tdesktop/Telegram/SourceFiles/calls/calls_call.h

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/*
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"
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#include "mtproto/mtproto_auth_key.h"
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namespace Media {
namespace Audio {
class Track;
} // namespace Audio
} // namespace Media
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namespace tgcalls {
class Instance;
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class VideoCaptureInterface;
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enum class State;
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enum class VideoState;
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enum class AudioState;
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} // namespace tgcalls
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namespace Webrtc {
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enum class VideoState;
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class VideoTrack;
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} // namespace Webrtc
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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,
};
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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 {
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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;
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virtual ~Delegate() = default;
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};
static constexpr auto kSoundSampleMs = 100;
using Type = CallType;
Call(
not_null<Delegate*> delegate,
not_null<UserData*> user,
Type type,
bool video);
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[[nodiscard]] Type type() const {
return _type;
}
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[[nodiscard]] not_null<UserData*> user() const {
return _user;
}
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[[nodiscard]] CallId id() const {
return _id;
}
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[[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,
WaitingUserConfirmation,
};
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[[nodiscard]] State state() const {
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return _state.current();
}
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[[nodiscard]] rpl::producer<State> stateValue() const {
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return _state.value();
}
[[nodiscard]] rpl::producer<Error> errors() const {
return _errors.events();
}
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enum class RemoteAudioState {
Muted,
Active,
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};
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[[nodiscard]] RemoteAudioState remoteAudioState() const {
return _remoteAudioState.current();
}
[[nodiscard]] auto remoteAudioStateValue() const
-> rpl::producer<RemoteAudioState> {
return _remoteAudioState.value();
}
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[[nodiscard]] Webrtc::VideoState remoteVideoState() const {
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return _remoteVideoState.current();
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}
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[[nodiscard]] auto remoteVideoStateValue() const
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-> rpl::producer<Webrtc::VideoState> {
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return _remoteVideoState.value();
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}
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static constexpr auto kSignalBarStarting = -1;
static constexpr auto kSignalBarFinished = -2;
static constexpr auto kSignalBarCount = 4;
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[[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();
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}
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[[nodiscard]] not_null<Webrtc::VideoTrack*> videoIncoming() const;
[[nodiscard]] not_null<Webrtc::VideoTrack*> videoOutgoing() const;
crl::time getDurationMs() const;
float64 getWaitingSoundPeakValue() const;
void applyUserConfirmation();
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);
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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);
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[[nodiscard]] rpl::lifetime &lifetime() {
return _lifetime;
}
~Call();
private:
enum class FinishType {
None,
Ended,
Failed,
};
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void handleRequestError(const QString &error);
void handleControllerError(const QString &error);
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void finish(
FinishType type,
const MTPPhoneCallDiscardReason &reason
= MTP_phoneCallDiscardReasonDisconnect());
void startOutgoing();
void startIncoming();
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void startWaitingTrack();
void sendSignalingData(const QByteArray &data);
void generateModExpFirst(bytes::const_span randomSeed);
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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);
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void setSignalBarCount(int count);
void destroyController();
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void setupOutgoingVideo();
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void updateRemoteMediaState(
tgcalls::AudioState audio,
tgcalls::VideoState video);
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const not_null<Delegate*> _delegate;
const not_null<UserData*> _user;
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MTP::Sender _api;
Type _type = Type::Outgoing;
rpl::variable<State> _state = State::WaitingUserConfirmation;
rpl::variable<RemoteAudioState> _remoteAudioState =
RemoteAudioState::Active;
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rpl::variable<Webrtc::VideoState> _remoteVideoState;
rpl::event_stream<Error> _errors;
FinishType _finishAfterRequestingCall = FinishType::None;
bool _answerAfterDhConfigReceived = false;
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rpl::variable<int> _signalBarCount = kSignalBarStarting;
crl::time _startTime = 0;
base::DelayedCallTimer _finishByTimeoutTimer;
base::Timer _discardByTimeoutTimer;
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rpl::variable<bool> _muted = false;
DhConfig _dhConfig;
bytes::vector _ga;
bytes::vector _gb;
bytes::vector _gaHash;
bytes::vector _randomPower;
MTP::AuthKey::Data _authKey;
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CallId _id = 0;
uint64 _accessHash = 0;
uint64 _keyFingerprint = 0;
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std::unique_ptr<tgcalls::Instance> _instance;
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std::shared_ptr<tgcalls::VideoCaptureInterface> _videoCapture;
QString _videoCaptureDeviceId;
bool _videoCaptureIsScreencast = false;
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const std::unique_ptr<Webrtc::VideoTrack> _videoIncoming;
const std::unique_ptr<Webrtc::VideoTrack> _videoOutgoing;
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std::unique_ptr<Media::Audio::Track> _waitingTrack;
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rpl::lifetime _lifetime;
};
void UpdateConfig(const std::string &data);
} // namespace Calls