298 lines
7.1 KiB
C++
298 lines
7.1 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
|
|
*/
|
|
#pragma once
|
|
|
|
#include "base/weak_ptr.h"
|
|
#include "base/timer.h"
|
|
#include "base/bytes.h"
|
|
#include "mtproto/sender.h"
|
|
#include "mtproto/mtproto_auth_key.h"
|
|
|
|
namespace Media {
|
|
namespace Audio {
|
|
class Track;
|
|
} // namespace Audio
|
|
} // namespace Media
|
|
|
|
namespace tgcalls {
|
|
class Instance;
|
|
class VideoCaptureInterface;
|
|
enum class State;
|
|
enum class VideoState;
|
|
enum class AudioState;
|
|
} // namespace tgcalls
|
|
|
|
namespace Webrtc {
|
|
enum class VideoState;
|
|
class VideoTrack;
|
|
} // namespace Webrtc
|
|
|
|
namespace Calls {
|
|
|
|
struct DhConfig {
|
|
int32 version = 0;
|
|
int32 g = 0;
|
|
bytes::vector p;
|
|
};
|
|
|
|
enum class ErrorType {
|
|
NoCamera,
|
|
NoMicrophone,
|
|
NotStartedCall,
|
|
NotVideoCall,
|
|
Unknown,
|
|
};
|
|
|
|
struct Error {
|
|
ErrorType type = ErrorType::Unknown;
|
|
QString details;
|
|
};
|
|
|
|
enum class CallType {
|
|
Incoming,
|
|
Outgoing,
|
|
};
|
|
|
|
class Call : public base::has_weak_ptr {
|
|
public:
|
|
class Delegate {
|
|
public:
|
|
virtual DhConfig getDhConfig() const = 0;
|
|
virtual void callFinished(not_null<Call*> call) = 0;
|
|
virtual void callFailed(not_null<Call*> call) = 0;
|
|
virtual void callRedial(not_null<Call*> call) = 0;
|
|
|
|
enum class CallSound {
|
|
Connecting,
|
|
Busy,
|
|
Ended,
|
|
};
|
|
virtual void callPlaySound(CallSound sound) = 0;
|
|
virtual void callRequestPermissionsOrFail(
|
|
Fn<void()> onSuccess,
|
|
bool video) = 0;
|
|
|
|
virtual auto callGetVideoCapture(
|
|
const QString &deviceId,
|
|
bool isScreenCapture)
|
|
-> std::shared_ptr<tgcalls::VideoCaptureInterface> = 0;
|
|
|
|
virtual ~Delegate() = default;
|
|
|
|
};
|
|
|
|
static constexpr auto kSoundSampleMs = 100;
|
|
|
|
using Type = CallType;
|
|
Call(
|
|
not_null<Delegate*> delegate,
|
|
not_null<UserData*> user,
|
|
Type type,
|
|
bool video);
|
|
|
|
[[nodiscard]] Type type() const {
|
|
return _type;
|
|
}
|
|
[[nodiscard]] not_null<UserData*> user() const {
|
|
return _user;
|
|
}
|
|
[[nodiscard]] bool isIncomingWaiting() const;
|
|
|
|
void start(bytes::const_span random);
|
|
bool handleUpdate(const MTPPhoneCall &call);
|
|
bool handleSignalingData(const MTPDupdatePhoneCallSignalingData &data);
|
|
|
|
enum State {
|
|
Starting,
|
|
WaitingInit,
|
|
WaitingInitAck,
|
|
Established,
|
|
FailedHangingUp,
|
|
Failed,
|
|
HangingUp,
|
|
Ended,
|
|
EndedByOtherDevice,
|
|
ExchangingKeys,
|
|
Waiting,
|
|
Requesting,
|
|
WaitingIncoming,
|
|
Ringing,
|
|
Busy,
|
|
};
|
|
[[nodiscard]] State state() const {
|
|
return _state.current();
|
|
}
|
|
[[nodiscard]] rpl::producer<State> stateValue() const {
|
|
return _state.value();
|
|
}
|
|
|
|
[[nodiscard]] rpl::producer<Error> errors() const {
|
|
return _errors.events();
|
|
}
|
|
|
|
enum class RemoteAudioState {
|
|
Muted,
|
|
Active,
|
|
};
|
|
[[nodiscard]] RemoteAudioState remoteAudioState() const {
|
|
return _remoteAudioState.current();
|
|
}
|
|
[[nodiscard]] auto remoteAudioStateValue() const
|
|
-> rpl::producer<RemoteAudioState> {
|
|
return _remoteAudioState.value();
|
|
}
|
|
|
|
[[nodiscard]] Webrtc::VideoState remoteVideoState() const {
|
|
return _remoteVideoState.current();
|
|
}
|
|
[[nodiscard]] auto remoteVideoStateValue() const
|
|
-> rpl::producer<Webrtc::VideoState> {
|
|
return _remoteVideoState.value();
|
|
}
|
|
|
|
static constexpr auto kSignalBarStarting = -1;
|
|
static constexpr auto kSignalBarFinished = -2;
|
|
static constexpr auto kSignalBarCount = 4;
|
|
[[nodiscard]] rpl::producer<int> signalBarCountValue() const {
|
|
return _signalBarCount.value();
|
|
}
|
|
|
|
void setMuted(bool mute);
|
|
[[nodiscard]] bool muted() const {
|
|
return _muted.current();
|
|
}
|
|
[[nodiscard]] rpl::producer<bool> mutedValue() const {
|
|
return _muted.value();
|
|
}
|
|
|
|
[[nodiscard]] not_null<Webrtc::VideoTrack*> videoIncoming() const;
|
|
[[nodiscard]] not_null<Webrtc::VideoTrack*> videoOutgoing() const;
|
|
|
|
crl::time getDurationMs() const;
|
|
float64 getWaitingSoundPeakValue() const;
|
|
|
|
void answer();
|
|
void hangup();
|
|
void redial();
|
|
|
|
bool isKeyShaForFingerprintReady() const;
|
|
bytes::vector getKeyShaForFingerprint() const;
|
|
|
|
QString getDebugLog() const;
|
|
|
|
void setCurrentAudioDevice(bool input, const QString &deviceId);
|
|
//void setAudioVolume(bool input, float level);
|
|
void setAudioDuckingEnabled(bool enabled);
|
|
|
|
void setCurrentCameraDevice(const QString &deviceId);
|
|
[[nodiscard]] QString videoDeviceId() const {
|
|
return _videoCaptureDeviceId;
|
|
}
|
|
|
|
[[nodiscard]] bool isSharingVideo() const;
|
|
[[nodiscard]] bool isSharingCamera() const;
|
|
[[nodiscard]] bool isSharingScreen() const;
|
|
[[nodiscard]] QString cameraSharingDeviceId() const;
|
|
[[nodiscard]] QString screenSharingDeviceId() const;
|
|
void toggleCameraSharing(bool enabled);
|
|
void toggleScreenSharing(std::optional<QString> uniqueId);
|
|
|
|
[[nodiscard]] rpl::lifetime &lifetime() {
|
|
return _lifetime;
|
|
}
|
|
|
|
~Call();
|
|
|
|
private:
|
|
enum class FinishType {
|
|
None,
|
|
Ended,
|
|
Failed,
|
|
};
|
|
|
|
void handleRequestError(const MTP::Error &error);
|
|
void handleControllerError(const QString &error);
|
|
void finish(
|
|
FinishType type,
|
|
const MTPPhoneCallDiscardReason &reason
|
|
= MTP_phoneCallDiscardReasonDisconnect());
|
|
void startOutgoing();
|
|
void startIncoming();
|
|
void startWaitingTrack();
|
|
void sendSignalingData(const QByteArray &data);
|
|
|
|
void generateModExpFirst(bytes::const_span randomSeed);
|
|
void handleControllerStateChange(tgcalls::State state);
|
|
void handleControllerBarCountChange(int count);
|
|
void createAndStartController(const MTPDphoneCall &call);
|
|
|
|
template <typename T>
|
|
bool checkCallCommonFields(const T &call);
|
|
bool checkCallFields(const MTPDphoneCall &call);
|
|
bool checkCallFields(const MTPDphoneCallAccepted &call);
|
|
|
|
void actuallyAnswer();
|
|
void confirmAcceptedCall(const MTPDphoneCallAccepted &call);
|
|
void startConfirmedCall(const MTPDphoneCall &call);
|
|
void setState(State state);
|
|
void setStateQueued(State state);
|
|
void setFailedQueued(const QString &error);
|
|
void setSignalBarCount(int count);
|
|
void destroyController();
|
|
|
|
void setupOutgoingVideo();
|
|
void updateRemoteMediaState(
|
|
tgcalls::AudioState audio,
|
|
tgcalls::VideoState video);
|
|
|
|
const not_null<Delegate*> _delegate;
|
|
const not_null<UserData*> _user;
|
|
MTP::Sender _api;
|
|
Type _type = Type::Outgoing;
|
|
rpl::variable<State> _state = State::Starting;
|
|
rpl::variable<RemoteAudioState> _remoteAudioState = RemoteAudioState::Active;
|
|
rpl::variable<Webrtc::VideoState> _remoteVideoState;
|
|
rpl::event_stream<Error> _errors;
|
|
FinishType _finishAfterRequestingCall = FinishType::None;
|
|
bool _answerAfterDhConfigReceived = false;
|
|
rpl::variable<int> _signalBarCount = kSignalBarStarting;
|
|
crl::time _startTime = 0;
|
|
base::DelayedCallTimer _finishByTimeoutTimer;
|
|
base::Timer _discardByTimeoutTimer;
|
|
|
|
rpl::variable<bool> _muted = false;
|
|
|
|
DhConfig _dhConfig;
|
|
bytes::vector _ga;
|
|
bytes::vector _gb;
|
|
bytes::vector _gaHash;
|
|
bytes::vector _randomPower;
|
|
MTP::AuthKey::Data _authKey;
|
|
MTPPhoneCallProtocol _protocol;
|
|
|
|
uint64 _id = 0;
|
|
uint64 _accessHash = 0;
|
|
uint64 _keyFingerprint = 0;
|
|
|
|
std::unique_ptr<tgcalls::Instance> _instance;
|
|
std::shared_ptr<tgcalls::VideoCaptureInterface> _videoCapture;
|
|
QString _videoCaptureDeviceId;
|
|
bool _videoCaptureIsScreencast = false;
|
|
const std::unique_ptr<Webrtc::VideoTrack> _videoIncoming;
|
|
const std::unique_ptr<Webrtc::VideoTrack> _videoOutgoing;
|
|
|
|
std::unique_ptr<Media::Audio::Track> _waitingTrack;
|
|
|
|
rpl::lifetime _lifetime;
|
|
|
|
};
|
|
|
|
void UpdateConfig(const std::string &data);
|
|
|
|
} // namespace Calls
|