1325 lines
37 KiB
C++
1325 lines
37 KiB
C++
/*
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This file is part of Telegram Desktop,
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the official desktop application for the Telegram messaging service.
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For license and copyright information please follow this link:
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https://github.com/telegramdesktop/tdesktop/blob/master/LEGAL
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*/
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#include "calls/calls_call.h"
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#include "main/main_session.h"
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#include "main/main_account.h"
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#include "main/main_app_config.h"
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#include "apiwrap.h"
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#include "lang/lang_keys.h"
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#include "ui/boxes/confirm_box.h"
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#include "ui/boxes/rate_call_box.h"
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#include "calls/calls_instance.h"
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#include "base/openssl_help.h"
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#include "base/random.h"
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#include "mtproto/mtproto_dh_utils.h"
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#include "mtproto/mtproto_config.h"
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#include "core/application.h"
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#include "core/core_settings.h"
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#include "media/audio/media_audio_track.h"
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#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"
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#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 InstanceV2Impl;
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class InstanceV2ReferenceImpl;
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class InstanceV2_4_0_0Impl;
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class InstanceImplLegacy;
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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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constexpr auto kAuthKeySize = 256;
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const auto kDefaultVersion = "2.4.4"_q;
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const auto Register = tgcalls::Register<tgcalls::InstanceImpl>();
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const auto RegisterV2 = tgcalls::Register<tgcalls::InstanceV2Impl>();
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const auto RegV2Ref = tgcalls::Register<tgcalls::InstanceV2ReferenceImpl>();
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const auto RegisterV240 = tgcalls::Register<tgcalls::InstanceV2_4_0_0Impl>();
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const auto RegisterLegacy = tgcalls::Register<tgcalls::InstanceImplLegacy>();
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[[nodiscard]] base::flat_set<int64> CollectEndpointIds(
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const QVector<MTPPhoneConnection> &list) {
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auto result = base::flat_set<int64>();
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result.reserve(list.size());
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for (const auto &connection : list) {
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connection.match([&](const MTPDphoneConnection &data) {
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result.emplace(int64(data.vid().v));
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}, [](const MTPDphoneConnectionWebrtc &) {
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});
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}
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return result;
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}
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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 || data.is_tcp()) {
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return;
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}
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tgcalls::Endpoint endpoint = {
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.endpointId = (int64_t)data.vid().v,
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.host = tgcalls::EndpointHost{
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.ipv4 = data.vip().v.toStdString(),
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.ipv6 = data.vipv6().v.toStdString() },
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.port = (uint16_t)data.vport().v,
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.type = tgcalls::EndpointType::UdpRelay,
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};
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const auto tag = data.vpeer_tag().v;
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if (tag.size() >= 16) {
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memcpy(endpoint.peerTag, tag.data(), 16);
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}
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list.push_back(std::move(endpoint));
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}, [&](const MTPDphoneConnectionWebrtc &data) {
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});
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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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const base::flat_set<int64> &ids) {
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connection.match([&](const MTPDphoneConnection &data) {
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const auto hex = [](const QByteArray &value) {
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const auto digit = [](uchar c) {
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return char((c < 10) ? ('0' + c) : ('a' + c - 10));
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};
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auto result = std::string();
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result.reserve(value.size() * 2);
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for (const auto ch : value) {
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result += digit(uchar(ch) / 16);
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result += digit(uchar(ch) % 16);
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}
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return result;
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};
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const auto host = data.vip().v;
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const auto hostv6 = data.vipv6().v;
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const auto port = uint16_t(data.vport().v);
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const auto username = std::string("reflector");
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const auto password = hex(data.vpeer_tag().v);
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const auto i = ids.find(int64(data.vid().v));
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Assert(i != end(ids));
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const auto id = uint8_t((i - begin(ids)) + 1);
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const auto pushTurn = [&](const QString &host) {
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list.push_back(tgcalls::RtcServer{
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.id = id,
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.host = host.toStdString(),
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.port = port,
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.login = username,
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.password = password,
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.isTurn = true,
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});
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};
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pushTurn(host);
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pushTurn(hostv6);
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}, [&](const MTPDphoneConnectionWebrtc &data) {
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const auto host = qs(data.vip());
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const auto hostv6 = qs(data.vipv6());
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const auto port = uint16_t(data.vport().v);
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if (data.is_stun()) {
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const auto pushStun = [&](const QString &host) {
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if (host.isEmpty()) {
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return;
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}
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list.push_back(tgcalls::RtcServer{
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.host = host.toStdString(),
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.port = port,
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.isTurn = false
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});
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};
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pushStun(host);
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pushStun(hostv6);
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}
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const auto username = qs(data.vusername());
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const auto password = qs(data.vpassword());
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if (data.is_turn() && !username.isEmpty() && !password.isEmpty()) {
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const auto pushTurn = [&](const QString &host) {
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list.push_back(tgcalls::RtcServer{
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.host = host.toStdString(),
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.port = port,
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.login = username.toStdString(),
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.password = password.toStdString(),
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.isTurn = true,
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});
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};
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pushTurn(host);
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pushTurn(hostv6);
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}
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});
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}
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constexpr auto kFingerprintDataSize = 256;
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uint64 ComputeFingerprint(bytes::const_span authKey) {
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Expects(authKey.size() == kFingerprintDataSize);
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auto hash = openssl::Sha1(authKey);
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return (gsl::to_integer<uint64>(hash[19]) << 56)
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| (gsl::to_integer<uint64>(hash[18]) << 48)
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| (gsl::to_integer<uint64>(hash[17]) << 40)
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| (gsl::to_integer<uint64>(hash[16]) << 32)
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| (gsl::to_integer<uint64>(hash[15]) << 24)
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| (gsl::to_integer<uint64>(hash[14]) << 16)
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| (gsl::to_integer<uint64>(hash[13]) << 8)
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| (gsl::to_integer<uint64>(hash[12]));
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}
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[[nodiscard]] QVector<MTPstring> WrapVersions(
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const std::vector<std::string> &data) {
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return ranges::views::all(
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data
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) | ranges::views::transform([=](const std::string &string) {
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return MTP_string(string);
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}) | ranges::to<QVector<MTPstring>>;
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}
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[[nodiscard]] QVector<MTPstring> CollectVersionsForApi() {
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return WrapVersions(tgcalls::Meta::Versions() | ranges::actions::reverse);
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}
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[[nodiscard]] Webrtc::VideoState StartVideoState(bool enabled) {
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using State = Webrtc::VideoState;
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return enabled ? State::Active : State::Inactive;
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}
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} // 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)
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, _user(user)
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, _api(&_user->session().mtp())
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, _type(type)
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, _discardByTimeoutTimer([=] { hangup(); })
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, _videoIncoming(
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std::make_unique<Webrtc::VideoTrack>(
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StartVideoState(video)))
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, _videoOutgoing(
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std::make_unique<Webrtc::VideoTrack>(
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StartVideoState(video))) {
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if (_type == Type::Outgoing) {
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setState(State::Requesting);
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} else {
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const auto &config = _user->session().serverConfig();
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_discardByTimeoutTimer.callOnce(config.callRingTimeoutMs);
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startWaitingTrack();
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}
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setupOutgoingVideo();
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}
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void Call::generateModExpFirst(bytes::const_span randomSeed) {
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auto first = MTP::CreateModExp(_dhConfig.g, _dhConfig.p, randomSeed);
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if (first.modexp.empty()) {
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LOG(("Call Error: Could not compute mod-exp first."));
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finish(FinishType::Failed);
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return;
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}
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_randomPower = std::move(first.randomPower);
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if (_type == Type::Incoming) {
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_gb = std::move(first.modexp);
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} else {
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_ga = std::move(first.modexp);
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_gaHash = openssl::Sha256(_ga);
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}
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}
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bool Call::isIncomingWaiting() const {
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if (type() != Call::Type::Incoming) {
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return false;
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}
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return (state() == State::Starting)
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|| (state() == State::WaitingIncoming);
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}
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void Call::start(bytes::const_span random) {
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// Save config here, because it is possible that it changes between
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// different usages inside the same call.
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_dhConfig = _delegate->getDhConfig();
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Assert(_dhConfig.g != 0);
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Assert(!_dhConfig.p.empty());
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generateModExpFirst(random);
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const auto state = _state.current();
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if (state == State::Starting || state == State::Requesting) {
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if (_type == Type::Outgoing) {
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startOutgoing();
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} else {
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startIncoming();
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}
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} else if (state == State::ExchangingKeys
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&& _answerAfterDhConfigReceived) {
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answer();
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}
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}
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void Call::startOutgoing() {
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Expects(_type == Type::Outgoing);
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Expects(_state.current() == State::Requesting);
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Expects(_gaHash.size() == kSha256Size);
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const auto flags = _videoCapture
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? MTPphone_RequestCall::Flag::f_video
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: MTPphone_RequestCall::Flag(0);
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_api.request(MTPphone_RequestCall(
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MTP_flags(flags),
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_user->inputUser,
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MTP_int(base::RandomValue<int32>()),
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MTP_bytes(_gaHash),
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MTP_phoneCallProtocol(
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MTP_flags(MTPDphoneCallProtocol::Flag::f_udp_p2p
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| MTPDphoneCallProtocol::Flag::f_udp_reflector),
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MTP_int(kMinLayer),
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MTP_int(tgcalls::Meta::MaxLayer()),
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MTP_vector(CollectVersionsForApi()))
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)).done([=](const MTPphone_PhoneCall &result) {
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Expects(result.type() == mtpc_phone_phoneCall);
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setState(State::Waiting);
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const auto &call = result.c_phone_phoneCall();
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_user->session().data().processUsers(call.vusers());
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if (call.vphone_call().type() != mtpc_phoneCallWaiting) {
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LOG(("Call Error: Expected phoneCallWaiting in response to "
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"phone.requestCall()"));
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finish(FinishType::Failed);
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return;
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}
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const auto &phoneCall = call.vphone_call();
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const auto &waitingCall = phoneCall.c_phoneCallWaiting();
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_id = waitingCall.vid().v;
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_accessHash = waitingCall.vaccess_hash().v;
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if (_finishAfterRequestingCall != FinishType::None) {
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if (_finishAfterRequestingCall == FinishType::Failed) {
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finish(_finishAfterRequestingCall);
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} else {
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hangup();
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}
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return;
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}
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const auto &config = _user->session().serverConfig();
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_discardByTimeoutTimer.callOnce(config.callReceiveTimeoutMs);
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handleUpdate(phoneCall);
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}).fail([this](const MTP::Error &error) {
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handleRequestError(error.type());
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}).send();
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}
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void Call::startIncoming() {
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Expects(_type == Type::Incoming);
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Expects(_state.current() == State::Starting);
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_api.request(MTPphone_ReceivedCall(
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MTP_inputPhoneCall(MTP_long(_id), MTP_long(_accessHash))
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)).done([=] {
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if (_state.current() == State::Starting) {
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setState(State::WaitingIncoming);
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}
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}).fail([=](const MTP::Error &error) {
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handleRequestError(error.type());
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}).send();
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}
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void Call::answer() {
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const auto video = isSharingVideo();
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_delegate->callRequestPermissionsOrFail(crl::guard(this, [=] {
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actuallyAnswer();
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}), video);
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}
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void Call::actuallyAnswer() {
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Expects(_type == Type::Incoming);
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const auto state = _state.current();
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if (state != State::Starting && state != State::WaitingIncoming) {
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if (state != State::ExchangingKeys
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|| !_answerAfterDhConfigReceived) {
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return;
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}
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}
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setState(State::ExchangingKeys);
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if (_gb.empty()) {
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_answerAfterDhConfigReceived = true;
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return;
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} else {
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_answerAfterDhConfigReceived = false;
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}
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_api.request(MTPphone_AcceptCall(
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MTP_inputPhoneCall(MTP_long(_id), MTP_long(_accessHash)),
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MTP_bytes(_gb),
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MTP_phoneCallProtocol(
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MTP_flags(MTPDphoneCallProtocol::Flag::f_udp_p2p
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| MTPDphoneCallProtocol::Flag::f_udp_reflector),
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MTP_int(kMinLayer),
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MTP_int(tgcalls::Meta::MaxLayer()),
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MTP_vector(CollectVersionsForApi()))
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)).done([=](const MTPphone_PhoneCall &result) {
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Expects(result.type() == mtpc_phone_phoneCall);
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const auto &call = result.c_phone_phoneCall();
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_user->session().data().processUsers(call.vusers());
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if (call.vphone_call().type() != mtpc_phoneCallWaiting) {
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LOG(("Call Error: "
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"Not phoneCallWaiting in response to phone.acceptCall."));
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finish(FinishType::Failed);
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return;
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}
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handleUpdate(call.vphone_call());
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}).fail([=](const MTP::Error &error) {
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handleRequestError(error.type());
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}).send();
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}
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void Call::setMuted(bool mute) {
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_muted = mute;
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if (_instance) {
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_instance->setMuteMicrophone(mute);
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}
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}
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void Call::setupOutgoingVideo() {
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static const auto hasDevices = [] {
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return !Webrtc::GetVideoInputList().empty();
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};
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const auto started = _videoOutgoing->state();
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if (!hasDevices()) {
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_videoOutgoing->setState(Webrtc::VideoState::Inactive);
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}
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_videoOutgoing->stateValue(
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) | rpl::start_with_next([=](Webrtc::VideoState state) {
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if (state != Webrtc::VideoState::Inactive
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&& !hasDevices()
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&& !_videoCaptureIsScreencast) {
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_errors.fire({ ErrorType::NoCamera });
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_videoOutgoing->setState(Webrtc::VideoState::Inactive);
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} else if (_state.current() != State::Established
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&& (state != Webrtc::VideoState::Inactive)
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&& (started == Webrtc::VideoState::Inactive)) {
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_errors.fire({ ErrorType::NotStartedCall });
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_videoOutgoing->setState(Webrtc::VideoState::Inactive);
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} else if (state != Webrtc::VideoState::Inactive
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&& _instance
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&& !_instance->supportsVideo()) {
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_errors.fire({ ErrorType::NotVideoCall });
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_videoOutgoing->setState(Webrtc::VideoState::Inactive);
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} else if (state != Webrtc::VideoState::Inactive) {
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// Paused not supported right now.
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Assert(state == Webrtc::VideoState::Active);
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if (!_videoCapture) {
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_videoCapture = _delegate->callGetVideoCapture(
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_videoCaptureDeviceId,
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_videoCaptureIsScreencast);
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_videoCapture->setOutput(_videoOutgoing->sink());
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}
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_videoCapture->setState(tgcalls::VideoState::Active);
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if (_instance) {
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_instance->setVideoCapture(_videoCapture);
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}
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} else if (_videoCapture) {
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_videoCapture->setState(tgcalls::VideoState::Inactive);
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if (_instance) {
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_instance->setVideoCapture(nullptr);
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}
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}
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}, _lifetime);
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}
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not_null<Webrtc::VideoTrack*> Call::videoIncoming() const {
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return _videoIncoming.get();
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}
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not_null<Webrtc::VideoTrack*> Call::videoOutgoing() const {
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return _videoOutgoing.get();
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}
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crl::time Call::getDurationMs() const {
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return _startTime ? (crl::now() - _startTime) : 0;
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}
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void Call::hangup() {
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const auto state = _state.current();
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if (state == State::Busy) {
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_delegate->callFinished(this);
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} else {
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const auto missed = (state == State::Ringing
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|| (state == State::Waiting && _type == Type::Outgoing));
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const auto declined = isIncomingWaiting();
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const auto reason = missed
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? MTP_phoneCallDiscardReasonMissed()
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: declined
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? MTP_phoneCallDiscardReasonBusy()
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: MTP_phoneCallDiscardReasonHangup();
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finish(FinishType::Ended, reason);
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}
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}
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void Call::redial() {
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if (_state.current() != State::Busy) {
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return;
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}
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Assert(_instance == nullptr);
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_type = Type::Outgoing;
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setState(State::Requesting);
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_answerAfterDhConfigReceived = false;
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startWaitingTrack();
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_delegate->callRedial(this);
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}
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QString Call::getDebugLog() const {
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return _instance
|
|
? QString::fromStdString(_instance->getDebugInfo())
|
|
: QString();
|
|
}
|
|
|
|
void Call::startWaitingTrack() {
|
|
_waitingTrack = Media::Audio::Current().createTrack();
|
|
const 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.type());
|
|
}).send();
|
|
}
|
|
|
|
float64 Call::getWaitingSoundPeakValue() const {
|
|
if (_waitingTrack) {
|
|
const 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: {
|
|
const 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;
|
|
const 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: {
|
|
const 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: {
|
|
const 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: {
|
|
const 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: {
|
|
const auto &data = call.c_phoneCallDiscarded();
|
|
if (data.vid().v != _id) {
|
|
return false;
|
|
}
|
|
if (data.is_need_debug()) {
|
|
const 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) {
|
|
const auto session = &_user->session();
|
|
const auto callId = _id;
|
|
const auto callAccessHash = _accessHash;
|
|
const auto box = Ui::show(Box<Ui::RateCallBox>(
|
|
Core::App().settings().sendSubmitWay()));
|
|
const auto sender = box->lifetime().make_state<MTP::Sender>(
|
|
&session->mtp());
|
|
box->sends(
|
|
) | rpl::take(
|
|
1 // Instead of keeping requestId.
|
|
) | rpl::start_with_next([=](const Ui::RateCallBox::Result &r) {
|
|
sender->request(MTPphone_SetCallRating(
|
|
MTP_flags(0),
|
|
MTP_inputPhoneCall(
|
|
MTP_long(callId),
|
|
MTP_long(callAccessHash)),
|
|
MTP_int(r.rating),
|
|
MTP_string(r.comment)
|
|
)).done([=](const MTPUpdates &updates) {
|
|
session->api().applyUpdates(updates);
|
|
box->closeBox();
|
|
}).fail([=] {
|
|
box->closeBox();
|
|
}).send();
|
|
}, box->lifetime());
|
|
}
|
|
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: {
|
|
const 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);
|
|
|
|
const 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.type());
|
|
}).send();
|
|
}
|
|
|
|
void Call::startConfirmedCall(const MTPDphoneCall &call) {
|
|
Expects(_type == Type::Incoming);
|
|
|
|
const 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());
|
|
|
|
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);
|
|
|
|
createAndStartController(call);
|
|
}
|
|
|
|
void Call::createAndStartController(const MTPDphoneCall &call) {
|
|
_discardByTimeoutTimer.cancel();
|
|
if (!checkCallFields(call) || _authKey.size() != kAuthKeySize) {
|
|
return;
|
|
}
|
|
|
|
const auto &protocol = call.vprotocol().c_phoneCallProtocol();
|
|
const auto &serverConfig = _user->session().serverConfig();
|
|
|
|
auto encryptionKeyValue = std::make_shared<std::array<
|
|
uint8_t,
|
|
kAuthKeySize>>();
|
|
memcpy(encryptionKeyValue->data(), _authKey.data(), kAuthKeySize);
|
|
|
|
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(call.is_p2p_allowed())));
|
|
|
|
const auto versionString = version.toStdString();
|
|
const auto &settings = Core::App().settings();
|
|
const auto weak = base::make_weak(this);
|
|
tgcalls::Descriptor descriptor = {
|
|
.version = versionString,
|
|
.config = tgcalls::Config{
|
|
.initializationTimeout =
|
|
serverConfig.callConnectTimeoutMs / 1000.,
|
|
.receiveTimeout = serverConfig.callPacketTimeoutMs / 1000.,
|
|
.dataSaving = tgcalls::DataSaving::Never,
|
|
.enableP2P = call.is_p2p_allowed(),
|
|
.enableAEC = false,
|
|
.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()) {
|
|
const auto callLogFolder = cWorkingDir() + qsl("DebugLogs");
|
|
const auto callLogPath = callLogFolder + qsl("/last_call_log.txt");
|
|
const 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);
|
|
}
|
|
|
|
const auto ids = CollectEndpointIds(call.vconnections().v);
|
|
for (const auto &connection : call.vconnections().v) {
|
|
AppendEndpoint(descriptor.endpoints, connection);
|
|
}
|
|
for (const auto &connection : call.vconnections().v) {
|
|
AppendServer(descriptor.rtcServers, connection, ids);
|
|
}
|
|
|
|
{
|
|
const 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();
|
|
}
|
|
}
|
|
}
|
|
_instance = tgcalls::Meta::Create(versionString, 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: {
|
|
const 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) {
|
|
const auto checkFailed = [this] {
|
|
finish(FinishType::Failed);
|
|
return false;
|
|
};
|
|
if (call.vaccess_hash().v != _accessHash) {
|
|
LOG(("Call Error: Wrong call access_hash."));
|
|
return checkFailed();
|
|
}
|
|
const auto adminId = (_type == Type::Outgoing)
|
|
? _user->session().userId()
|
|
: peerToUser(_user->id);
|
|
const 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);
|
|
_discardByTimeoutTimer.cancel();
|
|
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::setCurrentCameraDevice(const QString &deviceId) {
|
|
if (!_videoCaptureIsScreencast) {
|
|
_videoCaptureDeviceId = deviceId;
|
|
if (_videoCapture) {
|
|
_videoCapture->switchToDevice(deviceId.toStdString(), false);
|
|
if (_instance) {
|
|
_instance->sendVideoDeviceUpdated();
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
//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);
|
|
}
|
|
}
|
|
|
|
bool Call::isSharingVideo() const {
|
|
return (_videoOutgoing->state() != Webrtc::VideoState::Inactive);
|
|
}
|
|
|
|
bool Call::isSharingCamera() const {
|
|
return !_videoCaptureIsScreencast && isSharingVideo();
|
|
}
|
|
|
|
bool Call::isSharingScreen() const {
|
|
return _videoCaptureIsScreencast && isSharingVideo();
|
|
}
|
|
|
|
QString Call::cameraSharingDeviceId() const {
|
|
return isSharingCamera() ? _videoCaptureDeviceId : QString();
|
|
}
|
|
|
|
QString Call::screenSharingDeviceId() const {
|
|
return isSharingScreen() ? _videoCaptureDeviceId : QString();
|
|
}
|
|
|
|
void Call::toggleCameraSharing(bool enabled) {
|
|
if (isSharingCamera() == enabled) {
|
|
return;
|
|
} else if (!enabled) {
|
|
if (_videoCapture) {
|
|
_videoCapture->setState(tgcalls::VideoState::Inactive);
|
|
}
|
|
_videoOutgoing->setState(Webrtc::VideoState::Inactive);
|
|
_videoCaptureDeviceId = QString();
|
|
return;
|
|
}
|
|
_delegate->callRequestPermissionsOrFail(crl::guard(this, [=] {
|
|
toggleScreenSharing(std::nullopt);
|
|
const auto deviceId = Core::App().settings().callVideoInputDeviceId();
|
|
_videoCaptureDeviceId = deviceId;
|
|
if (_videoCapture) {
|
|
_videoCapture->switchToDevice(deviceId.toStdString(), false);
|
|
if (_instance) {
|
|
_instance->sendVideoDeviceUpdated();
|
|
}
|
|
}
|
|
_videoOutgoing->setState(Webrtc::VideoState::Active);
|
|
}), true);
|
|
}
|
|
|
|
void Call::toggleScreenSharing(std::optional<QString> uniqueId) {
|
|
if (!uniqueId) {
|
|
if (isSharingScreen()) {
|
|
if (_videoCapture) {
|
|
_videoCapture->setState(tgcalls::VideoState::Inactive);
|
|
}
|
|
_videoOutgoing->setState(Webrtc::VideoState::Inactive);
|
|
}
|
|
_videoCaptureDeviceId = QString();
|
|
_videoCaptureIsScreencast = false;
|
|
return;
|
|
} else if (screenSharingDeviceId() == *uniqueId) {
|
|
return;
|
|
}
|
|
toggleCameraSharing(false);
|
|
_videoCaptureIsScreencast = true;
|
|
_videoCaptureDeviceId = *uniqueId;
|
|
if (_videoCapture) {
|
|
_videoCapture->switchToDevice(uniqueId->toStdString(), true);
|
|
if (_instance) {
|
|
_instance->sendVideoDeviceUpdated();
|
|
}
|
|
}
|
|
_videoOutgoing->setState(Webrtc::VideoState::Active);
|
|
}
|
|
|
|
void Call::finish(FinishType type, const MTPPhoneCallDiscardReason &reason) {
|
|
Expects(type != FinishType::None);
|
|
|
|
setSignalBarCount(kSignalBarFinished);
|
|
|
|
const auto finalState = (type == FinishType::Ended)
|
|
? State::Ended
|
|
: State::Failed;
|
|
const 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);
|
|
const auto duration = getDurationMs() / 1000;
|
|
const auto connectionId = _instance
|
|
? _instance->getPreferredRelayId()
|
|
: 0;
|
|
_finishByTimeoutTimer.call(kHangupTimeoutMs, [this, finalState] {
|
|
setState(finalState);
|
|
});
|
|
|
|
using Video = Webrtc::VideoState;
|
|
const auto flags = ((_videoIncoming->state() != Video::Inactive)
|
|
|| (_videoOutgoing->state() != Video::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] {
|
|
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 QString &error) {
|
|
const auto inform = (error == u"USER_PRIVACY_RESTRICTED"_q)
|
|
? tr::lng_call_error_not_available(tr::now, lt_user, _user->name)
|
|
: (error == u"PARTICIPANT_VERSION_OUTDATED"_q)
|
|
? tr::lng_call_error_outdated(tr::now, lt_user, _user->name)
|
|
: (error == u"CALL_PROTOCOL_LAYER_INVALID"_q)
|
|
? Lang::Hard::CallErrorIncompatible().replace("{user}", _user->name)
|
|
: QString();
|
|
if (!inform.isEmpty()) {
|
|
Ui::show(Ui::MakeInformBox(inform));
|
|
}
|
|
finish(FinishType::Failed);
|
|
}
|
|
|
|
void Call::handleControllerError(const QString &error) {
|
|
const auto inform = (error == u"ERROR_INCOMPATIBLE"_q)
|
|
? Lang::Hard::CallErrorIncompatible().replace("{user}", _user->name)
|
|
: (error == u"ERROR_AUDIO_IO"_q)
|
|
? tr::lng_call_error_audio_io(tr::now)
|
|
: QString();
|
|
if (!inform.isEmpty()) {
|
|
Ui::show(Ui::MakeInformBox(inform));
|
|
}
|
|
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
|