push_back(tgcalls::RtcServer{
 | |
| 					.host = host.toStdString(),
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| 					.port = port,
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| 					.login = username.toStdString(),
 | |
| 					.password = password.toStdString(),
 | |
| 					.isTurn = true,
 | |
| 				});
 | |
| 			};
 | |
| 			pushTurn(host);
 | |
| 			pushTurn(hostv6);
 | |
| 		}
 | |
| 	});
 | |
| }
 | |
| 
 | |
| constexpr auto kFingerprintDataSize = 256;
 | |
| uint64 ComputeFingerprint(bytes::const_span authKey) {
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| 	Expects(authKey.size() == kFingerprintDataSize);
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| 
 | |
| 	auto hash = openssl::Sha1(authKey);
 | |
| 	return (gsl::to_integer<uint64>(hash[19]) << 56)
 | |
| 		| (gsl::to_integer<uint64>(hash[18]) << 48)
 | |
| 		| (gsl::to_integer<uint64>(hash[17]) << 40)
 | |
| 		| (gsl::to_integer<uint64>(hash[16]) << 32)
 | |
| 		| (gsl::to_integer<uint64>(hash[15]) << 24)
 | |
| 		| (gsl::to_integer<uint64>(hash[14]) << 16)
 | |
| 		| (gsl::to_integer<uint64>(hash[13]) << 8)
 | |
| 		| (gsl::to_integer<uint64>(hash[12]));
 | |
| }
 | |
| 
 | |
| [[nodiscard]] QVector<MTPstring> WrapVersions(
 | |
| 		const std::vector<std::string> &data) {
 | |
| 	auto result = QVector<MTPstring>();
 | |
| 	result.reserve(data.size());
 | |
| 	for (const auto &version : data) {
 | |
| 		result.push_back(MTP_string(version));
 | |
| 	}
 | |
| 	return result;
 | |
| }
 | |
| 
 | |
| [[nodiscard]] QVector<MTPstring> CollectVersionsForApi() {
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| 	return WrapVersions(tgcalls::Meta::Versions() | ranges::actions::reverse);
 | |
| }
 | |
| 
 | |
| [[nodiscard]] Webrtc::VideoState StartVideoState(bool enabled) {
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| 	using State = Webrtc::VideoState;
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| 	return enabled ? State::Active : State::Inactive;
 | |
| }
 | |
| 
 | |
| } // namespace
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| 
 | |
| Call::Call(
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| 	not_null<Delegate*> delegate,
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| 	not_null<UserData*> user,
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| 	Type type,
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| 	bool video)
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| : _delegate(delegate)
 | |
| , _user(user)
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| , _api(&_user->session().mtp())
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| , _type(type)
 | |
| , _videoIncoming(
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| 	std::make_unique<Webrtc::VideoTrack>(
 | |
| 		StartVideoState(video)))
 | |
| , _videoOutgoing(
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| 	std::make_unique<Webrtc::VideoTrack>(
 | |
| 		StartVideoState(video))) {
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| 	_discardByTimeoutTimer.setCallback([=] { hangup(); });
 | |
| 
 | |
| 	if (_type == Type::Outgoing) {
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| 		setState(State::Requesting);
 | |
| 	} else {
 | |
| 		startWaitingTrack();
 | |
| 	}
 | |
| 	setupOutgoingVideo();
 | |
| }
 | |
| 
 | |
| void Call::generateModExpFirst(bytes::const_span randomSeed) {
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| 	auto first = MTP::CreateModExp(_dhConfig.g, _dhConfig.p, randomSeed);
 | |
| 	if (first.modexp.empty()) {
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| 		LOG(("Call Error: Could not compute mod-exp first."));
 | |
| 		finish(FinishType::Failed);
 | |
| 		return;
 | |
| 	}
 | |
| 
 | |
| 	_randomPower = std::move(first.randomPower);
 | |
| 	if (_type == Type::Incoming) {
 | |
| 		_gb = std::move(first.modexp);
 | |
| 	} else {
 | |
| 		_ga = std::move(first.modexp);
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| 		_gaHash = openssl::Sha256(_ga);
 | |
| 	}
 | |
| }
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| 
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| bool Call::isIncomingWaiting() const {
 | |
| 	if (type() != Call::Type::Incoming) {
 | |
| 		return false;
 | |
| 	}
 | |
| 	return (state() == State::Starting)
 | |
| 		|| (state() == State::WaitingIncoming);
 | |
| }
 | |
| 
 | |
| void Call::start(bytes::const_span random) {
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| 	// Save config here, because it is possible that it changes between
 | |
| 	// different usages inside the same call.
 | |
| 	_dhConfig = _delegate->getDhConfig();
 | |
| 	Assert(_dhConfig.g != 0);
 | |
| 	Assert(!_dhConfig.p.empty());
 | |
| 
 | |
| 	generateModExpFirst(random);
 | |
| 	const auto state = _state.current();
 | |
| 	if (state == State::Starting || state == State::Requesting) {
 | |
| 		if (_type == Type::Outgoing) {
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| 			startOutgoing();
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| 		} else {
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| 			startIncoming();
 | |
| 		}
 | |
| 	} else if (state == State::ExchangingKeys
 | |
| 		&& _answerAfterDhConfigReceived) {
 | |
| 		answer();
 | |
| 	}
 | |
| }
 | |
| 
 | |
| void Call::startOutgoing() {
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| 	Expects(_type == Type::Outgoing);
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| 	Expects(_state.current() == State::Requesting);
 | |
| 	Expects(_gaHash.size() == kSha256Size);
 | |
| 
 | |
| 	const auto flags = _videoCapture
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| 		? MTPphone_RequestCall::Flag::f_video
 | |
| 		: MTPphone_RequestCall::Flag(0);
 | |
| 	_api.request(MTPphone_RequestCall(
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| 		MTP_flags(flags),
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| 		_user->inputUser,
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| 		MTP_int(openssl::RandomValue<int32>()),
 | |
| 		MTP_bytes(_gaHash),
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| 		MTP_phoneCallProtocol(
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| 			MTP_flags(MTPDphoneCallProtocol::Flag::f_udp_p2p
 | |
| 				| MTPDphoneCallProtocol::Flag::f_udp_reflector),
 | |
| 			MTP_int(kMinLayer),
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| 			MTP_int(tgcalls::Meta::MaxLayer()),
 | |
| 			MTP_vector(CollectVersionsForApi()))
 | |
| 	)).done([=](const MTPphone_PhoneCall &result) {
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| 		Expects(result.type() == mtpc_phone_phoneCall);
 | |
| 
 | |
| 		setState(State::Waiting);
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| 
 | |
| 		auto &call = result.c_phone_phoneCall();
 | |
| 		_user->session().data().processUsers(call.vusers());
 | |
| 		if (call.vphone_call().type() != mtpc_phoneCallWaiting) {
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| 			LOG(("Call Error: Expected phoneCallWaiting in response to phone.requestCall()"));
 | |
| 			finish(FinishType::Failed);
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| 			return;
 | |
| 		}
 | |
| 
 | |
| 		auto &phoneCall = call.vphone_call();
 | |
| 		auto &waitingCall = phoneCall.c_phoneCallWaiting();
 | |
| 		_id = waitingCall.vid().v;
 | |
| 		_accessHash = waitingCall.vaccess_hash().v;
 | |
| 		if (_finishAfterRequestingCall != FinishType::None) {
 | |
| 			if (_finishAfterRequestingCall == FinishType::Failed) {
 | |
| 				finish(_finishAfterRequestingCall);
 | |
| 			} else {
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| 				hangup();
 | |
| 			}
 | |
| 			return;
 | |
| 		}
 | |
| 
 | |
| 		const auto &config = _user->session().serverConfig();
 | |
| 		_discardByTimeoutTimer.callOnce(config.callReceiveTimeoutMs);
 | |
| 		handleUpdate(phoneCall);
 | |
| 	}).fail([this](const MTP::Error &error) {
 | |
| 		handleRequestError(error);
 | |
| 	}).send();
 | |
| }
 | |
| 
 | |
| void Call::startIncoming() {
 | |
| 	Expects(_type == Type::Incoming);
 | |
| 	Expects(_state.current() == State::Starting);
 | |
| 
 | |
| 	_api.request(MTPphone_ReceivedCall(
 | |
| 		MTP_inputPhoneCall(MTP_long(_id), MTP_long(_accessHash))
 | |
| 	)).done([=](const MTPBool &result) {
 | |
| 		if (_state.current() == State::Starting) {
 | |
| 			setState(State::WaitingIncoming);
 | |
| 		}
 | |
| 	}).fail([=](const MTP::Error &error) {
 | |
| 		handleRequestError(error);
 | |
| 	}).send();
 | |
| }
 | |
| 
 | |
| void Call::switchVideoOutgoing() {
 | |
| 	const auto video = _videoOutgoing->state() == Webrtc::VideoState::Active;
 | |
| 	_delegate->callRequestPermissionsOrFail(crl::guard(this, [=] {
 | |
| 		videoOutgoing()->setState(StartVideoState(!video));
 | |
| 	}), true);
 | |
| 
 | |
| }
 | |
| 
 | |
| void Call::answer() {
 | |
| 	const auto video = _videoOutgoing->state() == Webrtc::VideoState::Active;
 | |
| 	_delegate->callRequestPermissionsOrFail(crl::guard(this, [=] {
 | |
| 		actuallyAnswer();
 | |
| 	}), video);
 | |
| }
 | |
| 
 | |
| void Call::actuallyAnswer() {
 | |
| 	Expects(_type == Type::Incoming);
 | |
| 
 | |
| 	const auto state = _state.current();
 | |
| 	if (state != State::Starting && state != State::WaitingIncoming) {
 | |
| 		if (state != State::ExchangingKeys
 | |
| 			|| !_answerAfterDhConfigReceived) {
 | |
| 			return;
 | |
| 		}
 | |
| 	}
 | |
| 	setState(State::ExchangingKeys);
 | |
| 	if (_gb.empty()) {
 | |
| 		_answerAfterDhConfigReceived = true;
 | |
| 		return;
 | |
| 	} else {
 | |
| 		_answerAfterDhConfigReceived = false;
 | |
| 	}
 | |
| 	_api.request(MTPphone_AcceptCall(
 | |
| 		MTP_inputPhoneCall(MTP_long(_id), MTP_long(_accessHash)),
 | |
| 		MTP_bytes(_gb),
 | |
| 		MTP_phoneCallProtocol(
 | |
| 			MTP_flags(MTPDphoneCallProtocol::Flag::f_udp_p2p
 | |
| 				| MTPDphoneCallProtocol::Flag::f_udp_reflector),
 | |
| 			MTP_int(kMinLayer),
 | |
| 			MTP_int(tgcalls::Meta::MaxLayer()),
 | |
| 			MTP_vector(CollectVersionsForApi()))
 | |
| 	)).done([=](const MTPphone_PhoneCall &result) {
 | |
| 		Expects(result.type() == mtpc_phone_phoneCall);
 | |
| 
 | |
| 		auto &call = result.c_phone_phoneCall();
 | |
| 		_user->session().data().processUsers(call.vusers());
 | |
| 		if (call.vphone_call().type() != mtpc_phoneCallWaiting) {
 | |
| 			LOG(("Call Error: "
 | |
| 				"Not phoneCallWaiting in response to phone.acceptCall."));
 | |
| 			finish(FinishType::Failed);
 | |
| 			return;
 | |
| 		}
 | |
| 
 | |
| 		handleUpdate(call.vphone_call());
 | |
| 	}).fail([=](const MTP::Error &error) {
 | |
| 		handleRequestError(error);
 | |
| 	}).send();
 | |
| }
 | |
| 
 | |
| void Call::setMuted(bool mute) {
 | |
| 	_muted = mute;
 | |
| 	if (_instance) {
 | |
| 		_instance->setMuteMicrophone(mute);
 | |
| 	}
 | |
| }
 | |
| 
 | |
| void Call::setupOutgoingVideo() {
 | |
| 	static const auto hasDevices = [] {
 | |
| 		return !Webrtc::GetVideoInputList().empty();
 | |
| 	};
 | |
| 	const auto started = _videoOutgoing->state();
 | |
| 	if (!hasDevices()) {
 | |
| 		_videoOutgoing->setState(Webrtc::VideoState::Inactive);
 | |
| 	}
 | |
| 	_videoOutgoing->stateValue(
 | |
| 	) | rpl::start_with_next([=](Webrtc::VideoState state) {
 | |
| 		if (state != Webrtc::VideoState::Inactive && !hasDevices()) {
 | |
| 			_errors.fire({ ErrorType::NoCamera });
 | |
| 			_videoOutgoing->setState(Webrtc::VideoState::Inactive);
 | |
| 		} else if (_state.current() != State::Established
 | |
| 			&& (state != Webrtc::VideoState::Inactive)
 | |
| 			&& (started == Webrtc::VideoState::Inactive)) {
 | |
| 			_errors.fire({ ErrorType::NotStartedCall });
 | |
| 			_videoOutgoing->setState(Webrtc::VideoState::Inactive);
 | |
| 		} else if (state != Webrtc::VideoState::Inactive
 | |
| 			&& _instance
 | |
| 			&& !_instance->supportsVideo()) {
 | |
| 			_errors.fire({ ErrorType::NotVideoCall });
 | |
| 			_videoOutgoing->setState(Webrtc::VideoState::Inactive);
 | |
| 		} else if (state != Webrtc::VideoState::Inactive) {
 | |
| 			// Paused not supported right now.
 | |
| 			Assert(state == Webrtc::VideoState::Active);
 | |
| 			if (!_videoCapture) {
 | |
| 				_videoCapture = _delegate->callGetVideoCapture();
 | |
| 				_videoCapture->setOutput(_videoOutgoing->sink());
 | |
| 			}
 | |
| 			if (_instance) {
 | |
| 				_instance->setVideoCapture(_videoCapture);
 | |
| 			}
 | |
| 			_videoCapture->setState(tgcalls::VideoState::Active);
 | |
| 		} else if (_videoCapture) {
 | |
| 			_videoCapture->setState(tgcalls::VideoState::Inactive);
 | |
| 		}
 | |
| 	}, _lifetime);
 | |
| }
 | |
| 
 | |
| not_null<Webrtc::VideoTrack*> Call::videoIncoming() const {
 | |
| 	return _videoIncoming.get();
 | |
| }
 | |
| 
 | |
| not_null<Webrtc::VideoTrack*> Call::videoOutgoing() const {
 | |
| 	return _videoOutgoing.get();
 | |
| }
 | |
| 
 | |
| crl::time Call::getDurationMs() const {
 | |
| 	return _startTime ? (crl::now() - _startTime) : 0;
 | |
| }
 | |
| 
 | |
| void Call::hangup() {
 | |
| 	const auto state = _state.current();
 | |
| 	if (state == State::Busy) {
 | |
| 		_delegate->callFinished(this);
 | |
| 	} else {
 | |
| 		auto missed = (state == State::Ringing || (state == State::Waiting && _type == Type::Outgoing));
 | |
| 		auto declined = isIncomingWaiting();
 | |
| 		auto reason = missed ? MTP_phoneCallDiscardReasonMissed() :
 | |
| 			declined ? MTP_phoneCallDiscardReasonBusy() : MTP_phoneCallDiscardReasonHangup();
 | |
| 		finish(FinishType::Ended, reason);
 | |
| 	}
 | |
| }
 | |
| 
 | |
| void Call::redial() {
 | |
| 	if (_state.current() != State::Busy) {
 | |
| 		return;
 | |
| 	}
 | |
| 	Assert(_instance == nullptr);
 | |
| 	_type = Type::Outgoing;
 | |
| 	setState(State::Requesting);
 | |
| 	_answerAfterDhConfigReceived = false;
 | |
| 	startWaitingTrack();
 | |
| 	_delegate->callRedial(this);
 | |
| }
 | |
| 
 | |
| QString Call::getDebugLog() const {
 | |
| 	return _instance
 | |
| 		? QString::fromStdString(_instance->getDebugInfo())
 | |
| 		: QString();
 | |
| }
 | |
| 
 | |
| void Call::startWaitingTrack() {
 | |
| 	_waitingTrack = Media::Audio::Current().createTrack();
 | |
| 	auto trackFileName = Core::App().settings().getSoundPath(
 | |
| 		(_type == Type::Outgoing)
 | |
| 		? qsl("call_outgoing")
 | |
| 		: qsl("call_incoming"));
 | |
| 	_waitingTrack->samplePeakEach(kSoundSampleMs);
 | |
| 	_waitingTrack->fillFromFile(trackFileName);
 | |
| 	_waitingTrack->playInLoop();
 | |
| }
 | |
| 
 | |
| void Call::sendSignalingData(const QByteArray &data) {
 | |
| 	_api.request(MTPphone_SendSignalingData(
 | |
| 		MTP_inputPhoneCall(
 | |
| 			MTP_long(_id),
 | |
| 			MTP_long(_accessHash)),
 | |
| 		MTP_bytes(data)
 | |
| 	)).done([=](const MTPBool &result) {
 | |
| 		if (!mtpIsTrue(result)) {
 | |
| 			finish(FinishType::Failed);
 | |
| 		}
 | |
| 	}).fail([=](const MTP::Error &error) {
 | |
| 		handleRequestError(error);
 | |
| 	}).send();
 | |
| }
 | |
| 
 | |
| float64 Call::getWaitingSoundPeakValue() const {
 | |
| 	if (_waitingTrack) {
 | |
| 		auto when = crl::now() + kSoundSampleMs / 4;
 | |
| 		return _waitingTrack->getPeakValue(when);
 | |
| 	}
 | |
| 	return 0.;
 | |
| }
 | |
| 
 | |
| bool Call::isKeyShaForFingerprintReady() const {
 | |
| 	return (_keyFingerprint != 0);
 | |
| }
 | |
| 
 | |
| bytes::vector Call::getKeyShaForFingerprint() const {
 | |
| 	Expects(isKeyShaForFingerprintReady());
 | |
| 	Expects(!_ga.empty());
 | |
| 
 | |
| 	auto encryptedChatAuthKey = bytes::vector(_authKey.size() + _ga.size(), gsl::byte {});
 | |
| 	bytes::copy(gsl::make_span(encryptedChatAuthKey).subspan(0, _authKey.size()), _authKey);
 | |
| 	bytes::copy(gsl::make_span(encryptedChatAuthKey).subspan(_authKey.size(), _ga.size()), _ga);
 | |
| 	return openssl::Sha256(encryptedChatAuthKey);
 | |
| }
 | |
| 
 | |
| bool Call::handleUpdate(const MTPPhoneCall &call) {
 | |
| 	switch (call.type()) {
 | |
| 	case mtpc_phoneCallRequested: {
 | |
| 		auto &data = call.c_phoneCallRequested();
 | |
| 		if (_type != Type::Incoming
 | |
| 			|| _id != 0
 | |
| 			|| peerToUser(_user->id) != UserId(data.vadmin_id())) {
 | |
| 			Unexpected("phoneCallRequested call inside an existing call handleUpdate()");
 | |
| 		}
 | |
| 		if (_user->session().userId() != UserId(data.vparticipant_id())) {
 | |
| 			LOG(("Call Error: Wrong call participant_id %1, expected %2."
 | |
| 				).arg(data.vparticipant_id().v
 | |
| 				).arg(_user->session().userId().bare));
 | |
| 			finish(FinishType::Failed);
 | |
| 			return true;
 | |
| 		}
 | |
| 
 | |
| 		_id = data.vid().v;
 | |
| 		_accessHash = data.vaccess_hash().v;
 | |
| 		auto gaHashBytes = bytes::make_span(data.vg_a_hash().v);
 | |
| 		if (gaHashBytes.size() != kSha256Size) {
 | |
| 			LOG(("Call Error: Wrong g_a_hash size %1, expected %2."
 | |
| 				).arg(gaHashBytes.size()
 | |
| 				).arg(kSha256Size));
 | |
| 			finish(FinishType::Failed);
 | |
| 			return true;
 | |
| 		}
 | |
| 		_gaHash = bytes::make_vector(gaHashBytes);
 | |
| 	} return true;
 | |
| 
 | |
| 	case mtpc_phoneCallEmpty: {
 | |
| 		auto &data = call.c_phoneCallEmpty();
 | |
| 		if (data.vid().v != _id) {
 | |
| 			return false;
 | |
| 		}
 | |
| 		LOG(("Call Error: phoneCallEmpty received."));
 | |
| 		finish(FinishType::Failed);
 | |
| 	} return true;
 | |
| 
 | |
| 	case mtpc_phoneCallWaiting: {
 | |
| 		auto &data = call.c_phoneCallWaiting();
 | |
| 		if (data.vid().v != _id) {
 | |
| 			return false;
 | |
| 		}
 | |
| 		if (_type == Type::Outgoing
 | |
| 			&& _state.current() == State::Waiting
 | |
| 			&& data.vreceive_date().value_or_empty() != 0) {
 | |
| 			const auto &config = _user->session().serverConfig();
 | |
| 			_discardByTimeoutTimer.callOnce(config.callRingTimeoutMs);
 | |
| 			setState(State::Ringing);
 | |
| 			startWaitingTrack();
 | |
| 		}
 | |
| 	} return true;
 | |
| 
 | |
| 	case mtpc_phoneCall: {
 | |
| 		auto &data = call.c_phoneCall();
 | |
| 		if (data.vid().v != _id) {
 | |
| 			return false;
 | |
| 		}
 | |
| 		if (_type == Type::Incoming
 | |
| 			&& _state.current() == State::ExchangingKeys
 | |
| 			&& !_instance) {
 | |
| 			startConfirmedCall(data);
 | |
| 		}
 | |
| 	} return true;
 | |
| 
 | |
| 	case mtpc_phoneCallDiscarded: {
 | |
| 		auto &data = call.c_phoneCallDiscarded();
 | |
| 		if (data.vid().v != _id) {
 | |
| 			return false;
 | |
| 		}
 | |
| 		if (data.is_need_debug()) {
 | |
| 			auto debugLog = _instance
 | |
| 				? _instance->getDebugInfo()
 | |
| 				: std::string();
 | |
| 			if (!debugLog.empty()) {
 | |
| 				user()->session().api().request(MTPphone_SaveCallDebug(
 | |
| 					MTP_inputPhoneCall(
 | |
| 						MTP_long(_id),
 | |
| 						MTP_long(_accessHash)),
 | |
| 					MTP_dataJSON(MTP_string(debugLog))
 | |
| 				)).send();
 | |
| 			}
 | |
| 		}
 | |
| 		if (data.is_need_rating() && _id && _accessHash) {
 | |
| 			Ui::show(Box<RateCallBox>(&_user->session(), _id, _accessHash));
 | |
| 		}
 | |
| 		const auto reason = data.vreason();
 | |
| 		if (reason && reason->type() == mtpc_phoneCallDiscardReasonDisconnect) {
 | |
| 			LOG(("Call Info: Discarded with DISCONNECT reason."));
 | |
| 		}
 | |
| 		if (reason && reason->type() == mtpc_phoneCallDiscardReasonBusy) {
 | |
| 			setState(State::Busy);
 | |
| 		} else if (_type == Type::Outgoing
 | |
| 			|| _state.current() == State::HangingUp) {
 | |
| 			setState(State::Ended);
 | |
| 		} else {
 | |
| 			setState(State::EndedByOtherDevice);
 | |
| 		}
 | |
| 	} return true;
 | |
| 
 | |
| 	case mtpc_phoneCallAccepted: {
 | |
| 		auto &data = call.c_phoneCallAccepted();
 | |
| 		if (data.vid().v != _id) {
 | |
| 			return false;
 | |
| 		}
 | |
| 		if (_type != Type::Outgoing) {
 | |
| 			LOG(("Call Error: "
 | |
| 				"Unexpected phoneCallAccepted for an incoming call."));
 | |
| 			finish(FinishType::Failed);
 | |
| 		} else if (checkCallFields(data)) {
 | |
| 			confirmAcceptedCall(data);
 | |
| 		}
 | |
| 	} return true;
 | |
| 	}
 | |
| 
 | |
| 	Unexpected("phoneCall type inside an existing call handleUpdate()");
 | |
| }
 | |
| 
 | |
| void Call::updateRemoteMediaState(
 | |
| 		tgcalls::AudioState audio,
 | |
| 		tgcalls::VideoState video) {
 | |
| 	_remoteAudioState = [&] {
 | |
| 		using From = tgcalls::AudioState;
 | |
| 		using To = RemoteAudioState;
 | |
| 		switch (audio) {
 | |
| 		case From::Active: return To::Active;
 | |
| 		case From::Muted: return To::Muted;
 | |
| 		}
 | |
| 		Unexpected("Audio state in remoteMediaStateUpdated.");
 | |
| 	}();
 | |
| 	_videoIncoming->setState([&] {
 | |
| 		using From = tgcalls::VideoState;
 | |
| 		using To = Webrtc::VideoState;
 | |
| 		switch (video) {
 | |
| 		case From::Inactive: return To::Inactive;
 | |
| 		case From::Paused: return To::Paused;
 | |
| 		case From::Active: return To::Active;
 | |
| 		}
 | |
| 		Unexpected("Video state in remoteMediaStateUpdated.");
 | |
| 	}());
 | |
| }
 | |
| 
 | |
| bool Call::handleSignalingData(
 | |
| 		const MTPDupdatePhoneCallSignalingData &data) {
 | |
| 	if (data.vphone_call_id().v != _id || !_instance) {
 | |
| 		return false;
 | |
| 	}
 | |
| 	auto prepared = ranges::views::all(
 | |
| 		data.vdata().v
 | |
| 	) | ranges::views::transform([](char byte) {
 | |
| 		return static_cast<uint8_t>(byte);
 | |
| 	}) | ranges::to_vector;
 | |
| 	_instance->receiveSignalingData(std::move(prepared));
 | |
| 	return true;
 | |
| }
 | |
| 
 | |
| void Call::confirmAcceptedCall(const MTPDphoneCallAccepted &call) {
 | |
| 	Expects(_type == Type::Outgoing);
 | |
| 
 | |
| 	if (_state.current() == State::ExchangingKeys
 | |
| 		|| _instance) {
 | |
| 		LOG(("Call Warning: Unexpected confirmAcceptedCall."));
 | |
| 		return;
 | |
| 	}
 | |
| 
 | |
| 	const auto firstBytes = bytes::make_span(call.vg_b().v);
 | |
| 	const auto computedAuthKey = MTP::CreateAuthKey(
 | |
| 		firstBytes,
 | |
| 		_randomPower,
 | |
| 		_dhConfig.p);
 | |
| 	if (computedAuthKey.empty()) {
 | |
| 		LOG(("Call Error: Could not compute mod-exp final."));
 | |
| 		finish(FinishType::Failed);
 | |
| 		return;
 | |
| 	}
 | |
| 
 | |
| 	MTP::AuthKey::FillData(_authKey, computedAuthKey);
 | |
| 	_keyFingerprint = ComputeFingerprint(_authKey);
 | |
| 
 | |
| 	setState(State::ExchangingKeys);
 | |
| 	_api.request(MTPphone_ConfirmCall(
 | |
| 		MTP_inputPhoneCall(MTP_long(_id), MTP_long(_accessHash)),
 | |
| 		MTP_bytes(_ga),
 | |
| 		MTP_long(_keyFingerprint),
 | |
| 		MTP_phoneCallProtocol(
 | |
| 			MTP_flags(MTPDphoneCallProtocol::Flag::f_udp_p2p
 | |
| 				| MTPDphoneCallProtocol::Flag::f_udp_reflector),
 | |
| 			MTP_int(kMinLayer),
 | |
| 			MTP_int(tgcalls::Meta::MaxLayer()),
 | |
| 			MTP_vector(CollectVersionsForApi()))
 | |
| 	)).done([=](const MTPphone_PhoneCall &result) {
 | |
| 		Expects(result.type() == mtpc_phone_phoneCall);
 | |
| 
 | |
| 		auto &call = result.c_phone_phoneCall();
 | |
| 		_user->session().data().processUsers(call.vusers());
 | |
| 		if (call.vphone_call().type() != mtpc_phoneCall) {
 | |
| 			LOG(("Call Error: Expected phoneCall in response to phone.confirmCall()"));
 | |
| 			finish(FinishType::Failed);
 | |
| 			return;
 | |
| 		}
 | |
| 
 | |
| 		createAndStartController(call.vphone_call().c_phoneCall());
 | |
| 	}).fail([=](const MTP::Error &error) {
 | |
| 		handleRequestError(error);
 | |
| 	}).send();
 | |
| }
 | |
| 
 | |
| void Call::startConfirmedCall(const MTPDphoneCall &call) {
 | |
| 	Expects(_type == Type::Incoming);
 | |
| 
 | |
| 	auto firstBytes = bytes::make_span(call.vg_a_or_b().v);
 | |
| 	if (_gaHash != openssl::Sha256(firstBytes)) {
 | |
| 		LOG(("Call Error: Wrong g_a hash received."));
 | |
| 		finish(FinishType::Failed);
 | |
| 		return;
 | |
| 	}
 | |
| 	_ga = bytes::vector(firstBytes.begin(), firstBytes.end());
 | |
| 
 | |
| 	auto computedAuthKey = MTP::CreateAuthKey(firstBytes, _randomPower, _dhConfig.p);
 | |
| 	if (computedAuthKey.empty()) {
 | |
| 		LOG(("Call Error: Could not compute mod-exp final."));
 | |
| 		finish(FinishType::Failed);
 | |
| 		return;
 | |
| 	}
 | |
| 
 | |
| 	MTP::AuthKey::FillData(_authKey, computedAuthKey);
 | |
| 	_keyFingerprint = ComputeFingerprint(_authKey);
 | |
| 
 | |
| 	createAndStartController(call);
 | |
| }
 | |
| 
 | |
| void Call::createAndStartController(const MTPDphoneCall &call) {
 | |
| 	_discardByTimeoutTimer.cancel();
 | |
| 	if (!checkCallFields(call) || _authKey.size() != 256) {
 | |
| 		return;
 | |
| 	}
 | |
| 
 | |
| 	const auto &protocol = call.vprotocol().c_phoneCallProtocol();
 | |
| 	const auto &serverConfig = _user->session().serverConfig();
 | |
| 
 | |
| 	auto encryptionKeyValue = std::make_shared<std::array<uint8_t, 256>>();
 | |
| 	memcpy(encryptionKeyValue->data(), _authKey.data(), 256);
 | |
| 
 | |
| 	const auto &settings = Core::App().settings();
 | |
| 
 | |
| 	const auto weak = base::make_weak(this);
 | |
| 	tgcalls::Descriptor descriptor = {
 | |
| 		.config = tgcalls::Config{
 | |
| 			.initializationTimeout = serverConfig.callConnectTimeoutMs / 1000.,
 | |
| 			.receiveTimeout = serverConfig.callPacketTimeoutMs / 1000.,
 | |
| 			.dataSaving = tgcalls::DataSaving::Never,
 | |
| 			.enableP2P = call.is_p2p_allowed(),
 | |
| 			.enableAEC = !Platform::IsMac10_7OrGreater(),
 | |
| 			.enableNS = true,
 | |
| 			.enableAGC = true,
 | |
| 			.enableVolumeControl = true,
 | |
| 			.maxApiLayer = protocol.vmax_layer().v,
 | |
| 		},
 | |
| 		.encryptionKey = tgcalls::EncryptionKey(
 | |
| 			std::move(encryptionKeyValue),
 | |
| 			(_type == Type::Outgoing)),
 | |
| 		.mediaDevicesConfig = tgcalls::MediaDevicesConfig{
 | |
| 			.audioInputId = settings.callInputDeviceId().toStdString(),
 | |
| 			.audioOutputId = settings.callOutputDeviceId().toStdString(),
 | |
| 			.inputVolume = 1.f,//settings.callInputVolume() / 100.f,
 | |
| 			.outputVolume = 1.f,//settings.callOutputVolume() / 100.f,
 | |
| 		},
 | |
| 		.videoCapture = _videoCapture,
 | |
| 		.stateUpdated = [=](tgcalls::State state) {
 | |
| 			crl::on_main(weak, [=] {
 | |
| 				handleControllerStateChange(state);
 | |
| 			});
 | |
| 		},
 | |
| 		.signalBarsUpdated = [=](int count) {
 | |
| 			crl::on_main(weak, [=] {
 | |
| 				handleControllerBarCountChange(count);
 | |
| 			});
 | |
| 		},
 | |
| 		.remoteMediaStateUpdated = [=](tgcalls::AudioState audio, tgcalls::VideoState video) {
 | |
| 			crl::on_main(weak, [=] {
 | |
| 				updateRemoteMediaState(audio, video);
 | |
| 			});
 | |
| 		},
 | |
| 		.signalingDataEmitted = [=](const std::vector<uint8_t> &data) {
 | |
| 			const auto bytes = QByteArray(
 | |
| 				reinterpret_cast<const char*>(data.data()),
 | |
| 				data.size());
 | |
| 			crl::on_main(weak, [=] {
 | |
| 				sendSignalingData(bytes);
 | |
| 			});
 | |
| 		},
 | |
| 		.createAudioDeviceModule = Webrtc::AudioDeviceModuleCreator(
 | |
| 			settings.callAudioBackend()),
 | |
| 	};
 | |
| 	if (Logs::DebugEnabled()) {
 | |
| 		auto callLogFolder = cWorkingDir() + qsl("DebugLogs");
 | |
| 		auto callLogPath = callLogFolder + qsl("/last_call_log.txt");
 | |
| 		auto callLogNative = QDir::toNativeSeparators(callLogPath);
 | |
| #ifdef Q_OS_WIN
 | |
| 		descriptor.config.logPath.data = callLogNative.toStdWString();
 | |
| #else // Q_OS_WIN
 | |
| 		const auto callLogUtf = QFile::encodeName(callLogNative);
 | |
| 		descriptor.config.logPath.data.resize(callLogUtf.size());
 | |
| 		ranges::copy(callLogUtf, descriptor.config.logPath.data.begin());
 | |
| #endif // Q_OS_WIN
 | |
| 		QFile(callLogPath).remove();
 | |
| 		QDir().mkpath(callLogFolder);
 | |
| 	}
 | |
| 
 | |
| 	for (const auto &connection : call.vconnections().v) {
 | |
| 		AppendEndpoint(descriptor.endpoints, connection);
 | |
| 	}
 | |
| 	for (const auto &connection : call.vconnections().v) {
 | |
| 		AppendServer(descriptor.rtcServers, connection);
 | |
| 	}
 | |
| 
 | |
| 	{
 | |
| 		auto &settingsProxy = Core::App().settings().proxy();
 | |
| 		using ProxyData = MTP::ProxyData;
 | |
| 		if (settingsProxy.useProxyForCalls() && settingsProxy.isEnabled()) {
 | |
| 			const auto &selected = settingsProxy.selected();
 | |
| 			if (selected.supportsCalls() && !selected.host.isEmpty()) {
 | |
| 				Assert(selected.type == ProxyData::Type::Socks5);
 | |
| 				descriptor.proxy = std::make_unique<tgcalls::Proxy>();
 | |
| 				descriptor.proxy->host = selected.host.toStdString();
 | |
| 				descriptor.proxy->port = selected.port;
 | |
| 				descriptor.proxy->login = selected.user.toStdString();
 | |
| 				descriptor.proxy->password = selected.password.toStdString();
 | |
| 			}
 | |
| 		}
 | |
| 	}
 | |
| 
 | |
| 	const auto version = call.vprotocol().match([&](
 | |
| 			const MTPDphoneCallProtocol &data) {
 | |
| 		return data.vlibrary_versions().v;
 | |
| 	}).value(0, MTP_bytes(kDefaultVersion)).v;
 | |
| 
 | |
| 	LOG(("Call Info: Creating instance with version '%1', allowP2P: %2").arg(
 | |
| 		QString::fromUtf8(version),
 | |
| 		Logs::b(descriptor.config.enableP2P)));
 | |
| 	_instance = tgcalls::Meta::Create(
 | |
| 		version.toStdString(),
 | |
| 		std::move(descriptor));
 | |
| 	if (!_instance) {
 | |
| 		LOG(("Call Error: Wrong library version: %1."
 | |
| 			).arg(QString::fromUtf8(version)));
 | |
| 		finish(FinishType::Failed);
 | |
| 		return;
 | |
| 	}
 | |
| 
 | |
| 	const auto raw = _instance.get();
 | |
| 	if (_muted.current()) {
 | |
| 		raw->setMuteMicrophone(_muted.current());
 | |
| 	}
 | |
| 
 | |
| 	raw->setIncomingVideoOutput(_videoIncoming->sink());
 | |
| 	raw->setAudioOutputDuckingEnabled(settings.callAudioDuckingEnabled());
 | |
| }
 | |
| 
 | |
| void Call::handleControllerStateChange(tgcalls::State state) {
 | |
| 	switch (state) {
 | |
| 	case tgcalls::State::WaitInit: {
 | |
| 		DEBUG_LOG(("Call Info: State changed to WaitingInit."));
 | |
| 		setState(State::WaitingInit);
 | |
| 	} break;
 | |
| 
 | |
| 	case tgcalls::State::WaitInitAck: {
 | |
| 		DEBUG_LOG(("Call Info: State changed to WaitingInitAck."));
 | |
| 		setState(State::WaitingInitAck);
 | |
| 	} break;
 | |
| 
 | |
| 	case tgcalls::State::Established: {
 | |
| 		DEBUG_LOG(("Call Info: State changed to Established."));
 | |
| 		setState(State::Established);
 | |
| 	} break;
 | |
| 
 | |
| 	case tgcalls::State::Failed: {
 | |
| 		auto error = _instance
 | |
| 			? QString::fromStdString(_instance->getLastError())
 | |
| 			: QString();
 | |
| 		LOG(("Call Info: State changed to Failed, error: %1.").arg(error));
 | |
| 		handleControllerError(error);
 | |
| 	} break;
 | |
| 
 | |
| 	default: LOG(("Call Error: Unexpected state in handleStateChange: %1"
 | |
| 		).arg(int(state)));
 | |
| 	}
 | |
| }
 | |
| 
 | |
| void Call::handleControllerBarCountChange(int count) {
 | |
| 	setSignalBarCount(count);
 | |
| }
 | |
| 
 | |
| void Call::setSignalBarCount(int count) {
 | |
| 	_signalBarCount = count;
 | |
| }
 | |
| 
 | |
| template <typename T>
 | |
| bool Call::checkCallCommonFields(const T &call) {
 | |
| 	auto checkFailed = [this] {
 | |
| 		finish(FinishType::Failed);
 | |
| 		return false;
 | |
| 	};
 | |
| 	if (call.vaccess_hash().v != _accessHash) {
 | |
| 		LOG(("Call Error: Wrong call access_hash."));
 | |
| 		return checkFailed();
 | |
| 	}
 | |
| 	auto adminId = (_type == Type::Outgoing) ? _user->session().userId() : peerToUser(_user->id);
 | |
| 	auto participantId = (_type == Type::Outgoing) ? peerToUser(_user->id) : _user->session().userId();
 | |
| 	if (UserId(call.vadmin_id()) != adminId) {
 | |
| 		LOG(("Call Error: Wrong call admin_id %1, expected %2.").arg(call.vadmin_id().v).arg(adminId.bare));
 | |
| 		return checkFailed();
 | |
| 	}
 | |
| 	if (UserId(call.vparticipant_id()) != participantId) {
 | |
| 		LOG(("Call Error: Wrong call participant_id %1, expected %2.").arg(call.vparticipant_id().v).arg(participantId.bare));
 | |
| 		return checkFailed();
 | |
| 	}
 | |
| 	return true;
 | |
| }
 | |
| 
 | |
| bool Call::checkCallFields(const MTPDphoneCall &call) {
 | |
| 	if (!checkCallCommonFields(call)) {
 | |
| 		return false;
 | |
| 	}
 | |
| 	if (call.vkey_fingerprint().v != _keyFingerprint) {
 | |
| 		LOG(("Call Error: Wrong call fingerprint."));
 | |
| 		finish(FinishType::Failed);
 | |
| 		return false;
 | |
| 	}
 | |
| 	return true;
 | |
| }
 | |
| 
 | |
| bool Call::checkCallFields(const MTPDphoneCallAccepted &call) {
 | |
| 	return checkCallCommonFields(call);
 | |
| }
 | |
| 
 | |
| void Call::setState(State state) {
 | |
| 	if (_state.current() == State::Failed) {
 | |
| 		return;
 | |
| 	}
 | |
| 	if (_state.current() == State::FailedHangingUp && state != State::Failed) {
 | |
| 		return;
 | |
| 	}
 | |
| 	if (_state.current() != state) {
 | |
| 		_state = state;
 | |
| 
 | |
| 		if (true
 | |
| 			&& state != State::Starting
 | |
| 			&& state != State::Requesting
 | |
| 			&& state != State::Waiting
 | |
| 			&& state != State::WaitingIncoming
 | |
| 			&& state != State::Ringing) {
 | |
| 			_waitingTrack.reset();
 | |
| 		}
 | |
| 		if (false
 | |
| 			|| state == State::Ended
 | |
| 			|| state == State::EndedByOtherDevice
 | |
| 			|| state == State::Failed
 | |
| 			|| state == State::Busy) {
 | |
| 			// Destroy controller before destroying Call Panel,
 | |
| 			// so that the panel hide animation is smooth.
 | |
| 			destroyController();
 | |
| 		}
 | |
| 		switch (state) {
 | |
| 		case State::Established:
 | |
| 			_startTime = crl::now();
 | |
| 			break;
 | |
| 		case State::ExchangingKeys:
 | |
| 			_delegate->callPlaySound(Delegate::CallSound::Connecting);
 | |
| 			break;
 | |
| 		case State::Ended:
 | |
| 			_delegate->callPlaySound(Delegate::CallSound::Ended);
 | |
| 			[[fallthrough]];
 | |
| 		case State::EndedByOtherDevice:
 | |
| 			_delegate->callFinished(this);
 | |
| 			break;
 | |
| 		case State::Failed:
 | |
| 			_delegate->callPlaySound(Delegate::CallSound::Ended);
 | |
| 			_delegate->callFailed(this);
 | |
| 			break;
 | |
| 		case State::Busy:
 | |
| 			_delegate->callPlaySound(Delegate::CallSound::Busy);
 | |
| 			break;
 | |
| 		}
 | |
| 	}
 | |
| }
 | |
| 
 | |
| void Call::setCurrentAudioDevice(bool input, const QString &deviceId) {
 | |
| 	if (_instance) {
 | |
| 		const auto id = deviceId.toStdString();
 | |
| 		if (input) {
 | |
| 			_instance->setAudioInputDevice(id);
 | |
| 		} else {
 | |
| 			_instance->setAudioOutputDevice(id);
 | |
| 		}
 | |
| 	}
 | |
| }
 | |
| 
 | |
| void Call::setCurrentVideoDevice(const QString &deviceId) {
 | |
| 	if (_videoCapture) {
 | |
| 		_videoCapture->switchToDevice(deviceId.toStdString());
 | |
| 	}
 | |
| }
 | |
| 
 | |
| //void Call::setAudioVolume(bool input, float level) {
 | |
| //	if (_instance) {
 | |
| //		if (input) {
 | |
| //			_instance->setInputVolume(level);
 | |
| //		} else {
 | |
| //			_instance->setOutputVolume(level);
 | |
| //		}
 | |
| //	}
 | |
| //}
 | |
| 
 | |
| void Call::setAudioDuckingEnabled(bool enabled) {
 | |
| 	if (_instance) {
 | |
| 		_instance->setAudioOutputDuckingEnabled(enabled);
 | |
| 	}
 | |
| }
 | |
| 
 | |
| void Call::finish(FinishType type, const MTPPhoneCallDiscardReason &reason) {
 | |
| 	Expects(type != FinishType::None);
 | |
| 
 | |
| 	setSignalBarCount(kSignalBarFinished);
 | |
| 
 | |
| 	auto finalState = (type == FinishType::Ended) ? State::Ended : State::Failed;
 | |
| 	auto hangupState = (type == FinishType::Ended) ? State::HangingUp : State::FailedHangingUp;
 | |
| 	const auto state = _state.current();
 | |
| 	if (state == State::Requesting) {
 | |
| 		_finishByTimeoutTimer.call(kHangupTimeoutMs, [this, finalState] { setState(finalState); });
 | |
| 		_finishAfterRequestingCall = type;
 | |
| 		return;
 | |
| 	}
 | |
| 	if (state == State::HangingUp
 | |
| 		|| state == State::FailedHangingUp
 | |
| 		|| state == State::EndedByOtherDevice
 | |
| 		|| state == State::Ended
 | |
| 		|| state == State::Failed) {
 | |
| 		return;
 | |
| 	}
 | |
| 	if (!_id) {
 | |
| 		setState(finalState);
 | |
| 		return;
 | |
| 	}
 | |
| 
 | |
| 	setState(hangupState);
 | |
| 	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))
 | |
| 		? MTPphone_DiscardCall::Flag::f_video
 | |
| 		: MTPphone_DiscardCall::Flag(0);
 | |
| 
 | |
| 	// We want to discard request still being sent and processed even if
 | |
| 	// the call is already destroyed.
 | |
| 	const auto session = &_user->session();
 | |
| 	const auto weak = base::make_weak(this);
 | |
| 	session->api().request(MTPphone_DiscardCall( // We send 'discard' here.
 | |
| 		MTP_flags(flags),
 | |
| 		MTP_inputPhoneCall(
 | |
| 			MTP_long(_id),
 | |
| 			MTP_long(_accessHash)),
 | |
| 		MTP_int(duration),
 | |
| 		reason,
 | |
| 		MTP_long(connectionId)
 | |
| 	)).done([=](const MTPUpdates &result) {
 | |
| 		// Here 'this' could be destroyed by updates, so we set Ended after
 | |
| 		// updates being handled, but in a guarded way.
 | |
| 		crl::on_main(weak, [=] { setState(finalState); });
 | |
| 		session->api().applyUpdates(result);
 | |
| 	}).fail(crl::guard(weak, [this, finalState](const MTP::Error &error) {
 | |
| 		setState(finalState);
 | |
| 	})).send();
 | |
| }
 | |
| 
 | |
| void Call::setStateQueued(State state) {
 | |
| 	crl::on_main(this, [=] {
 | |
| 		setState(state);
 | |
| 	});
 | |
| }
 | |
| 
 | |
| void Call::setFailedQueued(const QString &error) {
 | |
| 	crl::on_main(this, [=] {
 | |
| 		handleControllerError(error);
 | |
| 	});
 | |
| }
 | |
| 
 | |
| void Call::handleRequestError(const MTP::Error &error) {
 | |
| 	if (error.type() == qstr("USER_PRIVACY_RESTRICTED")) {
 | |
| 		Ui::show(Box<InformBox>(tr::lng_call_error_not_available(tr::now, lt_user, _user->name)));
 | |
| 	} else if (error.type() == qstr("PARTICIPANT_VERSION_OUTDATED")) {
 | |
| 		Ui::show(Box<InformBox>(tr::lng_call_error_outdated(tr::now, lt_user, _user->name)));
 | |
| 	} else if (error.type() == qstr("CALL_PROTOCOL_LAYER_INVALID")) {
 | |
| 		Ui::show(Box<InformBox>(Lang::Hard::CallErrorIncompatible().replace("{user}", _user->name)));
 | |
| 	}
 | |
| 	finish(FinishType::Failed);
 | |
| }
 | |
| 
 | |
| void Call::handleControllerError(const QString &error) {
 | |
| 	if (error == u"ERROR_INCOMPATIBLE"_q) {
 | |
| 		Ui::show(Box<InformBox>(
 | |
| 			Lang::Hard::CallErrorIncompatible().replace(
 | |
| 				"{user}",
 | |
| 				_user->name)));
 | |
| 	} else if (error == u"ERROR_AUDIO_IO"_q) {
 | |
| 		Ui::show(Box<InformBox>(tr::lng_call_error_audio_io(tr::now)));
 | |
| 	}
 | |
| 	finish(FinishType::Failed);
 | |
| }
 | |
| 
 | |
| void Call::destroyController() {
 | |
| 	if (_instance) {
 | |
| 		_instance->stop([](tgcalls::FinalState) {
 | |
| 		});
 | |
| 
 | |
| 		DEBUG_LOG(("Call Info: Destroying call controller.."));
 | |
| 		_instance.reset();
 | |
| 		DEBUG_LOG(("Call Info: Call controller destroyed."));
 | |
| 	}
 | |
| 	setSignalBarCount(kSignalBarFinished);
 | |
| }
 | |
| 
 | |
| Call::~Call() {
 | |
| 	destroyController();
 | |
| }
 | |
| 
 | |
| void UpdateConfig(const std::string &data) {
 | |
| 	tgcalls::SetLegacyGlobalServerConfig(data);
 | |
| }
 | |
| 
 | |
| } // namespace Calls
 | 
