init

Telegram/ThirdParty/tgcalls/.clang-format

@@ -0,0 +1,2 @@
+DisableFormat: true
+SortIncludes: false

Telegram/ThirdParty/tgcalls/.gitignore

@@ -0,0 +1,5 @@
+.DS_Store
+._*
+*.*~
+SharedHeaders/*
+.build

Telegram/ThirdParty/tgcalls/.swiftpm/xcode/package.xcworkspace/contents.xcworkspacedata

@@ -0,0 +1,7 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<Workspace
+   version = "1.0">
+   <FileRef
+      location = "self:">
+   </FileRef>
+</Workspace>

Telegram/ThirdParty/tgcalls/.swiftpm/xcode/xcshareddata/xcschemes/TgVoipWebrtc.xcscheme

@@ -0,0 +1,67 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<Scheme
+   LastUpgradeVersion = "1310"
+   version = "1.3">
+   <BuildAction
+      parallelizeBuildables = "YES"
+      buildImplicitDependencies = "YES">
+      <BuildActionEntries>
+         <BuildActionEntry
+            buildForTesting = "YES"
+            buildForRunning = "YES"
+            buildForProfiling = "YES"
+            buildForArchiving = "YES"
+            buildForAnalyzing = "YES">
+            <BuildableReference
+               BuildableIdentifier = "primary"
+               BlueprintIdentifier = "TgVoipWebrtc"
+               BuildableName = "TgVoipWebrtc"
+               BlueprintName = "TgVoipWebrtc"
+               ReferencedContainer = "container:">
+            </BuildableReference>
+         </BuildActionEntry>
+      </BuildActionEntries>
+   </BuildAction>
+   <TestAction
+      buildConfiguration = "Debug"
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.LLDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.LLDB"
+      shouldUseLaunchSchemeArgsEnv = "YES">
+      <Testables>
+      </Testables>
+   </TestAction>
+   <LaunchAction
+      buildConfiguration = "Debug"
+      selectedDebuggerIdentifier = "Xcode.DebuggerFoundation.Debugger.LLDB"
+      selectedLauncherIdentifier = "Xcode.DebuggerFoundation.Launcher.LLDB"
+      launchStyle = "0"
+      useCustomWorkingDirectory = "NO"
+      ignoresPersistentStateOnLaunch = "NO"
+      debugDocumentVersioning = "YES"
+      debugServiceExtension = "internal"
+      allowLocationSimulation = "YES">
+   </LaunchAction>
+   <ProfileAction
+      buildConfiguration = "Release"
+      shouldUseLaunchSchemeArgsEnv = "YES"
+      savedToolIdentifier = ""
+      useCustomWorkingDirectory = "NO"
+      debugDocumentVersioning = "YES">
+      <MacroExpansion>
+         <BuildableReference
+            BuildableIdentifier = "primary"
+            BlueprintIdentifier = "TgVoipWebrtc"
+            BuildableName = "TgVoipWebrtc"
+            BlueprintName = "TgVoipWebrtc"
+            ReferencedContainer = "container:">
+         </BuildableReference>
+      </MacroExpansion>
+   </ProfileAction>
+   <AnalyzeAction
+      buildConfiguration = "Debug">
+   </AnalyzeAction>
+   <ArchiveAction
+      buildConfiguration = "Release"
+      revealArchiveInOrganizer = "YES">
+   </ArchiveAction>
+</Scheme>

Telegram/ThirdParty/tgcalls/LICENSE

@@ -0,0 +1,165 @@
+                   GNU LESSER GENERAL PUBLIC LICENSE
+                       Version 3, 29 June 2007
+
+ Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+
+  This version of the GNU Lesser General Public License incorporates
+the terms and conditions of version 3 of the GNU General Public
+License, supplemented by the additional permissions listed below.
+
+  0. Additional Definitions.
+
+  As used herein, "this License" refers to version 3 of the GNU Lesser
+General Public License, and the "GNU GPL" refers to version 3 of the GNU
+General Public License.
+
+  "The Library" refers to a covered work governed by this License,
+other than an Application or a Combined Work as defined below.
+
+  An "Application" is any work that makes use of an interface provided
+by the Library, but which is not otherwise based on the Library.
+Defining a subclass of a class defined by the Library is deemed a mode
+of using an interface provided by the Library.
+
+  A "Combined Work" is a work produced by combining or linking an
+Application with the Library.  The particular version of the Library
+with which the Combined Work was made is also called the "Linked
+Version".
+
+  The "Minimal Corresponding Source" for a Combined Work means the
+Corresponding Source for the Combined Work, excluding any source code
+for portions of the Combined Work that, considered in isolation, are
+based on the Application, and not on the Linked Version.
+
+  The "Corresponding Application Code" for a Combined Work means the
+object code and/or source code for the Application, including any data
+and utility programs needed for reproducing the Combined Work from the
+Application, but excluding the System Libraries of the Combined Work.
+
+  1. Exception to Section 3 of the GNU GPL.
+
+  You may convey a covered work under sections 3 and 4 of this License
+without being bound by section 3 of the GNU GPL.
+
+  2. Conveying Modified Versions.
+
+  If you modify a copy of the Library, and, in your modifications, a
+facility refers to a function or data to be supplied by an Application
+that uses the facility (other than as an argument passed when the
+facility is invoked), then you may convey a copy of the modified
+version:
+
+   a) under this License, provided that you make a good faith effort to
+   ensure that, in the event an Application does not supply the
+   function or data, the facility still operates, and performs
+   whatever part of its purpose remains meaningful, or
+
+   b) under the GNU GPL, with none of the additional permissions of
+   this License applicable to that copy.
+
+  3. Object Code Incorporating Material from Library Header Files.
+
+  The object code form of an Application may incorporate material from
+a header file that is part of the Library.  You may convey such object
+code under terms of your choice, provided that, if the incorporated
+material is not limited to numerical parameters, data structure
+layouts and accessors, or small macros, inline functions and templates
+(ten or fewer lines in length), you do both of the following:
+
+   a) Give prominent notice with each copy of the object code that the
+   Library is used in it and that the Library and its use are
+   covered by this License.
+
+   b) Accompany the object code with a copy of the GNU GPL and this license
+   document.
+
+  4. Combined Works.
+
+  You may convey a Combined Work under terms of your choice that,
+taken together, effectively do not restrict modification of the
+portions of the Library contained in the Combined Work and reverse
+engineering for debugging such modifications, if you also do each of
+the following:
+
+   a) Give prominent notice with each copy of the Combined Work that
+   the Library is used in it and that the Library and its use are
+   covered by this License.
+
+   b) Accompany the Combined Work with a copy of the GNU GPL and this license
+   document.
+
+   c) For a Combined Work that displays copyright notices during
+   execution, include the copyright notice for the Library among
+   these notices, as well as a reference directing the user to the
+   copies of the GNU GPL and this license document.
+
+   d) Do one of the following:
+
+       0) Convey the Minimal Corresponding Source under the terms of this
+       License, and the Corresponding Application Code in a form
+       suitable for, and under terms that permit, the user to
+       recombine or relink the Application with a modified version of
+       the Linked Version to produce a modified Combined Work, in the
+       manner specified by section 6 of the GNU GPL for conveying
+       Corresponding Source.
+
+       1) Use a suitable shared library mechanism for linking with the
+       Library.  A suitable mechanism is one that (a) uses at run time
+       a copy of the Library already present on the user's computer
+       system, and (b) will operate properly with a modified version
+       of the Library that is interface-compatible with the Linked
+       Version.
+
+   e) Provide Installation Information, but only if you would otherwise
+   be required to provide such information under section 6 of the
+   GNU GPL, and only to the extent that such information is
+   necessary to install and execute a modified version of the
+   Combined Work produced by recombining or relinking the
+   Application with a modified version of the Linked Version. (If
+   you use option 4d0, the Installation Information must accompany
+   the Minimal Corresponding Source and Corresponding Application
+   Code. If you use option 4d1, you must provide the Installation
+   Information in the manner specified by section 6 of the GNU GPL
+   for conveying Corresponding Source.)
+
+  5. Combined Libraries.
+
+  You may place library facilities that are a work based on the
+Library side by side in a single library together with other library
+facilities that are not Applications and are not covered by this
+License, and convey such a combined library under terms of your
+choice, if you do both of the following:
+
+   a) Accompany the combined library with a copy of the same work based
+   on the Library, uncombined with any other library facilities,
+   conveyed under the terms of this License.
+
+   b) Give prominent notice with the combined library that part of it
+   is a work based on the Library, and explaining where to find the
+   accompanying uncombined form of the same work.
+
+  6. Revised Versions of the GNU Lesser General Public License.
+
+  The Free Software Foundation may publish revised and/or new versions
+of the GNU Lesser General Public License from time to time. Such new
+versions will be similar in spirit to the present version, but may
+differ in detail to address new problems or concerns.
+
+  Each version is given a distinguishing version number. If the
+Library as you received it specifies that a certain numbered version
+of the GNU Lesser General Public License "or any later version"
+applies to it, you have the option of following the terms and
+conditions either of that published version or of any later version
+published by the Free Software Foundation. If the Library as you
+received it does not specify a version number of the GNU Lesser
+General Public License, you may choose any version of the GNU Lesser
+General Public License ever published by the Free Software Foundation.
+
+  If the Library as you received it specifies that a proxy can decide
+whether future versions of the GNU Lesser General Public License shall
+apply, that proxy's public statement of acceptance of any version is
+permanent authorization for you to choose that version for the
+Library.

Telegram/ThirdParty/tgcalls/Package.swift

@@ -0,0 +1,91 @@
+// swift-tools-version:5.5
+// The swift-tools-version declares the minimum version of Swift required to build this package.
+
+import PackageDescription
+
+let package = Package(
+    name: "TgVoipWebrtc",
+    platforms: [.macOS(.v10_13)],
+    products: [
+        .library(
+            name: "TgVoipWebrtc",
+            targets: ["TgVoipWebrtc"]),
+    ],
+    dependencies: [
+    ],
+    targets: [
+        .target(
+            name: "TgVoipWebrtc",
+            dependencies: [],
+            path: ".",
+            exclude: ["LICENSE",
+                      "README.md",
+                      "tgcalls/platform/android",
+                      "tgcalls/platform/tdesktop",
+                      "tgcalls/platform/uwp",
+                      "tgcalls/platform/fake",
+                      "tgcalls/platform/darwin/iOS",
+                      "tgcalls/platform/darwin/VideoCaptureView.mm",
+                      "tgcalls/platform/darwin/VideoCaptureView.h",
+                      "tgcalls/platform/darwin/VideoMetalView.mm",
+                      "tgcalls/platform/darwin/VideoMetalView.h",
+                      "tgcalls/platform/darwin/VideoSampleBufferView.mm",
+                      "tgcalls/platform/darwin/VideoSampleBufferView.h",
+                      "tgcalls/platform/darwin/AudioDeviceModuleIOS.h",
+                      "tgcalls/platform/darwin/VideoCameraCapturer.mm",
+                      "tgcalls/platform/darwin/VideoCameraCapturer.h",
+                      "tgcalls/platform/darwin/CustomExternalCapturer.mm",
+                      "tgcalls/platform/darwin/CustomExternalCapturer.h",
+                      "tgcalls/legacy/InstanceImplLegacy.h",
+                      "tgcalls/legacy/InstanceImplLegacy.cpp"
+                     ],
+            publicHeadersPath: "macos/PublicHeaders",
+            cxxSettings: [
+                .headerSearchPath("."),
+                .headerSearchPath("tgcalls"),
+                .headerSearchPath("macos/PublicHeaders"),
+                .headerSearchPath("SharedHeaders/webrtc"),
+                .headerSearchPath("SharedHeaders/webrtc/third_party/abseil-cpp"),
+                .headerSearchPath("SharedHeaders/webrtc/sdk/objc"),
+                .headerSearchPath("SharedHeaders/webrtc/sdk/objc/components/renderer/metal"),
+                .headerSearchPath("SharedHeaders/webrtc/sdk/objc/components/video_codec"),
+                .headerSearchPath("SharedHeaders/webrtc/sdk/objc/base"),
+                .headerSearchPath("SharedHeaders/webrtc/sdk/objc/api/video_codec"),
+                .headerSearchPath("SharedHeaders/webrtc/third_party/libyuv/include"),
+                .headerSearchPath("SharedHeaders/webrtc/sdk/objc/components/renderer/opengl"),
+                .headerSearchPath("SharedHeaders/openssl/include"),
+                .headerSearchPath("SharedHeaders/libopus/include"),
+                .headerSearchPath("SharedHeaders/ffmpeg/include"),
+                .headerSearchPath("SharedHeaders/rnoise"),
+
+                
+                .define("WEBRTC_POSIX", to: "1", nil),
+                .define("WEBRTC_MAC", to: "1", nil),
+                //NDEBUG FLAG MUST BE THE SAME AS WEBRTC BUILT OTHERWISE IT WILL CRASH SOMEWHERE , USE _DEBUG FOR DEBUG, NDEBUG FOR RELEASE
+                .define("NDEBUG", to: "1", nil),
+                .define("RTC_ENABLE_VP9", to: "1", nil),
+                .define("WEBRTC_USE_BUILTIN_ISAC_FLOAT", to: "", nil),
+                .define("ABSL_ALLOCATOR_NOTHROW", to: "1", nil),
+                .define("BWE_TEST_LOGGING_COMPILE_TIME_ENABLE", to: "0", nil),
+                .define("HAVE_NETINET_IN_H", to: "", nil),
+                .define("HAVE_SCTP", to: "", nil),
+                .define("HAVE_WEBRTC_VIDEO", to: "", nil),
+                .define("NO_MAIN_THREAD_WRAPPING", to: "", nil),
+                .define("RTC_DISABLE_TRACE_EVENTS", to: "", nil),
+                .define("WEBRTC_APM_DEBUG_DUMP", to: "0", nil),
+                .define("WEBRTC_ARCH_ARM64", to: "", nil),
+                .define("WEBRTC_ENABLE_PROTOBUF", to: "0", nil),
+                .define("WEBRTC_HAS_NEON", to: "", nil),
+                .define("WEBRTC_HAVE_DCSCTP", to: "", nil),
+                .define("WEBRTC_HAVE_SCTP", to: "", nil),
+                .define("WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE", to: "", nil),
+                .define("WEBRTC_LIBRARY_IMPL", to: "", nil),
+                .define("WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS", to: "1", nil),
+                .define("WEBRTC_OPUS_SUPPORT_120MS_PTIME", to: "1", nil),
+                .define("WEBRTC_OPUS_VARIABLE_COMPLEXITY", to: "0", nil),
+                .define("WEBRTC_USE_H264", to: "", nil),
+
+            ]),
+    ],
+    cxxLanguageStandard: .cxx20
+)

Telegram/ThirdParty/tgcalls/README.md

@@ -0,0 +1,6 @@
+# Telegram Calls Library
+
+Copyright (c) 2020 The Telegram Calls Library Authors.
+
+This code is published under the terms of the GNU Lesser General Public License version 3.
+Full license text can be found here: [https://github.com/TelegramMessenger/tgcalls/blob/master/LICENSE](https://github.com/TelegramMessenger/tgcalls/blob/master/LICENSE)

Telegram/ThirdParty/tgcalls/macos/PublicHeaders/TgVoipWebrtc/LibYuvBinding.h

@@ -0,0 +1,24 @@
+#ifndef TgVoipWebrtc_LibYuvBinding_h
+#define TgVoipWebrtc_LibYuvBinding_h
+
+#import <Foundation/Foundation.h>
+
+@interface LibYUVConverter : NSObject
+
++ (BOOL)I420ToNV12WithSrcY:(const uint8_t *)src_y
+               srcStrideY:(int)src_stride_y
+                    srcU:(const uint8_t *)src_u
+               srcStrideU:(int)src_stride_u
+                    srcV:(const uint8_t *)src_v
+               srcStrideV:(int)src_stride_v
+                    dstY:(uint8_t *)dst_y
+               dstStrideY:(int)dst_stride_y
+                   dstUV:(uint8_t *)dst_uv
+               dstStrideUV:(int)dst_stride_uv
+                    width:(int)width
+                   height:(int)height;
+
+@end
+
+
+#endif

Telegram/ThirdParty/tgcalls/macos/PublicHeaders/TgVoipWebrtc/MediaStreaming.h

@@ -0,0 +1,23 @@
+#ifndef TgVoipWebrtc_MediaStreaming_h
+#define TgVoipWebrtc_MediaStreaming_h
+
+#import <Foundation/Foundation.h>
+
+#import <TgVoipWebrtc/OngoingCallThreadLocalContext.h>
+
+@interface MediaStreamingContext : NSObject
+
+- (instancetype _Nonnull)initWithQueue:(id<OngoingCallThreadLocalContextQueueWebrtc> _Nonnull)queue
+    requestCurrentTime:(id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull)(void (^ _Nonnull)(int64_t)))requestAudioBroadcastPart
+    requestAudioBroadcastPart:(id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull)(int64_t, int64_t, void (^ _Nonnull)(OngoingGroupCallBroadcastPart * _Nullable)))requestAudioBroadcastPart
+    requestVideoBroadcastPart:(id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull)(int64_t, int64_t, int32_t, OngoingGroupCallRequestedVideoQuality, void (^ _Nonnull)(OngoingGroupCallBroadcastPart * _Nullable)))requestVideoBroadcastPart;
+
+- (void)start;
+- (void)stop;
+
+- (GroupCallDisposable * _Nonnull)addVideoOutput:(void (^_Nonnull)(CallVideoFrameData * _Nonnull))sink;
+- (void)getAudio:(int16_t * _Nonnull)audioSamples numSamples:(NSInteger)numSamples numChannels:(NSInteger)numChannels samplesPerSecond:(NSInteger)samplesPerSecond;
+
+@end
+
+#endif

Telegram/ThirdParty/tgcalls/macos/PublicHeaders/TgVoipWebrtc/OngoingCallThreadLocalContext.h

@@ -0,0 +1,437 @@
+#ifndef OngoingCallContext_h
+#define OngoingCallContext_h
+
+#import <Foundation/Foundation.h>
+
+#if TARGET_OS_IOS
+#import <UIKit/UIKit.h>
+#import <CoreMedia/CoreMedia.h>
+#else
+#import <AppKit/AppKit.h>
+#define UIView NSView
+#endif
+
+@interface CallAudioTone : NSObject
+
+@property (nonatomic, strong, readonly) NSData * _Nonnull samples;
+@property (nonatomic, readonly) NSInteger sampleRate;
+@property (nonatomic, readonly) NSInteger loopCount;
+
+- (instancetype _Nonnull)initWithSamples:(NSData * _Nonnull)samples sampleRate:(NSInteger)sampleRate loopCount:(NSInteger)loopCount;
+
+@end
+
+@interface SharedCallAudioDevice : NSObject
+
+- (instancetype _Nonnull)initWithDisableRecording:(bool)disableRecording enableSystemMute:(bool)enableSystemMute;
+
++ (void)setupAudioSession;
+
+- (void)setManualAudioSessionIsActive:(bool)isAudioSessionActive;
+
+- (void)setTone:(CallAudioTone * _Nullable)tone;
+
+@end
+
+@interface OngoingCallConnectionDescriptionWebrtc : NSObject
+
+@property (nonatomic, readonly) uint8_t reflectorId;
+@property (nonatomic, readonly) bool hasStun;
+@property (nonatomic, readonly) bool hasTurn;
+@property (nonatomic, readonly) bool hasTcp;
+@property (nonatomic, strong, readonly) NSString * _Nonnull ip;
+@property (nonatomic, readonly) int32_t port;
+@property (nonatomic, strong, readonly) NSString * _Nonnull username;
+@property (nonatomic, strong, readonly) NSString * _Nonnull password;
+
+- (instancetype _Nonnull)initWithReflectorId:(uint8_t)reflectorId hasStun:(bool)hasStun hasTurn:(bool)hasTurn hasTcp:(bool)hasTcp ip:(NSString * _Nonnull)ip port:(int32_t)port username:(NSString * _Nonnull)username password:(NSString * _Nonnull)password;
+
+@end
+
+typedef NS_ENUM(int32_t, OngoingCallStateWebrtc) {
+    OngoingCallStateInitializing,
+    OngoingCallStateConnected,
+    OngoingCallStateFailed,
+    OngoingCallStateReconnecting
+};
+
+typedef NS_ENUM(int32_t, OngoingCallVideoStateWebrtc) {
+    OngoingCallVideoStateInactive,
+    OngoingCallVideoStateActive,
+    OngoingCallVideoStatePaused
+};
+
+typedef NS_ENUM(int32_t, OngoingCallRemoteVideoStateWebrtc) {
+    OngoingCallRemoteVideoStateInactive,
+    OngoingCallRemoteVideoStateActive,
+    OngoingCallRemoteVideoStatePaused
+};
+
+typedef NS_ENUM(int32_t, OngoingCallRemoteAudioStateWebrtc) {
+    OngoingCallRemoteAudioStateMuted,
+    OngoingCallRemoteAudioStateActive,
+};
+
+typedef NS_ENUM(int32_t, OngoingCallRemoteBatteryLevelWebrtc) {
+    OngoingCallRemoteBatteryLevelNormal,
+    OngoingCallRemoteBatteryLevelLow
+};
+
+typedef NS_ENUM(int32_t, OngoingCallVideoOrientationWebrtc) {
+    OngoingCallVideoOrientation0,
+    OngoingCallVideoOrientation90,
+    OngoingCallVideoOrientation180,
+    OngoingCallVideoOrientation270
+};
+
+typedef NS_ENUM(int32_t, OngoingCallNetworkTypeWebrtc) {
+    OngoingCallNetworkTypeWifi,
+    OngoingCallNetworkTypeCellularGprs,
+    OngoingCallNetworkTypeCellularEdge,
+    OngoingCallNetworkTypeCellular3g,
+    OngoingCallNetworkTypeCellularLte
+};
+
+typedef NS_ENUM(int32_t, OngoingCallDataSavingWebrtc) {
+    OngoingCallDataSavingNever,
+    OngoingCallDataSavingCellular,
+    OngoingCallDataSavingAlways
+};
+
+@interface GroupCallDisposable : NSObject
+
+- (instancetype _Nonnull)initWithBlock:(dispatch_block_t _Nonnull)block;
+- (void)dispose;
+
+@end
+
+@protocol OngoingCallThreadLocalContextQueueWebrtc <NSObject>
+
+- (void)dispatch:(void (^ _Nonnull)())f;
+- (bool)isCurrent;
+
+- (GroupCallDisposable * _Nonnull)scheduleBlock:(void (^ _Nonnull)())f after:(double)timeout;
+
+@end
+
+@interface VoipProxyServerWebrtc : NSObject
+
+@property (nonatomic, strong, readonly) NSString * _Nonnull host;
+@property (nonatomic, readonly) int32_t port;
+@property (nonatomic, strong, readonly) NSString * _Nullable username;
+@property (nonatomic, strong, readonly) NSString * _Nullable password;
+
+- (instancetype _Nonnull)initWithHost:(NSString * _Nonnull)host port:(int32_t)port username:(NSString * _Nullable)username password:(NSString * _Nullable)password;
+
+@end
+
+@protocol OngoingCallThreadLocalContextWebrtcVideoView <NSObject>
+
+@property (nonatomic, readonly) OngoingCallVideoOrientationWebrtc orientation;
+@property (nonatomic, readonly) CGFloat aspect;
+
+- (void)setOnFirstFrameReceived:(void (^ _Nullable)(float))onFirstFrameReceived;
+- (void)setOnOrientationUpdated:(void (^ _Nullable)(OngoingCallVideoOrientationWebrtc, CGFloat))onOrientationUpdated;
+- (void)setOnIsMirroredUpdated:(void (^ _Nullable)(bool))onIsMirroredUpdated;
+- (void)updateIsEnabled:(bool)isEnabled;
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+- (void)setVideoContentMode:(CALayerContentsGravity _Nonnull )mode;
+- (void)setForceMirrored:(bool)forceMirrored;
+- (void)setIsPaused:(bool)paused;
+- (void)renderToSize:(NSSize)size animated: (bool)animated;
+#endif
+@end
+
+@protocol CallVideoFrameBuffer
+
+@end
+
+@interface CallVideoFrameNativePixelBuffer : NSObject<CallVideoFrameBuffer>
+
+@property (nonatomic, readonly) CVPixelBufferRef _Nonnull pixelBuffer;
+
+@end
+
+@interface CallVideoFrameNV12Buffer : NSObject<CallVideoFrameBuffer>
+
+@property (nonatomic, readonly) int width;
+@property (nonatomic, readonly) int height;
+
+@property (nonatomic, strong, readonly) NSData * _Nonnull y;
+@property (nonatomic, readonly) int strideY;
+
+@property (nonatomic, strong, readonly) NSData * _Nonnull uv;
+@property (nonatomic, readonly) int strideUV;
+
+@end
+
+@interface CallVideoFrameI420Buffer : NSObject<CallVideoFrameBuffer>
+
+@property (nonatomic, readonly) int width;
+@property (nonatomic, readonly) int height;
+
+@property (nonatomic, strong, readonly) NSData * _Nonnull y;
+@property (nonatomic, readonly) int strideY;
+
+@property (nonatomic, strong, readonly) NSData * _Nonnull u;
+@property (nonatomic, readonly) int strideU;
+
+@property (nonatomic, strong, readonly) NSData * _Nonnull v;
+@property (nonatomic, readonly) int strideV;
+
+@end
+
+@interface CallVideoFrameData : NSObject
+
+@property (nonatomic, strong, readonly) id<CallVideoFrameBuffer> _Nonnull buffer;
+@property (nonatomic, readonly) int width;
+@property (nonatomic, readonly) int height;
+@property (nonatomic, readonly) OngoingCallVideoOrientationWebrtc orientation;
+@property (nonatomic, readonly) bool hasDeviceRelativeOrientation;
+@property (nonatomic, readonly) OngoingCallVideoOrientationWebrtc deviceRelativeOrientation;
+@property (nonatomic, readonly) bool mirrorHorizontally;
+@property (nonatomic, readonly) bool mirrorVertically;
+
+@end
+
+@interface OngoingCallThreadLocalContextVideoCapturer : NSObject
+
+- (instancetype _Nonnull)initWithDeviceId:(NSString * _Nonnull)deviceId keepLandscape:(bool)keepLandscape;
+
+#if TARGET_OS_IOS
++ (instancetype _Nonnull)capturerWithExternalSampleBufferProvider;
+#endif
+
+- (void)switchVideoInput:(NSString * _Nonnull)deviceId;
+- (void)setIsVideoEnabled:(bool)isVideoEnabled;
+
+- (void)makeOutgoingVideoView:(bool)requestClone completion:(void (^_Nonnull)(UIView<OngoingCallThreadLocalContextWebrtcVideoView> * _Nullable, UIView<OngoingCallThreadLocalContextWebrtcVideoView> * _Nullable))completion;
+
+- (void)setOnFatalError:(dispatch_block_t _Nullable)onError;
+- (void)setOnPause:(void (^ _Nullable)(bool))onPause;
+- (void)setOnIsActiveUpdated:(void (^_Nonnull)(bool))onIsActiveUpdated;
+
+#if TARGET_OS_IOS
+- (void)submitPixelBuffer:(CVPixelBufferRef _Nonnull)pixelBuffer rotation:(OngoingCallVideoOrientationWebrtc)rotation;
+#endif
+
+- (GroupCallDisposable * _Nonnull)addVideoOutput:(void (^_Nonnull)(CallVideoFrameData * _Nonnull))sink;
+
+@end
+
+@protocol OngoingCallDirectConnection <NSObject>
+
+- (NSData * _Nonnull)addOnIncomingPacket:(void (^_Nonnull)(NSData * _Nonnull))addOnIncomingPacket;
+- (void)removeOnIncomingPacket:(NSData * _Nonnull)token;
+- (void)sendPacket:(NSData * _Nonnull)packet;
+
+@end
+
+@interface OngoingCallThreadLocalContextWebrtc : NSObject
+
++ (void)setupLoggingFunction:(void (* _Nullable)(NSString * _Nullable))loggingFunction;
++ (void)applyServerConfig:(NSString * _Nullable)data;
++ (int32_t)maxLayer;
++ (NSArray<NSString *> * _Nonnull)versionsWithIncludeReference:(bool)includeReference;
+
+@property (nonatomic, copy) void (^ _Nullable stateChanged)(OngoingCallStateWebrtc, OngoingCallVideoStateWebrtc, OngoingCallRemoteVideoStateWebrtc, OngoingCallRemoteAudioStateWebrtc, OngoingCallRemoteBatteryLevelWebrtc, float);
+@property (nonatomic, copy) void (^ _Nullable signalBarsChanged)(int32_t);
+@property (nonatomic, copy) void (^ _Nullable audioLevelUpdated)(float);
+
+- (instancetype _Nonnull)initWithVersion:(NSString * _Nonnull)version queue:(id<OngoingCallThreadLocalContextQueueWebrtc> _Nonnull)queue proxy:(VoipProxyServerWebrtc * _Nullable)proxy networkType:(OngoingCallNetworkTypeWebrtc)networkType dataSaving:(OngoingCallDataSavingWebrtc)dataSaving derivedState:(NSData * _Nonnull)derivedState key:(NSData * _Nonnull)key isOutgoing:(bool)isOutgoing connections:(NSArray<OngoingCallConnectionDescriptionWebrtc *> * _Nonnull)connections maxLayer:(int32_t)maxLayer allowP2P:(BOOL)allowP2P allowTCP:(BOOL)allowTCP enableStunMarking:(BOOL)enableStunMarking logPath:(NSString * _Nonnull)logPath statsLogPath:(NSString * _Nonnull)statsLogPath sendSignalingData:(void (^ _Nonnull)(NSData * _Nonnull))sendSignalingData videoCapturer:(OngoingCallThreadLocalContextVideoCapturer * _Nullable)videoCapturer preferredVideoCodec:(NSString * _Nullable)preferredVideoCodec inputDeviceId: (NSString * _Nonnull)inputDeviceId outputDeviceId: (NSString * _Nonnull)outputDeviceId;
+
+- (void)beginTermination;
+- (void)stop:(void (^_Nullable)(NSString * _Nullable debugLog, int64_t bytesSentWifi, int64_t bytesReceivedWifi, int64_t bytesSentMobile, int64_t bytesReceivedMobile))completion;
+
+- (bool)needRate;
+
+- (NSString * _Nullable)debugInfo;
+- (NSString * _Nullable)version;
+- (NSData * _Nonnull)getDerivedState;
+
+- (void)setIsMuted:(bool)isMuted;
+- (void)setIsLowBatteryLevel:(bool)isLowBatteryLevel;
+- (void)setNetworkType:(OngoingCallNetworkTypeWebrtc)networkType;
+- (void)makeIncomingVideoView:(void (^_Nonnull)(UIView<OngoingCallThreadLocalContextWebrtcVideoView> * _Nullable))completion;
+- (void)requestVideo:(OngoingCallThreadLocalContextVideoCapturer * _Nullable)videoCapturer;
+- (void)setRequestedVideoAspect:(float)aspect;
+- (void)disableVideo;
+- (void)addSignalingData:(NSData * _Nonnull)data;
+- (void)switchAudioOutput:(NSString * _Nonnull)deviceId;
+- (void)switchAudioInput:(NSString * _Nonnull)deviceId;
+- (void)addExternalAudioData:(NSData * _Nonnull)data;
+
+- (GroupCallDisposable * _Nonnull)addVideoOutputWithIsIncoming:(bool)isIncoming sink:(void (^_Nonnull)(CallVideoFrameData * _Nonnull))sink;
+
+
+@end
+
+typedef struct {
+    bool isConnected;
+    bool isTransitioningFromBroadcastToRtc;
+} GroupCallNetworkState;
+
+typedef NS_ENUM(int32_t, OngoingGroupCallMediaChannelType) {
+    OngoingGroupCallMediaChannelTypeAudio,
+    OngoingGroupCallMediaChannelTypeVideo
+};
+
+@interface OngoingGroupCallMediaChannelDescription : NSObject
+@property (nonatomic, readonly) OngoingGroupCallMediaChannelType type;
+@property (nonatomic, readonly) uint64_t peerId;
+@property (nonatomic, readonly) uint32_t audioSsrc;
+@property (nonatomic, strong, readonly) NSString * _Nullable videoDescription;
+
+- (instancetype _Nonnull)initWithType:(OngoingGroupCallMediaChannelType)type
+    peerId:(int64_t)peerId
+    audioSsrc:(uint32_t)audioSsrc
+    videoDescription:(NSString * _Nullable)videoDescription;
+
+@end
+
+@protocol OngoingGroupCallBroadcastPartTask <NSObject>
+
+- (void)cancel;
+
+@end
+
+@protocol OngoingGroupCallMediaChannelDescriptionTask <NSObject>
+
+- (void)cancel;
+
+@end
+
+typedef NS_ENUM(int32_t, OngoingCallConnectionMode) {
+    OngoingCallConnectionModeNone,
+    OngoingCallConnectionModeRtc,
+    OngoingCallConnectionModeBroadcast
+};
+
+typedef NS_ENUM(int32_t, OngoingGroupCallBroadcastPartStatus) {
+    OngoingGroupCallBroadcastPartStatusSuccess,
+    OngoingGroupCallBroadcastPartStatusNotReady,
+    OngoingGroupCallBroadcastPartStatusResyncNeeded
+};
+
+typedef NS_ENUM(int32_t, OngoingGroupCallVideoContentType) {
+    OngoingGroupCallVideoContentTypeNone,
+    OngoingGroupCallVideoContentTypeGeneric,
+    OngoingGroupCallVideoContentTypeScreencast,
+};
+
+@interface OngoingGroupCallBroadcastPart : NSObject
+
+@property (nonatomic, readonly) int64_t timestampMilliseconds;
+@property (nonatomic, readonly) double responseTimestamp;
+@property (nonatomic, readonly) OngoingGroupCallBroadcastPartStatus status;
+@property (nonatomic, strong, readonly) NSData * _Nonnull oggData;
+
+- (instancetype _Nonnull)initWithTimestampMilliseconds:(int64_t)timestampMilliseconds responseTimestamp:(double)responseTimestamp status:(OngoingGroupCallBroadcastPartStatus)status oggData:(NSData * _Nonnull)oggData;
+
+@end
+
+typedef NS_ENUM(int32_t, OngoingGroupCallRequestedVideoQuality) {
+    OngoingGroupCallRequestedVideoQualityThumbnail,
+    OngoingGroupCallRequestedVideoQualityMedium,
+    OngoingGroupCallRequestedVideoQualityFull,
+};
+
+@interface OngoingGroupCallSsrcGroup : NSObject
+
+@property (nonatomic, strong, readonly) NSString * _Nonnull semantics;
+@property (nonatomic, strong, readonly) NSArray<NSNumber *> * _Nonnull ssrcs;
+
+- (instancetype _Nonnull)initWithSemantics:(NSString * _Nonnull)semantics ssrcs:(NSArray<NSNumber *> * _Nonnull)ssrcs;
+
+@end
+
+@interface OngoingGroupCallRequestedVideoChannel : NSObject
+
+@property (nonatomic, readonly) uint32_t audioSsrc;
+@property (nonatomic, readonly) int64_t userId;
+@property (nonatomic, strong, readonly) NSString * _Nonnull endpointId;
+@property (nonatomic, strong, readonly) NSArray<OngoingGroupCallSsrcGroup *> * _Nonnull ssrcGroups;
+
+@property (nonatomic, readonly) OngoingGroupCallRequestedVideoQuality minQuality;
+@property (nonatomic, readonly) OngoingGroupCallRequestedVideoQuality maxQuality;
+
+- (instancetype _Nonnull)initWithAudioSsrc:(uint32_t)audioSsrc  userId:(int64_t)userId endpointId:(NSString * _Nonnull)endpointId ssrcGroups:(NSArray<OngoingGroupCallSsrcGroup *> * _Nonnull)ssrcGroups minQuality:(OngoingGroupCallRequestedVideoQuality)minQuality maxQuality:(OngoingGroupCallRequestedVideoQuality)maxQuality;
+
+@end
+
+@interface OngoingGroupCallIncomingVideoStats : NSObject
+
+@property (nonatomic, readonly) int receivingQuality;
+@property (nonatomic, readonly) int availableQuality;
+
+- (instancetype _Nonnull)initWithReceivingQuality:(int)receivingQuality availableQuality:(int)availableQuality;
+
+@end
+
+@interface OngoingGroupCallStats : NSObject
+
+@property (nonatomic, strong, readonly) NSDictionary<NSString *, OngoingGroupCallIncomingVideoStats *> * _Nonnull incomingVideoStats;
+
+- (instancetype _Nonnull)initWithIncomingVideoStats:(NSDictionary<NSString *, OngoingGroupCallIncomingVideoStats *> * _Nonnull)incomingVideoStats;
+
+@end
+
+@interface GroupCallThreadLocalContext : NSObject
+
+@property (nonatomic, copy) void (^ _Nullable signalBarsChanged)(int32_t);
+
+- (instancetype _Nonnull)initWithQueue:(id<OngoingCallThreadLocalContextQueueWebrtc> _Nonnull)queue
+    networkStateUpdated:(void (^ _Nonnull)(GroupCallNetworkState))networkStateUpdated
+    audioLevelsUpdated:(void (^ _Nonnull)(NSArray<NSNumber *> * _Nonnull))audioLevelsUpdated
+    activityUpdated:(void (^ _Nonnull)(NSArray<NSNumber *> * _Nonnull))activityUpdated
+    inputDeviceId:(NSString * _Nonnull)inputDeviceId
+    outputDeviceId:(NSString * _Nonnull)outputDeviceId
+    videoCapturer:(OngoingCallThreadLocalContextVideoCapturer * _Nullable)videoCapturer
+    requestMediaChannelDescriptions:(id<OngoingGroupCallMediaChannelDescriptionTask> _Nonnull (^ _Nonnull)(NSArray<NSNumber *> * _Nonnull, void (^ _Nonnull)(NSArray<OngoingGroupCallMediaChannelDescription *> * _Nonnull)))requestMediaChannelDescriptions
+    requestCurrentTime:(id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull)(void (^ _Nonnull)(int64_t)))requestAudioBroadcastPart
+    requestAudioBroadcastPart:(id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull)(int64_t, int64_t, void (^ _Nonnull)(OngoingGroupCallBroadcastPart * _Nullable)))requestAudioBroadcastPart
+    requestVideoBroadcastPart:(id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull)(int64_t, int64_t, int32_t, OngoingGroupCallRequestedVideoQuality, void (^ _Nonnull)(OngoingGroupCallBroadcastPart * _Nullable)))requestVideoBroadcastPart
+    outgoingAudioBitrateKbit:(int32_t)outgoingAudioBitrateKbit
+    videoContentType:(OngoingGroupCallVideoContentType)videoContentType
+    enableNoiseSuppression:(bool)enableNoiseSuppression
+    disableAudioInput:(bool)disableAudioInput
+    prioritizeVP8:(bool)prioritizeVP8
+    logPath:(NSString * _Nonnull)logPath
+    statsLogPath:(NSString * _Nonnull)statsLogPath
+    audioDevice:(SharedCallAudioDevice * _Nullable)audioDevice
+    isConference:(bool)isConference
+    isActiveByDefault:(bool)isActiveByDefault
+    encryptDecrypt:(NSData * _Nullable (^ _Nullable)(NSData * _Nonnull, int64_t, bool, int32_t))encryptDecrypt;
+
+- (void)stop:(void (^ _Nullable)())completion;
+
+- (void)setManualAudioSessionIsActive:(bool)isAudioSessionActive;
+
+- (void)setTone:(CallAudioTone * _Nullable)tone;
+
+- (void)setConnectionMode:(OngoingCallConnectionMode)connectionMode keepBroadcastConnectedIfWasEnabled:(bool)keepBroadcastConnectedIfWasEnabled isUnifiedBroadcast:(bool)isUnifiedBroadcast;
+
+- (void)emitJoinPayload:(void (^ _Nonnull)(NSString * _Nonnull, uint32_t))completion;
+- (void)setJoinResponsePayload:(NSString * _Nonnull)payload;
+- (void)removeSsrcs:(NSArray<NSNumber *> * _Nonnull)ssrcs;
+- (void)removeIncomingVideoSource:(uint32_t)ssrc;
+- (void)setIsMuted:(bool)isMuted;
+- (void)setIsNoiseSuppressionEnabled:(bool)isNoiseSuppressionEnabled;
+- (void)requestVideo:(OngoingCallThreadLocalContextVideoCapturer * _Nullable)videoCapturer completion:(void (^ _Nonnull)(NSString * _Nonnull, uint32_t))completion;
+- (void)disableVideo:(void (^ _Nonnull)(NSString * _Nonnull, uint32_t))completion;
+
+- (void)setVolumeForSsrc:(uint32_t)ssrc volume:(double)volume;
+- (void)setRequestedVideoChannels:(NSArray<OngoingGroupCallRequestedVideoChannel *> * _Nonnull)requestedVideoChannels;
+
+- (void)switchAudioOutput:(NSString * _Nonnull)deviceId;
+- (void)switchAudioInput:(NSString * _Nonnull)deviceId;
+- (void)makeIncomingVideoViewWithEndpointId:(NSString * _Nonnull)endpointId requestClone:(bool)requestClone completion:(void (^_Nonnull)(UIView<OngoingCallThreadLocalContextWebrtcVideoView> * _Nullable, UIView<OngoingCallThreadLocalContextWebrtcVideoView> * _Nullable))completion;
+- (GroupCallDisposable * _Nonnull)addVideoOutputWithEndpointId:(NSString * _Nonnull)endpointId sink:(void (^_Nonnull)(CallVideoFrameData * _Nonnull))sink;
+
+- (void)addExternalAudioData:(NSData * _Nonnull)data;
+
+- (void)getStats:(void (^ _Nonnull)(OngoingGroupCallStats * _Nonnull))completion;
+
+@end
+
+#endif

Telegram/ThirdParty/tgcalls/macos/PublicHeaders/shim.h

@@ -0,0 +1 @@
+#import <../../tgcalls/platform/darwin/DesktopCaptureSourceViewMac.h>

Telegram/ThirdParty/tgcalls/macos/Sources/LibYuvBinding.mm

@@ -0,0 +1,28 @@
+#import <TgVoipWebrtc/LibYuvBinding.h>
+
+#include "third_party/libyuv/include/libyuv/convert_from.h"
+
+
+@implementation LibYUVConverter
+
++ (BOOL)I420ToNV12WithSrcY:(const uint8_t *)src_y
+               srcStrideY:(int)src_stride_y
+                    srcU:(const uint8_t *)src_u
+               srcStrideU:(int)src_stride_u
+                    srcV:(const uint8_t *)src_v
+               srcStrideV:(int)src_stride_v
+                    dstY:(uint8_t *)dst_y
+               dstStrideY:(int)dst_stride_y
+                   dstUV:(uint8_t *)dst_uv
+               dstStrideUV:(int)dst_stride_uv
+                    width:(int)width
+                   height:(int)height {
+    // Call the actual libyuv function
+    int result = libyuv::I420ToNV12(src_y, src_stride_y, src_u, src_stride_u, src_v, src_stride_v,
+                                   dst_y, dst_stride_y, dst_uv, dst_stride_uv, width, height);
+
+    // Return YES if the conversion was successful, NO otherwise
+    return result == 0;
+}
+
+@end

Telegram/ThirdParty/tgcalls/macos/Sources/MediaStreaming.mm

@@ -0,0 +1,275 @@
+#import <TgVoipWebrtc/MediaStreaming.h>
+
+#import "MediaUtils.h"
+
+#include "StaticThreads.h"
+#include "group/StreamingMediaContext.h"
+
+#include "api/video/video_sink_interface.h"
+#include "sdk/objc/native/src/objc_frame_buffer.h"
+#include "api/video/video_frame.h"
+
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+#import "platform/darwin/TGRTCCVPixelBuffer.h"
+
+#include <memory>
+
+namespace {
+
+class BroadcastPartTaskImpl : public tgcalls::BroadcastPartTask {
+public:
+    BroadcastPartTaskImpl(id<OngoingGroupCallBroadcastPartTask> task) {
+        _task = task;
+    }
+    
+    virtual ~BroadcastPartTaskImpl() {
+    }
+    
+    virtual void cancel() override {
+        [_task cancel];
+    }
+    
+private:
+    id<OngoingGroupCallBroadcastPartTask> _task;
+};
+
+class VideoSinkAdapter : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+public:
+    VideoSinkAdapter(void (^frameReceived)(webrtc::VideoFrame const &)) {
+        _frameReceived = [frameReceived copy];
+    }
+
+    void OnFrame(const webrtc::VideoFrame& nativeVideoFrame) override {
+        @autoreleasepool {
+            if (_frameReceived) {
+                _frameReceived(nativeVideoFrame);
+            }
+        }
+    }
+
+private:
+    void (^_frameReceived)(webrtc::VideoFrame const &);
+};
+
+}
+
+@interface MediaStreamingVideoSink : NSObject {
+    std::shared_ptr<VideoSinkAdapter> _adapter;
+}
+
+@end
+
+
+@implementation MediaStreamingVideoSink
+
+- (instancetype)initWithSink:(void (^_Nonnull)(CallVideoFrameData * _Nonnull))sink {
+    self = [super init];
+    if (self != nil) {
+        void (^storedSink)(CallVideoFrameData * _Nonnull) = [sink copy];
+
+        _adapter.reset(new VideoSinkAdapter(^(webrtc::VideoFrame const &videoFrame) {
+            id<CallVideoFrameBuffer> mappedBuffer = nil;
+
+            bool mirrorHorizontally = false;
+            bool mirrorVertically = false;
+
+            if (videoFrame.video_frame_buffer()->type() == webrtc::VideoFrameBuffer::Type::kNative) {
+                id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> nativeBuffer = static_cast<webrtc::ObjCFrameBuffer *>(videoFrame.video_frame_buffer().get())->wrapped_frame_buffer();
+                if ([nativeBuffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]) {
+                    RTCCVPixelBuffer *pixelBuffer = (RTCCVPixelBuffer *)nativeBuffer;
+                    mappedBuffer = [[CallVideoFrameNativePixelBuffer alloc] initWithPixelBuffer:pixelBuffer.pixelBuffer];
+                }
+                if ([nativeBuffer isKindOfClass:[TGRTCCVPixelBuffer class]]) {
+                    if (((TGRTCCVPixelBuffer *)nativeBuffer).shouldBeMirrored) {
+                        switch (videoFrame.rotation()) {
+                            case webrtc::kVideoRotation_0:
+                            case webrtc::kVideoRotation_180:
+                                mirrorHorizontally = true;
+                                break;
+                            case webrtc::kVideoRotation_90:
+                            case webrtc::kVideoRotation_270:
+                                mirrorVertically = true;
+                                break;
+                            default:
+                                break;
+                        }
+                    }
+                }
+            } else if (videoFrame.video_frame_buffer()->type() == webrtc::VideoFrameBuffer::Type::kNV12) {
+                rtc::scoped_refptr<webrtc::NV12BufferInterface> nv12Buffer(static_cast<webrtc::NV12BufferInterface *>(videoFrame.video_frame_buffer().get()));
+                
+                mappedBuffer = [[CallVideoFrameNV12Buffer alloc] initWithBuffer:nv12Buffer];
+            } else if (videoFrame.video_frame_buffer()->type() == webrtc::VideoFrameBuffer::Type::kI420) {
+                rtc::scoped_refptr<webrtc::I420BufferInterface> i420Buffer(static_cast<webrtc::I420BufferInterface *>(videoFrame.video_frame_buffer().get()));
+                
+                mappedBuffer = [[CallVideoFrameI420Buffer alloc] initWithBuffer:i420Buffer];
+            }
+
+            if (storedSink && mappedBuffer) {
+                storedSink([[CallVideoFrameData alloc] initWithBuffer:mappedBuffer frame:videoFrame mirrorHorizontally:mirrorHorizontally mirrorVertically:mirrorVertically]);
+            }
+        }));
+    }
+    return self;
+}
+
+- (std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>>)sink {
+    return _adapter;
+}
+
+@end
+
+@interface MediaStreamingContext () {
+    id<OngoingCallThreadLocalContextQueueWebrtc> _queue;
+    
+    id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull _requestCurrentTime)(void (^ _Nonnull)(int64_t));
+    id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull _requestAudioBroadcastPart)(int64_t, int64_t, void (^ _Nonnull)(OngoingGroupCallBroadcastPart * _Nullable));
+    id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull _requestVideoBroadcastPart)(int64_t, int64_t, int32_t, OngoingGroupCallRequestedVideoQuality, void (^ _Nonnull)(OngoingGroupCallBroadcastPart * _Nullable));
+    
+    std::unique_ptr<tgcalls::StreamingMediaContext> _context;
+    
+    int _nextSinkId;
+    NSMutableDictionary<NSNumber *, MediaStreamingVideoSink *> *_sinks;
+}
+
+@end
+
+@implementation MediaStreamingContext
+
+- (instancetype _Nonnull)initWithQueue:(id<OngoingCallThreadLocalContextQueueWebrtc> _Nonnull)queue
+    requestCurrentTime:(id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull)(void (^ _Nonnull)(int64_t)))requestCurrentTime
+    requestAudioBroadcastPart:(id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull)(int64_t, int64_t, void (^ _Nonnull)(OngoingGroupCallBroadcastPart * _Nullable)))requestAudioBroadcastPart
+    requestVideoBroadcastPart:(id<OngoingGroupCallBroadcastPartTask> _Nonnull (^ _Nonnull)(int64_t, int64_t, int32_t, OngoingGroupCallRequestedVideoQuality, void (^ _Nonnull)(OngoingGroupCallBroadcastPart * _Nullable)))requestVideoBroadcastPart {
+    self = [super init];
+    if (self != nil) {
+        _queue = queue;
+        
+        _requestCurrentTime = [requestCurrentTime copy];
+        _requestAudioBroadcastPart = [requestAudioBroadcastPart copy];
+        _requestVideoBroadcastPart = [requestVideoBroadcastPart copy];
+        
+        _sinks = [[NSMutableDictionary alloc] init];
+    }
+    return self;
+}
+
+- (void)dealloc {
+}
+
+- (void)resetContext {
+    tgcalls::StreamingMediaContext::StreamingMediaContextArguments arguments;
+    arguments.threads = tgcalls::StaticThreads::getThreads();
+    arguments.isUnifiedBroadcast = true;
+    arguments.requestCurrentTime = [requestCurrentTime = _requestCurrentTime](std::function<void(int64_t)> completion) -> std::shared_ptr<tgcalls::BroadcastPartTask> {
+        id<OngoingGroupCallBroadcastPartTask> task = requestCurrentTime(^(int64_t result) {
+            completion(result);
+        });
+        return std::make_shared<BroadcastPartTaskImpl>(task);
+    };
+    arguments.requestAudioBroadcastPart = nullptr;
+    arguments.requestVideoBroadcastPart = [requestVideoBroadcastPart = _requestVideoBroadcastPart](int64_t timestampMilliseconds, int64_t durationMilliseconds, int32_t channelId, tgcalls::VideoChannelDescription::Quality quality, std::function<void(tgcalls::BroadcastPart &&)> completion) -> std::shared_ptr<tgcalls::BroadcastPartTask> {
+        OngoingGroupCallRequestedVideoQuality mappedQuality;
+        switch (quality) {
+            case tgcalls::VideoChannelDescription::Quality::Thumbnail: {
+                mappedQuality = OngoingGroupCallRequestedVideoQualityThumbnail;
+                break;
+            }
+            case tgcalls::VideoChannelDescription::Quality::Medium: {
+                mappedQuality = OngoingGroupCallRequestedVideoQualityMedium;
+                break;
+            }
+            case tgcalls::VideoChannelDescription::Quality::Full: {
+                mappedQuality = OngoingGroupCallRequestedVideoQualityFull;
+                break;
+            }
+            default: {
+                mappedQuality = OngoingGroupCallRequestedVideoQualityThumbnail;
+                break;
+            }
+        }
+        id<OngoingGroupCallBroadcastPartTask> task = requestVideoBroadcastPart(timestampMilliseconds, durationMilliseconds, channelId, mappedQuality, ^(OngoingGroupCallBroadcastPart * _Nullable part) {
+            tgcalls::BroadcastPart parsedPart;
+            parsedPart.timestampMilliseconds = part.timestampMilliseconds;
+
+            parsedPart.responseTimestamp = part.responseTimestamp;
+
+            tgcalls::BroadcastPart::Status mappedStatus;
+            switch (part.status) {
+                case OngoingGroupCallBroadcastPartStatusSuccess: {
+                    mappedStatus = tgcalls::BroadcastPart::Status::Success;
+                    break;
+                }
+                case OngoingGroupCallBroadcastPartStatusNotReady: {
+                    mappedStatus = tgcalls::BroadcastPart::Status::NotReady;
+                    break;
+                }
+                case OngoingGroupCallBroadcastPartStatusResyncNeeded: {
+                    mappedStatus = tgcalls::BroadcastPart::Status::ResyncNeeded;
+                    break;
+                }
+                default: {
+                    mappedStatus = tgcalls::BroadcastPart::Status::NotReady;
+                    break;
+                }
+            }
+            parsedPart.status = mappedStatus;
+
+            parsedPart.data.resize(part.oggData.length);
+            [part.oggData getBytes:parsedPart.data.data() length:part.oggData.length];
+
+            completion(std::move(parsedPart));
+        });
+        return std::make_shared<BroadcastPartTaskImpl>(task);
+    };
+    
+    arguments.updateAudioLevel = nullptr;
+    
+    _context = std::make_unique<tgcalls::StreamingMediaContext>(std::move(arguments));
+    
+    for (MediaStreamingVideoSink *storedSink in _sinks.allValues) {
+        _context->addVideoSink("unified", [storedSink sink]);
+    }
+}
+
+- (void)start {
+    [self resetContext];
+}
+
+- (void)stop {
+    _context.reset();
+}
+
+- (GroupCallDisposable * _Nonnull)addVideoOutput:(void (^_Nonnull)(CallVideoFrameData * _Nonnull))sink {
+    int sinkId = _nextSinkId;
+    _nextSinkId += 1;
+
+    MediaStreamingVideoSink *storedSink = [[MediaStreamingVideoSink alloc] initWithSink:sink];
+    _sinks[@(sinkId)] = storedSink;
+
+    if (_context) {
+        _context->addVideoSink("unified", [storedSink sink]);
+    }
+
+    __weak MediaStreamingContext *weakSelf = self;
+    id<OngoingCallThreadLocalContextQueueWebrtc> queue = _queue;
+    return [[GroupCallDisposable alloc] initWithBlock:^{
+        [queue dispatch:^{
+            __strong MediaStreamingContext *strongSelf = weakSelf;
+            if (!strongSelf) {
+                return;
+            }
+
+            [strongSelf->_sinks removeObjectForKey:@(sinkId)];
+        }];
+    }];
+}
+
+- (void)getAudio:(int16_t * _Nonnull)audioSamples numSamples:(NSInteger)numSamples numChannels:(NSInteger)numChannels samplesPerSecond:(NSInteger)samplesPerSecond {
+    if (_context) {
+        _context->getAudio(audioSamples, (size_t)numSamples, (size_t)numChannels, (uint32_t)samplesPerSecond);
+    } else {
+        memset(audioSamples, 0, numSamples * numChannels * sizeof(int16_t));
+    }
+}
+
+@end

Telegram/ThirdParty/tgcalls/macos/Sources/MediaUtils.h

@@ -0,0 +1,57 @@
+#import <TgVoipWebrtc/OngoingCallThreadLocalContext.h>
+
+
+#import "Instance.h"
+#import "InstanceImpl.h"
+#import "v2/InstanceV2Impl.h"
+#include "StaticThreads.h"
+
+#import "VideoCaptureInterface.h"
+#import "platform/darwin/VideoCameraCapturer.h"
+
+#ifndef WEBRTC_IOS
+#import "platform/darwin/VideoMetalViewMac.h"
+#import "platform/darwin/VideoSampleBufferViewMac.h"
+#define UIViewContentModeScaleAspectFill kCAGravityResizeAspectFill
+#define UIViewContentModeScaleAspect kCAGravityResizeAspect
+
+#else
+#import "platform/darwin/VideoMetalView.h"
+#import "platform/darwin/GLVideoView.h"
+#import "platform/darwin/VideoSampleBufferView.h"
+#import "platform/darwin/VideoCaptureView.h"
+#import "platform/darwin/CustomExternalCapturer.h"
+#endif
+
+#import "group/GroupInstanceImpl.h"
+#import "group/GroupInstanceCustomImpl.h"
+
+#import "VideoCaptureInterfaceImpl.h"
+
+#include "sdk/objc/native/src/objc_frame_buffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+#import "platform/darwin/TGRTCCVPixelBuffer.h"
+
+@interface CallVideoFrameNativePixelBuffer (Initialization)
+
+- (instancetype _Nonnull)initWithPixelBuffer:(CVPixelBufferRef _Nonnull)pixelBuffer;
+
+@end
+
+@interface CallVideoFrameI420Buffer (Initialization)
+
+- (instancetype _Nonnull)initWithBuffer:(rtc::scoped_refptr<webrtc::I420BufferInterface>)i420Buffer;
+
+@end
+
+@interface CallVideoFrameNV12Buffer (Initialization)
+
+- (instancetype _Nonnull)initWithBuffer:(rtc::scoped_refptr<webrtc::NV12BufferInterface>)nv12Buffer;
+
+@end
+
+@interface CallVideoFrameData (Initialization)
+
+- (instancetype _Nonnull)initWithBuffer:(id<CallVideoFrameBuffer> _Nonnull)buffer frame:(webrtc::VideoFrame const &)frame mirrorHorizontally:(bool)mirrorHorizontally mirrorVertically:(bool)mirrorVertically;
+
+@end

Telegram/ThirdParty/tgcalls/tgcalls/AudioDeviceHelper.cpp

@@ -0,0 +1,142 @@
+#include "AudioDeviceHelper.h"
+
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/logging.h"
+
+namespace tgcalls {
+namespace {
+
+bool SkipDefaultDevice(const char *name) {
+	const auto utfName = std::string(name);
+#ifdef WEBRTC_WIN
+	return (utfName.rfind("Default - ", 0) == 0)
+		|| (utfName.rfind("Communication - ", 0) == 0);
+#elif defined WEBRTC_MAC
+	return (utfName.rfind("default (", 0) == 0)
+		&& (utfName.find(")", utfName.size() - 1) == utfName.size() - 1);
+#else
+	return false;
+#endif // WEBRTC_WIN || WEBRTC_MAC
+}
+
+} // namespace
+
+void SetAudioInputDeviceById(webrtc::AudioDeviceModule *adm, const std::string &id) {
+	const auto recording = adm->Recording() || adm->RecordingIsInitialized();
+	if (recording) {
+		adm->StopRecording();
+	}
+	auto specific = false;
+	const auto finish = [&] {
+		if (!specific) {
+			if (const auto result = adm->SetRecordingDevice(webrtc::AudioDeviceModule::kDefaultCommunicationDevice)) {
+				RTC_LOG(LS_ERROR) << "setAudioInputDevice(" << id << "): SetRecordingDevice(kDefaultCommunicationDevice) failed: " << result << ".";
+			} else {
+				RTC_LOG(LS_INFO) << "setAudioInputDevice(" << id << "): SetRecordingDevice(kDefaultCommunicationDevice) success.";
+			}
+		}
+		if (recording && adm->InitRecording() == 0) {
+			adm->StartRecording();
+		}
+	};
+	if (id == "default" || id.empty()) {
+		return finish();
+	}
+#ifdef TGCALLS_UWP_DESKTOP
+	const auto result = adm->SetRecordingDevice(id);
+	if (result != 0) {
+		RTC_LOG(LS_ERROR) << "setAudioInputDevice(" << id << ") failed: " << result << ".";
+	} else {
+		RTC_LOG(LS_INFO) << "setAudioInputDevice(" << id << ") success.";
+		specific = true;
+	}
+	return finish();
+#else // TGCALLS_UWP_DESKTOP
+	const auto count = adm
+		? adm->RecordingDevices()
+		: int16_t(-666);
+	if (count <= 0) {
+		RTC_LOG(LS_ERROR) << "setAudioInputDevice(" << id << "): Could not get recording devices count: " << count << ".";
+		return finish();
+	}
+
+        int16_t order = !id.empty() && id[0] == '#' ? static_cast<int16_t>(std::stoi(id.substr(1))) : -1;
+        for (auto i = 0; i != count; ++i) {
+		char name[webrtc::kAdmMaxDeviceNameSize + 1] = { 0 };
+		char guid[webrtc::kAdmMaxGuidSize + 1] = { 0 };
+		adm->RecordingDeviceName(i, name, guid);
+		if ((!SkipDefaultDevice(name) && id == guid) || order == i) {
+			const auto result = adm->SetRecordingDevice(i);
+			if (result != 0) {
+				RTC_LOG(LS_ERROR) << "setAudioInputDevice(" << id << ") name '" << std::string(name) << "' failed: " << result << ".";
+			} else {
+				RTC_LOG(LS_INFO) << "setAudioInputDevice(" << id << ") name '" << std::string(name) << "' success.";
+				specific = true;
+			}
+			return finish();
+		}
+	}
+	RTC_LOG(LS_ERROR) << "setAudioInputDevice(" << id << "): Could not find recording device.";
+	return finish();
+#endif // TGCALLS_UWP_DESKTOP
+}
+
+void SetAudioOutputDeviceById(webrtc::AudioDeviceModule *adm, const std::string &id) {
+	if (adm->Playing()) {
+		adm->StopPlayout();
+	}
+	auto specific = false;
+	const auto finish = [&] {
+		if (!specific) {
+			if (const auto result = adm->SetPlayoutDevice(webrtc::AudioDeviceModule::kDefaultCommunicationDevice)) {
+				RTC_LOG(LS_ERROR) << "setAudioOutputDevice(" << id << "): SetPlayoutDevice(kDefaultCommunicationDevice) failed: " << result << ".";
+			} else {
+				RTC_LOG(LS_INFO) << "setAudioOutputDevice(" << id << "): SetPlayoutDevice(kDefaultCommunicationDevice) success.";
+			}
+		}
+		if (adm->InitPlayout() == 0) {
+			adm->StartPlayout();
+		}
+	};
+	if (id == "default" || id.empty()) {
+		return finish();
+	}
+#ifdef TGCALLS_UWP_DESKTOP
+	const auto result = adm->SetPlayoutDevice(id);
+	if (result != 0) {
+		RTC_LOG(LS_ERROR) << "setAudioOutputDevice(" << id << ") failed: " << result << ".";
+	} else {
+		RTC_LOG(LS_INFO) << "setAudioOutputDevice(" << id << ") success.";
+		specific = true;
+	}
+	return finish();
+#else // TGCALLS_UWP_DESKTOP
+	const auto count = adm
+		? adm->PlayoutDevices()
+		: int16_t(-666);
+	if (count <= 0) {
+		RTC_LOG(LS_ERROR) << "setAudioOutputDevice(" << id << "): Could not get playout devices count: " << count << ".";
+		return finish();
+	}
+        int16_t order = !id.empty() && id[0] == '#' ? static_cast<int16_t>(std::stoi(id.substr(1))) : -1;
+	for (auto i = 0; i != count; ++i) {
+		char name[webrtc::kAdmMaxDeviceNameSize + 1] = { 0 };
+		char guid[webrtc::kAdmMaxGuidSize + 1] = { 0 };
+		adm->PlayoutDeviceName(i, name, guid);
+		if ((!SkipDefaultDevice(name) && id == guid) || order == i) {
+			const auto result = adm->SetPlayoutDevice(i);
+			if (result != 0) {
+				RTC_LOG(LS_ERROR) << "setAudioOutputDevice(" << id << ") name '" << std::string(name) << "' failed: " << result << ".";
+			} else {
+				RTC_LOG(LS_INFO) << "setAudioOutputDevice(" << id << ") name '" << std::string(name) << "' success.";
+				specific = true;
+			}
+			return finish();
+		}
+	}
+	RTC_LOG(LS_ERROR) << "setAudioOutputDevice(" << id << "): Could not find playout device.";
+	return finish();
+#endif // TGCALLS_UWP_DESKTOP
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/AudioDeviceHelper.h

@@ -0,0 +1,17 @@
+#ifndef TGCALLS_AUDIO_DEVICE_HELPER_H
+#define TGCALLS_AUDIO_DEVICE_HELPER_H
+
+#include <string>
+
+namespace webrtc {
+class AudioDeviceModule;
+} // namespace webrtc
+
+namespace tgcalls {
+
+void SetAudioInputDeviceById(webrtc::AudioDeviceModule *adm, const std::string &id);
+void SetAudioOutputDeviceById(webrtc::AudioDeviceModule *adm, const std::string &id);
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/AudioFrame.h

@@ -0,0 +1,16 @@
+#pragma once
+
+#include <cinttypes>
+#include <cstring>
+
+namespace tgcalls {
+struct AudioFrame {
+  const int16_t *audio_samples;
+  size_t num_samples;
+  size_t bytes_per_sample;
+  size_t num_channels;
+  uint32_t samples_per_sec;
+  int64_t elapsed_time_ms;
+  int64_t ntp_time_ms;
+};
+}  // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/ChannelManager.cpp

@@ -0,0 +1,161 @@
+#include "ChannelManager.h"
+#include <utility>
+#include "absl/algorithm/container.h"
+#include "absl/memory/memory.h"
+#include "absl/strings/match.h"
+#include "api/media_types.h"
+#include "api/sequence_checker.h"
+#include "media/base/media_constants.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+namespace tgcalls {
+// static
+std::unique_ptr<ChannelManager> ChannelManager::Create(
+    std::unique_ptr<cricket::MediaEngineInterface> media_engine,
+    rtc::Thread* worker_thread,
+    rtc::Thread* network_thread) {
+  RTC_DCHECK(network_thread);
+  RTC_DCHECK(worker_thread);
+  return absl::WrapUnique(new ChannelManager(
+      std::move(media_engine), worker_thread, network_thread));
+}
+ChannelManager::ChannelManager(
+    std::unique_ptr<cricket::MediaEngineInterface> media_engine,
+    rtc::Thread* worker_thread,
+    rtc::Thread* network_thread)
+    : media_engine_(std::move(media_engine)),
+      signaling_thread_(rtc::Thread::Current()),
+      worker_thread_(worker_thread),
+      network_thread_(network_thread) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  RTC_DCHECK(worker_thread_);
+  RTC_DCHECK(network_thread_);
+  if (media_engine_) {
+    // TODO(tommi): Change VoiceEngine to do ctor time initialization so that
+    // this isn't necessary.
+    worker_thread_->BlockingCall([&] { media_engine_->Init(); });
+  }
+}
+ChannelManager::~ChannelManager() {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  worker_thread_->BlockingCall([&] {
+    RTC_DCHECK_RUN_ON(worker_thread_);
+    RTC_DCHECK(voice_channels_.empty());
+    RTC_DCHECK(video_channels_.empty());
+    // While `media_engine_` is const throughout the ChannelManager's lifetime,
+    // it requires destruction to happen on the worker thread. Instead of
+    // marking the pointer as non-const, we live with this const_cast<> in the
+    // destructor.
+    const_cast<std::unique_ptr<cricket::MediaEngineInterface>&>(media_engine_).reset();
+  });
+}
+cricket::VoiceChannel* ChannelManager::CreateVoiceChannel(
+    webrtc::Call* call,
+    const cricket::MediaConfig& media_config,
+    const std::string& mid,
+    bool srtp_required,
+    const webrtc::CryptoOptions& crypto_options,
+    const cricket::AudioOptions& options) {
+  RTC_DCHECK(call);
+  RTC_DCHECK(media_engine_);
+  // TODO(bugs.webrtc.org/11992): Remove this workaround after updates in
+  // PeerConnection and add the expectation that we're already on the right
+  // thread.
+  if (!worker_thread_->IsCurrent()) {
+    cricket::VoiceChannel* temp = nullptr;
+    worker_thread_->BlockingCall([&] {
+      temp = CreateVoiceChannel(call, media_config, mid, srtp_required,
+                                crypto_options, options);
+    });
+    return temp;
+  }
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_media_channel = media_engine_->voice().CreateSendChannel(
+      call, media_config, options, crypto_options, webrtc::AudioCodecPairId::Create());
+  if (!send_media_channel) {
+    return nullptr;
+  }
+  std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface> receive_media_channel = media_engine_->voice().CreateReceiveChannel(
+        call, media_config, options, crypto_options, webrtc::AudioCodecPairId::Create());
+  if (!receive_media_channel) {
+    return nullptr;
+  }
+  auto voice_channel = std::make_unique<cricket::VoiceChannel>(
+      worker_thread_, network_thread_, signaling_thread_,
+      std::move(send_media_channel), std::move(receive_media_channel), mid, srtp_required, crypto_options,
+      &ssrc_generator_);
+  cricket::VoiceChannel* voice_channel_ptr = voice_channel.get();
+  voice_channels_.push_back(std::move(voice_channel));
+  return voice_channel_ptr;
+}
+void ChannelManager::DestroyVoiceChannel(cricket::VoiceChannel* channel) {
+  TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  voice_channels_.erase(absl::c_find_if(
+      voice_channels_, [&](const auto& p) { return p.get() == channel; }));
+}
+cricket::VideoChannel* ChannelManager::CreateVideoChannel(
+    webrtc::Call* call,
+    const cricket::MediaConfig& media_config,
+    const std::string& mid,
+    bool srtp_required,
+    const webrtc::CryptoOptions& crypto_options,
+    const cricket::VideoOptions& options,
+    webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory) {
+  RTC_DCHECK(call);
+  RTC_DCHECK(media_engine_);
+  // TODO(bugs.webrtc.org/11992): Remove this workaround after updates in
+  // PeerConnection and add the expectation that we're already on the right
+  // thread.
+  if (!worker_thread_->IsCurrent()) {
+    cricket::VideoChannel* temp = nullptr;
+    worker_thread_->BlockingCall([&] {
+      temp = CreateVideoChannel(call, media_config, mid, srtp_required,
+                                crypto_options, options,
+                                video_bitrate_allocator_factory);
+    });
+    return temp;
+  }
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  std::unique_ptr<cricket::VideoMediaSendChannelInterface> send_media_channel = media_engine_->video().CreateSendChannel(
+      call, media_config, options, crypto_options,
+      video_bitrate_allocator_factory);
+  if (!send_media_channel) {
+    return nullptr;
+  }
+  std::unique_ptr<cricket::VideoMediaReceiveChannelInterface> receive_media_channel = media_engine_->video().CreateReceiveChannel(
+    call, media_config, options, crypto_options);
+  if (!receive_media_channel) {
+    return nullptr;
+  }
+  auto video_channel = std::make_unique<cricket::VideoChannel>(
+      worker_thread_, network_thread_, signaling_thread_,
+      std::move(send_media_channel), std::move(receive_media_channel), mid, srtp_required, crypto_options,
+      &ssrc_generator_);
+  cricket::VideoChannel* video_channel_ptr = video_channel.get();
+  video_channels_.push_back(std::move(video_channel));
+  return video_channel_ptr;
+}
+void ChannelManager::DestroyVideoChannel(cricket::VideoChannel* channel) {
+  TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  video_channels_.erase(absl::c_find_if(
+      video_channels_, [&](const auto& p) { return p.get() == channel; }));
+}
+void ChannelManager::DestroyChannel(cricket::ChannelInterface* channel) {
+  RTC_DCHECK(channel);
+  if (!worker_thread_->IsCurrent()) {
+    // TODO(tommi): Do this asynchronously when we have a way to make sure that
+    // the call to DestroyChannel runs before ~Call() runs, which today happens
+    // inside an Invoke from the signaling thread in PeerConnectin::Close().
+    worker_thread_->BlockingCall([&] { DestroyChannel(channel); });
+    return;
+  }
+  if (channel->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+    DestroyVoiceChannel(static_cast<cricket::VoiceChannel*>(channel));
+  } else {
+    RTC_DCHECK_EQ(channel->media_type(), cricket::MEDIA_TYPE_VIDEO);
+    DestroyVideoChannel(static_cast<cricket::VideoChannel*>(channel));
+  }
+}
+}  // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/ChannelManager.h

@@ -0,0 +1,95 @@
+#ifndef TGCALLS_CHANNEL_MANAGER_H_
+#define TGCALLS_CHANNEL_MANAGER_H_
+#include <stdint.h>
+#include <memory>
+#include <string>
+#include <vector>
+#include "api/audio_options.h"
+#include "api/crypto/crypto_options.h"
+#include "api/rtp_parameters.h"
+#include "api/video/video_bitrate_allocator_factory.h"
+#include "call/call.h"
+#include "media/base/codec.h"
+#include "media/base/media_channel.h"
+#include "media/base/media_config.h"
+#include "media/base/media_engine.h"
+#include "pc/channel.h"
+#include "pc/channel_interface.h"
+#include "pc/rtp_transport_internal.h"
+#include "pc/session_description.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/unique_id_generator.h"
+namespace tgcalls {
+// ChannelManager allows the MediaEngine to run on a separate thread, and takes
+// care of marshalling calls between threads. It also creates and keeps track of
+// voice and video channels; by doing so, it can temporarily pause all the
+// channels when a new audio or video device is chosen. The voice and video
+// channels are stored in separate vectors, to easily allow operations on just
+// voice or just video channels.
+// ChannelManager also allows the application to discover what devices it has
+// using device manager.
+class ChannelManager {
+ public:
+  // Returns an initialized instance of ChannelManager.
+  // If media_engine is non-nullptr, then the returned ChannelManager instance
+  // will own that reference and media engine initialization
+  static std::unique_ptr<ChannelManager> Create(
+      std::unique_ptr<cricket::MediaEngineInterface> media_engine,
+      rtc::Thread* worker_thread,
+      rtc::Thread* network_thread);
+  ChannelManager() = delete;
+  ~ChannelManager();
+  rtc::Thread* worker_thread() const { return worker_thread_; }
+  rtc::Thread* network_thread() const { return network_thread_; }
+  cricket::MediaEngineInterface* media_engine() { return media_engine_.get(); }
+  rtc::UniqueRandomIdGenerator& ssrc_generator() { return ssrc_generator_; }
+  // The operations below all occur on the worker thread.
+  // ChannelManager retains ownership of the created channels, so clients should
+  // call the appropriate Destroy*Channel method when done.
+  // Creates a voice channel, to be associated with the specified session.
+  cricket::VoiceChannel* CreateVoiceChannel(webrtc::Call* call,
+                                   const cricket::MediaConfig& media_config,
+                                   const std::string& mid,
+                                   bool srtp_required,
+                                   const webrtc::CryptoOptions& crypto_options,
+                                   const cricket::AudioOptions& options);
+  // Creates a video channel, synced with the specified voice channel, and
+  // associated with the specified session.
+  // Version of the above that takes PacketTransportInternal.
+  cricket::VideoChannel* CreateVideoChannel(
+      webrtc::Call* call,
+      const cricket::MediaConfig& media_config,
+      const std::string& mid,
+      bool srtp_required,
+      const webrtc::CryptoOptions& crypto_options,
+      const cricket::VideoOptions& options,
+      webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory);
+  void DestroyChannel(cricket::ChannelInterface* channel);
+ protected:
+  ChannelManager(std::unique_ptr<cricket::MediaEngineInterface> media_engine,
+                 rtc::Thread* worker_thread,
+                 rtc::Thread* network_thread);
+  // Destroys a voice channel created by CreateVoiceChannel.
+  void DestroyVoiceChannel(cricket::VoiceChannel* voice_channel);
+  // Destroys a video channel created by CreateVideoChannel.
+  void DestroyVideoChannel(cricket::VideoChannel* video_channel);
+ private:
+  const std::unique_ptr<cricket::MediaEngineInterface> media_engine_;  // Nullable.
+  rtc::Thread* const signaling_thread_;
+  rtc::Thread* const worker_thread_;
+  rtc::Thread* const network_thread_;
+  // This object should be used to generate any SSRC that is not explicitly
+  // specified by the user (or by the remote party).
+  // TODO(bugs.webrtc.org/12666): This variable is used from both the signaling
+  // and worker threads. See if we can't restrict usage to a single thread.
+  rtc::UniqueRandomIdGenerator ssrc_generator_;
+  // Vector contents are non-null.
+  std::vector<std::unique_ptr<cricket::VoiceChannel>> voice_channels_
+      RTC_GUARDED_BY(worker_thread_);
+  std::vector<std::unique_ptr<cricket::VideoChannel>> video_channels_
+      RTC_GUARDED_BY(worker_thread_);
+};
+}  // namespace tgcalls
+#endif  // TGCALLS_CHANNEL_MANAGER_H_

Telegram/ThirdParty/tgcalls/tgcalls/CodecSelectHelper.cpp

@@ -0,0 +1,303 @@
+#include "CodecSelectHelper.h"
+
+#include "platform/PlatformInterface.h"
+
+#include "media/base/media_constants.h"
+#include "media/base/codec.h"
+#include "absl/strings/match.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace tgcalls {
+namespace {
+
+using VideoFormat = webrtc::SdpVideoFormat;
+
+bool CompareFormats(const VideoFormat &a, const VideoFormat &b) {
+	if (a.name < b.name) {
+		return true;
+	} else if (b.name < a.name) {
+		return false;
+	} else {
+		return a.parameters < b.parameters;
+	}
+}
+
+int FormatPriority(const VideoFormat &format, const std::vector<std::string> &preferredCodecs) {
+	static const auto kCodecs = {
+		std::string(cricket::kAv1CodecName),
+#ifndef WEBRTC_DISABLE_H265
+		std::string(cricket::kH265CodecName),
+#endif
+		std::string(cricket::kH264CodecName),
+		std::string(cricket::kVp8CodecName),
+        std::string(cricket::kVp9CodecName),
+	};
+	static const auto kSupported = [] {
+		const auto platform = PlatformInterface::SharedInstance();
+
+		auto result = std::vector<std::string>();
+		result.reserve(kCodecs.size());
+		for (const auto &codec : kCodecs) {
+			if (platform->supportsEncoding(codec)) {
+				result.push_back(codec);
+			}
+		}
+		return result;
+	}();
+
+    for (int i = 0; i < preferredCodecs.size(); i++) {
+        for (const auto &name : kSupported) {
+            if (absl::EqualsIgnoreCase(format.name, preferredCodecs[i]) && absl::EqualsIgnoreCase(format.name, name)) {
+                return i;
+            }
+        }
+    }
+
+    auto result = (int)preferredCodecs.size();
+	for (const auto &name : kSupported) {
+		if (absl::EqualsIgnoreCase(format.name, name)) {
+			return result;
+		}
+		++result;
+	}
+	return -1;
+}
+
+bool ComparePriorities(const VideoFormat &a, const VideoFormat &b, const std::vector<std::string> &preferredCodecs) {
+	return FormatPriority(a, preferredCodecs) < FormatPriority(b, preferredCodecs);
+}
+
+std::vector<VideoFormat> FilterAndSortEncoders(std::vector<VideoFormat> list, const std::vector<std::string> &preferredCodecs) {
+	const auto listBegin = begin(list);
+	const auto listEnd = end(list);
+    std::sort(listBegin, listEnd, [&preferredCodecs](const VideoFormat &lhs, const VideoFormat &rhs) {
+        return ComparePriorities(lhs, rhs, preferredCodecs);
+    });
+	auto eraseFrom = listBegin;
+	auto eraseTill = eraseFrom;
+	while (eraseTill != listEnd && FormatPriority(*eraseTill, preferredCodecs) == -1) {
+		++eraseTill;
+	}
+	if (eraseTill != eraseFrom) {
+		list.erase(eraseFrom, eraseTill);
+	}
+	return list;
+}
+
+std::vector<VideoFormat> AppendUnique(
+		std::vector<VideoFormat> list,
+		std::vector<VideoFormat> other) {
+	if (list.empty()) {
+		return other;
+	}
+	list.reserve(list.size() + other.size());
+	const auto oldBegin = &list[0];
+	const auto oldEnd = oldBegin + list.size();
+	for (auto &format : other) {
+		if (std::find(oldBegin, oldEnd, format) == oldEnd) {
+			list.push_back(std::move(format));
+		}
+	}
+	return list;
+}
+
+std::vector<VideoFormat>::const_iterator FindEqualFormat(
+		const std::vector<VideoFormat> &list,
+		const VideoFormat &format) {
+	return std::find_if(list.begin(), list.end(), [&](const VideoFormat &other) {
+        return format.IsSameCodec(other);
+	});
+}
+
+void AddDefaultFeedbackParams(cricket::VideoCodec *codec) {
+	// Don't add any feedback params for RED and ULPFEC.
+	if (codec->name == cricket::kRedCodecName || codec->name == cricket::kUlpfecCodecName)
+		return;
+	codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamRemb, cricket::kParamValueEmpty));
+	codec->AddFeedbackParam(
+		cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
+	// Don't add any more feedback params for FLEXFEC.
+	if (codec->name == cricket::kFlexfecCodecName)
+		return;
+	codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir));
+	codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
+	codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamNack, cricket::kRtcpFbNackParamPli));
+	if (codec->name == cricket::kVp8CodecName &&
+		webrtc::field_trial::IsEnabled("WebRTC-RtcpLossNotification")) {
+		codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamLntf, cricket::kParamValueEmpty));
+	}
+}
+
+std::vector<VideoFormat> RemoveScalabilityModes(
+		std::vector<VideoFormat> list) {
+	auto changed = false;
+	for (auto &entry : list) {
+		if (!entry.scalability_modes.empty()) {
+			entry.scalability_modes = {};
+			changed = true;
+		}
+	}
+	if (changed && list.size() > 1) {
+		for (auto i = list.end() - 1; i != list.begin(); --i) {
+			if (std::find(list.begin(), i, *i) != i) {
+				i = list.erase(i);
+			}
+		}
+	}
+	return list;
+}
+
+} // namespace
+
+VideoFormatsMessage ComposeSupportedFormats(
+		std::vector<VideoFormat> encoders,
+		std::vector<VideoFormat> decoders,
+        const std::vector<std::string> &preferredCodecs) {
+	// We don't pass scalability_modes through signaling,
+	// So we have to remove them here, otherwise lists are different.
+	encoders = RemoveScalabilityModes(std::move(encoders));
+	decoders = RemoveScalabilityModes(std::move(decoders));
+
+	encoders = FilterAndSortEncoders(std::move(encoders), preferredCodecs);
+
+	auto result = VideoFormatsMessage();
+	result.encodersCount = (int)encoders.size();
+	result.formats = AppendUnique(std::move(encoders), std::move(decoders));
+	for (const auto &format : result.formats) {
+		RTC_LOG(LS_INFO) << "Format: " << format.ToString();
+	}
+	RTC_LOG(LS_INFO) << "First " << result.encodersCount << " formats are supported encoders.";
+	return result;
+}
+
+CommonFormats ComputeCommonFormats(
+		const VideoFormatsMessage &my,
+		VideoFormatsMessage their) {
+	assert(my.encodersCount <= my.formats.size());
+	assert(their.encodersCount <= their.formats.size());
+
+	for (const auto &format : their.formats) {
+		RTC_LOG(LS_INFO) << "Their format: " << format.ToString();
+	}
+	RTC_LOG(LS_INFO) << "Their first " << their.encodersCount << " formats are supported encoders.";
+
+	const auto myEncodersBegin = begin(my.formats);
+	const auto myEncodersEnd = myEncodersBegin + my.encodersCount;
+	const auto theirEncodersBegin = begin(their.formats);
+	const auto theirEncodersEnd = theirEncodersBegin + their.encodersCount;
+
+	auto result = CommonFormats();
+	const auto addUnique = [&](const VideoFormat &format) {
+		const auto already = std::find(
+			result.list.begin(),
+			result.list.end(),
+			format);
+		if (already == result.list.end()) {
+			result.list.push_back(format);
+		}
+	};
+	const auto addCommonAndFindFirst = [&](
+			std::vector<VideoFormat>::const_iterator begin,
+			std::vector<VideoFormat>::const_iterator end,
+			const std::vector<VideoFormat> &decoders) {
+		auto first = VideoFormat(std::string());
+		for (auto i = begin; i != end; ++i) {
+			const auto &format = *i;
+			const auto j = FindEqualFormat(decoders, format);
+			if (j != decoders.end()) {
+				if (first.name.empty()) {
+					first = format;
+				}
+				addUnique(format);
+				addUnique(*j);
+			};
+		}
+		return first;
+	};
+
+	result.list.reserve(my.formats.size() + their.formats.size());
+	auto myEncoderFormat = addCommonAndFindFirst(
+		myEncodersBegin,
+		myEncodersEnd,
+		their.formats);
+	auto theirEncoderFormat = addCommonAndFindFirst(
+		theirEncodersBegin,
+		theirEncodersEnd,
+		my.formats);
+	std::sort(begin(result.list), end(result.list), CompareFormats);
+	if (!myEncoderFormat.name.empty()) {
+		const auto i = std::find(begin(result.list), end(result.list), myEncoderFormat);
+		assert(i != end(result.list));
+		result.myEncoderIndex = (i - begin(result.list));
+	}
+
+	for (const auto &format : result.list) {
+		RTC_LOG(LS_INFO) << "Common format: " << format.ToString();
+	}
+	RTC_LOG(LS_INFO) << "My encoder: " << (result.myEncoderIndex >= 0 ? result.list[result.myEncoderIndex].ToString() : "(null)");
+	RTC_LOG(LS_INFO) << "Their encoder: " << (!theirEncoderFormat.name.empty() ? theirEncoderFormat.ToString() : "(null)");
+
+	return result;
+}
+
+CommonCodecs AssignPayloadTypesAndDefaultCodecs(CommonFormats &&formats) {
+	if (formats.list.empty()) {
+		return CommonCodecs();
+	}
+
+	constexpr int kFirstDynamicPayloadType = 96;
+	constexpr int kLastDynamicPayloadType = 127;
+
+	int payload_type = kFirstDynamicPayloadType;
+
+	formats.list.push_back(webrtc::SdpVideoFormat(cricket::kRedCodecName));
+	formats.list.push_back(webrtc::SdpVideoFormat(cricket::kUlpfecCodecName));
+
+	if (true) {
+		webrtc::SdpVideoFormat flexfec_format(cricket::kFlexfecCodecName);
+		// This value is currently arbitrarily set to 10 seconds. (The unit
+		// is microseconds.) This parameter MUST be present in the SDP, but
+		// we never use the actual value anywhere in our code however.
+		// TODO(brandtr): Consider honouring this value in the sender and receiver.
+		flexfec_format.parameters = { {cricket::kFlexfecFmtpRepairWindow, "10000000"} };
+		formats.list.push_back(flexfec_format);
+	}
+
+	auto inputIndex = 0;
+	auto result = CommonCodecs();
+	result.list.reserve(2 * formats.list.size() - 2);
+	for (const auto &format : formats.list) {
+        cricket::VideoCodec codec = cricket::CreateVideoCodec(format);
+		codec.id = payload_type;
+		AddDefaultFeedbackParams(&codec);
+
+		if (inputIndex++ == formats.myEncoderIndex) {
+			result.myEncoderIndex = (int)result.list.size();
+		}
+		result.list.push_back(codec);
+
+		// Increment payload type.
+		++payload_type;
+		if (payload_type > kLastDynamicPayloadType) {
+			RTC_LOG(LS_ERROR) << "Out of dynamic payload types, skipping the rest.";
+			break;
+		}
+
+		// Add associated RTX codec for non-FEC codecs.
+		if (!absl::EqualsIgnoreCase(codec.name, cricket::kUlpfecCodecName) &&
+			!absl::EqualsIgnoreCase(codec.name, cricket::kFlexfecCodecName)) {
+			result.list.push_back(cricket::CreateVideoRtxCodec(payload_type, codec.id));
+
+			// Increment payload type.
+			++payload_type;
+			if (payload_type > kLastDynamicPayloadType) {
+				RTC_LOG(LS_ERROR) << "Out of dynamic payload types, skipping the rest.";
+				break;
+			}
+		}
+	}
+	return result;
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/CodecSelectHelper.h

@@ -0,0 +1,32 @@
+#ifndef TGCALLS_CODEC_SELECT_HELPER_H
+#define TGCALLS_CODEC_SELECT_HELPER_H
+
+#include "Message.h"
+#include "media/base/codec.h"
+
+namespace tgcalls {
+
+struct CommonFormats {
+	std::vector<webrtc::SdpVideoFormat> list;
+	int myEncoderIndex = -1;
+};
+
+struct CommonCodecs {
+	std::vector<cricket::VideoCodec> list;
+	int myEncoderIndex = -1;
+};
+
+VideoFormatsMessage ComposeSupportedFormats(
+	std::vector<webrtc::SdpVideoFormat> encoders,
+	std::vector<webrtc::SdpVideoFormat> decoders,
+    const std::vector<std::string> &preferredCodecs);
+
+CommonFormats ComputeCommonFormats(
+	const VideoFormatsMessage &my,
+	VideoFormatsMessage theirs);
+
+CommonCodecs AssignPayloadTypesAndDefaultCodecs(CommonFormats &&formats);
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/CryptoHelper.cpp

@@ -0,0 +1,61 @@
+#include "CryptoHelper.h"
+
+#include <cstring>
+#include <limits.h>
+
+namespace tgcalls {
+
+AesKeyIv PrepareAesKeyIv(const uint8_t *key, const uint8_t *msgKey, int x) {
+	auto result = AesKeyIv();
+
+	const auto sha256a = ConcatSHA256(
+		MemorySpan{ msgKey, 16 },
+		MemorySpan{ key + x, 36 });
+	const auto sha256b = ConcatSHA256(
+		MemorySpan{ key + 40 + x, 36 },
+		MemorySpan{ msgKey, 16 });
+	const auto aesKey = result.key.data();
+	const auto aesIv = result.iv.data();
+	memcpy(aesKey, sha256a.data(), 8);
+	memcpy(aesKey + 8, sha256b.data() + 8, 16);
+	memcpy(aesKey + 8 + 16, sha256a.data() + 24, 8);
+	memcpy(aesIv, sha256b.data(), 4);
+	memcpy(aesIv + 4, sha256a.data() + 8, 8);
+	memcpy(aesIv + 4 + 8, sha256b.data() + 24, 4);
+
+	return result;
+}
+
+void AesProcessCtr(MemorySpan from, void *to, AesKeyIv &&aesKeyIv) {
+	auto aes = AES_KEY();
+	AES_set_encrypt_key(
+		reinterpret_cast<const unsigned char*>(aesKeyIv.key.data()),
+		aesKeyIv.key.size() * CHAR_BIT,
+		&aes);
+
+	unsigned char ecountBuf[16] = { 0 };
+	unsigned int offsetInBlock = 0;
+
+#ifdef OPENSSL_IS_BORINGSSL
+	AES_ctr128_encrypt(
+			reinterpret_cast<const unsigned char*>(from.data),
+			reinterpret_cast<unsigned char*>(to),
+			from.size,
+			&aes,
+			reinterpret_cast<unsigned char*>(aesKeyIv.iv.data()),
+			ecountBuf,
+			&offsetInBlock);
+#else
+	CRYPTO_ctr128_encrypt(
+		reinterpret_cast<const unsigned char*>(from.data),
+		reinterpret_cast<unsigned char*>(to),
+		from.size,
+		&aes,
+		reinterpret_cast<unsigned char*>(aesKeyIv.iv.data()),
+		ecountBuf,
+		&offsetInBlock,
+		block128_f(AES_encrypt));
+#endif
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/CryptoHelper.h

@@ -0,0 +1,68 @@
+#ifndef TGCALLS_CRYPTO_HELPER_H
+#define TGCALLS_CRYPTO_HELPER_H
+
+extern "C" {
+#include <openssl/sha.h>
+#include <openssl/aes.h>
+#ifndef OPENSSL_IS_BORINGSSL
+#include <openssl/modes.h>
+#endif
+#include <openssl/rand.h>
+#include <openssl/crypto.h>
+} // extern "C"
+
+#include <array>
+
+namespace tgcalls {
+
+struct MemorySpan {
+	MemorySpan(const void *data, size_t size) :
+		data(data),
+		size(size) {
+	}
+
+	const void *data = nullptr;
+	size_t size = 0;
+};
+
+struct AesKeyIv {
+	std::array<uint8_t, 32> key;
+	std::array<uint8_t, 16> iv;
+};
+
+constexpr auto kSha256Size = size_t(SHA256_DIGEST_LENGTH);
+
+template <typename ...Parts>
+void SHA256Update(SHA256_CTX*, Parts &&...parts);
+
+inline void SHA256Update(SHA256_CTX*) {
+}
+
+template <typename First, typename ...Others>
+inline void SHA256Update(SHA256_CTX *context, First &&span, Others &&...others) {
+	static_assert(
+		std::is_same<std::decay_t<First>, MemorySpan>::value,
+		"Pass some MemorySpan-s here.");
+
+	SHA256_Update(context, span.data, span.size);
+	SHA256Update(context, std::forward<Others>(others)...);
+}
+
+template <typename ...Parts>
+inline std::array<uint8_t, kSha256Size> ConcatSHA256(Parts &&... parts) {
+	static_assert(sizeof...(parts) > 0, "empty list");
+
+	auto result = std::array<uint8_t, kSha256Size>();
+	auto context = SHA256_CTX();
+	SHA256_Init(&context);
+	SHA256Update(&context, std::forward<Parts>(parts)...);
+	SHA256_Final(result.data(), &context);
+	return result;
+}
+
+AesKeyIv PrepareAesKeyIv(const uint8_t *key, const uint8_t *msgKey, int x);
+void AesProcessCtr(MemorySpan from, void *to, AesKeyIv &&aesKeyIv);
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/DirectConnectionChannel.h

@@ -0,0 +1,21 @@
+#ifndef TGCALLS_DIRECT_CONNECTION_CHANNEL_H
+#define TGCALLS_DIRECT_CONNECTION_CHANNEL_H
+
+#include <functional>
+#include <memory>
+#include <vector>
+
+namespace tgcalls {
+
+class DirectConnectionChannel {
+public:
+    virtual ~DirectConnectionChannel() = default;
+
+    virtual std::vector<uint8_t> addOnIncomingPacket(std::function<void(std::shared_ptr<std::vector<uint8_t>>)> &&) = 0;
+    virtual void removeOnIncomingPacket(std::vector<uint8_t> &token) = 0;
+    virtual void sendPacket(std::unique_ptr<std::vector<uint8_t>> &&packet) = 0;
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/EncryptedConnection.cpp

@@ -0,0 +1,754 @@
+#include "EncryptedConnection.h"
+
+#include "CryptoHelper.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/byte_buffer.h"
+#include "rtc_base/time_utils.h"
+
+namespace tgcalls {
+namespace {
+
+constexpr auto kSingleMessagePacketSeqBit = (uint32_t(1) << 31);
+constexpr auto kMessageRequiresAckSeqBit = (uint32_t(1) << 30);
+constexpr auto kMaxAllowedCounter = std::numeric_limits<uint32_t>::max()
+    & ~kSingleMessagePacketSeqBit
+    & ~kMessageRequiresAckSeqBit;
+
+static_assert(kMaxAllowedCounter < kSingleMessagePacketSeqBit, "bad");
+static_assert(kMaxAllowedCounter < kMessageRequiresAckSeqBit, "bad");
+
+constexpr auto kAckSerializedSize = sizeof(uint32_t) + sizeof(uint8_t);
+constexpr auto kNotAckedMessagesLimit = 64 * 1024;
+constexpr auto kMaxIncomingPacketSize = 128 * 1024; // don't try decrypting more
+constexpr auto kKeepIncomingCountersCount = 64;
+constexpr auto kMaxFullPacketSize = 1500; // IP_PACKET_SIZE from webrtc.
+
+// Max seen turn_overhead is around 36.
+constexpr auto kMaxOuterPacketSize = kMaxFullPacketSize - 48;
+
+constexpr auto kMaxSignalingPacketSize = 16 * 1024;
+
+constexpr auto kServiceCauseAcks = 1;
+constexpr auto kServiceCauseResend = 2;
+
+static constexpr uint8_t kAckId = uint8_t(-1);
+static constexpr uint8_t kEmptyId = uint8_t(-2);
+static constexpr uint8_t kCustomId = uint8_t(127);
+
+void AppendSeq(rtc::CopyOnWriteBuffer &buffer, uint32_t seq) {
+    const auto bytes = rtc::HostToNetwork32(seq);
+    buffer.AppendData(reinterpret_cast<const char*>(&bytes), sizeof(bytes));
+}
+
+void WriteSeq(void *bytes, uint32_t seq) {
+    *reinterpret_cast<uint32_t*>(bytes) = rtc::HostToNetwork32(seq);
+}
+
+uint32_t ReadSeq(const void *bytes) {
+    return rtc::NetworkToHost32(*reinterpret_cast<const uint32_t*>(bytes));
+}
+
+uint32_t CounterFromSeq(uint32_t seq) {
+    return seq & ~kSingleMessagePacketSeqBit & ~kMessageRequiresAckSeqBit;
+}
+
+absl::nullopt_t LogError(
+        const char *message,
+        const std::string &additional = std::string()) {
+    RTC_LOG(LS_ERROR) << "ERROR! " << message << additional;
+    return absl::nullopt;
+}
+
+bool ConstTimeIsDifferent(const void *a, const void *b, size_t size) {
+    auto ca = reinterpret_cast<const char*>(a);
+    auto cb = reinterpret_cast<const char*>(b);
+    volatile auto different = false;
+    for (const auto ce = ca + size; ca != ce; ++ca, ++cb) {
+        different = different | (*ca != *cb);
+    }
+    return different;
+}
+
+rtc::CopyOnWriteBuffer SerializeRawMessageWithSeq(
+        const rtc::CopyOnWriteBuffer &message,
+        uint32_t seq,
+        bool singleMessagePacket) {
+    rtc::ByteBufferWriter writer;
+    writer.WriteUInt32(seq);
+    writer.WriteUInt8(kCustomId);
+    writer.WriteUInt32((uint32_t)message.size());
+    writer.WriteBytes((const uint8_t *)message.data(), message.size());
+
+    auto result = rtc::CopyOnWriteBuffer();
+    result.AppendData(writer.Data(), writer.Length());
+
+    return result;
+}
+
+} // namespace
+
+EncryptedConnection::EncryptedConnection(
+    Type type,
+    const EncryptionKey &key,
+    std::function<void(int delayMs, int cause)> requestSendService) :
+_type(type),
+_key(key),
+_delayIntervals(DelayIntervalsByType(type)),
+_requestSendService(std::move(requestSendService)) {
+    assert(_key.value != nullptr);
+}
+
+absl::optional<rtc::CopyOnWriteBuffer> EncryptedConnection::encryptRawPacket(rtc::CopyOnWriteBuffer const &buffer) {
+    auto seq = ++_counter;
+
+    rtc::ByteBufferWriter writer;
+    writer.WriteUInt32(seq);
+
+    auto result = rtc::CopyOnWriteBuffer();
+    result.AppendData(writer.Data(), writer.Length());
+
+    result.AppendData(buffer);
+
+    auto encryptedPacket = encryptPrepared(result);
+
+    rtc::CopyOnWriteBuffer encryptedBuffer;
+    encryptedBuffer.AppendData(encryptedPacket.bytes.data(), encryptedPacket.bytes.size());
+    return encryptedBuffer;
+}
+
+absl::optional<rtc::CopyOnWriteBuffer> EncryptedConnection::decryptRawPacket(rtc::CopyOnWriteBuffer const &buffer) {
+    if (buffer.size() < 21 || buffer.size() > kMaxIncomingPacketSize) {
+        return absl::nullopt;
+    }
+
+    const auto x = (_key.isOutgoing ? 8 : 0) + (_type == Type::Signaling ? 128 : 0);
+    const auto key = _key.value->data();
+    const auto msgKey = reinterpret_cast<const uint8_t*>(buffer.data());
+    const auto encryptedData = msgKey + 16;
+    const auto dataSize = buffer.size() - 16;
+
+    auto aesKeyIv = PrepareAesKeyIv(key, msgKey, x);
+
+    auto decryptionBuffer = rtc::Buffer(dataSize);
+    AesProcessCtr(
+        MemorySpan{ encryptedData, dataSize },
+        decryptionBuffer.data(),
+        std::move(aesKeyIv));
+
+    const auto msgKeyLarge = ConcatSHA256(
+        MemorySpan{ key + 88 + x, 32 },
+        MemorySpan{ decryptionBuffer.data(), decryptionBuffer.size() });
+    if (ConstTimeIsDifferent(msgKeyLarge.data() + 8, msgKey, 16)) {
+        return absl::nullopt;
+    }
+
+    const auto incomingSeq = ReadSeq(decryptionBuffer.data());
+    const auto incomingCounter = CounterFromSeq(incomingSeq);
+    if (!registerIncomingCounter(incomingCounter)) {
+        // We've received that packet already.
+        return absl::nullopt;
+    }
+
+    rtc::CopyOnWriteBuffer resultBuffer;
+    resultBuffer.AppendData(decryptionBuffer.data() + 4, decryptionBuffer.size() - 4);
+    return resultBuffer;
+}
+
+auto EncryptedConnection::prepareForSending(const Message &message)
+-> absl::optional<EncryptedPacket> {
+    const auto messageRequiresAck = absl::visit([](const auto &data) {
+        return std::decay_t<decltype(data)>::kRequiresAck;
+    }, message.data);
+    
+    // If message requires ack, then we can't serialize it as a single
+    // message packet, because later it may be sent as a part of big packet.
+    const auto singleMessagePacket = !haveAdditionalMessages() && !messageRequiresAck;
+    const auto maybeSeq = computeNextSeq(messageRequiresAck, singleMessagePacket);
+    if (!maybeSeq) {
+        return absl::nullopt;
+    }
+    const auto seq = *maybeSeq;
+    auto serialized = SerializeMessageWithSeq(message, seq, singleMessagePacket);
+    
+    return prepareForSendingMessageInternal(serialized, seq, messageRequiresAck);
+}
+
+absl::optional<EncryptedConnection::EncryptedPacket> EncryptedConnection::prepareForSendingRawMessage(rtc::CopyOnWriteBuffer &message, bool messageRequiresAck) {
+    // If message requires ack, then we can't serialize it as a single
+    // message packet, because later it may be sent as a part of big packet.
+    const auto singleMessagePacket = !haveAdditionalMessages() && !messageRequiresAck;
+    const auto maybeSeq = computeNextSeq(messageRequiresAck, singleMessagePacket);
+    if (!maybeSeq) {
+        return absl::nullopt;
+    }
+    const auto seq = *maybeSeq;
+    auto serialized = SerializeRawMessageWithSeq(message, seq, singleMessagePacket);
+    
+    return prepareForSendingMessageInternal(serialized, seq, messageRequiresAck);
+}
+    
+absl::optional<EncryptedConnection::EncryptedPacket> EncryptedConnection::prepareForSendingMessageInternal(rtc::CopyOnWriteBuffer &serialized, uint32_t seq, bool messageRequiresAck) {
+    if (!enoughSpaceInPacket(serialized, 0)) {
+        return LogError("Too large packet: ", std::to_string(serialized.size()));
+    }
+    const auto notYetAckedCopy = messageRequiresAck
+        ? serialized
+        : rtc::CopyOnWriteBuffer();
+    if (!messageRequiresAck) {
+        appendAdditionalMessages(serialized);
+        return encryptPrepared(serialized);
+    }
+    const auto type = uint8_t(serialized.cdata()[4]);
+    const auto sendEnqueued = !_myNotYetAckedMessages.empty();
+    if (sendEnqueued) {
+        // All requiring ack messages should always be sent in order within
+        // one packet, starting with the least not-yet-acked one.
+        // So if we still have those, we send an empty message with all
+        // requiring ack messages that will fit in correct order.
+        RTC_LOG(LS_INFO) << logHeader()
+            << "Enqueue SEND:type" << type << "#" << CounterFromSeq(seq);
+    } else {
+        RTC_LOG(LS_INFO) << logHeader()
+            << "Add SEND:type" << type << "#" << CounterFromSeq(seq);
+        appendAdditionalMessages(serialized);
+    }
+    _myNotYetAckedMessages.push_back({ notYetAckedCopy, rtc::TimeMillis() });
+    if (!sendEnqueued) {
+        return encryptPrepared(serialized);
+    }
+    for (auto &queued : _myNotYetAckedMessages) {
+        queued.lastSent = 0;
+    }
+    return prepareForSendingService(0);
+}
+
+auto EncryptedConnection::prepareForSendingService(int cause)
+-> absl::optional<EncryptedPacket> {
+    if (cause == kServiceCauseAcks) {
+        _sendAcksTimerActive = false;
+    } else if (cause == kServiceCauseResend) {
+        _resendTimerActive = false;
+    }
+    if (!haveAdditionalMessages()) {
+        return absl::nullopt;
+    }
+    const auto messageRequiresAck = false;
+    const auto singleMessagePacket = false;
+    const auto seq = computeNextSeq(messageRequiresAck, singleMessagePacket);
+    if (!seq) {
+        return absl::nullopt;
+    }
+    auto serialized = SerializeEmptyMessageWithSeq(*seq);
+    assert(enoughSpaceInPacket(serialized, 0));
+
+    RTC_LOG(LS_INFO) << logHeader()
+        << "SEND:empty#" << CounterFromSeq(*seq);
+
+    appendAdditionalMessages(serialized);
+    return encryptPrepared(serialized);
+}
+
+bool EncryptedConnection::haveAdditionalMessages() const {
+    return !_myNotYetAckedMessages.empty() || !_acksToSendSeqs.empty();
+}
+
+absl::optional<uint32_t> EncryptedConnection::computeNextSeq(
+        bool messageRequiresAck,
+        bool singleMessagePacket) {
+    if (messageRequiresAck && _myNotYetAckedMessages.size() >= kNotAckedMessagesLimit) {
+        return LogError("Too many not ACKed messages.");
+    } else if (_counter == kMaxAllowedCounter) {
+        return LogError("Outgoing packet limit reached.");
+    }
+
+    return (++_counter)
+        | (singleMessagePacket ? kSingleMessagePacketSeqBit : 0)
+        | (messageRequiresAck ? kMessageRequiresAckSeqBit : 0);
+}
+
+size_t EncryptedConnection::packetLimit() const {
+    switch (_type) {
+        case Type::Signaling:
+            return kMaxSignalingPacketSize;
+        default:
+            return kMaxOuterPacketSize;
+    }
+}
+
+bool EncryptedConnection::enoughSpaceInPacket(const rtc::CopyOnWriteBuffer &buffer, size_t amount) const {
+    const auto limit = packetLimit();
+    return (amount < limit)
+        && (16 + buffer.size() + amount <= limit);
+}
+
+void EncryptedConnection::appendAcksToSend(rtc::CopyOnWriteBuffer &buffer) {
+    auto i = _acksToSendSeqs.begin();
+    while ((i != _acksToSendSeqs.end())
+        && enoughSpaceInPacket(
+            buffer,
+            kAckSerializedSize)) {
+
+        RTC_LOG(LS_INFO) << logHeader()
+            << "Add ACK#" << CounterFromSeq(*i);
+
+        AppendSeq(buffer, *i);
+        buffer.AppendData(&kAckId, 1);
+        ++i;
+    }
+    _acksToSendSeqs.erase(_acksToSendSeqs.begin(), i);
+    for (const auto seq : _acksToSendSeqs) {
+        RTC_LOG(LS_INFO) << logHeader()
+            << "Skip ACK#" << CounterFromSeq(seq)
+            << " (no space, length: " << kAckSerializedSize << ", already: " << buffer.size() << ")";
+    }
+}
+
+size_t EncryptedConnection::fullNotAckedLength() const {
+    assert(_myNotYetAckedMessages.size() < kNotAckedMessagesLimit);
+
+    auto result = size_t();
+    for (const auto &message : _myNotYetAckedMessages) {
+        result += message.data.size();
+    }
+    return result;
+}
+
+void EncryptedConnection::appendAdditionalMessages(rtc::CopyOnWriteBuffer &buffer) {
+    appendAcksToSend(buffer);
+
+    if (_myNotYetAckedMessages.empty()) {
+        return;
+    }
+
+    const auto now = rtc::TimeMillis();
+    for (auto &resending : _myNotYetAckedMessages) {
+        const auto sent = resending.lastSent;
+        const auto when = sent
+            ? (sent + _delayIntervals.minDelayBeforeMessageResend)
+            : 0;
+
+        assert(resending.data.size() >= 5);
+        const auto counter = CounterFromSeq(ReadSeq(resending.data.data()));
+        const auto type = uint8_t(resending.data.data()[4]);
+        if (when > now) {
+            RTC_LOG(LS_INFO) << logHeader()
+                << "Skip RESEND:type" << type << "#" << counter
+                << " (wait " << (when - now) << "ms).";
+            break;
+        } else if (enoughSpaceInPacket(buffer, resending.data.size())) {
+            RTC_LOG(LS_INFO) << logHeader()
+                << "Add RESEND:type" << type << "#" << counter;
+            buffer.AppendData(resending.data);
+            resending.lastSent = now;
+        } else {
+            RTC_LOG(LS_INFO) << logHeader()
+                << "Skip RESEND:type" << type << "#" << counter
+                << " (no space, length: " << resending.data.size() << ", already: " << buffer.size() << ")";
+            break;
+        }
+    }
+    if (!_resendTimerActive) {
+        _resendTimerActive = true;
+        _requestSendService(
+            _delayIntervals.maxDelayBeforeMessageResend,
+            kServiceCauseResend);
+    }
+}
+
+auto EncryptedConnection::encryptPrepared(const rtc::CopyOnWriteBuffer &buffer)
+-> EncryptedPacket {
+    auto result = EncryptedPacket();
+    result.counter = CounterFromSeq(ReadSeq(buffer.data()));
+    result.bytes.resize(16 + buffer.size());
+
+    const auto x = (_key.isOutgoing ? 0 : 8) + (_type == Type::Signaling ? 128 : 0);
+    const auto key = _key.value->data();
+
+    const auto msgKeyLarge = ConcatSHA256(
+        MemorySpan{ key + 88 + x, 32 },
+        MemorySpan{ buffer.data(), buffer.size() });
+    const auto msgKey = result.bytes.data();
+    memcpy(msgKey, msgKeyLarge.data() + 8, 16);
+
+    auto aesKeyIv = PrepareAesKeyIv(key, msgKey, x);
+
+    AesProcessCtr(
+        MemorySpan{ buffer.data(), buffer.size() },
+        result.bytes.data() + 16,
+        std::move(aesKeyIv));
+
+    return result;
+}
+
+bool EncryptedConnection::registerIncomingCounter(uint32_t incomingCounter) {
+    auto &list = _largestIncomingCounters;
+
+    const auto position = std::lower_bound(list.begin(), list.end(), incomingCounter);
+    const auto largest = list.empty() ? 0 : list.back();
+    if (position != list.end() && *position == incomingCounter) {
+        // The packet is in the list already.
+        return false;
+    } else if (incomingCounter + kKeepIncomingCountersCount <= largest) {
+        // The packet is too old.
+        return false;
+    }
+    const auto eraseTill = std::find_if(list.begin(), list.end(), [&](uint32_t counter) {
+        return (counter + kKeepIncomingCountersCount > incomingCounter);
+    });
+    const auto eraseCount = eraseTill - list.begin();
+    const auto positionIndex = (position - list.begin()) - eraseCount;
+    list.erase(list.begin(), eraseTill);
+
+    assert(positionIndex >= 0 && positionIndex <= list.size());
+    list.insert(list.begin() + positionIndex, incomingCounter);
+    return true;
+}
+
+auto EncryptedConnection::handleIncomingPacket(const char *bytes, size_t size)
+-> absl::optional<DecryptedPacket> {
+    if (size < 21 || size > kMaxIncomingPacketSize) {
+        return LogError("Bad incoming packet size: ", std::to_string(size));
+    }
+
+    const auto x = (_key.isOutgoing ? 8 : 0) + (_type == Type::Signaling ? 128 : 0);
+    const auto key = _key.value->data();
+    const auto msgKey = reinterpret_cast<const uint8_t*>(bytes);
+    const auto encryptedData = msgKey + 16;
+    const auto dataSize = size - 16;
+
+    auto aesKeyIv = PrepareAesKeyIv(key, msgKey, x);
+
+    auto decryptionBuffer = rtc::Buffer(dataSize);
+    AesProcessCtr(
+        MemorySpan{ encryptedData, dataSize },
+        decryptionBuffer.data(),
+        std::move(aesKeyIv));
+
+    const auto msgKeyLarge = ConcatSHA256(
+        MemorySpan{ key + 88 + x, 32 },
+        MemorySpan{ decryptionBuffer.data(), decryptionBuffer.size() });
+    if (ConstTimeIsDifferent(msgKeyLarge.data() + 8, msgKey, 16)) {
+        return LogError("Bad incoming data hash.");
+    }
+
+    const auto incomingSeq = ReadSeq(decryptionBuffer.data());
+    const auto incomingCounter = CounterFromSeq(incomingSeq);
+    if (!registerIncomingCounter(incomingCounter)) {
+        // We've received that packet already.
+        return LogError("Already handled packet received.", std::to_string(incomingCounter));
+    }
+    return processPacket(decryptionBuffer, incomingSeq);
+}
+
+absl::optional<EncryptedConnection::DecryptedRawPacket> EncryptedConnection::handleIncomingRawPacket(const char *bytes, size_t size) {
+    if (size < 21 || size > kMaxIncomingPacketSize) {
+        return LogError("Bad incoming packet size: ", std::to_string(size));
+    }
+
+    const auto x = (_key.isOutgoing ? 8 : 0) + (_type == Type::Signaling ? 128 : 0);
+    const auto key = _key.value->data();
+    const auto msgKey = reinterpret_cast<const uint8_t*>(bytes);
+    const auto encryptedData = msgKey + 16;
+    const auto dataSize = size - 16;
+
+    auto aesKeyIv = PrepareAesKeyIv(key, msgKey, x);
+
+    auto decryptionBuffer = rtc::Buffer(dataSize);
+    AesProcessCtr(
+        MemorySpan{ encryptedData, dataSize },
+        decryptionBuffer.data(),
+        std::move(aesKeyIv));
+
+    const auto msgKeyLarge = ConcatSHA256(
+        MemorySpan{ key + 88 + x, 32 },
+        MemorySpan{ decryptionBuffer.data(), decryptionBuffer.size() });
+    if (ConstTimeIsDifferent(msgKeyLarge.data() + 8, msgKey, 16)) {
+        return LogError("Bad incoming data hash.");
+    }
+
+    const auto incomingSeq = ReadSeq(decryptionBuffer.data());
+    const auto incomingCounter = CounterFromSeq(incomingSeq);
+    if (!registerIncomingCounter(incomingCounter)) {
+        // We've received that packet already.
+        return LogError("Already handled packet received.", std::to_string(incomingCounter));
+    }
+    return processRawPacket(decryptionBuffer, incomingSeq);
+}
+
+auto EncryptedConnection::processPacket(
+    const rtc::Buffer &fullBuffer,
+    uint32_t packetSeq)
+-> absl::optional<DecryptedPacket> {
+    assert(fullBuffer.size() >= 5);
+
+    auto additionalMessage = false;
+    auto firstMessageRequiringAck = true;
+    auto newRequiringAckReceived = false;
+
+    auto currentSeq = packetSeq;
+    auto currentCounter = CounterFromSeq(currentSeq);
+    rtc::ByteBufferReader reader(rtc::ArrayView<const uint8_t>(
+        reinterpret_cast<const uint8_t *>(fullBuffer.data() + 4), // Skip seq.
+        fullBuffer.size() - 4));
+
+    auto result = absl::optional<DecryptedPacket>();
+    while (true) {
+        const auto type = uint8_t(*reader.Data());
+        const auto singleMessagePacket = ((currentSeq & kSingleMessagePacketSeqBit) != 0);
+        if (singleMessagePacket && additionalMessage) {
+            return LogError("Single message packet bit in not first message.");
+        }
+
+        if (type == kEmptyId) {
+            if (additionalMessage) {
+                return LogError("Empty message should be only the first one in the packet.");
+            }
+            RTC_LOG(LS_INFO) << logHeader()
+                << "Got RECV:empty" << "#" << currentCounter;
+            reader.Consume(1);
+        } else if (type == kAckId) {
+            if (!additionalMessage) {
+                return LogError("Ack message must not be the first one in the packet.");
+            }
+            ackMyMessage(currentSeq);
+            reader.Consume(1);
+        } else if (auto message = DeserializeMessage(reader, singleMessagePacket)) {
+            const auto messageRequiresAck = ((currentSeq & kMessageRequiresAckSeqBit) != 0);
+            const auto skipMessage = messageRequiresAck
+                ? !registerSentAck(currentCounter, firstMessageRequiringAck)
+                : (additionalMessage && !registerIncomingCounter(currentCounter));
+            if (messageRequiresAck) {
+                firstMessageRequiringAck = false;
+                if (!skipMessage) {
+                    newRequiringAckReceived = true;
+                }
+                sendAckPostponed(currentSeq);
+                RTC_LOG(LS_INFO) << logHeader()
+                    << (skipMessage ? "Repeated RECV:type" : "Got RECV:type") << type << "#" << currentCounter;
+            }
+            if (!skipMessage) {
+                appendReceivedMessage(result, std::move(*message), currentSeq);
+            }
+        } else {
+            return LogError("Could not parse message from packet, type: ", std::to_string(type));
+        }
+        if (!reader.Length()) {
+            break;
+        } else if (singleMessagePacket) {
+            return LogError("Single message didn't fill the entire packet.");
+        } else if (reader.Length() < 5) {
+            return LogError("Bad remaining data size: ", std::to_string(reader.Length()));
+        }
+        const auto success = reader.ReadUInt32(&currentSeq);
+        assert(success);
+        (void)success;
+        currentCounter = CounterFromSeq(currentSeq);
+
+        additionalMessage = true;
+    }
+
+    if (!_acksToSendSeqs.empty()) {
+        if (newRequiringAckReceived) {
+            _requestSendService(0, 0);
+        } else if (!_sendAcksTimerActive) {
+            _sendAcksTimerActive = true;
+            _requestSendService(
+                _delayIntervals.maxDelayBeforeAckResend,
+                kServiceCauseAcks);
+        }
+    }
+
+    return result;
+}
+
+auto EncryptedConnection::processRawPacket(
+    const rtc::Buffer &fullBuffer,
+    uint32_t packetSeq)
+-> absl::optional<DecryptedRawPacket> {
+    assert(fullBuffer.size() >= 5);
+
+    auto additionalMessage = false;
+    auto firstMessageRequiringAck = true;
+    auto newRequiringAckReceived = false;
+
+    auto currentSeq = packetSeq;
+    auto currentCounter = CounterFromSeq(currentSeq);
+    rtc::ByteBufferReader reader(rtc::ArrayView<const uint8_t>(
+        reinterpret_cast<const uint8_t *>(fullBuffer.data() + 4), // Skip seq.
+        fullBuffer.size() - 4));
+
+    auto result = absl::optional<DecryptedRawPacket>();
+    while (true) {
+        const auto type = uint8_t(*reader.Data());
+        const auto singleMessagePacket = ((currentSeq & kSingleMessagePacketSeqBit) != 0);
+        if (singleMessagePacket && additionalMessage) {
+            return LogError("Single message packet bit in not first message.");
+        }
+
+        if (type == kEmptyId) {
+            if (additionalMessage) {
+                return LogError("Empty message should be only the first one in the packet.");
+            }
+            RTC_LOG(LS_INFO) << logHeader()
+                << "Got RECV:empty" << "#" << currentCounter;
+            reader.Consume(1);
+        } else if (type == kAckId) {
+            if (!additionalMessage) {
+                return LogError("Ack message must not be the first one in the packet.");
+            }
+            ackMyMessage(currentSeq);
+            reader.Consume(1);
+        } else if (type == kCustomId) {
+            reader.Consume(1);
+            
+            if (auto message = DeserializeRawMessage(reader, singleMessagePacket)) {
+                const auto messageRequiresAck = ((currentSeq & kMessageRequiresAckSeqBit) != 0);
+                const auto skipMessage = messageRequiresAck
+                    ? !registerSentAck(currentCounter, firstMessageRequiringAck)
+                    : (additionalMessage && !registerIncomingCounter(currentCounter));
+                if (messageRequiresAck) {
+                    firstMessageRequiringAck = false;
+                    if (!skipMessage) {
+                        newRequiringAckReceived = true;
+                    }
+                    sendAckPostponed(currentSeq);
+                    RTC_LOG(LS_INFO) << logHeader()
+                        << (skipMessage ? "Repeated RECV:type" : "Got RECV:type") << type << "#" << currentCounter;
+                }
+                if (!skipMessage) {
+                    appendReceivedRawMessage(result, std::move(*message), currentSeq);
+                }
+            } else {
+                return LogError("Could not parse message from packet, type: ", std::to_string(type));
+            }
+        } else {
+            return LogError("Could not parse message from packet, type: ", std::to_string(type));
+        }
+        if (!reader.Length()) {
+            break;
+        } else if (singleMessagePacket) {
+            return LogError("Single message didn't fill the entire packet.");
+        } else if (reader.Length() < 5) {
+            return LogError("Bad remaining data size: ", std::to_string(reader.Length()));
+        }
+        const auto success = reader.ReadUInt32(&currentSeq);
+        assert(success);
+        (void)success;
+        currentCounter = CounterFromSeq(currentSeq);
+
+        additionalMessage = true;
+    }
+
+    if (!_acksToSendSeqs.empty()) {
+        if (newRequiringAckReceived) {
+            _requestSendService(0, 0);
+        } else if (!_sendAcksTimerActive) {
+            _sendAcksTimerActive = true;
+            _requestSendService(
+                _delayIntervals.maxDelayBeforeAckResend,
+                kServiceCauseAcks);
+        }
+    }
+
+    return result;
+}
+
+void EncryptedConnection::appendReceivedMessage(
+        absl::optional<DecryptedPacket> &to,
+        Message &&message,
+        uint32_t incomingSeq) {
+    auto decrypted = DecryptedMessage{
+        std::move(message),
+        CounterFromSeq(incomingSeq)
+    };
+    if (to) {
+        to->additional.push_back(std::move(decrypted));
+    } else {
+        to = DecryptedPacket{ std::move(decrypted) };
+    }
+}
+
+void EncryptedConnection::appendReceivedRawMessage(
+        absl::optional<DecryptedRawPacket> &to,
+        rtc::CopyOnWriteBuffer &&message,
+        uint32_t incomingSeq) {
+    auto decrypted = DecryptedRawMessage{
+        std::move(message),
+        CounterFromSeq(incomingSeq)
+    };
+    if (to) {
+        to->additional.push_back(std::move(decrypted));
+    } else {
+        to = DecryptedRawPacket{ std::move(decrypted) };
+    }
+}
+
+const char *EncryptedConnection::logHeader() const {
+    return (_type == Type::Signaling) ? "(signaling) " : "(transport) ";
+}
+
+bool EncryptedConnection::registerSentAck(uint32_t counter, bool firstInPacket) {
+    auto &list = _acksSentCounters;
+
+    const auto position = std::lower_bound(list.begin(), list.end(), counter);
+    const auto already = (position != list.end()) && (*position == counter);
+
+    const auto was = list;
+    if (firstInPacket) {
+        list.erase(list.begin(), position);
+        if (!already) {
+            list.insert(list.begin(), counter);
+        }
+    } else if (!already) {
+        list.insert(position, counter);
+    }
+    return !already;
+}
+
+void EncryptedConnection::sendAckPostponed(uint32_t incomingSeq) {
+    auto &list = _acksToSendSeqs;
+    const auto already = std::find(list.begin(), list.end(), incomingSeq);
+    if (already == list.end()) {
+        list.push_back(incomingSeq);
+    }
+}
+
+void EncryptedConnection::ackMyMessage(uint32_t seq) {
+    auto type = uint8_t(0);
+    auto &list = _myNotYetAckedMessages;
+    for (auto i = list.begin(), e = list.end(); i != e; ++i) {
+        assert(i->data.size() >= 5);
+        if (ReadSeq(i->data.cdata()) == seq) {
+            type = uint8_t(i->data.cdata()[4]);
+            list.erase(i);
+            break;
+        }
+    }
+    RTC_LOG(LS_INFO) << logHeader()
+        << (type ? "Got ACK:type" + std::to_string(type) + "#" : "Repeated ACK#")
+        << CounterFromSeq(seq);
+}
+
+auto EncryptedConnection::DelayIntervalsByType(Type type) -> DelayIntervals {
+    auto result = DelayIntervals();
+    const auto signaling = (type == Type::Signaling);
+
+    // Don't resend faster than min delay even if we have a packet we can attach to.
+    result.minDelayBeforeMessageResend = signaling ? 3000 : 300;
+
+    // When max delay elapsed we resend anyway, in a dedicated packet.
+    result.maxDelayBeforeMessageResend = signaling ? 5000 : 1000;
+    result.maxDelayBeforeAckResend = signaling ? 5000 : 1000;
+
+    return result;
+}
+
+rtc::CopyOnWriteBuffer EncryptedConnection::SerializeEmptyMessageWithSeq(uint32_t seq) {
+    auto result = rtc::CopyOnWriteBuffer(5);
+    auto bytes = result.MutableData();
+    WriteSeq(bytes, seq);
+    bytes[4] = kEmptyId;
+    return result;
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/EncryptedConnection.h

@@ -0,0 +1,104 @@
+#ifndef TGCALLS_ENCRYPTED_CONNECTION_H
+#define TGCALLS_ENCRYPTED_CONNECTION_H
+
+#include "Instance.h"
+#include "Message.h"
+
+namespace rtc {
+class ByteBufferReader;
+} // namespace rtc
+
+namespace tgcalls {
+
+class EncryptedConnection final {
+public:
+    enum class Type : uint8_t {
+        Signaling,
+        Transport,
+    };
+    EncryptedConnection(
+        Type type,
+        const EncryptionKey &key,
+        std::function<void(int delayMs, int cause)> requestSendService);
+
+    struct EncryptedPacket {
+        std::vector<uint8_t> bytes;
+        uint32_t counter = 0;
+    };
+    absl::optional<EncryptedPacket> prepareForSending(const Message &message);
+    absl::optional<EncryptedPacket> prepareForSendingRawMessage(rtc::CopyOnWriteBuffer &serialized, bool messageRequiresAck);
+    absl::optional<EncryptedPacket> prepareForSendingService(int cause);
+
+    struct DecryptedPacket {
+        DecryptedMessage main;
+        std::vector<DecryptedMessage> additional;
+    };
+    struct DecryptedRawPacket {
+        DecryptedRawMessage main;
+        std::vector<DecryptedRawMessage> additional;
+    };
+    absl::optional<DecryptedPacket> handleIncomingPacket(const char *bytes, size_t size);
+    absl::optional<DecryptedRawPacket> handleIncomingRawPacket(const char *bytes, size_t size);
+
+    absl::optional<rtc::CopyOnWriteBuffer> encryptRawPacket(rtc::CopyOnWriteBuffer const &buffer);
+    absl::optional<rtc::CopyOnWriteBuffer> decryptRawPacket(rtc::CopyOnWriteBuffer const &buffer);
+
+private:
+    struct DelayIntervals {
+        // In milliseconds.
+        int minDelayBeforeMessageResend = 0;
+        int maxDelayBeforeMessageResend = 0;
+        int maxDelayBeforeAckResend = 0;
+    };
+    struct MessageForResend {
+        rtc::CopyOnWriteBuffer data;
+        int64_t lastSent = 0;
+    };
+
+    bool enoughSpaceInPacket(const rtc::CopyOnWriteBuffer &buffer, size_t amount) const;
+    size_t packetLimit() const;
+    size_t fullNotAckedLength() const;
+    void appendAcksToSend(rtc::CopyOnWriteBuffer &buffer);
+    void appendAdditionalMessages(rtc::CopyOnWriteBuffer &buffer);
+    EncryptedPacket encryptPrepared(const rtc::CopyOnWriteBuffer &buffer);
+    bool registerIncomingCounter(uint32_t incomingCounter);
+    absl::optional<DecryptedPacket> processPacket(const rtc::Buffer &fullBuffer, uint32_t packetSeq);
+    absl::optional<DecryptedRawPacket> processRawPacket(const rtc::Buffer &fullBuffer, uint32_t packetSeq);
+    bool registerSentAck(uint32_t counter, bool firstInPacket);
+    void ackMyMessage(uint32_t counter);
+    void sendAckPostponed(uint32_t incomingSeq);
+    bool haveAdditionalMessages() const;
+    absl::optional<uint32_t> computeNextSeq(bool messageRequiresAck, bool singleMessagePacket);
+    void appendReceivedMessage(
+        absl::optional<DecryptedPacket> &to,
+        Message &&message,
+        uint32_t incomingSeq);
+    void appendReceivedRawMessage(
+        absl::optional<DecryptedRawPacket> &to,
+        rtc::CopyOnWriteBuffer &&message,
+        uint32_t incomingSeq);
+    absl::optional<EncryptedPacket> prepareForSendingMessageInternal(rtc::CopyOnWriteBuffer &serialized, uint32_t seq, bool messageRequiresAck);
+
+    const char *logHeader() const;
+
+    static DelayIntervals DelayIntervalsByType(Type type);
+    static rtc::CopyOnWriteBuffer SerializeEmptyMessageWithSeq(uint32_t seq);
+
+    Type _type = Type();
+    EncryptionKey _key;
+    uint32_t _counter = 0;
+    DelayIntervals _delayIntervals;
+    std::vector<uint32_t> _largestIncomingCounters;
+    std::vector<uint32_t> _ackedIncomingCounters;
+    std::vector<uint32_t> _acksToSendSeqs;
+    std::vector<uint32_t> _acksSentCounters;
+    std::vector<MessageForResend> _myNotYetAckedMessages;
+    std::function<void(int delayMs, int cause)> _requestSendService;
+    bool _resendTimerActive = false;
+    bool _sendAcksTimerActive = false;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/FakeAudioDeviceModule.cpp

@@ -0,0 +1,257 @@
+#include "FakeAudioDeviceModule.h"
+
+#include "modules/audio_device/include/audio_device_default.h"
+#include "rtc_base/ref_counted_object.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+
+#include <thread>
+#include <mutex>
+#include <condition_variable>
+
+namespace tgcalls {
+class FakeAudioDeviceModuleImpl : public webrtc::webrtc_impl::AudioDeviceModuleDefault<webrtc::AudioDeviceModule> {
+ public:
+  static webrtc::scoped_refptr<webrtc::AudioDeviceModule> Create(webrtc::TaskQueueFactory* taskQueueFactory,
+                                                              std::shared_ptr<FakeAudioDeviceModule::Renderer> renderer,
+                                                              std::shared_ptr<FakeAudioDeviceModule::Recorder> recorder,
+                                                              FakeAudioDeviceModule::Options options) {
+    return webrtc::scoped_refptr<webrtc::AudioDeviceModule>(
+        new rtc::RefCountedObject<FakeAudioDeviceModuleImpl>(taskQueueFactory, options, std::move(renderer), std::move(recorder)));
+  }
+
+  FakeAudioDeviceModuleImpl(webrtc::TaskQueueFactory*, FakeAudioDeviceModule::Options options,
+                            std::shared_ptr<FakeAudioDeviceModule::Renderer> renderer,
+                            std::shared_ptr<FakeAudioDeviceModule::Recorder> recorder)
+      : num_channels_{options.num_channels}, samples_per_sec_{options.samples_per_sec}, scheduler_(options.scheduler_),
+        renderer_(std::move(renderer)), recorder_(std::move(recorder)) {
+    if (!scheduler_) {
+      scheduler_ = [](auto f)  {
+        std::thread([f = std::move(f)]() {
+          while (true)  {
+            double wait = f();
+            if (wait < 0) {
+              return;
+            }
+            std::this_thread::sleep_for(std::chrono::microseconds (static_cast<int64_t>(wait * 1000000)));
+          }
+        }).detach();
+      };
+    }
+    RTC_CHECK(num_channels_ == 1 || num_channels_ == 2);
+    auto good_sample_rate = [](size_t sr) {
+      return sr == 8000 || sr == 16000 || sr == 32000 || sr == 44100 || sr == 48000;
+    };
+    RTC_CHECK(good_sample_rate(samples_per_sec_));
+    samples_per_frame_ = samples_per_sec_ / 100;
+    playout_buffer_.resize(samples_per_frame_ * 2 /* 2 in case stereo will be turned on later */, 0);
+  }
+
+  ~FakeAudioDeviceModuleImpl() override {
+    StopPlayout();
+  }
+
+  int32_t PlayoutIsAvailable(bool* available) override {
+    if (available) {
+      *available = true;
+    }
+    return 0;
+  }
+
+  int32_t StereoPlayoutIsAvailable(bool* available) const override {
+    if (available) {
+      *available = true;
+    }
+    return 0;
+  }
+  int32_t StereoPlayout(bool* enabled) const override {
+    if (enabled) {
+      *enabled = num_channels_ == 2;
+    }
+    return 0;
+  }
+  int32_t SetStereoPlayout(bool enable) override {
+    size_t new_num_channels = enable ? 2 : 1;
+    if (new_num_channels != num_channels_) {
+      return -1;
+    }
+    return 0;
+  }
+
+  int32_t Init() override {
+    return 0;
+  }
+
+  int32_t RegisterAudioCallback(webrtc::AudioTransport* callback) override {
+    std::unique_lock<std::mutex> lock(render_mutex_);
+    audio_callback_ = callback;
+    return 0;
+  }
+
+  int32_t StartPlayout() override {
+    std::unique_lock<std::mutex> lock(render_mutex_);
+    if (!renderer_) {
+      return 0;
+    }
+    if (rendering_) {
+      return 0;
+    }
+    need_rendering_ = true;
+    rendering_ = true;
+    scheduler_([this]{
+      return Render() / 1000000.0;
+    });
+    return 0;
+  }
+
+  int32_t StopPlayout() override {
+    if (!rendering_) {
+      return 0;
+    }
+
+    need_rendering_ = false;
+    std::unique_lock<std::mutex> lock(render_mutex_);
+    render_cond_.wait(lock, [this]{ return !rendering_; });
+
+    return 0;
+  }
+
+  bool Playing() const override {
+    return rendering_;
+  }
+
+  int32_t StartRecording() override {
+    std::unique_lock<std::mutex> lock(record_mutex_);
+    if (!recorder_) {
+      return 0;
+    }
+    if (recording_) {
+      return 0;
+    }
+    need_recording_ = true;
+    recording_ = true;
+    scheduler_([this]{
+      return Record() / 1000000.0;
+    });
+    return 0;
+  }
+ int32_t StopRecording() override {
+   if (!recording_) {
+     return 0;
+   }
+
+   need_recording_ = false;
+   std::unique_lock<std::mutex> lock(record_mutex_);
+   record_cond_.wait(lock, [this]{ return !recording_; });
+
+   return 0;
+ }
+  bool Recording() const override {
+    return recording_;
+  }
+
+private:
+
+  int32_t Render() {
+    std::unique_lock<std::mutex> lock(render_mutex_);
+    if (!need_rendering_) {
+      rendering_ = false;
+      render_cond_.notify_all();
+      return -1;
+    }
+
+    size_t samples_out = 0;
+    int64_t elapsed_time_ms = -1;
+    int64_t ntp_time_ms = -1;
+    size_t bytes_per_sample = 2 * num_channels_;
+
+    RTC_CHECK(audio_callback_);
+    if (renderer_) {
+      renderer_->BeginFrame(0);
+    }
+    audio_callback_->NeedMorePlayData(samples_per_frame_, bytes_per_sample, num_channels_, samples_per_sec_,
+                                      playout_buffer_.data(), samples_out, &elapsed_time_ms, &ntp_time_ms);
+    if (renderer_) {
+      renderer_->EndFrame();
+    }
+    if (samples_out != 0 && renderer_) {
+      AudioFrame frame;
+      frame.audio_samples = playout_buffer_.data();
+      frame.num_samples = samples_out;
+      frame.bytes_per_sample = bytes_per_sample;
+      frame.num_channels = num_channels_;
+      frame.samples_per_sec = samples_per_sec_;
+      frame.elapsed_time_ms = elapsed_time_ms;
+      frame.ntp_time_ms = ntp_time_ms;
+      renderer_->Render(frame);
+    }
+    int32_t wait_for_us = -1;
+    if (renderer_) {
+      wait_for_us = renderer_->WaitForUs();
+    }
+    return wait_for_us;
+  }
+
+  int32_t Record() {
+    std::unique_lock<std::mutex> lock(record_mutex_);
+    if (!need_recording_) {
+      recording_ = false;
+      record_cond_.notify_all();
+      return -1;
+    }
+
+    auto frame = recorder_->Record();
+    if (frame.num_samples != 0) {
+       uint32_t new_mic_level;
+       audio_callback_->RecordedDataIsAvailable(frame.audio_samples,
+         frame.num_samples, frame.bytes_per_sample, frame.num_channels,
+         frame.samples_per_sec, 0, 0, 0, false, new_mic_level);
+    }
+
+    int32_t wait_for_us = -1;
+    if (recorder_) {
+      wait_for_us = recorder_->WaitForUs();
+    }
+    return wait_for_us;
+  }
+
+  size_t num_channels_;
+  const uint32_t samples_per_sec_;
+  size_t samples_per_frame_{0};
+
+  std::function<void(FakeAudioDeviceModule::Task)> scheduler_;
+
+  mutable std::mutex render_mutex_;
+  std::atomic<bool> need_rendering_{false};
+  std::atomic<bool> rendering_{false};
+  std::condition_variable render_cond_;
+  std::unique_ptr<rtc::PlatformThread> renderThread_;
+
+  mutable std::mutex record_mutex_;
+  std::atomic<bool> need_recording_{false};
+  std::atomic<bool> recording_{false};
+  std::condition_variable record_cond_;
+  std::unique_ptr<rtc::PlatformThread> recordThread_;
+
+
+  webrtc::AudioTransport* audio_callback_{nullptr};
+  const std::shared_ptr<FakeAudioDeviceModule::Renderer> renderer_;
+  const std::shared_ptr<FakeAudioDeviceModule::Recorder> recorder_;
+  std::vector<int16_t> playout_buffer_;
+};
+
+std::function<webrtc::scoped_refptr<webrtc::AudioDeviceModule>(webrtc::TaskQueueFactory*)> FakeAudioDeviceModule::Creator(
+    std::shared_ptr<Renderer> renderer, std::shared_ptr<Recorder> recorder, Options options) {
+  bool is_renderer_empty = bool(renderer);
+  auto boxed_renderer = std::make_shared<std::shared_ptr<Renderer>>(std::move(renderer));
+  bool is_recorder_empty = bool(recorder);
+  auto boxed_recorder = std::make_shared<std::shared_ptr<Recorder>>(std::move(recorder));
+  return
+      [boxed_renderer = std::move(boxed_renderer), is_renderer_empty,
+       boxed_recorder = std::move(boxed_recorder), is_recorder_empty, options](webrtc::TaskQueueFactory* task_factory) {
+        RTC_CHECK(is_renderer_empty == bool(*boxed_renderer));  // call only once if renderer exists
+        RTC_CHECK(is_recorder_empty == bool(*boxed_recorder));  // call only once if recorder exists
+        return FakeAudioDeviceModuleImpl::Create(task_factory, std::move(*boxed_renderer), std::move(*boxed_recorder), options);
+      };
+}
+}  // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/FakeAudioDeviceModule.h

@@ -0,0 +1,54 @@
+#pragma once
+
+#include <functional>
+#include <memory>
+
+#include "AudioFrame.h"
+
+namespace webrtc {
+class AudioDeviceModule;
+class TaskQueueFactory;
+}  // namespace webrtc
+
+namespace webrtc {
+template <class T>
+class scoped_refptr;
+}
+
+namespace tgcalls {
+class FakeAudioDeviceModule {
+ public:
+  class Renderer {
+   public:
+    virtual ~Renderer() = default;
+    virtual bool Render(const AudioFrame &samples) = 0;
+    virtual void BeginFrame(double timestamp) {
+    }
+    virtual void AddFrameChannel(uint32_t ssrc, const tgcalls::AudioFrame &frame) {
+    }
+    virtual void EndFrame() {
+    }
+    virtual int32_t WaitForUs() {
+       return 10000;
+    }
+  };
+  class Recorder {
+  public:
+    virtual ~Recorder() = default;
+    virtual AudioFrame Record() = 0;
+    virtual int32_t WaitForUs() {
+      return 10000;
+    }
+  };
+  using Task = std::function<double()>;
+  struct Options {
+    uint32_t samples_per_sec{48000};
+    uint32_t num_channels{2};
+    std::function<void(Task)> scheduler_;
+  };
+  static std::function<webrtc::scoped_refptr<webrtc::AudioDeviceModule>(webrtc::TaskQueueFactory *)> Creator(
+      std::shared_ptr<Renderer> renderer,
+      std::shared_ptr<Recorder> recorder,
+      Options options);
+};
+}  // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/FakeVideoTrackSource.cpp

@@ -0,0 +1,173 @@
+#include "FakeVideoTrackSource.h"
+
+#include "api/video/i420_buffer.h"
+#include "media/base/video_broadcaster.h"
+#include "pc/video_track_source.h"
+
+#include "libyuv.h"
+
+#include <thread>
+
+namespace tgcalls {
+
+int WIDTH = 1280;
+int HEIGHT = 720;
+
+class ChessFrameSource : public FrameSource {
+public:
+  ChessFrameSource() {
+    int N = 100;
+    frames_.reserve(N);
+    for (int i = 0; i < N; i++) {
+      frames_.push_back(genFrame(i, N));
+    }
+  }
+  Info info() const override{
+    return Info{WIDTH, HEIGHT};
+  }
+//  webrtc::VideoFrame next_frame() override {
+//    i = (i + 1) % frames_.size();
+//    return frames_[i].frame;
+//  }
+  void next_frame_rgb0(char *buf, double *pts) override {
+    *pts = 0;
+    i = (i + 1) % frames_.size();
+    size_t size = WIDTH * HEIGHT * 4;
+    memcpy(buf, frames_[i].rbga.get(), size);
+  }
+
+private:
+  struct Frame {
+    webrtc::VideoFrame frame;
+    std::unique_ptr<std::uint8_t[]> rbga;
+  };
+  std::vector<Frame> frames_;
+  size_t i = 0;
+  Frame genFrame(int i, int n) {
+    int width = WIDTH;
+    int height = HEIGHT;
+    auto bytes_ptr = std::make_unique<std::uint8_t[]>(width * height * 4);
+    auto bytes = bytes_ptr.get();
+    auto set_rgb = [&](int x, int y, std::uint8_t r, std::uint8_t g, std::uint8_t b) {
+      auto dest = bytes + (x * width + y) * 4;
+      dest[0] = r;
+      dest[1] = g;
+      dest[2] = b;
+      dest[3] = 0;
+    };
+    auto angle = (double)i / n * M_PI;
+    auto co = cos(angle);
+    auto si = sin(angle);
+
+    for (int i = 0; i < height; i++) {
+      for (int j = 0; j < width; j++) {
+        double sx = (i - height / 2) * 20.0 / HEIGHT;
+        double sy = (j  - width / 2) * 20.0 / HEIGHT;
+
+        int x, y;
+        if (sx * sx + sy * sy < 10) {
+          x = int(floor(sx * co - sy * si));
+          y = int(floor(sx * si + sy * co));
+        } else {
+          x = int(floor(sx));
+          y = int(floor(sy));
+        }
+        std::uint8_t color = ((y & 1) ^ (x & 1)) * 255;
+        set_rgb(i, j, color, color, color);
+      }
+    }
+
+    webrtc::scoped_refptr<webrtc::I420Buffer> buffer = webrtc::I420Buffer::Create(width, height);
+
+    libyuv::RGBAToI420(bytes, width * 4, buffer->MutableDataY(), buffer->StrideY(), buffer->MutableDataU(),
+                       buffer->StrideU(), buffer->MutableDataV(), buffer->StrideV(), width, height);
+
+    return Frame{webrtc::VideoFrame::Builder().set_video_frame_buffer(buffer).build(), std::move(bytes_ptr)};
+  }
+
+};
+
+webrtc::VideoFrame FrameSource::next_frame() {
+  auto info = this->info();
+  auto height = info.height;
+  auto width = info.width;
+  auto bytes_ptr = std::make_unique<std::uint8_t[]>(width * height * 4);
+  double pts;
+  next_frame_rgb0(reinterpret_cast<char *>(bytes_ptr.get()), &pts);
+  webrtc::scoped_refptr<webrtc::I420Buffer> buffer = webrtc::I420Buffer::Create(width, height);
+  libyuv::ABGRToI420(bytes_ptr.get(), width * 4, buffer->MutableDataY(), buffer->StrideY(), buffer->MutableDataU(),
+                     buffer->StrideU(), buffer->MutableDataV(), buffer->StrideV(), width, height);
+  return webrtc::VideoFrame::Builder().set_timestamp_us(static_cast<int64_t>(pts * 1000000)).set_video_frame_buffer(buffer).build();
+}
+
+class FakeVideoSource : public rtc::VideoSourceInterface<webrtc::VideoFrame> {
+ public:
+  FakeVideoSource(std::unique_ptr<FrameSource> source) {
+    data_ = std::make_shared<Data>();
+    std::thread([data = data_, source = std::move(source)] {
+      std::uint32_t step = 0;
+      while (!data->flag_) {
+        step++;
+        std::this_thread::sleep_for(std::chrono::milliseconds(1000 / 30));
+        auto frame = source->next_frame();
+        frame.set_id(static_cast<std::uint16_t>(step));
+        frame.set_timestamp_us(rtc::TimeMicros());
+        data->broadcaster_.OnFrame(frame);
+      }
+    }).detach();
+  }
+  ~FakeVideoSource() {
+    data_->flag_ = true;
+  }
+  using VideoFrameT = webrtc::VideoFrame;
+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrameT> *sink, const rtc::VideoSinkWants &wants) override {
+    RTC_LOG(LS_WARNING) << "ADD";
+    data_->broadcaster_.AddOrUpdateSink(sink, wants);
+  }
+  // RemoveSink must guarantee that at the time the method returns,
+  // there is no current and no future calls to VideoSinkInterface::OnFrame.
+  void RemoveSink(rtc::VideoSinkInterface<VideoFrameT> *sink) override {
+    RTC_LOG(LS_WARNING) << "REMOVE";
+    data_->broadcaster_.RemoveSink(sink);
+  }
+
+ private:
+  struct Data {
+    std::atomic<bool> flag_;
+    rtc::VideoBroadcaster broadcaster_;
+  };
+  std::shared_ptr<Data> data_;
+};
+
+class FakeVideoTrackSourceImpl : public webrtc::VideoTrackSource {
+ public:
+  static webrtc::scoped_refptr<FakeVideoTrackSourceImpl> Create(std::unique_ptr<FrameSource> source) {
+    return webrtc::scoped_refptr<FakeVideoTrackSourceImpl>(new rtc::RefCountedObject<FakeVideoTrackSourceImpl>(std::move(source)));
+  }
+
+  explicit FakeVideoTrackSourceImpl(std::unique_ptr<FrameSource> source) : VideoTrackSource(false), source_(std::move(source)) {
+  }
+
+ protected:
+  FakeVideoSource source_;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *source() override {
+    return &source_;
+  }
+};
+
+std::function<webrtc::VideoTrackSourceInterface*()> FakeVideoTrackSource::create(std::unique_ptr<FrameSource> frame_source) {
+  auto source = FakeVideoTrackSourceImpl::Create(std::move(frame_source));
+  return [source] {
+    return source.get();
+  };
+}
+std::unique_ptr<FrameSource> FrameSource::chess(){
+  return std::make_unique<ChessFrameSource>();
+}
+
+void FrameSource::video_frame_to_rgb0(const webrtc::VideoFrame & src, char *dest){
+  auto buffer = src.video_frame_buffer()->ToI420();
+  libyuv::I420ToABGR(buffer->DataY(), buffer->StrideY(), buffer->DataU(),
+                     buffer->StrideU(), buffer->DataV(), buffer->StrideV( ), reinterpret_cast<uint8_t *>(dest), src.width() * 4,  src.width(), src.height());
+}
+}

Telegram/ThirdParty/tgcalls/tgcalls/FakeVideoTrackSource.h

@@ -0,0 +1,35 @@
+#pragma once
+
+#include <functional>
+#include <memory>
+#include <string>
+
+namespace webrtc {
+class VideoTrackSourceInterface;
+class VideoFrame;
+}
+
+namespace tgcalls {
+class FrameSource {
+public:
+  struct Info {
+    int32_t width;
+    int32_t height;
+  };
+
+  virtual ~FrameSource() = default;
+
+  virtual Info info() const = 0;
+  virtual webrtc::VideoFrame next_frame();
+  static void video_frame_to_rgb0(const webrtc::VideoFrame &src, char *dest);
+  virtual void next_frame_rgb0(char *buf, double *pt_in_seconds) = 0;
+
+  static std::unique_ptr<FrameSource> chess();
+  static std::unique_ptr<FrameSource> from_file(std::string path);
+};
+
+class FakeVideoTrackSource {
+ public:
+  static std::function<webrtc::VideoTrackSourceInterface*()> create(std::unique_ptr<FrameSource> source);
+};
+}

Telegram/ThirdParty/tgcalls/tgcalls/FieldTrialsConfig.cpp

@@ -0,0 +1,7 @@
+#include "FieldTrialsConfig.h"
+
+namespace tgcalls {
+
+webrtc::FieldTrialBasedConfig fieldTrialsBasedConfig;
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/FieldTrialsConfig.h

@@ -0,0 +1,12 @@
+#ifndef TGCALLS_FIELD_TRIALS_CONFIG_H
+#define TGCALLS_FIELD_TRIALS_CONFIG_H
+
+#include "api/transport/field_trial_based_config.h"
+
+namespace tgcalls {
+
+extern webrtc::FieldTrialBasedConfig fieldTrialsBasedConfig;
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/Instance.cpp

@@ -0,0 +1,66 @@
+#include "Instance.h"
+
+#include <algorithm>
+#include <stdarg.h>
+
+namespace tgcalls {
+namespace {
+
+std::function<void(std::string const &)> globalLoggingFunction;
+
+std::map<std::string, std::shared_ptr<Meta>> &MetaMap() {
+	static auto result = std::map<std::string, std::shared_ptr<Meta>>();
+	return result;
+}
+
+} // namespace
+
+std::vector<std::string> Meta::Versions() {
+	auto &map = MetaMap();
+	auto result = std::vector<std::string>();
+	result.reserve(map.size());
+	for (const auto &entry : map) {
+		result.push_back(entry.first);
+	}
+	return result;
+}
+
+int Meta::MaxLayer() {
+	auto result = 0;
+	for (const auto &entry : MetaMap()) {
+		result = std::max(result, entry.second->connectionMaxLayer());
+	}
+	return result;
+}
+
+std::unique_ptr<Instance> Meta::Create(
+		const std::string &version,
+		Descriptor &&descriptor) {
+	const auto i = MetaMap().find(version);
+
+	// Enforce correct protocol version.
+	if (version == "2.7.7") {
+		descriptor.config.protocolVersion = ProtocolVersion::V0;
+	} else if (version == "5.0.0") {
+		descriptor.config.protocolVersion = ProtocolVersion::V1;
+	}
+
+	return (i != MetaMap().end())
+		? i->second->construct(std::move(descriptor))
+		: nullptr;
+}
+
+void Meta::RegisterOne(std::shared_ptr<Meta> meta) {
+	if (meta) {
+		const auto versions = meta->versions();
+        for (auto &it : versions) {
+            MetaMap().emplace(it, meta);
+        }
+	}
+}
+
+void SetLoggingFunction(std::function<void(std::string const &)> loggingFunction) {
+	globalLoggingFunction = loggingFunction;
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/Instance.h

@@ -0,0 +1,292 @@
+#ifndef TGCALLS_INSTANCE_H
+#define TGCALLS_INSTANCE_H
+
+#include <functional>
+#include <vector>
+#include <string>
+#include <memory>
+#include <map>
+
+#include "Stats.h"
+#include "DirectConnectionChannel.h"
+
+namespace rtc {
+template <typename VideoFrameT>
+class VideoSinkInterface;
+} // namespace rtc
+
+namespace webrtc {
+class VideoFrame;
+class AudioDeviceModule;
+class TaskQueueFactory;
+template <class T>
+class scoped_refptr;
+} // namespace webrtc
+
+namespace tgcalls {
+
+class WrappedAudioDeviceModule;
+class VideoCaptureInterface;
+
+struct FilePath {
+#ifndef _WIN32
+	std::string data;
+#else
+	std::wstring data;
+#endif
+};
+
+struct Proxy {
+	std::string host;
+	uint16_t port = 0;
+	std::string login;
+	std::string password;
+};
+
+struct RtcServer {
+    uint8_t id = 0;
+	std::string host;
+	uint16_t port = 0;
+	std::string login;
+	std::string password;
+	bool isTurn = false;
+    bool isTcp = false;
+};
+
+enum class EndpointType {
+	Inet,
+	Lan,
+	UdpRelay,
+	TcpRelay
+};
+
+struct EndpointHost {
+	std::string ipv4;
+	std::string ipv6;
+};
+
+struct Endpoint {
+	int64_t endpointId = 0;
+	EndpointHost host;
+	uint16_t port = 0;
+	EndpointType type = EndpointType{};
+	unsigned char peerTag[16] = { 0 };
+};
+
+enum class ProtocolVersion {
+    V0,
+    V1 // Low-cost network negotiation
+};
+
+enum class NetworkType {
+	Unknown,
+	Gprs,
+	Edge,
+	ThirdGeneration,
+	Hspa,
+	Lte,
+	WiFi,
+	Ethernet,
+	OtherHighSpeed,
+	OtherLowSpeed,
+	OtherMobile,
+	Dialup
+};
+
+enum class DataSaving {
+	Never,
+	Mobile,
+	Always
+};
+
+struct PersistentState {
+	std::vector<uint8_t> value;
+};
+
+struct Config {
+	double initializationTimeout = 0.;
+	double receiveTimeout = 0.;
+	DataSaving dataSaving = DataSaving::Never;
+	bool enableP2P = false;
+    bool allowTCP = false;
+    bool enableStunMarking = false;
+	bool enableAEC = false;
+	bool enableNS = false;
+	bool enableAGC = false;
+	bool enableCallUpgrade = false;
+	bool enableVolumeControl = false;
+	FilePath logPath;
+    FilePath statsLogPath;
+	int maxApiLayer = 0;
+    bool enableHighBitrateVideo = false;
+    std::vector<std::string> preferredVideoCodecs;
+    ProtocolVersion protocolVersion = ProtocolVersion::V0;
+    std::string customParameters = "";
+};
+
+struct EncryptionKey {
+	static constexpr int kSize = 256;
+
+	std::shared_ptr<const std::array<uint8_t, kSize>> value;
+	bool isOutgoing = false;
+
+    EncryptionKey(
+		std::shared_ptr<const std::array<uint8_t, kSize>> const value,
+		bool isOutgoing)
+	: value(value), isOutgoing(isOutgoing) {
+    }
+};
+
+enum class State {
+	WaitInit,
+	WaitInitAck,
+	Established,
+	Failed,
+	Reconnecting
+};
+
+// Defined in VideoCaptureInterface.h
+enum class VideoState;
+
+enum class AudioState {
+	Muted,
+	Active,
+};
+
+struct TrafficStats {
+	uint64_t bytesSentWifi = 0;
+	uint64_t bytesReceivedWifi = 0;
+	uint64_t bytesSentMobile = 0;
+	uint64_t bytesReceivedMobile = 0;
+};
+
+struct FinalState {
+	PersistentState persistentState;
+	std::string debugLog;
+	TrafficStats trafficStats;
+    CallStats callStats;
+	bool isRatingSuggested = false;
+};
+
+struct MediaDevicesConfig {
+	std::string audioInputId;
+	std::string audioOutputId;
+	float inputVolume = 1.f;
+	float outputVolume = 1.f;
+};
+
+class Instance {
+protected:
+	Instance() = default;
+
+public:
+	virtual ~Instance() = default;
+
+	virtual void setNetworkType(NetworkType networkType) = 0;
+	virtual void setMuteMicrophone(bool muteMicrophone) = 0;
+	virtual void setAudioOutputGainControlEnabled(bool enabled) = 0;
+	virtual void setEchoCancellationStrength(int strength) = 0;
+
+	virtual bool supportsVideo() = 0;
+	virtual void setIncomingVideoOutput(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) = 0;
+
+	virtual void setAudioInputDevice(std::string id) = 0;
+	virtual void setAudioOutputDevice(std::string id) = 0;
+	virtual void setInputVolume(float level) = 0;
+	virtual void setOutputVolume(float level) = 0;
+	virtual void setAudioOutputDuckingEnabled(bool enabled) = 0;
+    virtual void addExternalAudioSamples(std::vector<uint8_t> &&samples) {
+    }
+
+    virtual void setIsLowBatteryLevel(bool isLowBatteryLevel) = 0;
+
+	virtual std::string getLastError() = 0;
+	virtual std::string getDebugInfo() = 0;
+	virtual int64_t getPreferredRelayId() = 0;
+	virtual TrafficStats getTrafficStats() = 0;
+	virtual PersistentState getPersistentState() = 0;
+
+	virtual void receiveSignalingData(const std::vector<uint8_t> &data) = 0;
+	virtual void setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture) = 0;
+	virtual void sendVideoDeviceUpdated() = 0;
+    virtual void setRequestedVideoAspect(float aspect) = 0;
+
+	virtual void stop(std::function<void(FinalState)> completion) = 0;
+
+};
+
+template <typename Implementation>
+bool Register();
+
+struct Descriptor {
+    std::string version;
+	Config config;
+	PersistentState persistentState;
+	std::vector<Endpoint> endpoints;
+	std::unique_ptr<Proxy> proxy;
+	std::vector<RtcServer> rtcServers;
+	NetworkType initialNetworkType = NetworkType();
+	EncryptionKey encryptionKey;
+	MediaDevicesConfig mediaDevicesConfig;
+	std::shared_ptr<VideoCaptureInterface> videoCapture;
+	std::function<void(State)> stateUpdated;
+	std::function<void(int)> signalBarsUpdated;
+    std::function<void(float)> audioLevelUpdated;
+    std::function<void(bool)> remoteBatteryLevelIsLowUpdated;
+	std::function<void(AudioState, VideoState)> remoteMediaStateUpdated;
+    std::function<void(float)> remotePrefferedAspectRatioUpdated;
+	std::function<void(const std::vector<uint8_t> &)> signalingDataEmitted;
+	std::function<webrtc::scoped_refptr<webrtc::AudioDeviceModule>(webrtc::TaskQueueFactory*)> createAudioDeviceModule;
+    std::function<webrtc::scoped_refptr<WrappedAudioDeviceModule>(webrtc::TaskQueueFactory*)> createWrappedAudioDeviceModule;
+    std::string initialInputDeviceId;
+    std::string initialOutputDeviceId;
+    std::shared_ptr<DirectConnectionChannel> directConnectionChannel;
+};
+
+class Meta {
+public:
+	virtual ~Meta() = default;
+
+	virtual std::unique_ptr<Instance> construct(Descriptor &&descriptor) = 0;
+	virtual int connectionMaxLayer() = 0;
+	virtual std::vector<std::string> versions() = 0;
+
+	static std::unique_ptr<Instance> Create(
+		const std::string &version,
+		Descriptor &&descriptor);
+	static std::vector<std::string> Versions();
+	static int MaxLayer();
+
+private:
+	template <typename Implementation>
+	friend bool Register();
+
+	template <typename Implementation>
+	static bool RegisterOne();
+	static void RegisterOne(std::shared_ptr<Meta> meta);
+
+};
+
+template <typename Implementation>
+bool Meta::RegisterOne() {
+	class MetaImpl final : public Meta {
+	public:
+		int connectionMaxLayer() override {
+			return Implementation::GetConnectionMaxLayer();
+		}
+		std::vector<std::string> versions() override {
+			return Implementation::GetVersions();
+		}
+		std::unique_ptr<Instance> construct(Descriptor &&descriptor) override {
+			return std::make_unique<Implementation>(std::move(descriptor));
+		}
+	};
+	RegisterOne(std::make_shared<MetaImpl>());
+	return true;
+}
+
+void SetLoggingFunction(std::function<void(std::string const &)> loggingFunction);
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/InstanceImpl.cpp

@@ -0,0 +1,202 @@
+#include "InstanceImpl.h"
+
+#include "LogSinkImpl.h"
+#include "Manager.h"
+#include "MediaManager.h"
+#include "VideoCaptureInterfaceImpl.h"
+#include "VideoCapturerInterface.h"
+
+namespace tgcalls {
+
+namespace {
+
+rtc::Thread *makeManagerThread() {
+	static std::unique_ptr<rtc::Thread> value = rtc::Thread::Create();
+	value->SetName("WebRTC-Manager", nullptr);
+	value->Start();
+	return value.get();
+}
+
+
+rtc::Thread *getManagerThread() {
+	static rtc::Thread *value = makeManagerThread();
+	return value;
+}
+
+} // namespace
+
+InstanceImpl::InstanceImpl(Descriptor &&descriptor)
+: _logSink(std::make_unique<LogSinkImpl>(descriptor.config.logPath)) {
+    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+    rtc::LogMessage::SetLogToStderr(false);
+	rtc::LogMessage::AddLogToStream(_logSink.get(), rtc::LS_INFO);
+
+    auto networkType = descriptor.initialNetworkType;
+
+	_manager.reset(new ThreadLocalObject<Manager>(getManagerThread(), [descriptor = std::move(descriptor)]() mutable {
+		return std::make_shared<Manager>(getManagerThread(), std::move(descriptor));
+	}));
+	_manager->perform([](Manager *manager) {
+		manager->start();
+	});
+
+    setNetworkType(networkType);
+}
+
+InstanceImpl::~InstanceImpl() {
+	rtc::LogMessage::RemoveLogToStream(_logSink.get());
+}
+
+void InstanceImpl::receiveSignalingData(const std::vector<uint8_t> &data) {
+	_manager->perform([data](Manager *manager) {
+		manager->receiveSignalingData(data);
+	});
+};
+
+void InstanceImpl::setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture) {
+    _manager->perform([videoCapture](Manager *manager) {
+        manager->setVideoCapture(videoCapture);
+    });
+}
+
+void InstanceImpl::sendVideoDeviceUpdated() {
+    _manager->perform([](Manager *manager) {
+        manager->sendVideoDeviceUpdated();
+    });
+}
+
+void InstanceImpl::setRequestedVideoAspect(float aspect) {
+    _manager->perform([aspect](Manager *manager) {
+        manager->setRequestedVideoAspect(aspect);
+    });
+}
+
+void InstanceImpl::setNetworkType(NetworkType networkType) {
+    bool isLowCostNetwork = false;
+    switch (networkType) {
+        case NetworkType::WiFi:
+        case NetworkType::Ethernet:
+            isLowCostNetwork = true;
+            break;
+        default:
+            break;
+    }
+
+    _manager->perform([isLowCostNetwork](Manager *manager) {
+        manager->setIsLocalNetworkLowCost(isLowCostNetwork);
+    });
+}
+
+void InstanceImpl::setMuteMicrophone(bool muteMicrophone) {
+	_manager->perform([muteMicrophone](Manager *manager) {
+		manager->setMuteOutgoingAudio(muteMicrophone);
+	});
+}
+
+void InstanceImpl::setIncomingVideoOutput(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+	_manager->perform([sink](Manager *manager) {
+		manager->setIncomingVideoOutput(sink);
+	});
+}
+
+void InstanceImpl::setAudioOutputGainControlEnabled(bool enabled) {
+}
+
+void InstanceImpl::setEchoCancellationStrength(int strength) {
+}
+
+void InstanceImpl::setAudioInputDevice(std::string id) {
+	_manager->perform([id](Manager *manager) {
+		manager->setAudioInputDevice(id);
+	});
+}
+
+void InstanceImpl::setAudioOutputDevice(std::string id) {
+	_manager->perform([id](Manager *manager) {
+		manager->setAudioOutputDevice(id);
+	});
+}
+
+void InstanceImpl::setInputVolume(float level) {
+	_manager->perform([level](Manager *manager) {
+		manager->setInputVolume(level);
+	});
+}
+
+void InstanceImpl::setOutputVolume(float level) {
+	_manager->perform([level](Manager *manager) {
+		manager->setOutputVolume(level);
+	});
+}
+
+void InstanceImpl::setAudioOutputDuckingEnabled(bool enabled) {
+	// TODO: not implemented
+}
+
+void InstanceImpl::addExternalAudioSamples(std::vector<uint8_t> &&samples) {
+    _manager->perform([samples = std::move(samples)](Manager *manager) mutable {
+        manager->addExternalAudioSamples(std::move(samples));
+    });
+}
+
+void InstanceImpl::setIsLowBatteryLevel(bool isLowBatteryLevel) {
+    _manager->perform([isLowBatteryLevel](Manager *manager) {
+        manager->setIsLowBatteryLevel(isLowBatteryLevel);
+    });
+}
+
+std::string InstanceImpl::getLastError() {
+	return "";  // TODO: not implemented
+}
+
+std::string InstanceImpl::getDebugInfo() {
+	return "";  // TODO: not implemented
+}
+
+int64_t InstanceImpl::getPreferredRelayId() {
+	return 0;  // we don't have endpoint ids
+}
+
+TrafficStats InstanceImpl::getTrafficStats() {
+	return TrafficStats{};  // TODO: not implemented
+}
+
+PersistentState InstanceImpl::getPersistentState() {
+	return PersistentState{};  // we dont't have such information
+}
+
+void InstanceImpl::stop(std::function<void(FinalState)> completion) {
+    RTC_LOG(LS_INFO) << "Stopping InstanceImpl";
+    
+    std::string debugLog = _logSink->result();
+
+    _manager->perform([completion, debugLog = std::move(debugLog)](Manager *manager) {
+        manager->getNetworkStats([completion, debugLog = std::move(debugLog)](TrafficStats stats, CallStats callStats) {
+            FinalState finalState;
+            finalState.debugLog = debugLog;
+            finalState.isRatingSuggested = false;
+            finalState.trafficStats = stats;
+            finalState.callStats = callStats;
+
+            completion(finalState);
+        });
+    });
+}
+
+int InstanceImpl::GetConnectionMaxLayer() {
+	return 92;
+}
+
+std::vector<std::string> InstanceImpl::GetVersions() {
+    std::vector<std::string> result;
+    result.push_back("2.7.7");
+    result.push_back("5.0.0");
+    return result;
+}
+
+template <>
+bool Register<InstanceImpl>() {
+	return Meta::RegisterOne<InstanceImpl>();
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/InstanceImpl.h

@@ -0,0 +1,56 @@
+#ifndef TGCALLS_INSTANCE_IMPL_H
+#define TGCALLS_INSTANCE_IMPL_H
+
+#include "Instance.h"
+
+namespace tgcalls {
+
+class LogSinkImpl;
+
+class Manager;
+template <typename T>
+class ThreadLocalObject;
+
+class InstanceImpl final : public Instance {
+public:
+	explicit InstanceImpl(Descriptor &&descriptor);
+	~InstanceImpl() override;
+
+	static int GetConnectionMaxLayer();
+	static std::vector<std::string> GetVersions();
+
+	void receiveSignalingData(const std::vector<uint8_t> &data) override;
+	void setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture) override;
+	void sendVideoDeviceUpdated() override;
+    void setRequestedVideoAspect(float aspect) override;
+	void setNetworkType(NetworkType networkType) override;
+	void setMuteMicrophone(bool muteMicrophone) override;
+	bool supportsVideo() override {
+		return true;
+	}
+	void setIncomingVideoOutput(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) override;
+	void setAudioOutputGainControlEnabled(bool enabled) override;
+	void setEchoCancellationStrength(int strength) override;
+	void setAudioInputDevice(std::string id) override;
+	void setAudioOutputDevice(std::string id) override;
+	void setInputVolume(float level) override;
+	void setOutputVolume(float level) override;
+	void setAudioOutputDuckingEnabled(bool enabled) override;
+    void addExternalAudioSamples(std::vector<uint8_t> &&samples) override;
+    void setIsLowBatteryLevel(bool isLowBatteryLevel) override;
+	std::string getLastError() override;
+	std::string getDebugInfo() override;
+	int64_t getPreferredRelayId() override;
+	TrafficStats getTrafficStats() override;
+	PersistentState getPersistentState() override;
+	void stop(std::function<void(FinalState)> completion) override;
+
+private:
+	std::unique_ptr<ThreadLocalObject<Manager>> _manager;
+	std::unique_ptr<LogSinkImpl> _logSink;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/LogSinkImpl.cpp

@@ -0,0 +1,71 @@
+#include "LogSinkImpl.h"
+
+#include "Instance.h"
+
+#ifdef WEBRTC_WIN
+#include "windows.h"
+#include <ctime>
+#else // WEBRTC_WIN
+#include <sys/time.h>
+#endif // WEBRTC_WIN
+
+namespace tgcalls {
+
+LogSinkImpl::LogSinkImpl(const FilePath &logPath) {
+	if (!logPath.data.empty()) {
+		_file.open(logPath.data);
+	}
+}
+
+void LogSinkImpl::OnLogMessage(const std::string &msg, rtc::LoggingSeverity severity, const char *tag) {
+	OnLogMessage(std::string(tag) + ": " + msg);
+}
+
+void LogSinkImpl::OnLogMessage(const std::string &message, rtc::LoggingSeverity severity) {
+	OnLogMessage(message);
+}
+
+void LogSinkImpl::OnLogMessage(const std::string &message) {
+	time_t rawTime;
+	time(&rawTime);
+	struct tm timeinfo;
+
+#ifdef WEBRTC_WIN
+	localtime_s(&timeinfo, &rawTime);
+
+	FILETIME ft;
+	unsigned __int64 full = 0;
+	GetSystemTimeAsFileTime(&ft);
+
+	full |= ft.dwHighDateTime;
+	full <<= 32;
+	full |= ft.dwLowDateTime;
+
+	const auto deltaEpochInMicrosecs = 11644473600000000Ui64;
+	full -= deltaEpochInMicrosecs;
+	full /= 10;
+	int32_t milliseconds = (long)(full % 1000000UL) / 1000;
+#else
+	timeval curTime = { 0 };
+	localtime_r(&rawTime, &timeinfo);
+	gettimeofday(&curTime, nullptr);
+	int32_t milliseconds = curTime.tv_usec / 1000;
+#endif
+
+	auto &stream = _file.is_open() ? (std::ostream&)_file : _data;
+	stream
+		<< (timeinfo.tm_year + 1900)
+		<< "-" << (timeinfo.tm_mon + 1)
+		<< "-" << (timeinfo.tm_mday)
+		<< " " << timeinfo.tm_hour
+		<< ":" << timeinfo.tm_min
+		<< ":" << timeinfo.tm_sec
+		<< ":" << milliseconds
+		<< " " << message;
+
+#if DEBUG
+    printf("%d-%d-%d %d:%d:%d:%d %s\n", timeinfo.tm_year + 1900, timeinfo.tm_mon + 1, timeinfo.tm_mday, timeinfo.tm_hour, timeinfo.tm_min, timeinfo.tm_sec, milliseconds, message.c_str());
+#endif
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/LogSinkImpl.h

@@ -0,0 +1,31 @@
+#ifndef TGCALLS_LOG_SINK_IMPL_H
+#define TGCALLS_LOG_SINK_IMPL_H
+
+#include "rtc_base/logging.h"
+#include <fstream>
+
+namespace tgcalls {
+
+struct FilePath;
+
+class LogSinkImpl final : public rtc::LogSink {
+public:
+	LogSinkImpl(const FilePath &logPath);
+
+	void OnLogMessage(const std::string &msg, rtc::LoggingSeverity severity, const char *tag) override;
+	void OnLogMessage(const std::string &message, rtc::LoggingSeverity severity) override;
+	void OnLogMessage(const std::string &message) override;
+
+	std::string result() const {
+		return _data.str();
+	}
+
+private:
+	std::ofstream _file;
+	std::ostringstream _data;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/Manager.cpp

@@ -0,0 +1,465 @@
+#include "Manager.h"
+
+#include "rtc_base/byte_buffer.h"
+#include "StaticThreads.h"
+
+#include <fstream>
+
+namespace tgcalls {
+namespace {
+
+void dumpStatsLog(const FilePath &path, const CallStats &stats) {
+	if (path.data.empty()) {
+		return;
+	}
+    std::ofstream file;
+    file.open(path.data);
+
+    file << "{";
+    file << "\"v\":\"" << 1 << "\"";
+    file << ",";
+
+    file << "\"codec\":\"" << stats.outgoingCodec << "\"";
+    file << ",";
+
+    file << "\"bitrate\":[";
+    bool addComma = false;
+    for (auto &it : stats.bitrateRecords) {
+        if (addComma) {
+            file << ",";
+        }
+        file << "{";
+        file << "\"t\":\"" << it.timestamp << "\"";
+        file << ",";
+        file << "\"b\":\"" << it.bitrate << "\"";
+        file << "}";
+        addComma = true;
+    }
+    file << "]";
+    file << ",";
+
+    file << "\"network\":[";
+    addComma = false;
+    for (auto &it : stats.networkRecords) {
+        if (addComma) {
+            file << ",";
+        }
+        file << "{";
+        file << "\"t\":\"" << it.timestamp << "\"";
+        file << ",";
+        file << "\"e\":\"" << (int)(it.endpointType) << "\"";
+        file << ",";
+        file << "\"w\":\"" << (it.isLowCost ? 1 : 0) << "\"";
+        file << "}";
+        addComma = true;
+    }
+    file << "]";
+
+    file << "}";
+
+    file.close();
+}
+
+} // namespace
+
+bool Manager::ResolvedNetworkStatus::operator==(const ResolvedNetworkStatus &rhs) const {
+    if (rhs.isLowCost != isLowCost) {
+        return false;
+    }
+    if (rhs.isLowDataRequested != isLowDataRequested) {
+        return false;
+    }
+    return true;
+}
+
+bool Manager::ResolvedNetworkStatus::operator!=(const ResolvedNetworkStatus &rhs) const {
+    return !(*this == rhs);
+}
+
+Manager::Manager(rtc::Thread *thread, Descriptor &&descriptor) :
+_thread(thread),
+_encryptionKey(descriptor.encryptionKey),
+_signaling(
+	EncryptedConnection::Type::Signaling,
+	_encryptionKey,
+	[=](int delayMs, int cause) { sendSignalingAsync(delayMs, cause); }),
+_enableP2P(descriptor.config.enableP2P),
+_enableTCP(descriptor.config.allowTCP),
+_enableStunMarking(descriptor.config.enableStunMarking),
+_protocolVersion(descriptor.config.protocolVersion),
+_statsLogPath(descriptor.config.statsLogPath),
+_rtcServers(std::move(descriptor.rtcServers)),
+_proxy(std::move(descriptor.proxy)),
+_mediaDevicesConfig(std::move(descriptor.mediaDevicesConfig)),
+_videoCapture(std::move(descriptor.videoCapture)),
+_stateUpdated(std::move(descriptor.stateUpdated)),
+_remoteMediaStateUpdated(std::move(descriptor.remoteMediaStateUpdated)),
+_remoteBatteryLevelIsLowUpdated(std::move(descriptor.remoteBatteryLevelIsLowUpdated)),
+_remotePrefferedAspectRatioUpdated(std::move(descriptor.remotePrefferedAspectRatioUpdated)),
+_signalingDataEmitted(std::move(descriptor.signalingDataEmitted)),
+_signalBarsUpdated(std::move(descriptor.signalBarsUpdated)),
+_audioLevelUpdated(std::move(descriptor.audioLevelUpdated)),
+_createAudioDeviceModule(std::move(descriptor.createAudioDeviceModule)),
+_enableHighBitrateVideo(descriptor.config.enableHighBitrateVideo),
+_dataSaving(descriptor.config.dataSaving) {
+	assert(_thread->IsCurrent());
+	assert(_stateUpdated != nullptr);
+	assert(_signalingDataEmitted != nullptr);
+
+    _preferredCodecs = descriptor.config.preferredVideoCodecs;
+
+	_sendSignalingMessage = [=](const Message &message) {
+		if (const auto prepared = _signaling.prepareForSending(message)) {
+			_signalingDataEmitted(prepared->bytes);
+			return prepared->counter;
+		}
+		return uint32_t(0);
+	};
+	_sendTransportMessage = [=](Message &&message) {
+		_networkManager->perform([message = std::move(message)](NetworkManager *networkManager) {
+			networkManager->sendMessage(message);
+		});
+	};
+}
+
+Manager::~Manager() {
+	assert(_thread->IsCurrent());
+}
+
+void Manager::sendSignalingAsync(int delayMs, int cause) {
+	auto task = [weak = std::weak_ptr<Manager>(shared_from_this()), cause] {
+		const auto strong = weak.lock();
+		if (!strong) {
+			return;
+		}
+		if (const auto prepared = strong->_signaling.prepareForSendingService(cause)) {
+			strong->_signalingDataEmitted(prepared->bytes);
+		}
+	};
+	if (delayMs) {
+		_thread->PostDelayedTask(std::move(task), webrtc::TimeDelta::Millis(delayMs));
+	} else {
+		_thread->PostTask(std::move(task));
+	}
+}
+
+void Manager::start() {
+	const auto weak = std::weak_ptr<Manager>(shared_from_this());
+	const auto thread = _thread;
+	const auto sendSignalingMessage = [=](Message &&message) {
+		thread->PostTask([=, message = std::move(message)]() mutable {
+			const auto strong = weak.lock();
+			if (!strong) {
+				return;
+			}
+			strong->_sendSignalingMessage(std::move(message));
+		});
+	};
+	_networkManager.reset(new ThreadLocalObject<NetworkManager>(StaticThreads::getNetworkThread(), [weak, thread, sendSignalingMessage, encryptionKey = _encryptionKey, enableP2P = _enableP2P, enableTCP = _enableTCP, enableStunMarking = _enableStunMarking, rtcServers = _rtcServers, proxy = std::move(_proxy)] () mutable {
+		return std::make_shared<NetworkManager>(
+            StaticThreads::getNetworkThread(),
+			encryptionKey,
+			enableP2P,
+            enableTCP,
+            enableStunMarking,
+			rtcServers,
+            std::move(proxy),
+			[=](const NetworkManager::State &state) {
+				thread->PostTask([=] {
+					const auto strong = weak.lock();
+					if (!strong) {
+						return;
+					}
+                    State mappedState;
+                    if (state.isFailed) {
+                        mappedState = State::Failed;
+                    } else {
+                        mappedState = state.isReadyToSendData
+                            ? State::Established
+                            : State::Reconnecting;
+                    }
+                    bool isFirstConnection = false;
+					if (state.isReadyToSendData) {
+						if (!strong->_didConnectOnce) {
+							strong->_didConnectOnce = true;
+                            isFirstConnection = true;
+						}
+					}
+					strong->_state = mappedState;
+					strong->_stateUpdated(mappedState);
+
+					strong->_mediaManager->perform([=](MediaManager *mediaManager) {
+						mediaManager->setIsConnected(state.isReadyToSendData);
+					});
+
+                    if (isFirstConnection) {
+                        strong->sendInitialSignalingMessages();
+                    }
+				});
+			},
+			[=](DecryptedMessage &&message) {
+				thread->PostTask([=, message = std::move(message)]() mutable {
+					if (const auto strong = weak.lock()) {
+						strong->receiveMessage(std::move(message));
+					}
+				});
+			},
+			sendSignalingMessage,
+			[=](int delayMs, int cause) {
+				const auto task = [=] {
+					if (const auto strong = weak.lock()) {
+						strong->_networkManager->perform([=](NetworkManager *networkManager) {
+							networkManager->sendTransportService(cause);
+							});
+					}
+				};
+				if (delayMs) {
+					thread->PostDelayedTask(task, webrtc::TimeDelta::Millis(delayMs));
+				} else {
+					thread->PostTask(task);
+				}
+			});
+	}));
+	bool isOutgoing = _encryptionKey.isOutgoing;
+	_mediaManager.reset(new ThreadLocalObject<MediaManager>(StaticThreads::getMediaThread(), [weak, isOutgoing, protocolVersion = _protocolVersion, thread, sendSignalingMessage, videoCapture = _videoCapture, mediaDevicesConfig = _mediaDevicesConfig, enableHighBitrateVideo = _enableHighBitrateVideo, signalBarsUpdated = _signalBarsUpdated, audioLevelUpdated = _audioLevelUpdated, preferredCodecs = _preferredCodecs, createAudioDeviceModule = _createAudioDeviceModule]() {
+		return std::make_shared<MediaManager>(
+            StaticThreads::getMediaThread(),
+			isOutgoing,
+            protocolVersion,
+			mediaDevicesConfig,
+			videoCapture,
+			sendSignalingMessage,
+			[=](Message &&message) {
+				thread->PostTask([=, message = std::move(message)]() mutable {
+					const auto strong = weak.lock();
+					if (!strong) {
+						return;
+					}
+					strong->_sendTransportMessage(std::move(message));
+				});
+			},
+            signalBarsUpdated,
+            audioLevelUpdated,
+			createAudioDeviceModule,
+			enableHighBitrateVideo,
+            preferredCodecs);
+	}));
+    _networkManager->perform([](NetworkManager *networkManager) {
+        networkManager->start();
+    });
+	_mediaManager->perform([](MediaManager *mediaManager) {
+		mediaManager->start();
+	});
+}
+
+void Manager::receiveSignalingData(const std::vector<uint8_t> &data) {
+	if (auto decrypted = _signaling.handleIncomingPacket((const char*)data.data(), data.size())) {
+		receiveMessage(std::move(decrypted->main));
+		for (auto &message : decrypted->additional) {
+			receiveMessage(std::move(message));
+		}
+	}
+}
+
+void Manager::receiveMessage(DecryptedMessage &&message) {
+	const auto data = &message.message.data;
+	if (absl::get_if<CandidatesListMessage>(data)) {
+		_networkManager->perform([message = std::move(message)](NetworkManager *networkManager) mutable {
+			networkManager->receiveSignalingMessage(std::move(message));
+		});
+	} else if (absl::get_if<VideoFormatsMessage>(data)) {
+		_mediaManager->perform([message = std::move(message)](MediaManager *mediaManager) mutable {
+			mediaManager->receiveMessage(std::move(message));
+		});
+    } else if (const auto remoteMediaState = absl::get_if<RemoteMediaStateMessage>(data)) {
+		if (_remoteMediaStateUpdated) {
+			_remoteMediaStateUpdated(
+				remoteMediaState->audio,
+				remoteMediaState->video);
+		}
+        _mediaManager->perform([video = remoteMediaState->video](MediaManager *mediaManager) {
+            mediaManager->remoteVideoStateUpdated(video);
+        });
+	} else if (const auto remoteBatteryLevelIsLow = absl::get_if<RemoteBatteryLevelIsLowMessage>(data)) {
+        if (_remoteBatteryLevelIsLowUpdated) {
+			_remoteBatteryLevelIsLowUpdated(remoteBatteryLevelIsLow->batteryLow);
+        }
+    } else if (const auto remoteNetworkStatus = absl::get_if<RemoteNetworkStatusMessage>(data)) {
+        _remoteNetworkIsLowCost = remoteNetworkStatus->isLowCost;
+        _remoteIsLowDataRequested = remoteNetworkStatus->isLowDataRequested;
+        updateCurrentResolvedNetworkStatus();
+    } else {
+        if (const auto videoParameters = absl::get_if<VideoParametersMessage>(data)) {
+            float value = ((float)videoParameters->aspectRatio) / 1000.0;
+			if (_remotePrefferedAspectRatioUpdated) {
+				_remotePrefferedAspectRatioUpdated(value);
+			}
+        }
+		_mediaManager->perform([=, message = std::move(message)](MediaManager *mediaManager) mutable {
+			mediaManager->receiveMessage(std::move(message));
+		});
+	}
+}
+
+void Manager::setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture) {
+	assert(_didConnectOnce);
+
+	if (_videoCapture == videoCapture) {
+		return;
+	}
+    _videoCapture = videoCapture;
+    _mediaManager->perform([videoCapture](MediaManager *mediaManager) {
+        mediaManager->setSendVideo(videoCapture);
+    });
+}
+
+void Manager::sendVideoDeviceUpdated() {
+    _mediaManager->perform([](MediaManager *mediaManager) {
+        mediaManager->sendVideoDeviceUpdated();
+    });
+}
+
+void Manager::setRequestedVideoAspect(float aspect) {
+    _mediaManager->perform([aspect](MediaManager *mediaManager) {
+        mediaManager->setRequestedVideoAspect(aspect);
+    });
+}
+
+void Manager::setMuteOutgoingAudio(bool mute) {
+	_mediaManager->perform([mute](MediaManager *mediaManager) {
+		mediaManager->setMuteOutgoingAudio(mute);
+	});
+}
+
+void Manager::setIncomingVideoOutput(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+	_mediaManager->perform([sink](MediaManager *mediaManager) {
+		mediaManager->setIncomingVideoOutput(sink);
+	});
+}
+
+void Manager::setIsLowBatteryLevel(bool isLowBatteryLevel) {
+    _sendTransportMessage({ RemoteBatteryLevelIsLowMessage{ isLowBatteryLevel } });
+}
+
+void Manager::setIsLocalNetworkLowCost(bool isLocalNetworkLowCost) {
+    if (isLocalNetworkLowCost != _localNetworkIsLowCost) {
+        _networkManager->perform([isLocalNetworkLowCost](NetworkManager *networkManager) {
+            networkManager->setIsLocalNetworkLowCost(isLocalNetworkLowCost);
+        });
+
+        _localNetworkIsLowCost = isLocalNetworkLowCost;
+        updateCurrentResolvedNetworkStatus();
+    }
+}
+
+void Manager::getNetworkStats(std::function<void (TrafficStats, CallStats)> completion) {
+    _networkManager->perform([thread = _thread, weak = std::weak_ptr<Manager>(shared_from_this()), completion = std::move(completion), statsLogPath = _statsLogPath](NetworkManager *networkManager) {
+        auto networkStats = networkManager->getNetworkStats();
+
+        CallStats callStats;
+        networkManager->fillCallStats(callStats);
+
+        thread->PostTask([weak, networkStats, completion = std::move(completion), callStats = std::move(callStats), statsLogPath = statsLogPath] {
+            const auto strong = weak.lock();
+            if (!strong) {
+                return;
+            }
+
+            strong->_mediaManager->perform([networkStats, completion = std::move(completion), callStatsValue = std::move(callStats), statsLogPath = statsLogPath](MediaManager *mediaManager) {
+                CallStats callStats = std::move(callStatsValue);
+                mediaManager->fillCallStats(callStats);
+                dumpStatsLog(statsLogPath, callStats);
+                completion(networkStats, callStats);
+            });
+        });
+    });
+}
+
+void Manager::updateCurrentResolvedNetworkStatus() {
+    bool localIsLowDataRequested = false;
+    switch (_dataSaving) {
+        case DataSaving::Never:
+            localIsLowDataRequested = false;
+            break;
+        case DataSaving::Mobile:
+            localIsLowDataRequested = !_localNetworkIsLowCost;
+            break;
+        case DataSaving::Always:
+            localIsLowDataRequested = true;
+        default:
+            break;
+    }
+
+    ResolvedNetworkStatus localStatus;
+    localStatus.isLowCost = _localNetworkIsLowCost;
+    localStatus.isLowDataRequested = localIsLowDataRequested;
+
+    if (!_currentResolvedLocalNetworkStatus.has_value() || *_currentResolvedLocalNetworkStatus != localStatus) {
+        _currentResolvedLocalNetworkStatus = localStatus;
+
+        switch (_protocolVersion) {
+            case ProtocolVersion::V1:
+                if (_didConnectOnce) {
+                    _sendTransportMessage({ RemoteNetworkStatusMessage{ localStatus.isLowCost, localStatus.isLowDataRequested } });
+                }
+                break;
+            default:
+                break;
+        }
+    }
+
+    ResolvedNetworkStatus status;
+    status.isLowCost = _localNetworkIsLowCost && _remoteNetworkIsLowCost;
+    status.isLowDataRequested = localIsLowDataRequested || _remoteIsLowDataRequested;
+
+    if (!_currentResolvedNetworkStatus.has_value() || *_currentResolvedNetworkStatus != status) {
+        _currentResolvedNetworkStatus = status;
+        _mediaManager->perform([status](MediaManager *mediaManager) {
+            mediaManager->setNetworkParameters(status.isLowCost, status.isLowDataRequested);
+        });
+    }
+}
+
+void Manager::sendInitialSignalingMessages() {
+    if (_currentResolvedLocalNetworkStatus.has_value()) {
+        switch (_protocolVersion) {
+        case ProtocolVersion::V1:
+            _sendTransportMessage({ RemoteNetworkStatusMessage{ _currentResolvedLocalNetworkStatus->isLowCost, _currentResolvedLocalNetworkStatus->isLowDataRequested } });
+                break;
+            default:
+                break;
+        }
+    }
+}
+
+void Manager::setAudioInputDevice(std::string id) {
+	_mediaManager->perform([id](MediaManager *mediaManager) {
+		mediaManager->setAudioInputDevice(id);
+	});
+}
+
+void Manager::setAudioOutputDevice(std::string id) {
+	_mediaManager->perform([id](MediaManager *mediaManager) {
+		mediaManager->setAudioOutputDevice(id);
+	});
+}
+
+void Manager::setInputVolume(float level) {
+	_mediaManager->perform([level](MediaManager *mediaManager) {
+		mediaManager->setInputVolume(level);
+	});
+}
+
+void Manager::setOutputVolume(float level) {
+	_mediaManager->perform([level](MediaManager *mediaManager) {
+		mediaManager->setOutputVolume(level);
+	});
+}
+
+void Manager::addExternalAudioSamples(std::vector<uint8_t> &&samples) {
+    _mediaManager->perform([samples = std::move(samples)](MediaManager *mediaManager) mutable {
+        mediaManager->addExternalAudioSamples(std::move(samples));
+    });
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/Manager.h

@@ -0,0 +1,93 @@
+#ifndef TGCALLS_MANAGER_H
+#define TGCALLS_MANAGER_H
+
+#include "ThreadLocalObject.h"
+#include "EncryptedConnection.h"
+#include "NetworkManager.h"
+#include "MediaManager.h"
+#include "Instance.h"
+#include "Stats.h"
+
+namespace tgcalls {
+
+class Manager final : public std::enable_shared_from_this<Manager> {
+private:
+    struct ResolvedNetworkStatus {
+        bool isLowCost = false;
+        bool isLowDataRequested = false;
+
+        bool operator==(const ResolvedNetworkStatus &rhs) const;
+        bool operator!=(const ResolvedNetworkStatus &rhs) const;
+    };
+
+public:
+	static rtc::Thread *getMediaThread();
+
+	Manager(rtc::Thread *thread, Descriptor &&descriptor);
+	~Manager();
+
+	void start();
+	void receiveSignalingData(const std::vector<uint8_t> &data);
+	void setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture);
+    void sendVideoDeviceUpdated();
+    void setRequestedVideoAspect(float aspect);
+    void setMuteOutgoingAudio(bool mute);
+	void setIncomingVideoOutput(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+    void setIsLowBatteryLevel(bool isLowBatteryLevel);
+    void setIsLocalNetworkLowCost(bool isLocalNetworkLowCost);
+    void getNetworkStats(std::function<void(TrafficStats, CallStats)> completion);
+
+
+	void setAudioInputDevice(std::string id);
+	void setAudioOutputDevice(std::string id);
+	void setInputVolume(float level);
+	void setOutputVolume(float level);
+
+    void addExternalAudioSamples(std::vector<uint8_t> &&samples);
+
+private:
+	void sendSignalingAsync(int delayMs, int cause);
+	void receiveMessage(DecryptedMessage &&message);
+    void updateCurrentResolvedNetworkStatus();
+    void sendInitialSignalingMessages();
+
+	rtc::Thread *_thread;
+	EncryptionKey _encryptionKey;
+	EncryptedConnection _signaling;
+	bool _enableP2P = false;
+    bool _enableTCP = false;
+    bool _enableStunMarking = false;
+    ProtocolVersion _protocolVersion = ProtocolVersion::V0;
+    FilePath _statsLogPath;
+	std::vector<RtcServer> _rtcServers;
+    std::unique_ptr<Proxy> _proxy;
+	MediaDevicesConfig _mediaDevicesConfig;
+	std::shared_ptr<VideoCaptureInterface> _videoCapture;
+	std::function<void(State)> _stateUpdated;
+	std::function<void(AudioState, VideoState)> _remoteMediaStateUpdated;
+    std::function<void(bool)> _remoteBatteryLevelIsLowUpdated;
+    std::function<void(float)> _remotePrefferedAspectRatioUpdated;
+	std::function<void(const std::vector<uint8_t> &)> _signalingDataEmitted;
+    std::function<void(int)> _signalBarsUpdated;
+    std::function<void(float)> _audioLevelUpdated;
+	std::function<webrtc::scoped_refptr<webrtc::AudioDeviceModule>(webrtc::TaskQueueFactory*)> _createAudioDeviceModule;
+	std::function<uint32_t(const Message &)> _sendSignalingMessage;
+	std::function<void(Message&&)> _sendTransportMessage;
+	std::unique_ptr<ThreadLocalObject<NetworkManager>> _networkManager;
+	std::unique_ptr<ThreadLocalObject<MediaManager>> _mediaManager;
+	State _state = State::Reconnecting;
+    bool _didConnectOnce = false;
+    bool _enableHighBitrateVideo = false;
+    DataSaving _dataSaving = DataSaving::Never;
+    std::vector<std::string> _preferredCodecs;
+    bool _localNetworkIsLowCost = false;
+    bool _remoteNetworkIsLowCost = false;
+    bool _remoteIsLowDataRequested = false;
+    absl::optional<ResolvedNetworkStatus> _currentResolvedLocalNetworkStatus;
+    absl::optional<ResolvedNetworkStatus> _currentResolvedNetworkStatus;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/MediaManager.h

@@ -0,0 +1,194 @@
+#ifndef TGCALLS_MEDIA_MANAGER_H
+#define TGCALLS_MEDIA_MANAGER_H
+
+#include "rtc_base/thread.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+#include "api/transport/field_trial_based_config.h"
+#include "pc/rtp_sender.h"
+#include "media/base/media_channel.h"
+#include "pc/media_factory.h"
+#include "api/environment/environment.h"
+
+#include "Instance.h"
+#include "Message.h"
+#include "VideoCaptureInterface.h"
+#include "Stats.h"
+
+#include <functional>
+#include <memory>
+
+namespace webrtc {
+class Call;
+class RtcEventLogNull;
+class TaskQueueFactory;
+class VideoBitrateAllocatorFactory;
+class VideoTrackSourceInterface;
+class AudioDeviceModule;
+} // namespace webrtc
+
+namespace cricket {
+class MediaEngineInterface;
+class VoiceMediaChannel;
+class VideoMediaChannel;
+} // namespace cricket
+
+namespace tgcalls {
+
+class VideoSinkInterfaceProxyImpl;
+
+class MediaManager : public sigslot::has_slots<>, public std::enable_shared_from_this<MediaManager> {
+public:
+	static rtc::Thread *getWorkerThread();
+
+	MediaManager(
+		rtc::Thread *thread,
+		bool isOutgoing,
+        ProtocolVersion protocolVersion,
+		const MediaDevicesConfig &devicesConfig,
+		std::shared_ptr<VideoCaptureInterface> videoCapture,
+		std::function<void(Message &&)> sendSignalingMessage,
+		std::function<void(Message &&)> sendTransportMessage,
+        std::function<void(int)> signalBarsUpdated,
+        std::function<void(float)> audioLevelUpdated,
+		std::function<webrtc::scoped_refptr<webrtc::AudioDeviceModule>(webrtc::TaskQueueFactory*)> createAudioDeviceModule,
+        bool enableHighBitrateVideo,
+        std::vector<std::string> preferredCodecs);
+	~MediaManager();
+
+	void start();
+	void setIsConnected(bool isConnected);
+	void notifyPacketSent(const rtc::SentPacket &sentPacket);
+	void setSendVideo(std::shared_ptr<VideoCaptureInterface> videoCapture);
+	void sendVideoDeviceUpdated();
+    void setRequestedVideoAspect(float aspect);
+	void setMuteOutgoingAudio(bool mute);
+	void setIncomingVideoOutput(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+	void receiveMessage(DecryptedMessage &&message);
+    void remoteVideoStateUpdated(VideoState videoState);
+    void setNetworkParameters(bool isLowCost, bool isDataSavingActive);
+    void fillCallStats(CallStats &callStats);
+
+	void setAudioInputDevice(std::string id);
+	void setAudioOutputDevice(std::string id);
+	void setInputVolume(float level);
+	void setOutputVolume(float level);
+
+    void addExternalAudioSamples(std::vector<uint8_t> &&samples);
+
+private:
+	struct SSRC {
+		uint32_t incoming = 0;
+		uint32_t outgoing = 0;
+		uint32_t fecIncoming = 0;
+		uint32_t fecOutgoing = 0;
+	};
+
+	class NetworkInterfaceImpl : public cricket::MediaChannelNetworkInterface {
+	public:
+		NetworkInterfaceImpl(MediaManager *mediaManager, bool isVideo);
+        
+		bool SendPacket(rtc::CopyOnWriteBuffer *packet, const rtc::PacketOptions& options) override;
+		bool SendRtcp(rtc::CopyOnWriteBuffer *packet, const rtc::PacketOptions& options) override;
+		int SetOption(SocketType type, rtc::Socket::Option opt, int option) override;
+
+	private:
+		bool sendTransportMessage(rtc::CopyOnWriteBuffer *packet, const rtc::PacketOptions& options);
+
+		MediaManager *_mediaManager = nullptr;
+		bool _isVideo = false;
+
+	};
+
+	friend class MediaManager::NetworkInterfaceImpl;
+
+	void setPeerVideoFormats(VideoFormatsMessage &&peerFormats);
+
+	bool computeIsSendingVideo() const;
+    void configureSendingVideoIfNeeded();
+	void checkIsSendingVideoChanged(bool wasSending);
+	bool videoCodecsNegotiated() const;
+
+    int getMaxVideoBitrate() const;
+    int getMaxAudioBitrate() const;
+    void adjustBitratePreferences(bool resetStartBitrate);
+    bool computeIsReceivingVideo() const;
+    void checkIsReceivingVideoChanged(bool wasReceiving);
+
+	void setOutgoingVideoState(VideoState state);
+	void setOutgoingAudioState(AudioState state);
+	void sendVideoParametersMessage();
+	void sendOutgoingMediaStateMessage();
+
+	webrtc::scoped_refptr<webrtc::AudioDeviceModule> createAudioDeviceModule();
+
+    void beginStatsTimer(int timeoutMs);
+    void beginLevelsTimer(int timeoutMs);
+    void collectStats();
+
+	rtc::Thread *_thread = nullptr;
+	std::unique_ptr<webrtc::RtcEventLogNull> _eventLog;
+
+	std::function<void(Message &&)> _sendSignalingMessage;
+	std::function<void(Message &&)> _sendTransportMessage;
+    std::function<void(int)> _signalBarsUpdated;
+    std::function<void(float)> _audioLevelUpdated;
+	std::function<webrtc::scoped_refptr<webrtc::AudioDeviceModule>(webrtc::TaskQueueFactory*)> _createAudioDeviceModule;
+
+	SSRC _ssrcAudio;
+	SSRC _ssrcVideo;
+	bool _enableFlexfec = true;
+
+    ProtocolVersion _protocolVersion;
+
+	bool _isConnected = false;
+    bool _didConnectOnce = false;
+	bool _readyToReceiveVideo = false;
+    bool _didConfigureVideo = false;
+	AudioState _outgoingAudioState = AudioState::Active;
+	VideoState _outgoingVideoState = VideoState::Inactive;
+
+	VideoFormatsMessage _myVideoFormats;
+	std::vector<cricket::VideoCodec> _videoCodecs;
+	absl::optional<cricket::VideoCodec> _videoCodecOut;
+
+    webrtc::Environment _webrtcEnvironment;
+    std::unique_ptr<webrtc::MediaFactory> _mediaFactory;
+    std::unique_ptr<cricket::MediaEngineInterface> _mediaEngine;
+	std::unique_ptr<webrtc::Call> _call;
+	webrtc::LocalAudioSinkAdapter _audioSource;
+	webrtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
+	std::unique_ptr<cricket::VoiceMediaSendChannelInterface> _audioSendChannel;
+    std::unique_ptr<cricket::VoiceMediaReceiveChannelInterface> _audioReceiveChannel;
+	std::unique_ptr<cricket::VideoMediaSendChannelInterface> _videoSendChannel;
+    bool _haveVideoSendChannel = false;
+    std::unique_ptr<cricket::VideoMediaReceiveChannelInterface> _videoReceiveChannel;
+	std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> _videoBitrateAllocatorFactory;
+	std::shared_ptr<VideoCaptureInterface> _videoCapture;
+	std::shared_ptr<bool> _videoCaptureGuard;
+    bool _isScreenCapture = false;
+    std::shared_ptr<VideoSinkInterfaceProxyImpl> _incomingVideoSinkProxy;
+    webrtc::RtpHeaderExtensionMap _audioRtpHeaderExtensionMap;
+    webrtc::RtpHeaderExtensionMap _videoRtpHeaderExtensionMap;
+
+    float _localPreferredVideoAspectRatio = 0.0f;
+    float _preferredAspectRatio = 0.0f;
+    bool _enableHighBitrateVideo = false;
+    bool _isLowCostNetwork = false;
+    bool _isDataSavingActive = false;
+
+    float _currentAudioLevel = 0.0f;
+    float _currentMyAudioLevel = 0.0f;
+
+	std::unique_ptr<MediaManager::NetworkInterfaceImpl> _audioNetworkInterface;
+	std::unique_ptr<MediaManager::NetworkInterfaceImpl> _videoNetworkInterface;
+
+    std::vector<CallStatsBitrateRecord> _bitrateRecords;
+
+    std::vector<float> _externalAudioSamples;
+    webrtc::Mutex _externalAudioSamplesMutex;
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/Message.cpp

@@ -0,0 +1,406 @@
+#include "Message.h"
+
+#include "rtc_base/byte_buffer.h"
+#include "api/jsep_ice_candidate.h"
+
+namespace tgcalls {
+namespace {
+
+constexpr auto kMaxStringLength = 65536;
+
+void Serialize(rtc::ByteBufferWriter &to, const std::string &from) {
+	assert(from.size() < kMaxStringLength);
+
+	to.WriteUInt32(uint32_t(from.size()));
+	to.WriteString(from);
+}
+
+bool Deserialize(std::string &to, rtc::ByteBufferReader &from) {
+	uint32_t length = 0;
+	if (!from.ReadUInt32(&length)) {
+		RTC_LOG(LS_ERROR) << "Could not read string length.";
+		return false;
+	} else if (length >= kMaxStringLength) {
+		RTC_LOG(LS_ERROR) << "Invalid string length: " << length;
+		return false;
+	} else if (!from.ReadString(&to, length)) {
+		RTC_LOG(LS_ERROR) << "Could not read string data.";
+		return false;
+	}
+	return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const webrtc::SdpVideoFormat &from) {
+	assert(from.parameters.size() < std::numeric_limits<uint8_t>::max());
+
+	Serialize(to, from.name);
+	to.WriteUInt8(uint8_t(from.parameters.size()));
+	for (const auto &pair : from.parameters) {
+		Serialize(to, pair.first);
+		Serialize(to, pair.second);
+	}
+}
+
+bool Deserialize(webrtc::SdpVideoFormat &to, rtc::ByteBufferReader &from) {
+	if (!Deserialize(to.name, from)) {
+		RTC_LOG(LS_ERROR) << "Could not read video format name.";
+		return false;
+	}
+	auto count = uint8_t();
+	if (!from.ReadUInt8(&count)) {
+		RTC_LOG(LS_ERROR) << "Could not read video format parameters count.";
+		return false;
+	}
+	for (uint32_t i = 0; i != count; ++i) {
+		auto key = std::string();
+		auto value = std::string();
+		if (!Deserialize(key, from)) {
+			RTC_LOG(LS_ERROR) << "Could not read video format parameter key.";
+			return false;
+		} else if (!Deserialize(value, from)) {
+			RTC_LOG(LS_ERROR) << "Could not read video format parameter value.";
+			return false;
+		}
+		to.parameters.emplace(key, value);
+	}
+	return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const cricket::Candidate &from) {
+	webrtc::JsepIceCandidate iceCandidate{ std::string(), 0 };
+	iceCandidate.SetCandidate(from);
+	std::string serialized;
+	const auto success = iceCandidate.ToString(&serialized);
+	assert(success);
+	(void)success;
+	Serialize(to, serialized);
+}
+
+bool Deserialize(cricket::Candidate &to, rtc::ByteBufferReader &from) {
+	std::string candidate;
+	if (!Deserialize(candidate, from)) {
+		RTC_LOG(LS_ERROR) << "Could not read candidate string.";
+		return false;
+	}
+	webrtc::JsepIceCandidate parseCandidate{ std::string(), 0 };
+	if (!parseCandidate.Initialize(candidate, nullptr)) {
+		RTC_LOG(LS_ERROR) << "Could not parse candidate: " << candidate;
+		return false;
+	}
+	to = parseCandidate.candidate();
+	return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const RequestVideoMessage &from, bool singleMessagePacket) {
+}
+
+bool Deserialize(RequestVideoMessage &to, rtc::ByteBufferReader &reader, bool singleMessagePacket) {
+	return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const RemoteMediaStateMessage &from, bool singleMessagePacket) {
+	uint8_t state = (uint8_t(from.video) << 1) | uint8_t(from.audio);
+	to.WriteUInt8(state);
+}
+
+bool Deserialize(RemoteMediaStateMessage &to, rtc::ByteBufferReader &reader, bool singleMessagePacket) {
+	uint8_t state = 0;
+	if (!reader.ReadUInt8(&state)) {
+		RTC_LOG(LS_ERROR) << "Could not read remote media state.";
+		return false;
+	}
+	to.audio = AudioState(state & 0x01);
+	to.video = VideoState((state >> 1) & 0x03);
+	if (to.video == VideoState(0x03)) {
+		RTC_LOG(LS_ERROR) << "Invalid value for remote video state.";
+		return false;
+	}
+	return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const CandidatesListMessage &from, bool singleMessagePacket) {
+	assert(from.candidates.size() < std::numeric_limits<uint8_t>::max());
+
+	to.WriteUInt8(uint8_t(from.candidates.size()));
+	for (const auto &candidate : from.candidates) {
+		Serialize(to, candidate);
+	}
+
+    Serialize(to, from.iceParameters.ufrag);
+    Serialize(to, from.iceParameters.pwd);
+}
+
+bool Deserialize(CandidatesListMessage &to, rtc::ByteBufferReader &reader, bool singleMessagePacket) {
+	auto count = uint8_t();
+	if (!reader.ReadUInt8(&count)) {
+		RTC_LOG(LS_ERROR) << "Could not read candidates count.";
+		return false;
+	}
+	for (uint32_t i = 0; i != count; ++i) {
+		auto candidate = cricket::Candidate();
+		if (!Deserialize(candidate, reader)) {
+			RTC_LOG(LS_ERROR) << "Could not read candidate.";
+			return false;
+		}
+		to.candidates.push_back(std::move(candidate));
+	}
+    if (!Deserialize(to.iceParameters.ufrag, reader)) {
+        return false;
+    }
+    if (!Deserialize(to.iceParameters.pwd, reader)) {
+        return false;
+    }
+	return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const VideoFormatsMessage &from, bool singleMessagePacket) {
+	assert(from.formats.size() < std::numeric_limits<uint8_t>::max());
+	assert(from.encodersCount <= from.formats.size());
+
+	to.WriteUInt8(uint8_t(from.formats.size()));
+	for (const auto &format : from.formats) {
+		Serialize(to, format);
+	}
+	to.WriteUInt8(uint8_t(from.encodersCount));
+}
+
+bool Deserialize(VideoFormatsMessage &to, rtc::ByteBufferReader &from, bool singleMessagePacket) {
+	auto count = uint8_t();
+	if (!from.ReadUInt8(&count)) {
+		RTC_LOG(LS_ERROR) << "Could not read video formats count.";
+		return false;
+	}
+	for (uint32_t i = 0; i != count; ++i) {
+		auto format = webrtc::SdpVideoFormat(std::string());
+		if (!Deserialize(format, from)) {
+			RTC_LOG(LS_ERROR) << "Could not read video format.";
+			return false;
+		}
+		to.formats.push_back(std::move(format));
+	}
+	auto encoders = uint8_t();
+	if (!from.ReadUInt8(&encoders)) {
+		RTC_LOG(LS_ERROR) << "Could not read encoders count.";
+		return false;
+	} else if (encoders > to.formats.size()) {
+		RTC_LOG(LS_ERROR) << "Invalid encoders count: " << encoders << ", full formats count: " << to.formats.size();
+		return false;
+	}
+	to.encodersCount = encoders;
+	return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const rtc::CopyOnWriteBuffer &from, bool singleMessagePacket) {
+	if (!singleMessagePacket) {
+		assert(from.size() <= UINT16_MAX);
+		to.WriteUInt16(from.size());
+	}
+	to.WriteBytes(reinterpret_cast<const uint8_t*>(from.cdata()), from.size());
+}
+
+bool Deserialize(rtc::CopyOnWriteBuffer &to, rtc::ByteBufferReader &from, bool singleMessagePacket) {
+	auto length = uint16_t(from.Length());
+	if (!singleMessagePacket) {
+		if (!from.ReadUInt16(&length)) {
+			RTC_LOG(LS_ERROR) << "Could not read buffer length.";
+			return false;
+		} else if (from.Length() < length) {
+			RTC_LOG(LS_ERROR) << "Invalid buffer length: " << length << ", available: " << from.Length();
+			return false;
+		}
+	}
+	to.AppendData(from.Data(), length);
+	from.Consume(length);
+	return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const AudioDataMessage &from, bool singleMessagePacket) {
+	Serialize(to, from.data, singleMessagePacket);
+}
+
+bool Deserialize(AudioDataMessage &to, rtc::ByteBufferReader &from, bool singleMessagePacket) {
+	return Deserialize(to.data, from, singleMessagePacket);
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const VideoDataMessage &from, bool singleMessagePacket) {
+	Serialize(to, from.data, singleMessagePacket);
+}
+
+bool Deserialize(VideoDataMessage &to, rtc::ByteBufferReader &from, bool singleMessagePacket) {
+	return Deserialize(to.data, from, singleMessagePacket);
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const UnstructuredDataMessage &from, bool singleMessagePacket) {
+    Serialize(to, from.data, singleMessagePacket);
+}
+
+bool Deserialize(UnstructuredDataMessage &to, rtc::ByteBufferReader &from, bool singleMessagePacket) {
+    return Deserialize(to.data, from, singleMessagePacket);
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const VideoParametersMessage &from, bool singleMessagePacket) {
+    to.WriteUInt32(from.aspectRatio);
+}
+
+bool Deserialize(VideoParametersMessage &to, rtc::ByteBufferReader &from, bool singleMessagePacket) {
+    uint32_t aspectRatio = 0;
+    if (!from.ReadUInt32(&aspectRatio)) {
+        return false;
+    }
+    to.aspectRatio = aspectRatio;
+    return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const RemoteBatteryLevelIsLowMessage &from, bool singleMessagePacket) {
+    to.WriteUInt8(from.batteryLow ? 1 : 0);
+}
+
+bool Deserialize(RemoteBatteryLevelIsLowMessage &to, rtc::ByteBufferReader &reader, bool singleMessagePacket) {
+    uint8_t value = 0;
+    if (!reader.ReadUInt8(&value)) {
+        RTC_LOG(LS_ERROR) << "Could not read batteryLevelIsLow.";
+        return false;
+    }
+    to.batteryLow = (value != 0);
+    return true;
+}
+
+void Serialize(rtc::ByteBufferWriter &to, const RemoteNetworkStatusMessage &from, bool singleMessagePacket) {
+    to.WriteUInt8(from.isLowCost ? 1 : 0);
+    to.WriteUInt8(from.isLowDataRequested ? 1 : 0);
+}
+
+bool Deserialize(RemoteNetworkStatusMessage &to, rtc::ByteBufferReader &reader, bool singleMessagePacket) {
+    uint8_t value = 0;
+    if (!reader.ReadUInt8(&value)) {
+        RTC_LOG(LS_ERROR) << "Could not read isLowCost.";
+        return false;
+    }
+    to.isLowCost = (value != 0);
+    if (reader.ReadUInt8(&value)) {
+        to.isLowDataRequested = (value != 0);
+    }
+    return true;
+}
+
+enum class TryResult : uint8_t {
+	Success,
+	TryNext,
+	Abort,
+};
+
+template <typename T>
+TryResult TryDeserialize(
+		absl::optional<Message> &to,
+		rtc::ByteBufferReader &reader,
+		bool singleMessagePacket) {
+	assert(reader.Length() != 0);
+
+	constexpr auto id = T::kId;
+	if (uint8_t(*reader.Data()) != id) {
+		return TryResult::TryNext;
+	}
+	reader.Consume(1);
+	auto parsed = T();
+	if (!Deserialize(parsed, reader, singleMessagePacket)) {
+		RTC_LOG(LS_ERROR) << "Could not read message with kId: " << id;
+		return TryResult::Abort;
+	}
+	to = Message{ parsed };
+	return TryResult::Success;
+}
+
+template <typename ...Types>
+struct TryDeserializeNext;
+
+template <>
+struct TryDeserializeNext<> {
+	static bool Call(
+			absl::optional<Message> &to,
+			rtc::ByteBufferReader &reader,
+			bool singleMessagePacket) {
+		return false;
+	}
+};
+
+template <typename T, typename ...Other>
+struct TryDeserializeNext<T, Other...> {
+	static bool Call(
+			absl::optional<Message> &to,
+			rtc::ByteBufferReader &reader,
+			bool singleMessagePacket) {
+		const auto result = TryDeserialize<T>(to, reader, singleMessagePacket);
+		return (result == TryResult::TryNext)
+			? TryDeserializeNext<Other...>::Call(to, reader, singleMessagePacket)
+			: (result == TryResult::Success);
+	}
+};
+
+template <typename ...Types>
+bool TryDeserializeRecursive(
+		absl::optional<Message> &to,
+		rtc::ByteBufferReader &reader,
+		bool singleMessagePacket,
+		absl::variant<Types...> *) {
+	return TryDeserializeNext<Types...>::Call(to, reader, singleMessagePacket);
+}
+
+} // namespace
+
+
+rtc::CopyOnWriteBuffer SerializeMessageWithSeq(
+		const Message &message,
+		uint32_t seq,
+		bool singleMessagePacket) {
+	rtc::ByteBufferWriter writer;
+	writer.WriteUInt32(seq);
+	absl::visit([&](const auto &data) {
+		writer.WriteUInt8(std::decay_t<decltype(data)>::kId);
+		Serialize(writer, data, singleMessagePacket);
+	}, message.data);
+
+	auto result = rtc::CopyOnWriteBuffer();
+	result.AppendData(writer.Data(), writer.Length());
+
+	return result;
+}
+
+absl::optional<Message> DeserializeMessage(
+		rtc::ByteBufferReader &reader,
+		bool singleMessagePacket) {
+	if (!reader.Length()) {
+		return absl::nullopt;
+	}
+	using Variant = decltype(std::declval<Message>().data);
+	auto result = absl::make_optional<Message>();
+	return TryDeserializeRecursive(result, reader, singleMessagePacket, (Variant*)nullptr)
+		? result
+		: absl::nullopt;
+}
+
+absl::optional<rtc::CopyOnWriteBuffer> DeserializeRawMessage(
+    rtc::ByteBufferReader &reader,
+    bool singleMessagePacket) {
+    if (!reader.Length()) {
+        return absl::nullopt;
+    }
+
+    uint32_t length = 0;
+    if (!reader.ReadUInt32(&length)) {
+        return absl::nullopt;
+    }
+
+    if (/*length < 0 || */length > 1024 * 1024) {
+        return absl::nullopt;
+    }
+
+    rtc::CopyOnWriteBuffer result;
+    result.SetSize(length);
+    if (!reader.ReadBytes(rtc::ArrayView<uint8_t>((uint8_t *)result.MutableData(), result.size()))) {
+        return absl::nullopt;
+    }
+
+    return result;
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/Message.h

@@ -0,0 +1,148 @@
+#ifndef TGCALLS_MESSAGE_H
+#define TGCALLS_MESSAGE_H
+
+#include "api/candidate.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "absl/types/variant.h"
+#include "absl/types/optional.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/byte_buffer.h"
+
+#include <vector>
+
+namespace tgcalls {
+
+enum class VideoState;
+enum class AudioState;
+
+struct PeerIceParameters {
+    std::string ufrag;
+    std::string pwd;
+    bool supportsRenomination = false;
+    
+    PeerIceParameters() = default;
+
+    PeerIceParameters(std::string ufrag_, std::string pwd_, bool supportsRenomination_) :
+    ufrag(ufrag_),
+    pwd(pwd_),
+    supportsRenomination(supportsRenomination_) {
+    }
+
+};
+
+struct CandidatesListMessage {
+	static constexpr uint8_t kId = 1;
+	static constexpr bool kRequiresAck = true;
+
+	std::vector<cricket::Candidate> candidates;
+    PeerIceParameters iceParameters;
+};
+
+struct VideoFormatsMessage {
+	static constexpr uint8_t kId = 2;
+	static constexpr bool kRequiresAck = true;
+
+	std::vector<webrtc::SdpVideoFormat> formats;
+	int encodersCount = 0;
+};
+
+struct RequestVideoMessage {
+	static constexpr uint8_t kId = 3;
+	static constexpr bool kRequiresAck = true;
+};
+
+struct RemoteMediaStateMessage {
+	static constexpr uint8_t kId = 4;
+	static constexpr bool kRequiresAck = true;
+
+	AudioState audio = AudioState();
+	VideoState video = VideoState();
+};
+
+struct AudioDataMessage {
+	static constexpr uint8_t kId = 5;
+	static constexpr bool kRequiresAck = false;
+
+	rtc::CopyOnWriteBuffer data;
+};
+
+struct VideoDataMessage {
+	static constexpr uint8_t kId = 6;
+	static constexpr bool kRequiresAck = false;
+
+	rtc::CopyOnWriteBuffer data;
+};
+
+struct UnstructuredDataMessage {
+    static constexpr uint8_t kId = 7;
+    static constexpr bool kRequiresAck = true;
+
+    rtc::CopyOnWriteBuffer data;
+};
+
+struct VideoParametersMessage {
+    static constexpr uint8_t kId = 8;
+    static constexpr bool kRequiresAck = true;
+
+    uint32_t aspectRatio;
+};
+
+struct RemoteBatteryLevelIsLowMessage {
+    static constexpr uint8_t kId = 9;
+    static constexpr bool kRequiresAck = true;
+
+    bool batteryLow = false;
+};
+
+struct RemoteNetworkStatusMessage {
+    static constexpr uint8_t kId = 10;
+    static constexpr bool kRequiresAck = true;
+
+    bool isLowCost = false;
+    bool isLowDataRequested = false;
+};
+
+// To add a new message you should:
+// 1. Add the message struct.
+// 2. Add the message to the variant in Message struct.
+// 3. Add Serialize/Deserialize methods in Message module.
+
+struct Message {
+	absl::variant<
+		CandidatesListMessage,
+		VideoFormatsMessage,
+        RequestVideoMessage,
+		RemoteMediaStateMessage,
+		AudioDataMessage,
+		VideoDataMessage,
+        UnstructuredDataMessage,
+        VideoParametersMessage,
+        RemoteBatteryLevelIsLowMessage,
+        RemoteNetworkStatusMessage> data;
+};
+
+rtc::CopyOnWriteBuffer SerializeMessageWithSeq(
+	const Message &message,
+	uint32_t seq,
+	bool singleMessagePacket);
+absl::optional<Message> DeserializeMessage(
+	rtc::ByteBufferReader &reader,
+	bool singleMessagePacket);
+absl::optional<rtc::CopyOnWriteBuffer> DeserializeRawMessage(
+    rtc::ByteBufferReader &reader,
+    bool singleMessagePacket);
+
+struct DecryptedMessage {
+	Message message;
+	uint32_t counter = 0;
+};
+
+struct DecryptedRawMessage {
+    rtc::CopyOnWriteBuffer message;
+    uint32_t counter = 0;
+};
+
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/NetworkManager.cpp

@@ -0,0 +1,397 @@
+#include "NetworkManager.h"
+
+#include "Message.h"
+
+#include "p2p/base/basic_packet_socket_factory.h"
+#include "v2/ReflectorRelayPortFactory.h"
+#include "p2p/client/basic_port_allocator.h"
+#include "p2p/base/p2p_transport_channel.h"
+#include "p2p/base/basic_async_resolver_factory.h"
+#include "api/packet_socket_factory.h"
+#include "p2p/base/ice_credentials_iterator.h"
+#include "api/jsep_ice_candidate.h"
+#include "rtc_base/network_monitor_factory.h"
+
+#include "TurnCustomizerImpl.h"
+#include "platform/PlatformInterface.h"
+
+extern "C" {
+#include <openssl/sha.h>
+#include <openssl/aes.h>
+#ifndef OPENSSL_IS_BORINGSSL
+#include <openssl/modes.h>
+#endif
+#include <openssl/rand.h>
+#include <openssl/crypto.h>
+} // extern "C"
+
+namespace tgcalls {
+
+class TgCallsCryptStringImpl : public rtc::CryptStringImpl {
+public:
+    TgCallsCryptStringImpl(std::string const &value) :
+    _value(value) {
+    }
+    
+    virtual ~TgCallsCryptStringImpl() override {
+    }
+    
+    virtual size_t GetLength() const override {
+        return _value.size();
+    }
+    
+    virtual void CopyTo(char* dest, bool nullterminate) const override {
+        memcpy(dest, _value.data(), _value.size());
+        if (nullterminate) {
+            dest[_value.size()] = 0;
+        }
+    }
+    virtual std::string UrlEncode() const override {
+        return _value;
+    }
+    virtual CryptStringImpl* Copy() const override {
+        return new TgCallsCryptStringImpl(_value);
+    }
+    
+    virtual void CopyRawTo(std::vector<unsigned char>* dest) const override {
+        dest->resize(_value.size());
+        memcpy(dest->data(), _value.data(), _value.size());
+    }
+    
+private:
+    std::string _value;
+};
+
+NetworkManager::NetworkManager(
+	rtc::Thread *thread,
+	EncryptionKey encryptionKey,
+	bool enableP2P,
+    bool enableTCP,
+    bool enableStunMarking,
+	std::vector<RtcServer> const &rtcServers,
+    std::unique_ptr<Proxy> proxy,
+	std::function<void(const NetworkManager::State &)> stateUpdated,
+	std::function<void(DecryptedMessage &&)> transportMessageReceived,
+	std::function<void(Message &&)> sendSignalingMessage,
+	std::function<void(int delayMs, int cause)> sendTransportServiceAsync) :
+_thread(thread),
+_enableP2P(enableP2P),
+_enableTCP(enableTCP),
+_enableStunMarking(enableStunMarking),
+_rtcServers(rtcServers),
+_proxy(std::move(proxy)),
+_transport(
+	EncryptedConnection::Type::Transport,
+	encryptionKey,
+	[=](int delayMs, int cause) { sendTransportServiceAsync(delayMs, cause); }),
+_isOutgoing(encryptionKey.isOutgoing),
+_stateUpdated(std::move(stateUpdated)),
+_transportMessageReceived(std::move(transportMessageReceived)),
+_sendSignalingMessage(std::move(sendSignalingMessage)),
+_localIceParameters(rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH), rtc::CreateRandomString(cricket::ICE_PWD_LENGTH), false) {
+	assert(_thread->IsCurrent());
+
+    _networkMonitorFactory = PlatformInterface::SharedInstance()->createNetworkMonitorFactory();
+}
+
+NetworkManager::~NetworkManager() {
+	assert(_thread->IsCurrent());
+    
+    RTC_LOG(LS_INFO) << "NetworkManager::~NetworkManager()";
+
+	_transportChannel.reset();
+	_asyncResolverFactory.reset();
+	_portAllocator.reset();
+	_networkManager.reset();
+	_socketFactory.reset();
+    _networkMonitorFactory.reset();
+}
+
+void NetworkManager::start() {
+    _socketFactory.reset(new rtc::BasicPacketSocketFactory(_thread->socketserver()));
+
+    _networkManager = std::make_unique<rtc::BasicNetworkManager>(_networkMonitorFactory.get(), _thread->socketserver());
+    
+    if (_enableStunMarking) {
+        _turnCustomizer.reset(new TurnCustomizerImpl());
+    }
+    
+    _relayPortFactory.reset(new ReflectorRelayPortFactory(_rtcServers, false, 0, _thread->socketserver()));
+    
+    _portAllocator.reset(new cricket::BasicPortAllocator(_networkManager.get(), _socketFactory.get(), _turnCustomizer.get(), _relayPortFactory.get()));
+
+    uint32_t flags = _portAllocator->flags();
+    
+    flags |=
+        //cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
+        cricket::PORTALLOCATOR_ENABLE_IPV6 |
+        cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
+    
+    if (!_enableTCP) {
+        flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
+    }
+    if (!_enableP2P) {
+        flags |= cricket::PORTALLOCATOR_DISABLE_UDP;
+        flags |= cricket::PORTALLOCATOR_DISABLE_STUN;
+        uint32_t candidateFilter = _portAllocator->candidate_filter();
+        candidateFilter &= ~(cricket::CF_REFLEXIVE);
+        _portAllocator->SetCandidateFilter(candidateFilter);
+    }
+    
+    _portAllocator->set_step_delay(cricket::kMinimumStepDelay);
+    
+    if (_proxy) {
+        rtc::ProxyInfo proxyInfo;
+        proxyInfo.type = rtc::ProxyType::PROXY_SOCKS5;
+        proxyInfo.address = rtc::SocketAddress(_proxy->host, _proxy->port);
+        proxyInfo.username = _proxy->login;
+        proxyInfo.password = rtc::CryptString(TgCallsCryptStringImpl(_proxy->password));
+        _portAllocator->set_proxy("t/1.0", proxyInfo);
+    }
+    
+    _portAllocator->set_flags(flags);
+    _portAllocator->Initialize();
+
+    cricket::ServerAddresses stunServers;
+    std::vector<cricket::RelayServerConfig> turnServers;
+
+    for (auto &server : _rtcServers) {
+        if (server.isTcp) {
+            continue;
+        }
+        
+        if (server.isTurn) {
+            turnServers.push_back(cricket::RelayServerConfig(
+                rtc::SocketAddress(server.host, server.port),
+                server.login,
+                server.password,
+                cricket::PROTO_UDP
+            ));
+        } else {
+            rtc::SocketAddress stunAddress = rtc::SocketAddress(server.host, server.port);
+            stunServers.insert(stunAddress);
+        }
+    }
+
+    _portAllocator->SetConfiguration(stunServers, turnServers, 2, webrtc::NO_PRUNE, _turnCustomizer.get());
+
+    _asyncResolverFactory = std::make_unique<webrtc::BasicAsyncDnsResolverFactory>();
+
+    webrtc::IceTransportInit iceTransportInit;
+    iceTransportInit.set_port_allocator(_portAllocator.get());
+    iceTransportInit.set_async_dns_resolver_factory(_asyncResolverFactory.get());
+
+    _transportChannel = cricket::P2PTransportChannel::Create("transport", 0, std::move(iceTransportInit));
+
+    cricket::IceConfig iceConfig;
+    iceConfig.continual_gathering_policy = cricket::GATHER_CONTINUALLY;
+    iceConfig.prioritize_most_likely_candidate_pairs = true;
+    iceConfig.regather_on_failed_networks_interval = cricket::REGATHER_ON_FAILED_NETWORKS_INTERVAL;
+    _transportChannel->SetIceConfig(iceConfig);
+
+    cricket::IceParameters localIceParameters(
+        _localIceParameters.ufrag,
+        _localIceParameters.pwd,
+        false
+    );
+
+    _transportChannel->SetIceParameters(localIceParameters);
+    _transportChannel->SetIceRole(_isOutgoing ? cricket::ICEROLE_CONTROLLING : cricket::ICEROLE_CONTROLLED);
+
+    _transportChannel->SignalCandidateGathered.connect(this, &NetworkManager::candidateGathered);
+    _transportChannel->SignalGatheringState.connect(this, &NetworkManager::candidateGatheringState);
+    _transportChannel->SignalIceTransportStateChanged.connect(this, &NetworkManager::transportStateChanged);
+    _transportChannel->SignalReadPacket.connect(this, &NetworkManager::transportPacketReceived);
+    _transportChannel->SignalNetworkRouteChanged.connect(this, &NetworkManager::transportRouteChanged);
+
+    _transportChannel->MaybeStartGathering();
+
+    _transportChannel->SetRemoteIceMode(cricket::ICEMODE_FULL);
+    
+    _lastNetworkActivityMs = rtc::TimeMillis();
+    
+    checkConnectionTimeout();
+}
+
+void NetworkManager::receiveSignalingMessage(DecryptedMessage &&message) {
+	const auto list = absl::get_if<CandidatesListMessage>(&message.message.data);
+	assert(list != nullptr);
+
+    if (!_remoteIceParameters.has_value()) {
+        PeerIceParameters parameters(list->iceParameters.ufrag, list->iceParameters.pwd, false);
+        _remoteIceParameters = parameters;
+
+        cricket::IceParameters remoteIceParameters(
+            parameters.ufrag,
+            parameters.pwd,
+            false
+        );
+
+        _transportChannel->SetRemoteIceParameters(remoteIceParameters);
+    }
+
+	for (const auto &candidate : list->candidates) {
+		_transportChannel->AddRemoteCandidate(candidate);
+	}
+}
+
+uint32_t NetworkManager::sendMessage(const Message &message) {
+	if (const auto prepared = _transport.prepareForSending(message)) {
+		rtc::PacketOptions packetOptions;
+		_transportChannel->SendPacket((const char *)prepared->bytes.data(), prepared->bytes.size(), packetOptions, 0);
+        addTrafficStats(prepared->bytes.size(), false);
+		return prepared->counter;
+	}
+	return 0;
+}
+
+void NetworkManager::sendTransportService(int cause) {
+	if (const auto prepared = _transport.prepareForSendingService(cause)) {
+		rtc::PacketOptions packetOptions;
+		_transportChannel->SendPacket((const char *)prepared->bytes.data(), prepared->bytes.size(), packetOptions, 0);
+        addTrafficStats(prepared->bytes.size(), false);
+	}
+}
+
+void NetworkManager::setIsLocalNetworkLowCost(bool isLocalNetworkLowCost) {
+    _isLocalNetworkLowCost = isLocalNetworkLowCost;
+    
+    logCurrentNetworkState();
+}
+
+TrafficStats NetworkManager::getNetworkStats() {
+    TrafficStats stats;
+    stats.bytesSentWifi = _trafficStatsWifi.outgoing;
+    stats.bytesReceivedWifi = _trafficStatsWifi.incoming;
+    stats.bytesSentMobile = _trafficStatsCellular.outgoing;
+    stats.bytesReceivedMobile = _trafficStatsCellular.incoming;
+    return stats;
+}
+
+void NetworkManager::fillCallStats(CallStats &callStats) {
+    callStats.networkRecords = _networkRecords;
+}
+
+void NetworkManager::logCurrentNetworkState() {
+    if (!_currentEndpointType.has_value()) {
+        return;
+    }
+    
+    CallStatsNetworkRecord record;
+    record.timestamp = (int32_t)(rtc::TimeMillis() / 1000);
+    record.endpointType = *_currentEndpointType;
+    record.isLowCost = _isLocalNetworkLowCost;
+    _networkRecords.push_back(std::move(record));
+}
+
+void NetworkManager::checkConnectionTimeout() {
+    const auto weak = std::weak_ptr<NetworkManager>(shared_from_this());
+    _thread->PostDelayedTask([weak]() {
+        auto strong = weak.lock();
+        if (!strong) {
+            return;
+        }
+        
+        int64_t currentTimestamp = rtc::TimeMillis();
+        const int64_t maxTimeout = 20000;
+        
+        if (strong->_lastNetworkActivityMs + maxTimeout < currentTimestamp) {
+            NetworkManager::State emitState;
+            emitState.isReadyToSendData = false;
+            emitState.isFailed = true;
+            strong->_stateUpdated(emitState);
+        }
+        
+        strong->checkConnectionTimeout();
+    }, webrtc::TimeDelta::Millis(1000));
+}
+
+void NetworkManager::candidateGathered(cricket::IceTransportInternal *transport, const cricket::Candidate &candidate) {
+	assert(_thread->IsCurrent());
+	_sendSignalingMessage({ CandidatesListMessage{ { 1, candidate }, _localIceParameters } });
+}
+
+void NetworkManager::candidateGatheringState(cricket::IceTransportInternal *transport) {
+	assert(_thread->IsCurrent());
+}
+
+void NetworkManager::transportStateChanged(cricket::IceTransportInternal *transport) {
+	assert(_thread->IsCurrent());
+
+	auto state = transport->GetIceTransportState();
+	bool isConnected = false;
+	switch (state) {
+		case webrtc::IceTransportState::kConnected:
+		case webrtc::IceTransportState::kCompleted:
+			isConnected = true;
+			break;
+		default:
+			break;
+	}
+	NetworkManager::State emitState;
+	emitState.isReadyToSendData = isConnected;
+	_stateUpdated(emitState);
+}
+
+void NetworkManager::transportReadyToSend(cricket::IceTransportInternal *transport) {
+	assert(_thread->IsCurrent());
+}
+
+void NetworkManager::transportPacketReceived(rtc::PacketTransportInternal *transport, const char *bytes, size_t size, const int64_t &timestamp, int unused) {
+	assert(_thread->IsCurrent());
+    
+    _lastNetworkActivityMs = rtc::TimeMillis();
+    
+    addTrafficStats(size, true);
+
+	if (auto decrypted = _transport.handleIncomingPacket(bytes, size)) {
+		if (_transportMessageReceived) {
+			_transportMessageReceived(std::move(decrypted->main));
+			for (auto &message : decrypted->additional) {
+				_transportMessageReceived(std::move(message));
+			}
+		}
+	}
+}
+
+void NetworkManager::transportRouteChanged(absl::optional<rtc::NetworkRoute> route) {
+    assert(_thread->IsCurrent());
+    
+    if (route.has_value()) {
+        RTC_LOG(LS_INFO) << "NetworkManager route changed: " << route->DebugString();
+        
+        bool localIsWifi = route->local.adapter_type() == rtc::AdapterType::ADAPTER_TYPE_WIFI;
+        bool remoteIsWifi = route->remote.adapter_type() == rtc::AdapterType::ADAPTER_TYPE_WIFI;
+        
+        RTC_LOG(LS_INFO) << "NetworkManager is wifi: local=" << localIsWifi << ", remote=" << remoteIsWifi;
+        
+        CallStatsConnectionEndpointType endpointType;
+        if (route->local.uses_turn()) {
+            endpointType = CallStatsConnectionEndpointType::ConnectionEndpointTURN;
+        } else {
+            endpointType = CallStatsConnectionEndpointType::ConnectionEndpointP2P;
+        }
+        if (!_currentEndpointType.has_value() || _currentEndpointType != endpointType) {
+            _currentEndpointType = endpointType;
+            logCurrentNetworkState();
+        }
+    }
+}
+
+void NetworkManager::addTrafficStats(int64_t byteCount, bool isIncoming) {
+    if (_isLocalNetworkLowCost) {
+        if (isIncoming) {
+            _trafficStatsWifi.incoming += byteCount;
+        } else {
+            _trafficStatsWifi.outgoing += byteCount;
+        }
+    } else {
+        if (isIncoming) {
+            _trafficStatsCellular.incoming += byteCount;
+        } else {
+            _trafficStatsCellular.outgoing += byteCount;
+        }
+    }
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/NetworkManager.h

@@ -0,0 +1,122 @@
+#ifndef TGCALLS_NETWORK_MANAGER_H
+#define TGCALLS_NETWORK_MANAGER_H
+
+#include "rtc_base/thread.h"
+
+#include "EncryptedConnection.h"
+#include "Instance.h"
+#include "Message.h"
+#include "Stats.h"
+
+#include "rtc_base/copy_on_write_buffer.h"
+#include "api/candidate.h"
+#include "rtc_base/network_monitor_factory.h"
+#include "api/async_dns_resolver.h"
+
+#include <functional>
+#include <memory>
+
+namespace rtc {
+class BasicPacketSocketFactory;
+class BasicNetworkManager;
+class PacketTransportInternal;
+struct NetworkRoute;
+} // namespace rtc
+
+namespace cricket {
+class BasicPortAllocator;
+class P2PTransportChannel;
+class IceTransportInternal;
+class RelayPortFactoryInterface;
+} // namespace cricket
+
+namespace webrtc {
+class BasicAsyncResolverFactory;
+class TurnCustomizer;
+} // namespace webrtc
+
+namespace tgcalls {
+
+struct Message;
+
+class NetworkManager : public sigslot::has_slots<>, public std::enable_shared_from_this<NetworkManager> {
+public:
+	struct State {
+		bool isReadyToSendData = false;
+        bool isFailed = false;
+	};
+    
+    struct InterfaceTrafficStats {
+        int64_t incoming = 0;
+        int64_t outgoing = 0;
+    };
+
+	NetworkManager(
+		rtc::Thread *thread,
+		EncryptionKey encryptionKey,
+		bool enableP2P,
+        bool enableTCP,
+        bool enableStunMarking,
+		std::vector<RtcServer> const &rtcServers,
+        std::unique_ptr<Proxy> proxy,
+		std::function<void(const State &)> stateUpdated,
+		std::function<void(DecryptedMessage &&)> transportMessageReceived,
+		std::function<void(Message &&)> sendSignalingMessage,
+		std::function<void(int delayMs, int cause)> sendTransportServiceAsync);
+	~NetworkManager();
+
+    void start();
+	void receiveSignalingMessage(DecryptedMessage &&message);
+	uint32_t sendMessage(const Message &message);
+	void sendTransportService(int cause);
+    void setIsLocalNetworkLowCost(bool isLocalNetworkLowCost);
+    TrafficStats getNetworkStats();
+    void fillCallStats(CallStats &callStats);
+    void logCurrentNetworkState();
+
+private:
+    void checkConnectionTimeout();
+	void candidateGathered(cricket::IceTransportInternal *transport, const cricket::Candidate &candidate);
+	void candidateGatheringState(cricket::IceTransportInternal *transport);
+	void transportStateChanged(cricket::IceTransportInternal *transport);
+	void transportReadyToSend(cricket::IceTransportInternal *transport);
+	void transportPacketReceived(rtc::PacketTransportInternal *transport, const char *bytes, size_t size, const int64_t &timestamp, int unused);
+    void transportRouteChanged(absl::optional<rtc::NetworkRoute> route);
+    void addTrafficStats(int64_t byteCount, bool isIncoming);
+
+	rtc::Thread *_thread = nullptr;
+    bool _enableP2P = false;
+    bool _enableTCP = false;
+    bool _enableStunMarking = false;
+    std::vector<RtcServer> _rtcServers;
+    std::unique_ptr<Proxy> _proxy;
+	EncryptedConnection _transport;
+	bool _isOutgoing = false;
+	std::function<void(const NetworkManager::State &)> _stateUpdated;
+	std::function<void(DecryptedMessage &&)> _transportMessageReceived;
+	std::function<void(Message &&)> _sendSignalingMessage;
+
+    std::unique_ptr<rtc::NetworkMonitorFactory> _networkMonitorFactory;
+	std::unique_ptr<rtc::BasicPacketSocketFactory> _socketFactory;
+	std::unique_ptr<rtc::BasicNetworkManager> _networkManager;
+    std::unique_ptr<webrtc::TurnCustomizer> _turnCustomizer;
+    std::unique_ptr<cricket::RelayPortFactoryInterface> _relayPortFactory;
+	std::unique_ptr<cricket::BasicPortAllocator> _portAllocator;
+	std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface> _asyncResolverFactory;
+	std::unique_ptr<cricket::P2PTransportChannel> _transportChannel;
+
+    PeerIceParameters _localIceParameters;
+    absl::optional<PeerIceParameters> _remoteIceParameters;
+    
+    bool _isLocalNetworkLowCost = false;
+    int64_t _lastNetworkActivityMs = 0;
+    InterfaceTrafficStats _trafficStatsWifi;
+    InterfaceTrafficStats _trafficStatsCellular;
+    
+    absl::optional<CallStatsConnectionEndpointType> _currentEndpointType;
+    std::vector<CallStatsNetworkRecord> _networkRecords;
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/PlatformContext.h

@@ -0,0 +1,15 @@
+#ifndef TGCALLS_PLATFORM_CONTEXT_H
+#define TGCALLS_PLATFORM_CONTEXT_H
+
+namespace tgcalls {
+
+class PlatformContext {
+
+public:
+    virtual ~PlatformContext() = default;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/SctpDataChannelProviderInterfaceImpl.cpp

@@ -0,0 +1,160 @@
+#include "SctpDataChannelProviderInterfaceImpl.h"
+
+#include "p2p/base/dtls_transport.h"
+#include "api/transport/field_trial_based_config.h"
+#include "FieldTrialsConfig.h"
+
+namespace tgcalls {
+
+SctpDataChannelProviderInterfaceImpl::SctpDataChannelProviderInterfaceImpl(
+    rtc::PacketTransportInternal *transportChannel,
+    bool isOutgoing,
+    std::function<void(bool)> onStateChanged,
+    std::function<void()> onTerminated,
+    std::function<void(std::string const &)> onMessageReceived,
+    std::shared_ptr<Threads> threads
+) :
+_weakFactory(this),
+_threads(std::move(threads)),
+_onStateChanged(onStateChanged),
+_onTerminated(onTerminated),
+_onMessageReceived(onMessageReceived) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    _sctpTransportFactory.reset(new cricket::SctpTransportFactory(_threads->getNetworkThread()));
+
+    _sctpTransport = _sctpTransportFactory->CreateSctpTransport(transportChannel);
+    _sctpTransport->SetDataChannelSink(this);
+
+    // TODO: should we disconnect the data channel sink?
+
+    webrtc::InternalDataChannelInit dataChannelInit;
+    dataChannelInit.id = 0;
+    dataChannelInit.open_handshake_role = isOutgoing ? webrtc::InternalDataChannelInit::kOpener : webrtc::InternalDataChannelInit::kAcker;
+    
+    _dataChannel = webrtc::SctpDataChannel::Create(
+        _weakFactory.GetWeakPtr(),
+        "data",
+        true,
+        dataChannelInit,
+        _threads->getNetworkThread(),
+        _threads->getNetworkThread()
+    );
+
+    _dataChannel->RegisterObserver(this);
+    
+    AddSctpDataStream(webrtc::StreamId(0));
+}
+
+SctpDataChannelProviderInterfaceImpl::~SctpDataChannelProviderInterfaceImpl() {
+    assert(_threads->getNetworkThread()->IsCurrent());
+    
+    _weakFactory.InvalidateWeakPtrs();
+
+    _dataChannel->UnregisterObserver();
+    _dataChannel->Close();
+    _dataChannel = nullptr;
+
+    _sctpTransport = nullptr;
+    _sctpTransportFactory.reset();
+}
+
+bool SctpDataChannelProviderInterfaceImpl::IsOkToCallOnTheNetworkThread() {
+    return true;
+}
+
+void SctpDataChannelProviderInterfaceImpl::sendDataChannelMessage(std::string const &message) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    if (_isDataChannelOpen) {
+        RTC_LOG(LS_INFO) << "Outgoing DataChannel message: " << message;
+
+        webrtc::DataBuffer buffer(message);
+        _dataChannel->Send(buffer);
+    } else {
+        RTC_LOG(LS_INFO) << "Could not send an outgoing DataChannel message: the channel is not open";
+    }
+}
+
+void SctpDataChannelProviderInterfaceImpl::OnStateChange() {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    auto state = _dataChannel->state();
+    bool isDataChannelOpen = state == webrtc::DataChannelInterface::DataState::kOpen;
+    if (_isDataChannelOpen != isDataChannelOpen) {
+        _isDataChannelOpen = isDataChannelOpen;
+        _onStateChanged(_isDataChannelOpen);
+    }
+}
+
+void SctpDataChannelProviderInterfaceImpl::OnMessage(const webrtc::DataBuffer& buffer) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    if (!buffer.binary) {
+        std::string messageText(buffer.data.data(), buffer.data.data() + buffer.data.size());
+        RTC_LOG(LS_INFO) << "Incoming DataChannel message: " << messageText;
+
+        _onMessageReceived(messageText);
+    }
+}
+
+void SctpDataChannelProviderInterfaceImpl::updateIsConnected(bool isConnected) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    if (isConnected) {
+        if (!_isSctpTransportStarted) {
+            _isSctpTransportStarted = true;
+            _sctpTransport->Start(5000, 5000, 262144);
+        }
+    }
+}
+
+void SctpDataChannelProviderInterfaceImpl::OnReadyToSend() {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    _dataChannel->OnTransportReady();
+}
+
+void SctpDataChannelProviderInterfaceImpl::OnTransportClosed(webrtc::RTCError error) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    if (_onTerminated) {
+        _onTerminated();
+    }
+}
+
+void SctpDataChannelProviderInterfaceImpl::OnDataReceived(int channel_id, webrtc::DataMessageType type, const rtc::CopyOnWriteBuffer& buffer) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    _dataChannel->OnDataReceived(type, buffer);
+}
+
+webrtc::RTCError SctpDataChannelProviderInterfaceImpl::SendData(
+    webrtc::StreamId sid,
+    const webrtc::SendDataParams& params,
+    const rtc::CopyOnWriteBuffer& payload
+) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    return _sctpTransport->SendData(sid.stream_id_int(), params, payload);
+}
+
+void SctpDataChannelProviderInterfaceImpl::AddSctpDataStream(webrtc::StreamId sid) {
+  assert(_threads->getNetworkThread()->IsCurrent());
+
+    _sctpTransport->OpenStream(sid.stream_id_int());
+}
+
+void SctpDataChannelProviderInterfaceImpl::RemoveSctpDataStream(webrtc::StreamId sid) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    _threads->getNetworkThread()->BlockingCall([this, sid]() {
+        _sctpTransport->ResetStream(sid.stream_id_int());
+    });
+}
+
+void SctpDataChannelProviderInterfaceImpl::OnChannelStateChanged(webrtc::SctpDataChannel *data_channel, webrtc::DataChannelInterface::DataState state) {
+    
+}
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/SctpDataChannelProviderInterfaceImpl.h

@@ -0,0 +1,74 @@
+#ifndef TGCALLS_SCTP_DATA_CHANNEL_PROVIDER_IMPL_H
+#define TGCALLS_SCTP_DATA_CHANNEL_PROVIDER_IMPL_H
+
+#include "rtc_base/weak_ptr.h"
+#include "api/turn_customizer.h"
+#include "api/data_channel_interface.h"
+#include "pc/sctp_data_channel.h"
+#include "media/sctp/sctp_transport_factory.h"
+
+#include "StaticThreads.h"
+
+namespace cricket {
+class DtlsTransport;
+} // namespace cricket
+
+namespace tgcalls {
+
+class SctpDataChannelProviderInterfaceImpl : public sigslot::has_slots<>, public webrtc::SctpDataChannelControllerInterface, public webrtc::DataChannelObserver, public webrtc::DataChannelSink {
+public:
+    SctpDataChannelProviderInterfaceImpl(
+        rtc::PacketTransportInternal *transportChannel,
+        bool isOutgoing,
+        std::function<void(bool)> onStateChanged,
+        std::function<void()> onTerminated,
+        std::function<void(std::string const &)> onMessageReceived,
+        std::shared_ptr<Threads> threads
+    );
+    virtual ~SctpDataChannelProviderInterfaceImpl();
+
+    virtual bool IsOkToCallOnTheNetworkThread() override;
+    
+    void updateIsConnected(bool isConnected);
+    void sendDataChannelMessage(std::string const &message);
+
+    virtual void OnStateChange() override;
+    virtual void OnMessage(const webrtc::DataBuffer& buffer) override;
+    virtual webrtc::RTCError SendData(
+        webrtc::StreamId sid,
+        const webrtc::SendDataParams& params,
+        const rtc::CopyOnWriteBuffer& payload) override;
+    
+    virtual void AddSctpDataStream(webrtc::StreamId sid) override;
+    virtual void RemoveSctpDataStream(webrtc::StreamId sid) override;
+    virtual void OnChannelStateChanged(webrtc::SctpDataChannel *data_channel, webrtc::DataChannelInterface::DataState state) override;
+
+    virtual void OnDataReceived(int channel_id,
+                                webrtc::DataMessageType type,
+                                const rtc::CopyOnWriteBuffer& buffer) override;
+    virtual void OnReadyToSend() override;
+    virtual void OnTransportClosed(webrtc::RTCError error) override;
+
+    // Unused
+    virtual void OnChannelClosing(int channel_id) override{}
+    virtual void OnChannelClosed(int channel_id) override{}
+
+private:
+    rtc::WeakPtrFactory<SctpDataChannelProviderInterfaceImpl> _weakFactory;
+    std::shared_ptr<Threads> _threads;
+    std::function<void(bool)> _onStateChanged;
+    std::function<void()> _onTerminated;
+    std::function<void(std::string const &)> _onMessageReceived;
+
+    std::unique_ptr<cricket::SctpTransportFactory> _sctpTransportFactory;
+    std::unique_ptr<cricket::SctpTransportInternal> _sctpTransport;
+    webrtc::scoped_refptr<webrtc::SctpDataChannel> _dataChannel;
+
+    bool _isSctpTransportStarted = false;
+    bool _isDataChannelOpen = false;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/StaticThreads.cpp

@@ -0,0 +1,144 @@
+#include "StaticThreads.h"
+
+#include "rtc_base/thread.h"
+#include "call/call.h"
+
+#include <mutex>
+#include <algorithm>
+
+namespace tgcalls {
+
+template <class ValueT, class CreatorT>
+class Pool : public std::enable_shared_from_this<Pool<ValueT, CreatorT>> {
+  struct Entry {
+    std::unique_ptr<ValueT> value;
+    size_t refcnt;
+
+    bool operator < (const Entry &other) const {
+      return refcnt < other.refcnt;
+    }
+  };
+
+public:
+  explicit Pool(CreatorT creator) : creator_(std::move(creator)) {
+  }
+  std::shared_ptr<ValueT> get() {
+    std::unique_lock<std::mutex> lock(mutex_);
+    set_pool_size_locked(1);
+    auto i = std::min_element(entries_.begin(), entries_.end()) - entries_.begin();
+    return std::shared_ptr<ValueT>(entries_[i].value.get(),
+      [i, self = this->shared_from_this()](auto *ptr) {
+        self->dec_ref(i);
+      });
+  }
+
+  void set_pool_size(size_t size) {
+    std::unique_lock<std::mutex> lock(mutex_);
+    set_pool_size_locked(size);
+  }
+
+  void dec_ref(size_t i) {
+    std::unique_lock<std::mutex> lock(mutex_);
+    entries_.at(i).refcnt--;
+  }
+
+private:
+  std::mutex mutex_;
+  std::vector<Entry> entries_;
+
+  CreatorT creator_;
+
+  void set_pool_size_locked(size_t size) {
+    for (size_t i = entries_.size(); i < size; i++) {
+      entries_.emplace_back(Entry{creator_(i + 1), 0});
+    }
+  }
+};
+
+class ThreadsImpl : public Threads {
+  using Thread = std::unique_ptr<rtc::Thread>;
+public:
+  explicit ThreadsImpl(size_t i) {
+    auto suffix = i == 0 ? "" : "#" + std::to_string(i);
+    media_ = create("tgc-media" + suffix);
+    worker_ = create("tgc-work" + suffix);
+    network_ = create_network("tgc-net" + suffix);
+      
+    media_->AllowInvokesToThread(worker_.get());
+    media_->AllowInvokesToThread(network_.get());
+    worker_->AllowInvokesToThread(network_.get());
+      
+    //network_->DisallowAllInvokes();
+    //worker_->DisallowAllInvokes();
+    //worker_->AllowInvokesToThread(network_.get());
+  }
+
+  rtc::Thread *getNetworkThread() override {
+    return network_.get();
+  }
+  rtc::Thread *getMediaThread() override {
+    return media_.get();
+  }
+  rtc::Thread *getWorkerThread() override {
+    return worker_.get();
+  }
+
+private:
+  Thread network_;
+  Thread media_;
+  Thread worker_;
+
+  static Thread create(const std::string &name) {
+    return init(std::unique_ptr<rtc::Thread>(rtc::Thread::Create()), name);
+  }
+  static Thread create_network(const std::string &name) {
+    return init(std::unique_ptr<rtc::Thread>(rtc::Thread::CreateWithSocketServer()), name);
+  }
+
+  static Thread init(Thread value, const std::string &name) {
+    value->SetName(name, nullptr);
+    value->Start();
+    return value;
+  }
+};
+
+class ThreadsCreator {
+public:
+  std::unique_ptr<Threads> operator()(size_t i) {
+   return std::make_unique<ThreadsImpl>(i);
+  }
+};
+
+Pool<Threads, ThreadsCreator> &get_pool() {
+  static auto pool = std::make_shared<Pool<Threads, ThreadsCreator>>(ThreadsCreator());
+  return *pool;
+}
+
+void Threads::setPoolSize(size_t size){
+  get_pool().set_pool_size(size);
+}
+std::shared_ptr<Threads> Threads::getThreads(){
+  return get_pool().get();
+}
+
+namespace StaticThreads {
+
+rtc::Thread *getNetworkThread() {
+  return getThreads()->getNetworkThread();
+}
+
+rtc::Thread *getMediaThread() {
+  return getThreads()->getMediaThread();
+}
+
+rtc::Thread *getWorkerThread() {
+  return getThreads()->getWorkerThread();
+}
+
+std::shared_ptr<Threads> &getThreads() {
+  static std::shared_ptr<Threads> threads = std::make_shared<ThreadsImpl>(0);
+  return threads;
+}
+};
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/StaticThreads.h

@@ -0,0 +1,31 @@
+#pragma once
+
+#include <cstddef>
+#include <memory>
+
+namespace rtc {
+class Thread;
+}
+
+namespace tgcalls {
+
+class Threads {
+public:
+  virtual ~Threads() = default;
+  virtual rtc::Thread *getNetworkThread() = 0;
+  virtual rtc::Thread *getMediaThread() = 0;
+  virtual rtc::Thread *getWorkerThread() = 0;
+
+  // it is not possible to decrease pool size
+  static void setPoolSize(size_t size);
+  static std::shared_ptr<Threads> getThreads();
+};
+
+namespace StaticThreads {
+rtc::Thread *getNetworkThread();
+rtc::Thread *getMediaThread();
+rtc::Thread *getWorkerThread();
+std::shared_ptr<Threads> &getThreads();
+}
+
+};

Telegram/ThirdParty/tgcalls/tgcalls/Stats.h

@@ -0,0 +1,30 @@
+#ifndef TGCALLS_STATS_H
+#define TGCALLS_STATS_H
+
+namespace tgcalls {
+
+enum class CallStatsConnectionEndpointType {
+    ConnectionEndpointP2P = 0,
+    ConnectionEndpointTURN = 1
+};
+
+struct CallStatsNetworkRecord {
+    int32_t timestamp = 0;
+    CallStatsConnectionEndpointType endpointType = CallStatsConnectionEndpointType::ConnectionEndpointP2P;
+    bool isLowCost = false;
+};
+
+struct CallStatsBitrateRecord {
+    int32_t timestamp = 0;
+    int32_t bitrate = 0;
+};
+
+struct CallStats {
+    std::string outgoingCodec;
+    std::vector<CallStatsNetworkRecord> networkRecords;
+    std::vector<CallStatsBitrateRecord> bitrateRecords;
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/ThreadLocalObject.cpp

@@ -0,0 +1 @@
+

Telegram/ThirdParty/tgcalls/tgcalls/ThreadLocalObject.h

@@ -0,0 +1,58 @@
+#ifndef TGCALLS_THREAD_LOCAL_OBJECT_H
+#define TGCALLS_THREAD_LOCAL_OBJECT_H
+
+#include "rtc_base/thread.h"
+
+#include <functional>
+#include <memory>
+
+namespace tgcalls {
+
+template <typename T>
+class ThreadLocalObject {
+public:
+	template <
+		typename Generator,
+		typename = std::enable_if_t<std::is_same<std::shared_ptr<T>, decltype(std::declval<Generator>()())>::value>>
+	ThreadLocalObject(rtc::Thread *thread, Generator &&generator) :
+	_thread(thread),
+	_valueHolder(std::make_unique<ValueHolder>()) {
+		assert(_thread != nullptr);
+		_thread->PostTask([valueHolder = _valueHolder.get(), generator = std::forward<Generator>(generator)]() mutable {
+			valueHolder->_value = generator();
+		});
+	}
+
+	~ThreadLocalObject() {
+		_thread->PostTask([valueHolder = std::move(_valueHolder)](){
+			valueHolder->_value.reset();
+		});
+	}
+
+	template <typename FunctorT>
+	void perform(FunctorT &&functor) {
+		_thread->PostTask([valueHolder = _valueHolder.get(), f = std::forward<FunctorT>(functor)]() mutable {
+			assert(valueHolder->_value != nullptr);
+			f(valueHolder->_value.get());
+		});
+	}
+
+	T *getSyncAssumingSameThread() {
+		assert(_thread->IsCurrent());
+		assert(_valueHolder->_value != nullptr);
+		return _valueHolder->_value.get();
+	}
+
+private:
+	struct ValueHolder {
+		std::shared_ptr<T> _value;
+	};
+
+	rtc::Thread *_thread = nullptr;
+	std::unique_ptr<ValueHolder> _valueHolder;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/TurnCustomizerImpl.cpp

@@ -0,0 +1,21 @@
+#include "TurnCustomizerImpl.h"
+
+#include "api/transport/stun.h"
+
+namespace tgcalls {
+
+TurnCustomizerImpl::TurnCustomizerImpl() {
+}
+
+TurnCustomizerImpl::~TurnCustomizerImpl() {
+}
+
+void TurnCustomizerImpl::MaybeModifyOutgoingStunMessage(cricket::PortInterface* port, cricket::StunMessage* message) {
+    message->AddAttribute(std::make_unique<cricket::StunByteStringAttribute>(cricket::STUN_ATTR_SOFTWARE, "Telegram "));
+}
+
+bool TurnCustomizerImpl::AllowChannelData(cricket::PortInterface* port, const void *data, size_t size, bool payload) {
+    return true;
+}
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/TurnCustomizerImpl.h

@@ -0,0 +1,19 @@
+#ifndef TGCALLS_TURN_CUSTOMIZER_H
+#define TGCALLS_TURN_CUSTOMIZER_H
+
+#include "api/turn_customizer.h"
+
+namespace tgcalls {
+
+class TurnCustomizerImpl : public webrtc::TurnCustomizer {
+public:
+    TurnCustomizerImpl();
+    virtual ~TurnCustomizerImpl();
+
+    void MaybeModifyOutgoingStunMessage(cricket::PortInterface* port, cricket::StunMessage* message) override;
+    bool AllowChannelData(cricket::PortInterface* port, const void *data, size_t size, bool payload) override;
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/VideoCaptureInterface.cpp

@@ -0,0 +1,15 @@
+#include "VideoCaptureInterface.h"
+
+#include "VideoCaptureInterfaceImpl.h"
+
+namespace tgcalls {
+
+std::unique_ptr<VideoCaptureInterface> VideoCaptureInterface::Create(
+   std::shared_ptr<Threads> threads, std::string deviceId, bool isScreenCapture,
+   std::shared_ptr<PlatformContext> platformContext) {
+	return std::make_unique<VideoCaptureInterfaceImpl>(deviceId, isScreenCapture, platformContext, std::move(threads));
+}
+
+VideoCaptureInterface::~VideoCaptureInterface() = default;
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/VideoCaptureInterface.h

@@ -0,0 +1,62 @@
+#ifndef TGCALLS_VIDEO_CAPTURE_INTERFACE_H
+#define TGCALLS_VIDEO_CAPTURE_INTERFACE_H
+
+#include <string>
+#include <memory>
+#include <functional>
+
+namespace rtc {
+template <typename VideoFrameT>
+class VideoSinkInterface;
+} // namespace rtc
+
+namespace webrtc {
+class VideoFrame;
+} // namespace webrtc
+
+namespace tgcalls {
+
+class PlatformContext;
+class Threads;
+
+enum class VideoState {
+	Inactive,
+	Paused,
+	Active,
+};
+
+
+class VideoCaptureInterface {
+protected:
+	VideoCaptureInterface() = default;
+
+public:
+	static std::unique_ptr<VideoCaptureInterface> Create(
+                std::shared_ptr<Threads> threads,
+                std::string deviceId = std::string(),
+                bool isScreenCapture = false,
+		std::shared_ptr<PlatformContext> platformContext = nullptr);
+
+	virtual ~VideoCaptureInterface();
+
+	virtual void switchToDevice(std::string deviceId, bool isScreenCapture) = 0;
+	virtual void setState(VideoState state) = 0;
+    virtual void setPreferredAspectRatio(float aspectRatio) = 0;
+	virtual void setOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) = 0;
+    virtual void setOnFatalError(std::function<void()> error) {
+      // TODO: make this function pure virtual when everybody implements it.
+    }
+    virtual void setOnPause(std::function<void(bool)> pause) {
+      // TODO: make this function pure virtual when everybody implements it.
+    }
+    virtual void setOnIsActiveUpdated(std::function<void(bool)> onIsActiveUpdated) {
+      // TODO: make this function pure virtual when everybody implements it.
+    }
+    virtual void withNativeImplementation(std::function<void(void *)> completion) {
+        completion(nullptr);
+    }
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/VideoCaptureInterfaceImpl.cpp

@@ -0,0 +1,227 @@
+#include "VideoCaptureInterfaceImpl.h"
+
+#include "VideoCapturerInterface.h"
+#include "Manager.h"
+#include "MediaManager.h"
+#include "platform/PlatformInterface.h"
+#include "StaticThreads.h"
+
+namespace tgcalls {
+
+VideoCaptureInterfaceObject::VideoCaptureInterfaceObject(std::string deviceId, bool isScreenCapture, std::shared_ptr<PlatformContext> platformContext, Threads &threads)
+: _videoSource(PlatformInterface::SharedInstance()->makeVideoSource(threads.getMediaThread(), threads.getWorkerThread())) {
+	_platformContext = platformContext;
+
+	switchToDevice(deviceId, isScreenCapture);
+}
+
+VideoCaptureInterfaceObject::~VideoCaptureInterfaceObject() {
+	if (_videoCapturer) {
+		_videoCapturer->setUncroppedOutput(nullptr);
+	}
+}
+
+webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface> VideoCaptureInterfaceObject::source() {
+	return _videoSource;
+}
+
+int VideoCaptureInterfaceObject::getRotation() {
+    if (_videoCapturer) {
+        return _videoCapturer->getRotation();
+    } else {
+        return 0;
+    }
+}
+
+bool VideoCaptureInterfaceObject::isScreenCapture() {
+    return _isScreenCapture;
+}
+
+void VideoCaptureInterfaceObject::switchToDevice(std::string deviceId, bool isScreenCapture) {
+    if (_videoCapturer) {
+		_videoCapturer->setUncroppedOutput(nullptr);
+    }
+    _isScreenCapture = isScreenCapture;
+	if (_videoSource) {
+        //this should outlive the capturer
+        _videoCapturer = nullptr;
+		_videoCapturer = PlatformInterface::SharedInstance()->makeVideoCapturer(_videoSource, deviceId, [this](VideoState state) {
+			if (this->_stateUpdated) {
+				this->_stateUpdated(state);
+			}
+            if (this->_onIsActiveUpdated) {
+                switch (state) {
+                    case VideoState::Active: {
+                        this->_onIsActiveUpdated(true);
+                        break;
+                    }
+                    default: {
+                        this->_onIsActiveUpdated(false);
+                        break;
+                    }
+                }
+            }
+        }, [this](PlatformCaptureInfo info) {
+            if (this->_shouldBeAdaptedToReceiverAspectRate != info.shouldBeAdaptedToReceiverAspectRate) {
+                this->_shouldBeAdaptedToReceiverAspectRate = info.shouldBeAdaptedToReceiverAspectRate;
+            }
+            if (this->_rotationUpdated) {
+                this->_rotationUpdated(info.rotation);
+            }
+            this->updateAspectRateAdaptation();
+        }, _platformContext, _videoCapturerResolution);
+	}
+	if (_videoCapturer) {
+		if (_preferredAspectRatio > 0) {
+			_videoCapturer->setPreferredCaptureAspectRatio(_preferredAspectRatio);
+		}
+		if (const auto currentUncroppedSink = _currentUncroppedSink.lock()) {
+			_videoCapturer->setUncroppedOutput(currentUncroppedSink);
+		}
+        if (_onFatalError) {
+            _videoCapturer->setOnFatalError(_onFatalError);
+        }
+        if (_onPause) {
+            _videoCapturer->setOnPause(_onPause);
+        }
+		_videoCapturer->setState(_state);
+	}
+}
+
+void VideoCaptureInterfaceObject::withNativeImplementation(std::function<void(void *)> completion) {
+    if (_videoCapturer) {
+        _videoCapturer->withNativeImplementation(completion);
+    } else {
+        completion(nullptr);
+    }
+}
+
+void VideoCaptureInterfaceObject::setState(VideoState state) {
+	if (_state != state) {
+		_state = state;
+		if (_videoCapturer) {
+			_videoCapturer->setState(state);
+		}
+	}
+}
+
+void VideoCaptureInterfaceObject::setPreferredAspectRatio(float aspectRatio) {
+    _preferredAspectRatio = aspectRatio;
+    updateAspectRateAdaptation();
+}
+
+void VideoCaptureInterfaceObject::updateAspectRateAdaptation() {
+    if (_videoCapturer) {
+        if (_videoCapturerResolution.first != 0 && _videoCapturerResolution.second != 0) {
+            if (_preferredAspectRatio > 0.01 && _shouldBeAdaptedToReceiverAspectRate) {
+                float originalWidth = (float)_videoCapturerResolution.first;
+                float originalHeight = (float)_videoCapturerResolution.second;
+
+                float aspectRatio = _preferredAspectRatio;
+
+                float width = (originalWidth > aspectRatio * originalHeight)
+                    ? int(std::round(aspectRatio * originalHeight))
+                    : originalWidth;
+                float height = (originalWidth > aspectRatio * originalHeight)
+                    ? originalHeight
+                    : int(std::round(originalHeight / aspectRatio));
+
+                PlatformInterface::SharedInstance()->adaptVideoSource(_videoSource, (int)width, (int)height, 25);
+            } else {
+                PlatformInterface::SharedInstance()->adaptVideoSource(_videoSource, _videoCapturerResolution.first, _videoCapturerResolution.second, 25);
+            }
+        }
+    }
+}
+
+void VideoCaptureInterfaceObject::setOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+	if (_videoCapturer) {
+		_videoCapturer->setUncroppedOutput(sink);
+	}
+	_currentUncroppedSink = sink;
+}
+
+void VideoCaptureInterfaceObject::setOnFatalError(std::function<void()> error) {
+    if (_videoCapturer) {
+        _videoCapturer->setOnFatalError(error);
+    }
+    _onFatalError = error;
+}
+void VideoCaptureInterfaceObject::setOnPause(std::function<void(bool)> pause) {
+    if (_videoCapturer) {
+        _videoCapturer->setOnPause(pause);
+    }
+    _onPause = pause;
+}
+
+void VideoCaptureInterfaceObject::setOnIsActiveUpdated(std::function<void(bool)> onIsActiveUpdated) {
+    _onIsActiveUpdated = onIsActiveUpdated;
+}
+
+void VideoCaptureInterfaceObject::setStateUpdated(std::function<void(VideoState)> stateUpdated) {
+	_stateUpdated = stateUpdated;
+}
+
+void VideoCaptureInterfaceObject::setRotationUpdated(std::function<void(int)> rotationUpdated) {
+    _rotationUpdated = rotationUpdated;
+}
+
+VideoCaptureInterfaceImpl::VideoCaptureInterfaceImpl(std::string deviceId, bool isScreenCapture, std::shared_ptr<PlatformContext> platformContext, std::shared_ptr<Threads> threads) :
+_impl(threads->getMediaThread(), [deviceId, isScreenCapture, platformContext, threads]() {
+	return std::make_shared<VideoCaptureInterfaceObject>(deviceId, isScreenCapture, platformContext, *threads);
+}) {
+}
+
+VideoCaptureInterfaceImpl::~VideoCaptureInterfaceImpl() = default;
+
+void VideoCaptureInterfaceImpl::switchToDevice(std::string deviceId, bool isScreenCapture) {
+	_impl.perform([deviceId, isScreenCapture](VideoCaptureInterfaceObject *impl) {
+		impl->switchToDevice(deviceId, isScreenCapture);
+	});
+}
+
+void VideoCaptureInterfaceImpl::withNativeImplementation(std::function<void(void *)> completion) {
+    _impl.perform([completion](VideoCaptureInterfaceObject *impl) {
+        impl->withNativeImplementation(completion);
+    });
+}
+
+void VideoCaptureInterfaceImpl::setState(VideoState state) {
+	_impl.perform([state](VideoCaptureInterfaceObject *impl) {
+		impl->setState(state);
+	});
+}
+
+void VideoCaptureInterfaceImpl::setPreferredAspectRatio(float aspectRatio) {
+    _impl.perform([aspectRatio](VideoCaptureInterfaceObject *impl) {
+        impl->setPreferredAspectRatio(aspectRatio);
+    });
+}
+void VideoCaptureInterfaceImpl::setOnFatalError(std::function<void()> error) {
+    _impl.perform([error](VideoCaptureInterfaceObject *impl) {
+        impl->setOnFatalError(error);
+    });
+}
+void VideoCaptureInterfaceImpl::setOnPause(std::function<void(bool)> pause) {
+    _impl.perform([pause](VideoCaptureInterfaceObject *impl) {
+        impl->setOnPause(pause);
+    });
+}
+
+void VideoCaptureInterfaceImpl::setOnIsActiveUpdated(std::function<void(bool)> onIsActiveUpdated) {
+    _impl.perform([onIsActiveUpdated](VideoCaptureInterfaceObject *impl) {
+        impl->setOnIsActiveUpdated(onIsActiveUpdated);
+    });
+}
+
+void VideoCaptureInterfaceImpl::setOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+	_impl.perform([sink](VideoCaptureInterfaceObject *impl) {
+		impl->setOutput(sink);
+	});
+}
+
+ThreadLocalObject<VideoCaptureInterfaceObject> *VideoCaptureInterfaceImpl::object() {
+	return &_impl;
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/VideoCaptureInterfaceImpl.h

@@ -0,0 +1,76 @@
+#ifndef TGCALLS_VIDEO_CAPTURE_INTERFACE_IMPL_H
+#define TGCALLS_VIDEO_CAPTURE_INTERFACE_IMPL_H
+
+#include "VideoCaptureInterface.h"
+#include <memory>
+#include "ThreadLocalObject.h"
+#include "api/media_stream_interface.h"
+#include "platform/PlatformInterface.h"
+
+namespace tgcalls {
+
+class VideoCapturerInterface;
+class Threads;
+
+class VideoCaptureInterfaceObject {
+public:
+	VideoCaptureInterfaceObject(std::string deviceId, bool isScreenCapture, std::shared_ptr<PlatformContext> platformContext, Threads &threads);
+	~VideoCaptureInterfaceObject();
+
+	void switchToDevice(std::string deviceId, bool isScreenCapture);
+    void withNativeImplementation(std::function<void(void *)> completion);
+	void setState(VideoState state);
+    void setPreferredAspectRatio(float aspectRatio);
+	void setOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+	void setStateUpdated(std::function<void(VideoState)> stateUpdated);
+    void setRotationUpdated(std::function<void(int)> rotationUpdated);
+    void setOnFatalError(std::function<void()> error);
+    void setOnPause(std::function<void(bool)> pause);
+    void setOnIsActiveUpdated(std::function<void(bool)> onIsActiveUpdated);
+    webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source();
+    int getRotation();
+    bool isScreenCapture();
+
+private:
+    void updateAspectRateAdaptation();
+
+    webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface> _videoSource;
+	std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> _currentUncroppedSink;
+	std::shared_ptr<PlatformContext> _platformContext;
+    std::pair<int, int> _videoCapturerResolution;
+	std::unique_ptr<VideoCapturerInterface> _videoCapturer;
+	std::function<void(VideoState)> _stateUpdated;
+    std::function<void()> _onFatalError;
+    std::function<void(bool)> _onPause;
+    std::function<void(bool)> _onIsActiveUpdated;
+    std::function<void(int)> _rotationUpdated;
+	VideoState _state = VideoState::Active;
+    float _preferredAspectRatio = 0.0f;
+    bool _shouldBeAdaptedToReceiverAspectRate = true;
+    bool _isScreenCapture = false;
+};
+
+class VideoCaptureInterfaceImpl : public VideoCaptureInterface {
+public:
+	VideoCaptureInterfaceImpl(std::string deviceId, bool isScreenCapture, std::shared_ptr<PlatformContext> platformContext, std::shared_ptr<Threads> threads);
+	virtual ~VideoCaptureInterfaceImpl();
+
+	void switchToDevice(std::string deviceId, bool isScreenCapture) override;
+    void withNativeImplementation(std::function<void(void *)> completion) override;
+	void setState(VideoState state) override;
+    void setPreferredAspectRatio(float aspectRatio) override;
+	void setOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) override;
+    void setOnFatalError(std::function<void()> error) override;
+    void setOnPause(std::function<void(bool)> pause) override;
+    void setOnIsActiveUpdated(std::function<void(bool)> onIsActiveUpdated) override;
+
+	ThreadLocalObject<VideoCaptureInterfaceObject> *object();
+
+private:
+	ThreadLocalObject<VideoCaptureInterfaceObject> _impl;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/VideoCapturerInterface.h

@@ -0,0 +1,42 @@
+#ifndef TGCALLS_VIDEO_CAPTURER_INTERFACE_H
+#define TGCALLS_VIDEO_CAPTURER_INTERFACE_H
+
+#include "Instance.h"
+
+#include <memory>
+#include <functional>
+
+namespace rtc {
+template <typename VideoFrameT>
+class VideoSinkInterface;
+} // namespace rtc
+
+namespace webrtc {
+class VideoFrame;
+} // namespace webrtc
+
+namespace tgcalls {
+
+class VideoCapturerInterface {
+public:
+	virtual ~VideoCapturerInterface() = default;
+
+	virtual void setState(VideoState state) = 0;
+	virtual void setPreferredCaptureAspectRatio(float aspectRatio) = 0;
+	virtual void setUncroppedOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) = 0;
+    virtual int getRotation() = 0;
+    virtual void setOnFatalError(std::function<void()> error) {
+      // TODO: make this function pure virtual when everybody implements it.
+    }
+    virtual void setOnPause(std::function<void(bool)> pause) {
+      // TODO: make this function pure virtual when everybody implements it.
+    }
+    virtual void withNativeImplementation(std::function<void(void *)> completion) {
+        completion(nullptr);
+    }
+    
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/desktop_capturer/DesktopCaptureSource.cpp

@@ -0,0 +1,68 @@
+//
+//  DesktopCaptureSource.m
+//  TgVoipWebrtc
+//
+//  Created by Mikhail Filimonov on 29.12.2020.
+//  Copyright © 2020 Mikhail Filimonov. All rights reserved.
+//
+
+#include "tgcalls/desktop_capturer/DesktopCaptureSource.h"
+
+namespace tgcalls {
+
+std::string DesktopCaptureSourceData::cachedKey() const {
+    return std::to_string(aspectSize.width)
+        + 'x'
+        + std::to_string(aspectSize.height)
+        + ':'
+        + std::to_string(fps)
+        + ':'
+        + (captureMouse ? '1' : '0');
+}
+
+DesktopCaptureSource::DesktopCaptureSource(
+	long long uniqueId,
+	std::string title,
+    bool isWindow)
+: _uniqueId(uniqueId)
+, _title(std::move(title))
+, _isWindow(isWindow) {
+}
+
+long long DesktopCaptureSource::uniqueId() const {
+    return _uniqueId;
+}
+
+bool DesktopCaptureSource::isWindow() const {
+    return _isWindow;
+}
+
+std::string DesktopCaptureSource::deviceIdKey() const {
+    return std::string("desktop_capturer_")
+        + (_isWindow ? "window_" : "screen_")
+        + std::to_string(uniqueId());
+}
+
+std::string DesktopCaptureSource::title() const {
+    return _isWindow ? _title : "Screen";
+}
+
+std::string DesktopCaptureSource::uniqueKey() const {
+    return std::to_string(_uniqueId)
+        + ':'
+        + (_isWindow ? "Window" : "Screen");
+}
+
+std::string DesktopCaptureSource::deviceIdKey() {
+    return static_cast<const DesktopCaptureSource*>(this)->deviceIdKey();
+}
+
+std::string DesktopCaptureSource::title() {
+	return static_cast<const DesktopCaptureSource*>(this)->title();
+}
+
+std::string DesktopCaptureSource::uniqueKey() {
+	return static_cast<const DesktopCaptureSource*>(this)->uniqueKey();
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/desktop_capturer/DesktopCaptureSource.h

@@ -0,0 +1,85 @@
+//
+//  DesktopCaptureSource.h
+//  TgVoipWebrtc
+//
+//  Created by Mikhail Filimonov on 29.12.2020.
+//  Copyright © 2020 Mikhail Filimonov. All rights reserved.
+//
+#ifndef TGCALLS_DESKTOP_CAPTURE_SOURCE_H__
+#define TGCALLS_DESKTOP_CAPTURE_SOURCE_H__
+
+#include <string>
+
+#ifdef WEBRTC_WIN
+// Compiler errors in conflicting Windows headers if not included here.
+#include <winsock2.h>
+#endif // WEBRTC_WIN
+
+namespace tgcalls {
+
+class VideoSource {
+public:
+	virtual ~VideoSource() = default;
+
+	virtual std::string deviceIdKey() = 0;
+	virtual std::string title() = 0;
+	virtual std::string uniqueKey() = 0;
+};
+
+struct DesktopSize {
+	int width = 0;
+	int height = 0;
+};
+
+struct DesktopCaptureSourceData {
+	DesktopSize aspectSize;
+	double fps = 24.;
+	bool captureMouse = true;
+
+	std::string cachedKey() const;
+};
+
+class DesktopCaptureSource : public VideoSource {
+public:
+	DesktopCaptureSource(
+		long long uniqueId,
+		std::string title,
+		bool isWindow);
+
+	static DesktopCaptureSource Invalid() {
+		return InvalidTag{};
+	}
+
+	long long uniqueId() const;
+	bool isWindow() const;
+
+	std::string deviceIdKey() const;
+	std::string title() const;
+	std::string uniqueKey() const;
+
+	bool valid() const {
+		return _valid;
+	}
+	explicit operator bool() const {
+		return _valid;
+	}
+
+private:
+	struct InvalidTag {};
+	DesktopCaptureSource(InvalidTag) : _valid(false) {
+	}
+
+	std::string deviceIdKey() override;
+	std::string title() override;
+	std::string uniqueKey() override;
+
+	long long _uniqueId = 0;
+	std::string _title;
+	bool _isWindow = false;
+	bool _valid = true;
+
+};
+
+} // namespace tgcalls
+
+#endif // TGCALLS_DESKTOP_CAPTURE_SOURCE_H__

Telegram/ThirdParty/tgcalls/tgcalls/desktop_capturer/DesktopCaptureSourceHelper.cpp

@@ -0,0 +1,461 @@
+//
+//  DesktopCaptureSourceHelper.m
+//  TgVoipWebrtc
+//
+//  Created by Mikhail Filimonov on 28.12.2020.
+//  Copyright © 2020 Mikhail Filimonov. All rights reserved.
+//
+
+#include "tgcalls/desktop_capturer/DesktopCaptureSourceHelper.h"
+
+#include <iostream>
+#include <memory>
+#include <algorithm>
+#include <chrono>
+#include <iostream>
+#include <vector>
+#include <functional>
+
+#include "tgcalls/desktop_capturer/DesktopCaptureSourceManager.h"
+#include "rtc_base/thread.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_frame.h"
+#include "modules/desktop_capture/desktop_and_cursor_composer.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "system_wrappers/include/clock.h"
+#include "api/video/i420_buffer.h"
+#include "third_party/libyuv/include/libyuv.h"
+
+#ifdef WEBRTC_MAC
+#import <QuartzCore/QuartzCore.h>
+#endif // WEBRTC_MAC
+
+namespace tgcalls {
+namespace {
+
+#ifdef WEBRTC_MAC
+class CaptureScheduler {
+public:
+    void runAsync(std::function<void()> method) {
+		dispatch_async(dispatch_get_main_queue(), ^{
+			method();
+		});
+    }
+    void runDelayed(int delayMs, std::function<void()> method) {
+        const auto time = dispatch_time(
+            DISPATCH_TIME_NOW,
+            ((long long)delayMs * NSEC_PER_SEC) / 1000);
+        dispatch_after(time, dispatch_get_main_queue(), ^{
+            method();
+        });
+    }
+};
+#else // WEBRTC_MAC
+rtc::Thread *GlobalCapturerThread() {
+    static auto result = [] {
+        auto thread = rtc::Thread::Create();
+        thread->SetName("WebRTC-DesktopCapturer", nullptr);
+        thread->Start();
+        return thread;
+    }();
+    return result.get();
+}
+
+class CaptureScheduler {
+public:
+    CaptureScheduler() : _thread(GlobalCapturerThread()) {
+    }
+
+    void runAsync(std::function<void()> method) {
+        _thread->PostTask(std::move(method));
+    }
+    void runDelayed(int delayMs, std::function<void()> method) {
+        _thread->PostDelayedTask(std::move(method), webrtc::TimeDelta::Millis(delayMs));
+    }
+
+private:
+    rtc::Thread *_thread;
+
+};
+#endif // WEBRTC_MAC
+
+class SourceFrameCallbackImpl : public webrtc::DesktopCapturer::Callback {
+public:
+    SourceFrameCallbackImpl(DesktopSize size, int fps);
+
+    void OnCaptureResult(
+        webrtc::DesktopCapturer::Result result,
+        std::unique_ptr<webrtc::DesktopFrame> frame) override;
+	void setOutput(
+		std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+	void setSecondaryOutput(
+		std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+    void setOnFatalError(std::function<void ()>);
+    void setOnPause(std::function<void (bool)>);
+private:
+    rtc::scoped_refptr<webrtc::I420Buffer> i420_buffer_;
+	std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> _sink;
+	std::shared_ptr<
+        rtc::VideoSinkInterface<webrtc::VideoFrame>> _secondarySink;
+    DesktopSize size_;
+    std::function<void ()> _onFatalError;
+    std::function<void (bool)> _onPause;
+};
+
+class DesktopSourceRenderer {
+public:
+    DesktopSourceRenderer(
+        CaptureScheduler &scheduler,
+        DesktopCaptureSource source,
+        DesktopCaptureSourceData data);
+
+    void start();
+    void stop();
+    void setOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+    void setSecondaryOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+    void loop();
+    void setOnFatalError(std::function<void ()>);
+    void setOnPause(std::function<void (bool)>);
+private:
+    CaptureScheduler &_scheduler;
+    std::unique_ptr<webrtc::DesktopCapturer> _capturer;
+    SourceFrameCallbackImpl _callback;
+    std::shared_ptr<bool> _timerGuard;
+    std::function<void()> _onFatalError;
+    std::function<void(bool)> _onPause;
+    bool _isRunning = false;
+    bool _fatalError = false;
+    bool _currentlyOnPause = false;
+    double _delayMs = 0.;
+
+};
+
+SourceFrameCallbackImpl::SourceFrameCallbackImpl(DesktopSize size, int fps)
+: size_(size) {
+}
+
+void SourceFrameCallbackImpl::OnCaptureResult(
+	    webrtc::DesktopCapturer::Result result,
+	    std::unique_ptr<webrtc::DesktopFrame> frame) {
+
+    const auto failed = (result != webrtc::DesktopCapturer::Result::SUCCESS)
+        || !frame
+        || frame->size().equals({ 1, 1 });
+    if (failed) {
+        if (result == webrtc::DesktopCapturer::Result::ERROR_PERMANENT) {
+            if (_onFatalError) {
+                _onFatalError();
+            }
+        } else if (_onPause) {
+            _onPause(true);
+        }
+        return;
+    } else if (_onPause) {
+        _onPause(false);
+    }
+
+    const auto frameSize = frame->size();
+    auto fittedSize = (frameSize.width() >= size_.width * 2
+        || frameSize.height() >= size_.height * 2)
+        ? DesktopSize{ frameSize.width() / 2, frameSize.height() / 2 }
+        : DesktopSize{ frameSize.width(), frameSize.height() };
+
+    fittedSize.width -= (fittedSize.width % 4);
+    fittedSize.height -= (fittedSize.height % 4);
+
+    const auto outputSize = webrtc::DesktopSize{
+        fittedSize.width,
+        fittedSize.height
+    };
+
+    webrtc::BasicDesktopFrame outputFrame{ outputSize };
+
+	const auto outputRect = webrtc::DesktopRect::MakeSize(outputSize);
+
+	const auto outputRectData = outputFrame.data() +
+        outputFrame.stride() * outputRect.top() +
+		webrtc::DesktopFrame::kBytesPerPixel * outputRect.left();
+
+
+	libyuv::ARGBScale(
+		frame->data(),
+		frame->stride(),
+		frame->size().width(),
+		frame->size().height(),
+        outputRectData,
+        outputFrame.stride(),
+        outputSize.width(),
+        outputSize.height(),
+		libyuv::kFilterBilinear);
+
+	int width = outputFrame.size().width();
+	int height = outputFrame.size().height();
+	int stride_y = width;
+	int stride_uv = (width + 1) / 2;
+
+	if (!i420_buffer_
+        || i420_buffer_->width() != width
+        || i420_buffer_->height() != height) {
+		i420_buffer_ = webrtc::I420Buffer::Create(
+            width,
+            height,
+            stride_y,
+            stride_uv,
+            stride_uv);
+	}
+
+	int i420Result = libyuv::ConvertToI420(
+        outputFrame.data(),
+		width * height,
+		i420_buffer_->MutableDataY(), i420_buffer_->StrideY(),
+		i420_buffer_->MutableDataU(), i420_buffer_->StrideU(),
+		i420_buffer_->MutableDataV(), i420_buffer_->StrideV(),
+		0, 0,
+		width, height,
+		width, height,
+		libyuv::kRotate0,
+		libyuv::FOURCC_ARGB);
+
+
+	assert(i420Result == 0);
+	(void)i420Result;
+	webrtc::VideoFrame nativeVideoFrame = webrtc::VideoFrame(
+		i420_buffer_,
+		webrtc::kVideoRotation_0,
+        webrtc::Clock::GetRealTimeClock()->CurrentTime().us());
+	if (const auto sink = _sink.get()) {
+		sink->OnFrame(nativeVideoFrame);
+	}
+	if (const auto sink = _secondarySink.get()) {
+		sink->OnFrame(nativeVideoFrame);
+	}
+}
+
+void SourceFrameCallbackImpl::setOutput(
+        std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+    _sink = std::move(sink);
+}
+
+void SourceFrameCallbackImpl::setOnFatalError(std::function<void ()> error) {
+    _onFatalError = error;
+}
+void SourceFrameCallbackImpl::setOnPause(std::function<void (bool)> pause) {
+    _onPause = pause;
+}
+
+void SourceFrameCallbackImpl::setSecondaryOutput(
+        std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+    _secondarySink = std::move(sink);
+}
+
+DesktopSourceRenderer::DesktopSourceRenderer(
+    CaptureScheduler &scheduler,
+    DesktopCaptureSource source,
+    DesktopCaptureSourceData data)
+: _scheduler(scheduler)
+, _callback(data.aspectSize, data.fps)
+, _delayMs(1000. / data.fps) {
+    _callback.setOnFatalError([=, this] {
+		stop();
+		_fatalError = true;
+		if (_onFatalError) _onFatalError();
+	});
+
+    _callback.setOnPause([=, this] (bool pause) {
+        bool previousOnPause = _currentlyOnPause;
+        _currentlyOnPause = pause;
+        if (previousOnPause != _currentlyOnPause) {
+            if (_onPause) _onPause(pause);
+        }
+    });
+
+    auto options = webrtc::DesktopCaptureOptions::CreateDefault();
+    options.set_disable_effects(true);
+    options.set_detect_updated_region(true);
+
+#ifdef WEBRTC_WIN
+    options.set_allow_directx_capturer(true);
+#elif defined WEBRTC_MAC
+    options.set_allow_iosurface(true);
+#elif defined WEBRTC_USE_PIPEWIRE
+    options.set_allow_pipewire(true);
+#endif // WEBRTC_WIN || WEBRTC_MAC
+
+    _capturer = webrtc::DesktopCapturer::CreateGenericCapturer(options);
+    if (!_capturer) {
+        if (source.isWindow()) {
+            _capturer = webrtc::DesktopCapturer::CreateWindowCapturer(options);
+        } else {
+            _capturer = webrtc::DesktopCapturer::CreateScreenCapturer(options);
+        }
+        if (!_capturer) {
+            _fatalError = true;
+            return;
+        }
+    }
+    if (data.captureMouse) {
+        _capturer = std::make_unique<webrtc::DesktopAndCursorComposer>(
+            std::move(_capturer),
+            options);
+    }
+    _capturer->SelectSource(source.uniqueId());
+    _capturer->Start(&_callback);
+}
+
+void DesktopSourceRenderer::start() {
+    if (!_capturer || _isRunning) {
+        return;
+    }
+//    ++GlobalCount;
+//#ifdef WEBRTC_MAC
+//    NSLog(@"current capture count: %d", GlobalCount);
+//#endif // WEBRTC_MAC
+
+    _isRunning = true;
+    _timerGuard = std::make_shared<bool>(true);
+    loop();
+}
+
+void DesktopSourceRenderer::stop() {
+//    if (_isRunning) {
+//        GlobalCount--;
+//
+//#ifdef WEBRTC_MAC
+//        NSLog(@"current capture count: %d", GlobalCount);
+//#endif // WEBRTC_MAC
+//    }
+    _isRunning = false;
+    _timerGuard = nullptr;
+}
+
+void DesktopSourceRenderer::loop() {
+    if (!_capturer || !_isRunning) {
+        return;
+    }
+
+    _capturer->CaptureFrame();
+    const auto guard = std::weak_ptr<bool>(_timerGuard);
+    _scheduler.runDelayed(_delayMs, [=, this] {
+        if (guard.lock()) {
+            loop();
+        }
+    });
+}
+
+void DesktopSourceRenderer::setOnFatalError(std::function<void ()> error) {
+    if (_fatalError) {
+        error();
+    } else {
+        _onFatalError = std::move(error);
+    }
+}
+
+void DesktopSourceRenderer::setOnPause(std::function<void (bool)> pause) {
+    if (_currentlyOnPause) {
+        pause(true);
+    }
+    _onPause = std::move(pause);
+}
+
+void DesktopSourceRenderer::setOutput(
+        std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+    _callback.setOutput(std::move(sink));
+}
+
+void DesktopSourceRenderer::setSecondaryOutput(
+        std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+    _callback.setSecondaryOutput(std::move(sink));
+}
+
+} // namespace
+
+struct DesktopCaptureSourceHelper::Renderer {
+    CaptureScheduler scheduler;
+    std::unique_ptr<DesktopSourceRenderer> renderer;
+};
+
+DesktopCaptureSource DesktopCaptureSourceForKey(
+	    const std::string &uniqueKey) {
+    if (!ShouldBeDesktopCapture(uniqueKey)) {
+		return DesktopCaptureSource::Invalid();
+    }
+    if (uniqueKey == "desktop_capturer_pipewire") {
+        return DesktopCaptureSource(0, "pipewire", false);
+    }
+    const auto windowPrefix = std::string("desktop_capturer_window_");
+    const auto isWindow = (uniqueKey.find(windowPrefix) == 0);
+    DesktopCaptureSourceManager manager(isWindow
+        ? DesktopCaptureType::Window
+        : DesktopCaptureType::Screen);
+	const auto sources = manager.sources();
+
+    // "desktop_capturer_window_".size() == "desktop_capturer_screen_".size()
+    const auto keyId = std::stoll(uniqueKey.substr(windowPrefix.size()));
+    for (const auto &source : sources) {
+        if (source.uniqueId() == keyId) {
+            return source;
+        }
+    }
+    return DesktopCaptureSource::Invalid();
+}
+
+bool ShouldBeDesktopCapture(const std::string &uniqueKey) {
+    return (uniqueKey.find("desktop_capturer_") == 0);
+}
+
+DesktopCaptureSourceHelper::DesktopCaptureSourceHelper(
+	DesktopCaptureSource source,
+	DesktopCaptureSourceData data)
+: _renderer(std::make_shared<Renderer>()) {
+    _renderer->scheduler.runAsync([renderer = _renderer, source, data] {
+        renderer->renderer = std::make_unique<DesktopSourceRenderer>(
+            renderer->scheduler,
+            source,
+            data);
+    });
+}
+
+DesktopCaptureSourceHelper::~DesktopCaptureSourceHelper() {
+    _renderer->scheduler.runAsync([renderer = _renderer] {
+    });
+}
+
+void DesktopCaptureSourceHelper::setOutput(
+    std::shared_ptr<
+        rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) const {
+    _renderer->scheduler.runAsync([renderer = _renderer, sink] {
+        renderer->renderer->setOutput(sink);
+    });
+}
+
+void DesktopCaptureSourceHelper::setSecondaryOutput(
+    std::shared_ptr<
+        rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) const {
+	_renderer->scheduler.runAsync([renderer = _renderer, sink] {
+		renderer->renderer->setSecondaryOutput(sink);
+	});
+}
+
+void DesktopCaptureSourceHelper::start() const {
+	_renderer->scheduler.runAsync([renderer = _renderer] {
+		renderer->renderer->start();
+	});
+}
+void DesktopCaptureSourceHelper::setOnFatalError(std::function<void ()> error) const {
+    _renderer->scheduler.runAsync([renderer = _renderer, error = error] {
+        renderer->renderer->setOnFatalError(error);
+    });
+}
+void DesktopCaptureSourceHelper::setOnPause(std::function<void (bool)> pause) const {
+    _renderer->scheduler.runAsync([renderer = _renderer, pause = pause] {
+        renderer->renderer->setOnPause(pause);
+    });
+}
+
+void DesktopCaptureSourceHelper::stop() const {
+	_renderer->scheduler.runAsync([renderer = _renderer] {
+		renderer->renderer->stop();
+	});
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/desktop_capturer/DesktopCaptureSourceHelper.h

@@ -0,0 +1,56 @@
+//
+//  DesktopCaptureSourceHelper.h
+//  TgVoipWebrtc
+//
+//  Created by Mikhail Filimonov on 28.12.2020.
+//  Copyright © 2020 Mikhail Filimonov. All rights reserved.
+//
+#ifndef TGCALLS_DESKTOP_CAPTURE_SOURCE_HELPER_H__
+#define TGCALLS_DESKTOP_CAPTURE_SOURCE_HELPER_H__
+
+#include "tgcalls/desktop_capturer/DesktopCaptureSource.h"
+
+#include <memory>
+#include <functional>
+
+namespace webrtc {
+class VideoFrame;
+} // namespace webrtc
+
+namespace rtc {
+template <typename T>
+class VideoSinkInterface;
+} // namespace rtc
+
+namespace tgcalls {
+
+DesktopCaptureSource DesktopCaptureSourceForKey(
+	const std::string &uniqueKey);
+bool ShouldBeDesktopCapture(const std::string &uniqueKey);
+
+class DesktopCaptureSourceHelper {
+public:
+	DesktopCaptureSourceHelper(
+		DesktopCaptureSource source,
+		DesktopCaptureSourceData data);
+	~DesktopCaptureSourceHelper();
+
+	void setOutput(
+		std::shared_ptr<
+			rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) const;
+	void setSecondaryOutput(
+		std::shared_ptr<
+		rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) const;
+	void start() const;
+	void stop() const;
+    void setOnFatalError(std::function<void ()>) const;
+    void setOnPause(std::function<void (bool)>) const;
+private:
+	struct Renderer;
+	std::shared_ptr<Renderer> _renderer;
+
+};
+
+} // namespace tgcalls
+
+#endif // TGCALLS_DESKTOP_CAPTURE_SOURCE_HELPER_H__

Telegram/ThirdParty/tgcalls/tgcalls/desktop_capturer/DesktopCaptureSourceManager.cpp

@@ -0,0 +1,65 @@
+//
+//  DesktopCaptureSourceManager.m
+//  TgVoipWebrtc
+//
+//  Created by Mikhail Filimonov on 28.12.2020.
+//  Copyright © 2020 Mikhail Filimonov. All rights reserved.
+//
+
+#include "tgcalls/desktop_capturer/DesktopCaptureSourceManager.h"
+
+#include "modules/desktop_capture/desktop_and_cursor_composer.h"
+#include "modules/desktop_capture/desktop_capturer_differ_wrapper.h"
+#include "third_party/libyuv/include/libyuv.h"
+#include "api/video/i420_buffer.h"
+#include "tgcalls/desktop_capturer/DesktopCaptureSourceHelper.h"
+
+
+namespace tgcalls {
+
+DesktopCaptureSourceManager::DesktopCaptureSourceManager(
+    DesktopCaptureType type)
+: _capturer(CreateForType(type))
+ , _type(type) {
+}
+
+DesktopCaptureSourceManager::~DesktopCaptureSourceManager() = default;
+
+webrtc::DesktopCaptureOptions DesktopCaptureSourceManager::OptionsForType(
+    DesktopCaptureType type) {
+    auto result = webrtc::DesktopCaptureOptions::CreateDefault();
+#ifdef WEBRTC_WIN
+    result.set_allow_directx_capturer(true);
+#elif defined WEBRTC_MAC
+    result.set_allow_iosurface(type == DesktopCaptureType::Screen);
+#elif defined WEBRTC_USE_PIPEWIRE
+    result.set_allow_pipewire(true);
+#endif // WEBRTC_WIN || WEBRTC_MAC
+    result.set_detect_updated_region(true);
+    return result;
+}
+
+auto DesktopCaptureSourceManager::CreateForType(DesktopCaptureType type)
+-> std::unique_ptr<webrtc::DesktopCapturer> {
+    const auto options = OptionsForType(type);
+    if (auto result = webrtc::DesktopCapturer::CreateGenericCapturer(options)) {
+        return result;
+    }
+    return (type == DesktopCaptureType::Screen)
+        ? webrtc::DesktopCapturer::CreateScreenCapturer(options)
+        : webrtc::DesktopCapturer::CreateWindowCapturer(options);
+}
+
+std::vector<DesktopCaptureSource> DesktopCaptureSourceManager::sources() {
+    auto result = std::vector<DesktopCaptureSource>();
+	auto list = webrtc::DesktopCapturer::SourceList();
+    if (_capturer && _capturer->GetSourceList(&list)) {
+        const auto isWindow = (_type == DesktopCaptureType::Window);
+        for (const auto &source : list) {
+            result.emplace_back(source.id, source.title, isWindow);
+        }
+    }
+    return result;
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/desktop_capturer/DesktopCaptureSourceManager.h

@@ -0,0 +1,49 @@
+//
+//  DesktopCaptureSourceManager.h
+//  TgVoipWebrtc
+//
+//  Created by Mikhail Filimonov on 28.12.2020.
+//  Copyright © 2020 Mikhail Filimonov. All rights reserved.
+//
+#ifndef TGCALLS_DESKTOP_CAPTURE_SOURCE_MANAGER_H__
+#define TGCALLS_DESKTOP_CAPTURE_SOURCE_MANAGER_H__
+
+#include "tgcalls/desktop_capturer/DesktopCaptureSource.h"
+#include "tgcalls/desktop_capturer/DesktopCaptureSourceHelper.h"
+
+#include <map>
+#include <vector>
+
+namespace webrtc {
+class DesktopCapturer;
+class DesktopCaptureOptions;
+} // namespace webrtc
+
+namespace tgcalls {
+
+enum class DesktopCaptureType {
+	Screen,
+	Window,
+};
+
+class DesktopCaptureSourceManager {
+public:
+	explicit DesktopCaptureSourceManager(DesktopCaptureType type);
+	~DesktopCaptureSourceManager();
+
+	std::vector<DesktopCaptureSource> sources();
+
+private:
+	static webrtc::DesktopCaptureOptions OptionsForType(
+		DesktopCaptureType type);
+	static std::unique_ptr<webrtc::DesktopCapturer> CreateForType(
+		DesktopCaptureType type);
+
+	std::unique_ptr<webrtc::DesktopCapturer> _capturer;
+	DesktopCaptureType _type = DesktopCaptureType::Screen;
+
+};
+
+} // namespace tgcalls
+
+#endif // TGCALLS_DESKTOP_CAPTURE_SOURCE_MANAGER_H__

Telegram/ThirdParty/tgcalls/tgcalls/group/AVIOContextImpl.cpp

@@ -0,0 +1,64 @@
+#include "AVIOContextImpl.h"
+
+#include "rtc_base/logging.h"
+#include "rtc_base/third_party/base64/base64.h"
+#include "api/video/i420_buffer.h"
+
+#include <string>
+#include <set>
+#include <map>
+
+namespace tgcalls {
+
+namespace {
+
+int AVIOContextImplRead(void *opaque, unsigned char *buffer, int bufferSize) {
+    AVIOContextImpl *instance = static_cast<AVIOContextImpl *>(opaque);
+
+    int bytesToRead = std::min(bufferSize, ((int)instance->_fileData.size()) - instance->_fileReadPosition);
+    if (bytesToRead < 0) {
+        bytesToRead = 0;
+    }
+
+    if (bytesToRead > 0) {
+        memcpy(buffer, instance->_fileData.data() + instance->_fileReadPosition, bytesToRead);
+        instance->_fileReadPosition += bytesToRead;
+
+        return bytesToRead;
+    } else {
+        return AVERROR_EOF;
+    }
+}
+
+int64_t AVIOContextImplSeek(void *opaque, int64_t offset, int whence) {
+    AVIOContextImpl *instance = static_cast<AVIOContextImpl *>(opaque);
+
+    if (whence == 0x10000) {
+        return (int64_t)instance->_fileData.size();
+    } else {
+        int64_t seekOffset = std::min(offset, (int64_t)instance->_fileData.size());
+        if (seekOffset < 0) {
+            seekOffset = 0;
+        }
+        instance->_fileReadPosition = (int)seekOffset;
+        return seekOffset;
+    }
+}
+
+}
+
+AVIOContextImpl::AVIOContextImpl(std::vector<uint8_t> &&fileData) :
+_fileData(std::move(fileData)) {
+    _buffer.resize(4 * 1024);
+    _context = avio_alloc_context(_buffer.data(), (int)_buffer.size(), 0, this, &AVIOContextImplRead, NULL, &AVIOContextImplSeek);
+}
+
+AVIOContextImpl::~AVIOContextImpl() {
+    avio_context_free(&_context);
+}
+
+AVIOContext *AVIOContextImpl::getContext() const {
+    return _context;
+};
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/group/AVIOContextImpl.h

@@ -0,0 +1,40 @@
+#ifndef TGCALLS_AVIOCONTEXTIMPL_H
+#define TGCALLS_AVIOCONTEXTIMPL_H
+
+#include "absl/types/optional.h"
+#include <vector>
+#include <stdint.h>
+
+#include "api/video/video_frame.h"
+#include "absl/types/optional.h"
+
+// Fix build on Windows - this should appear before FFmpeg timestamp include.
+#define _USE_MATH_DEFINES
+#include <math.h>
+
+extern "C" {
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+#include <libavcodec/avcodec.h>
+}
+
+namespace tgcalls {
+
+class AVIOContextImpl {
+public:
+    AVIOContextImpl(std::vector<uint8_t> &&fileData);
+    ~AVIOContextImpl();
+
+    AVIOContext *getContext() const;
+
+public:
+    std::vector<uint8_t> _fileData;
+    int _fileReadPosition = 0;
+
+    std::vector<uint8_t> _buffer;
+    AVIOContext *_context = nullptr;
+};
+
+}
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/AudioStreamingPart.cpp

@@ -0,0 +1,181 @@
+#include "AudioStreamingPart.h"
+
+#include "AudioStreamingPartInternal.h"
+
+#include "rtc_base/logging.h"
+#include "rtc_base/third_party/base64/base64.h"
+
+#include <string>
+#include <bitset>
+#include <set>
+#include <map>
+
+namespace tgcalls {
+
+class AudioStreamingPartState {
+    struct ChannelMapping {
+        uint32_t ssrc = 0;
+        int channelIndex = 0;
+
+        ChannelMapping(uint32_t ssrc_, int channelIndex_) :
+            ssrc(ssrc_), channelIndex(channelIndex_) {
+        }
+    };
+
+public:
+    AudioStreamingPartState(std::vector<uint8_t> &&data, std::string const &container, bool isSingleChannel) :
+    _isSingleChannel(isSingleChannel),
+    _parsedPart(std::move(data), container) {
+        if (_parsedPart.getChannelUpdates().size() == 0 && !isSingleChannel) {
+            _didReadToEnd = true;
+            return;
+        }
+
+        _remainingMilliseconds = _parsedPart.getDurationInMilliseconds();
+
+        for (const auto &it : _parsedPart.getChannelUpdates()) {
+            _allSsrcs.insert(it.ssrc);
+        }
+    }
+
+    ~AudioStreamingPartState() {
+    }
+
+    std::map<std::string, int32_t> getEndpointMapping() const {
+        return _parsedPart.getEndpointMapping();
+    }
+
+    int getRemainingMilliseconds() const {
+        return _remainingMilliseconds;
+    }
+
+    std::vector<AudioStreamingPart::StreamingPartChannel> get10msPerChannel(AudioStreamingPartPersistentDecoder &persistentDecoder) {
+        if (_didReadToEnd) {
+            return {};
+        }
+
+        for (const auto &update : _parsedPart.getChannelUpdates()) {
+            if (update.frameIndex == _frameIndex) {
+                updateCurrentMapping(update.ssrc, update.id);
+            }
+        }
+
+        auto readResult = _parsedPart.readPcm(persistentDecoder, _pcm10ms);
+        if (readResult.numSamples <= 0) {
+            _didReadToEnd = true;
+            return {};
+        }
+
+        std::vector<AudioStreamingPart::StreamingPartChannel> resultChannels;
+
+        if (_isSingleChannel) {
+            for (int i = 0; i < readResult.numChannels; i++) {
+                AudioStreamingPart::StreamingPartChannel emptyPart;
+                emptyPart.ssrc = i + 1;
+                resultChannels.push_back(emptyPart);
+            }
+
+            for (int i = 0; i < readResult.numChannels; i++) {
+                auto channel = resultChannels.begin() + i;
+                int sourceChannelIndex = i;
+                for (int j = 0; j < readResult.numSamples; j++) {
+                    channel->pcmData.push_back(_pcm10ms[sourceChannelIndex + j * readResult.numChannels]);
+                }
+                channel->numSamples += readResult.numSamples;
+            }
+        } else {
+            for (const auto ssrc : _allSsrcs) {
+                AudioStreamingPart::StreamingPartChannel emptyPart;
+                emptyPart.ssrc = ssrc;
+                resultChannels.push_back(emptyPart);
+            }
+
+            for (auto &channel : resultChannels) {
+                auto mappedChannelIndex = getCurrentMappedChannelIndex(channel.ssrc);
+
+                if (mappedChannelIndex) {
+                    int sourceChannelIndex = mappedChannelIndex.value();
+                    for (int j = 0; j < readResult.numSamples; j++) {
+                        channel.pcmData.push_back(_pcm10ms[sourceChannelIndex + j * readResult.numChannels]);
+                    }
+                    channel.numSamples += readResult.numSamples;
+                } else {
+                    for (int j = 0; j < readResult.numSamples; j++) {
+                        channel.pcmData.push_back(0);
+                    }
+                    channel.numSamples += readResult.numSamples;
+                }
+            }
+        }
+
+        _remainingMilliseconds -= 10;
+        if (_remainingMilliseconds < 0) {
+            _remainingMilliseconds = 0;
+        }
+        _frameIndex++;
+
+        return resultChannels;
+    }
+
+private:
+    absl::optional<int> getCurrentMappedChannelIndex(uint32_t ssrc) {
+        for (const auto &it : _currentChannelMapping) {
+            if (it.ssrc == ssrc) {
+                return it.channelIndex;
+            }
+        }
+        return absl::nullopt;
+    }
+
+    void updateCurrentMapping(uint32_t ssrc, int channelIndex) {
+        for (int i = (int)_currentChannelMapping.size() - 1; i >= 0; i--) {
+            const auto &entry = _currentChannelMapping[i];
+            if (entry.ssrc == ssrc && entry.channelIndex == channelIndex) {
+                return;
+            } else if (entry.ssrc == ssrc || entry.channelIndex == channelIndex) {
+                _currentChannelMapping.erase(_currentChannelMapping.begin() + i);
+            }
+        }
+        _currentChannelMapping.emplace_back(ssrc, channelIndex);
+    }
+
+private:
+    bool _isSingleChannel = false;
+    AudioStreamingPartInternal _parsedPart;
+    std::set<uint32_t> _allSsrcs;
+
+    std::vector<int16_t> _pcm10ms;
+    std::vector<ChannelMapping> _currentChannelMapping;
+    int _frameIndex = 0;
+    int _remainingMilliseconds = 0;
+
+    bool _didReadToEnd = false;
+};
+
+AudioStreamingPart::AudioStreamingPart(std::vector<uint8_t> &&data, std::string const &container, bool isSingleChannel) {
+    if (!data.empty()) {
+        _state = new AudioStreamingPartState(std::move(data), container, isSingleChannel);
+    }
+}
+
+AudioStreamingPart::~AudioStreamingPart() {
+    if (_state) {
+        delete _state;
+    }
+}
+
+std::map<std::string, int32_t> AudioStreamingPart::getEndpointMapping() const {
+    return _state ? _state->getEndpointMapping() : std::map<std::string, int32_t>();
+}
+
+int AudioStreamingPart::getRemainingMilliseconds() const {
+    return _state ? _state->getRemainingMilliseconds() : 0;
+}
+
+std::vector<AudioStreamingPart::StreamingPartChannel> AudioStreamingPart::get10msPerChannel(AudioStreamingPartPersistentDecoder &persistentDecoder) {
+    return _state
+        ? _state->get10msPerChannel(persistentDecoder)
+        : std::vector<AudioStreamingPart::StreamingPartChannel>();
+}
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/group/AudioStreamingPart.h

@@ -0,0 +1,45 @@
+#ifndef TGCALLS_AUDIO_STREAMING_PART_H
+#define TGCALLS_AUDIO_STREAMING_PART_H
+
+#include "absl/types/optional.h"
+#include <vector>
+#include <string>
+#include <map>
+#include <stdint.h>
+
+#include "AudioStreamingPartPersistentDecoder.h"
+
+namespace tgcalls {
+
+class AudioStreamingPartState;
+
+class AudioStreamingPart {
+public:
+    struct StreamingPartChannel {
+        uint32_t ssrc = 0;
+        std::vector<int16_t> pcmData;
+        int numSamples = 0;
+    };
+    
+    explicit AudioStreamingPart(std::vector<uint8_t> &&data, std::string const &container, bool isSingleChannel);
+    ~AudioStreamingPart();
+    
+    AudioStreamingPart(const AudioStreamingPart&) = delete;
+    AudioStreamingPart(AudioStreamingPart&& other) {
+        _state = other._state;
+        other._state = nullptr;
+    }
+    AudioStreamingPart& operator=(const AudioStreamingPart&) = delete;
+    AudioStreamingPart& operator=(AudioStreamingPart&&) = delete;
+
+    std::map<std::string, int32_t> getEndpointMapping() const;
+    int getRemainingMilliseconds() const;
+    std::vector<StreamingPartChannel> get10msPerChannel(AudioStreamingPartPersistentDecoder &persistentDecoder);
+    
+private:
+    AudioStreamingPartState *_state = nullptr;
+};
+
+}
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/AudioStreamingPartInternal.cpp

@@ -0,0 +1,404 @@
+#include "AudioStreamingPartInternal.h"
+
+#include "rtc_base/logging.h"
+#include "rtc_base/third_party/base64/base64.h"
+
+extern "C" {
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+#include <libavcodec/avcodec.h>
+}
+
+#include <string>
+#include <bitset>
+#include <set>
+#include <map>
+
+namespace tgcalls {
+
+namespace {
+
+int16_t sampleFloatToInt16(float sample) {
+  return av_clip_int16 (static_cast<int32_t>(lrint(sample*32767)));
+}
+
+uint32_t stringToUInt32(std::string const &string) {
+    std::stringstream stringStream(string);
+    uint32_t value = 0;
+    stringStream >> value;
+    return value;
+}
+
+template <typename Out>
+void splitString(const std::string &s, char delim, Out result) {
+    std::istringstream iss(s);
+    std::string item;
+    while (std::getline(iss, item, delim)) {
+        *result++ = item;
+    }
+}
+
+std::vector<std::string> splitString(const std::string &s, char delim) {
+    std::vector<std::string> elems;
+    splitString(s, delim, std::back_inserter(elems));
+    return elems;
+}
+
+static absl::optional<uint32_t> readInt32(std::string const &data, int &offset) {
+    if (offset + 4 > data.length()) {
+        return absl::nullopt;
+    }
+
+    int32_t value = 0;
+    memcpy(&value, data.data() + offset, 4);
+    offset += 4;
+
+    return value;
+}
+
+std::vector<AudioStreamingPartInternal::ChannelUpdate> parseChannelUpdates(std::string const &data, int &offset) {
+    std::vector<AudioStreamingPartInternal::ChannelUpdate> result;
+
+    auto channels = readInt32(data, offset);
+    if (!channels) {
+        return {};
+    }
+
+    auto count = readInt32(data, offset);
+    if (!count) {
+        return {};
+    }
+
+    for (int i = 0; i < count.value(); i++) {
+        auto frameIndex = readInt32(data, offset);
+        if (!frameIndex) {
+            return {};
+        }
+
+        auto channelId = readInt32(data, offset);
+        if (!channelId) {
+            return {};
+        }
+
+        auto ssrc = readInt32(data, offset);
+        if (!ssrc) {
+            return {};
+        }
+
+        AudioStreamingPartInternal::ChannelUpdate update;
+        update.frameIndex = frameIndex.value();
+        update.id = channelId.value();
+        update.ssrc = ssrc.value();
+
+        result.push_back(update);
+    }
+
+    return result;
+}
+
+}
+
+AudioStreamingPartInternal::AudioStreamingPartInternal(std::vector<uint8_t> &&fileData, std::string const &container) :
+_avIoContext(std::move(fileData)) {
+    int ret = 0;
+
+    _frame = av_frame_alloc();
+
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+    const
+#endif
+    AVInputFormat *inputFormat = av_find_input_format(container.c_str());
+    if (!inputFormat) {
+        _didReadToEnd = true;
+        return;
+    }
+
+    _inputFormatContext = avformat_alloc_context();
+    if (!_inputFormatContext) {
+        _didReadToEnd = true;
+        return;
+    }
+
+    _inputFormatContext->pb = _avIoContext.getContext();
+
+    if ((ret = avformat_open_input(&_inputFormatContext, "", inputFormat, nullptr)) < 0) {
+        _didReadToEnd = true;
+        return;
+    }
+
+    if ((ret = avformat_find_stream_info(_inputFormatContext, nullptr)) < 0) {
+        _didReadToEnd = true;
+
+        avformat_close_input(&_inputFormatContext);
+        _inputFormatContext = nullptr;
+        return;
+    }
+
+    for (int i = 0; i < _inputFormatContext->nb_streams; i++) {
+        AVStream *inStream = _inputFormatContext->streams[i];
+
+        AVCodecParameters *inCodecpar = inStream->codecpar;
+        if (inCodecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
+            continue;
+        }
+        
+        _audioCodecParameters = avcodec_parameters_alloc();
+        avcodec_parameters_copy(_audioCodecParameters, inCodecpar);
+        
+        _streamId = i;
+
+        _durationInMilliseconds = (int)(inStream->duration * av_q2d(inStream->time_base) * 1000);
+
+        if (inStream->metadata) {
+            AVDictionaryEntry *entry = av_dict_get(inStream->metadata, "TG_META", nullptr, 0);
+            if (entry && entry->value) {
+                std::string result;
+                size_t data_used = 0;
+                std::string sourceBase64 = (const char *)entry->value;
+                rtc::Base64::Decode(sourceBase64, rtc::Base64::DO_LAX, &result, &data_used);
+
+                if (result.size() != 0) {
+                    int offset = 0;
+                    _channelUpdates = parseChannelUpdates(result, offset);
+                }
+            }
+
+            uint32_t videoChannelMask = 0;
+            entry = av_dict_get(inStream->metadata, "ACTIVE_MASK", nullptr, 0);
+            if (entry && entry->value) {
+                std::string sourceString = (const char *)entry->value;
+                videoChannelMask = stringToUInt32(sourceString);
+            }
+
+            std::vector<std::string> endpointList;
+            entry = av_dict_get(inStream->metadata, "ENDPOINTS", nullptr, 0);
+            if (entry && entry->value) {
+                std::string sourceString = (const char *)entry->value;
+                endpointList = splitString(sourceString, ' ');
+            }
+
+            std::bitset<32> videoChannels(videoChannelMask);
+            size_t endpointIndex = 0;
+            if (videoChannels.count() == endpointList.size()) {
+                for (size_t i = 0; i < videoChannels.size(); i++) {
+                    if (videoChannels[i]) {
+                        _endpointMapping.insert(std::make_pair(endpointList[endpointIndex], i));
+                        endpointIndex++;
+                    }
+                }
+            }
+        }
+
+        break;
+    }
+    
+    if (_streamId == -1) {
+        _didReadToEnd = true;
+    }
+}
+
+AudioStreamingPartInternal::~AudioStreamingPartInternal() {
+    if (_frame) {
+        av_frame_free(&_frame);
+    }
+    if (_inputFormatContext) {
+        avformat_close_input(&_inputFormatContext);
+    }
+    if (_audioCodecParameters) {
+        avcodec_parameters_free(&_audioCodecParameters);
+    }
+}
+
+AudioStreamingPartInternal::ReadPcmResult AudioStreamingPartInternal::readPcm(AudioStreamingPartPersistentDecoder &persistentDecoder, std::vector<int16_t> &outPcm) {
+    if (_didReadToEnd) {
+        return AudioStreamingPartInternal::ReadPcmResult();
+    }
+    
+    int outPcmSampleOffset = 0;
+    ReadPcmResult result;
+    
+    if (_pcmBufferSampleOffset >= _pcmBufferSampleSize) {
+        fillPcmBuffer(persistentDecoder);
+    }
+
+    if (outPcm.size() != 480 * _channelCount) {
+        outPcm.resize(480 * _channelCount);
+    }
+    int readSamples = 0;
+    if (_channelCount != 0) {
+        readSamples = (int)outPcm.size() / _channelCount;
+    }
+
+    while (outPcmSampleOffset < readSamples) {
+        if (_pcmBufferSampleOffset >= _pcmBufferSampleSize) {
+            fillPcmBuffer(persistentDecoder);
+
+            if (_pcmBufferSampleOffset >= _pcmBufferSampleSize) {
+                break;
+            }
+        }
+
+        int readFromPcmBufferSamples = std::min(_pcmBufferSampleSize - _pcmBufferSampleOffset, readSamples - outPcmSampleOffset);
+        if (readFromPcmBufferSamples != 0) {
+            std::copy(_pcmBuffer.begin() + _pcmBufferSampleOffset * _channelCount, _pcmBuffer.begin() + _pcmBufferSampleOffset * _channelCount + readFromPcmBufferSamples * _channelCount, outPcm.begin() + outPcmSampleOffset * _channelCount);
+            _pcmBufferSampleOffset += readFromPcmBufferSamples;
+            outPcmSampleOffset += readFromPcmBufferSamples;
+            result.numSamples += readFromPcmBufferSamples;
+            _readSampleCount += readFromPcmBufferSamples;
+        }
+    }
+    
+    result.numChannels = _channelCount;
+    
+    // Uncomment for debugging incomplete frames
+    /*if (result.numSamples != 480 && result.numSamples != 0) {
+        RTC_LOG(LS_INFO) << "result.numSamples = " << result.numSamples << ", _readSampleCount = " << _readSampleCount << ", duration = " << _inputFormatContext->streams[_streamId]->duration;
+    }*/
+
+    return result;
+}
+
+int AudioStreamingPartInternal::getDurationInMilliseconds() const {
+    return _durationInMilliseconds;
+}
+
+std::vector<AudioStreamingPartInternal::ChannelUpdate> const &AudioStreamingPartInternal::getChannelUpdates() const {
+    return _channelUpdates;
+}
+
+std::map<std::string, int32_t> AudioStreamingPartInternal::getEndpointMapping() const {
+    return _endpointMapping;
+}
+    
+void AudioStreamingPartInternal::fillPcmBuffer(AudioStreamingPartPersistentDecoder &persistentDecoder) {
+    _pcmBufferSampleSize = 0;
+    _pcmBufferSampleOffset = 0;
+
+    if (_didReadToEnd) {
+        return;
+    }
+    if (!_inputFormatContext) {
+        _didReadToEnd = true;
+        return;
+    }
+
+    int ret = 0;
+    while (true) {
+      ret = av_read_frame(_inputFormatContext, &_packet);
+      if (ret < 0) {
+        _didReadToEnd = true;
+        return;
+      }
+        
+      if (_packet.stream_index != _streamId) {
+        av_packet_unref(&_packet);
+        continue;
+      }
+        
+      ret = persistentDecoder.decode(_audioCodecParameters, _inputFormatContext->streams[_streamId]->time_base, _packet, _frame);
+      av_packet_unref(&_packet);
+
+      if (ret == AVERROR(EAGAIN)) {
+        continue;
+      }
+        
+      break;
+    }
+
+    if (ret != 0) {
+        _didReadToEnd = true;
+        return;
+    }
+    
+    if (_channelCount == 0) {
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+        _channelCount = _frame->ch_layout.nb_channels;
+#else
+        _channelCount = _frame->channels;
+#endif
+    }
+    
+    if (_channelCount == 0) {
+        _didReadToEnd = true;
+        return;
+    }
+    
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+    if (_frame->ch_layout.nb_channels != _channelCount || _frame->ch_layout.nb_channels > 8) {
+#else
+    if (_frame->channels != _channelCount || _frame->channels > 8) {
+#endif
+        _didReadToEnd = true;
+        return;
+    }
+
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+    if (_pcmBuffer.size() < _frame->nb_samples * _frame->ch_layout.nb_channels) {
+        _pcmBuffer.resize(_frame->nb_samples * _frame->ch_layout.nb_channels);
+    }
+#else
+    if (_pcmBuffer.size() < _frame->nb_samples * _frame->channels) {
+        _pcmBuffer.resize(_frame->nb_samples * _frame->channels);
+    }
+#endif
+
+    switch (_frame->format) {
+    case AV_SAMPLE_FMT_S16: {
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+        memcpy(_pcmBuffer.data(), _frame->data[0], _frame->nb_samples * 2 * _frame->ch_layout.nb_channels);
+#else
+        memcpy(_pcmBuffer.data(), _frame->data[0], _frame->nb_samples * 2 * _frame->channels);
+#endif
+    } break;
+
+    case AV_SAMPLE_FMT_S16P: {
+        int16_t *to = _pcmBuffer.data();
+        for (int sample = 0; sample < _frame->nb_samples; ++sample) {
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+            for (int channel = 0; channel < _frame->ch_layout.nb_channels; ++channel) {
+#else
+            for (int channel = 0; channel < _frame->channels; ++channel) {
+#endif
+                int16_t *shortChannel = (int16_t*)_frame->data[channel];
+                *to++ = shortChannel[sample];
+            }
+        }
+    } break;
+
+    case AV_SAMPLE_FMT_FLT: {
+        float *floatData = (float *)&_frame->data[0];
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+        for (int i = 0; i < _frame->nb_samples * _frame->ch_layout.nb_channels; i++) {
+            _pcmBuffer[i] = sampleFloatToInt16(floatData[i]);
+        }
+#else
+        for (int i = 0; i < _frame->nb_samples * _frame->channels; i++) {
+            _pcmBuffer[i] = sampleFloatToInt16(floatData[i]);
+        }
+#endif
+    } break;
+
+    case AV_SAMPLE_FMT_FLTP: {
+        int16_t *to = _pcmBuffer.data();
+        for (int sample = 0; sample < _frame->nb_samples; ++sample) {
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+            for (int channel = 0; channel < _frame->ch_layout.nb_channels; ++channel) {
+#else
+            for (int channel = 0; channel < _frame->channels; ++channel) {
+#endif
+                float *floatChannel = (float*)_frame->data[channel];
+                *to++ = sampleFloatToInt16(floatChannel[sample]);
+            }
+        }
+    } break;
+
+    default: {
+        RTC_FATAL() << "Unexpected sample_fmt";
+    } break;
+    }
+
+    _pcmBufferSampleSize = _frame->nb_samples;
+    _pcmBufferSampleOffset = 0;
+}
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/group/AudioStreamingPartInternal.h

@@ -0,0 +1,65 @@
+#ifndef TGCALLS_AUDIO_STREAMING_PART_INTERNAL_H
+#define TGCALLS_AUDIO_STREAMING_PART_INTERNAL_H
+
+#include "absl/types/optional.h"
+#include <vector>
+#include <string>
+#include <map>
+#include <stdint.h>
+
+#include "AVIOContextImpl.h"
+#include "AudioStreamingPartPersistentDecoder.h"
+
+namespace tgcalls {
+
+class AudioStreamingPartInternal {
+public:
+    struct ReadPcmResult {
+        int numSamples = 0;
+        int numChannels = 0;
+    };
+
+    struct ChannelUpdate {
+        int frameIndex = 0;
+        int id = 0;
+        uint32_t ssrc = 0;
+    };
+
+public:
+    AudioStreamingPartInternal(std::vector<uint8_t> &&fileData, std::string const &container);
+    ~AudioStreamingPartInternal();
+
+    ReadPcmResult readPcm(AudioStreamingPartPersistentDecoder &persistentDecoder, std::vector<int16_t> &outPcm);
+    int getDurationInMilliseconds() const;
+    std::vector<ChannelUpdate> const &getChannelUpdates() const;
+    std::map<std::string, int32_t> getEndpointMapping() const;
+
+private:
+    void fillPcmBuffer(AudioStreamingPartPersistentDecoder &persistentDecoder);
+
+private:
+    AVIOContextImpl _avIoContext;
+
+    AVFormatContext *_inputFormatContext = nullptr;
+    AVPacket _packet;
+    AVFrame *_frame = nullptr;
+    AVCodecParameters *_audioCodecParameters = nullptr;
+
+    bool _didReadToEnd = false;
+
+    int _durationInMilliseconds = 0;
+    int _streamId = -1;
+    int _channelCount = 0;
+
+    std::vector<ChannelUpdate> _channelUpdates;
+    std::map<std::string, int32_t> _endpointMapping;
+
+    std::vector<int16_t> _pcmBuffer;
+    int _pcmBufferSampleOffset = 0;
+    int _pcmBufferSampleSize = 0;
+    int _readSampleCount = 0;
+};
+
+}
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/AudioStreamingPartPersistentDecoder.cpp

@@ -0,0 +1,138 @@
+#include "AudioStreamingPartPersistentDecoder.h"
+
+#include "rtc_base/logging.h"
+#include "rtc_base/third_party/base64/base64.h"
+
+namespace tgcalls {
+
+WrappedCodecParameters::WrappedCodecParameters(AVCodecParameters const *codecParameters) {
+    _value = avcodec_parameters_alloc();
+    avcodec_parameters_copy(_value, codecParameters);
+}
+    
+WrappedCodecParameters::~WrappedCodecParameters() {
+    avcodec_parameters_free(&_value);
+}
+
+bool WrappedCodecParameters::isEqual(AVCodecParameters const *other) {
+    if (_value->codec_id != other->codec_id) {
+        return false;
+    }
+    if (_value->format != other->format) {
+        return false;
+    }
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+    if (_value->ch_layout.nb_channels != other->ch_layout.nb_channels) {
+#else
+    if (_value->channels != other->channels) {
+#endif
+        return false;
+    }
+
+    return true;
+}
+
+class AudioStreamingPartPersistentDecoderState {
+public:
+    AudioStreamingPartPersistentDecoderState(AVCodecParameters const *codecParameters, AVRational timeBase) :
+    _codecParameters(codecParameters),
+    _timeBase(timeBase) {
+        const AVCodec *codec = avcodec_find_decoder(codecParameters->codec_id);
+        if (codec) {
+            _codecContext = avcodec_alloc_context3(codec);
+            int ret = avcodec_parameters_to_context(_codecContext, codecParameters);
+            if (ret < 0) {
+                avcodec_free_context(&_codecContext);
+                _codecContext = nullptr;
+            } else {
+                _codecContext->pkt_timebase = timeBase;
+
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+                _channelCount = _codecContext->ch_layout.nb_channels;
+#else
+                _channelCount = _codecContext->channels;
+#endif
+
+                ret = avcodec_open2(_codecContext, codec, nullptr);
+                if (ret < 0) {
+                    avcodec_free_context(&_codecContext);
+                    _codecContext = nullptr;
+                }
+            }
+        }
+    }
+    
+    ~AudioStreamingPartPersistentDecoderState() {
+        if (_codecContext) {
+            #if LIBAVFORMAT_VERSION_MAJOR < 59
+            avcodec_close(_codecContext);
+            #endif
+            avcodec_free_context(&_codecContext);
+        }
+    }
+    
+    int decode(AVPacket &packet, AVFrame *frame) {
+        int ret = avcodec_send_packet(_codecContext, &packet);
+        if (ret < 0) {
+            return ret;
+        }
+
+        int bytesPerSample = av_get_bytes_per_sample(_codecContext->sample_fmt);
+        if (bytesPerSample != 2 && bytesPerSample != 4) {
+            return -1;
+        }
+
+        ret = avcodec_receive_frame(_codecContext, frame);
+        return ret;
+    }
+
+public:
+    WrappedCodecParameters _codecParameters;
+    AVRational _timeBase;
+    AVCodecContext *_codecContext = nullptr;
+    int _channelCount = 0;
+};
+
+AudioStreamingPartPersistentDecoder::AudioStreamingPartPersistentDecoder() {
+}
+
+AudioStreamingPartPersistentDecoder::~AudioStreamingPartPersistentDecoder() {
+    if (_state) {
+        delete _state;
+        _state = nullptr;
+    }
+}
+
+void AudioStreamingPartPersistentDecoder::maybeReset(AVCodecParameters const *codecParameters, AVRational timeBase) {
+    if (_state) {
+        bool isUpdated = false;
+        if (!_state->_codecParameters.isEqual(codecParameters)) {
+            isUpdated = true;
+        }
+        if (_state->_timeBase.num != timeBase.num || _state->_timeBase.den != timeBase.den) {
+            isUpdated = true;
+        }
+        if (!isUpdated) {
+            return;
+        }
+    }
+    
+    if (_state) {
+        delete _state;
+        _state = nullptr;
+    }
+    
+    _state = new AudioStreamingPartPersistentDecoderState(codecParameters, timeBase);
+}
+
+int AudioStreamingPartPersistentDecoder::decode(AVCodecParameters const *codecParameters, AVRational timeBase, AVPacket &packet, AVFrame *frame) {
+    maybeReset(codecParameters, timeBase);
+    
+    if (!_state) {
+        return -1;
+    }
+    
+    return _state->decode(packet, frame);
+}
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/group/AudioStreamingPartPersistentDecoder.h

@@ -0,0 +1,51 @@
+#ifndef TGCALLS_AUDIO_STREAMING_PART_PERSISTENT_DECODER_H
+#define TGCALLS_AUDIO_STREAMING_PART_PERSISTENT_DECODER_H
+
+#include "absl/types/optional.h"
+#include <vector>
+#include <string>
+#include <map>
+#include <stdint.h>
+
+// Fix build on Windows - this should appear before FFmpeg timestamp include.
+#define _USE_MATH_DEFINES
+#include <math.h>
+
+extern "C" {
+#include <libavutil/timestamp.h>
+#include <libavformat/avformat.h>
+#include <libavcodec/avcodec.h>
+}
+
+namespace tgcalls {
+
+class AudioStreamingPartPersistentDecoderState;
+
+class WrappedCodecParameters {
+public:
+    WrappedCodecParameters(AVCodecParameters const *codecParameters);
+    ~WrappedCodecParameters();
+
+    bool isEqual(AVCodecParameters const *other);
+
+private:
+    AVCodecParameters *_value = nullptr;
+};
+
+class AudioStreamingPartPersistentDecoder {
+public:
+    AudioStreamingPartPersistentDecoder();
+    ~AudioStreamingPartPersistentDecoder();
+
+    int decode(AVCodecParameters const *codecParameters, AVRational timeBase, AVPacket &packet, AVFrame *frame);
+
+private:
+    void maybeReset(AVCodecParameters const *codecParameters, AVRational timeBase);
+
+private:
+    AudioStreamingPartPersistentDecoderState *_state = nullptr;
+};
+
+}
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/GroupInstanceCustomImpl.h

@@ -0,0 +1,59 @@
+#ifndef TGCALLS_GROUP_INSTANCE_CUSTOM_IMPL_H
+#define TGCALLS_GROUP_INSTANCE_CUSTOM_IMPL_H
+
+#include <functional>
+#include <vector>
+#include <string>
+#include <memory>
+#include <map>
+
+#include "../Instance.h"
+#include "GroupInstanceImpl.h"
+
+namespace tgcalls {
+
+class LogSinkImpl;
+class GroupInstanceCustomInternal;
+class Threads;
+
+class GroupInstanceCustomImpl final : public GroupInstanceInterface {
+public:
+    explicit GroupInstanceCustomImpl(GroupInstanceDescriptor &&descriptor);
+    ~GroupInstanceCustomImpl();
+
+    void stop(std::function<void()> completion);
+    
+    void setConnectionMode(GroupConnectionMode connectionMode, bool keepBroadcastIfWasEnabled, bool isUnifiedBroadcast);
+
+    void emitJoinPayload(std::function<void(GroupJoinPayload const &)> completion);
+    void setJoinResponsePayload(std::string const &payload);
+    void removeSsrcs(std::vector<uint32_t> ssrcs);
+    void removeIncomingVideoSource(uint32_t ssrc);
+
+    void setIsMuted(bool isMuted);
+    void setIsNoiseSuppressionEnabled(bool isNoiseSuppressionEnabled);
+    void setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture);
+    void setVideoSource(std::function<webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface>()> getVideoSource);
+    void setAudioOutputDevice(std::string id);
+    void setAudioInputDevice(std::string id);
+    void addExternalAudioSamples(std::vector<uint8_t> &&samples);
+    
+    void addOutgoingVideoOutput(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+    void addIncomingVideoOutput(std::string const &endpointId, std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+    
+    void setVolume(uint32_t ssrc, double volume);
+    void setRequestedVideoChannels(std::vector<VideoChannelDescription> &&requestedVideoChannels);
+
+    void getStats(std::function<void(GroupInstanceStats)> completion);
+    void internal_addCustomNetworkEvent(bool isRemoteConnected);
+
+private:
+    std::shared_ptr<Threads> _threads;
+    std::unique_ptr<ThreadLocalObject<GroupInstanceCustomInternal>> _internal;
+    std::unique_ptr<LogSinkImpl> _logSink;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/GroupInstanceImpl.h

@@ -0,0 +1,237 @@
+#ifndef TGCALLS_GROUP_INSTANCE_IMPL_H
+#define TGCALLS_GROUP_INSTANCE_IMPL_H
+
+#include <functional>
+#include <vector>
+#include <string>
+#include <memory>
+#include <map>
+#include <optional>
+
+#include "../Instance.h"
+
+#include "../StaticThreads.h"
+#include "GroupJoinPayload.h"
+
+namespace webrtc {
+class AudioDeviceModule;
+class TaskQueueFactory;
+class VideoTrackSourceInterface;
+}
+
+namespace webrtc {
+template <class T>
+class scoped_refptr;
+}
+
+namespace tgcalls {
+
+class LogSinkImpl;
+class GroupInstanceManager;
+class WrappedAudioDeviceModule;
+struct AudioFrame;
+
+struct GroupConfig {
+    bool need_log{true};
+    FilePath logPath;
+};
+
+struct GroupLevelValue {
+    float level = 0.;
+    bool voice = false;
+    bool isMuted = false;
+};
+
+struct GroupLevelUpdate {
+    uint32_t ssrc = 0;
+    GroupLevelValue value;
+};
+
+struct GroupLevelsUpdate {
+    std::vector<GroupLevelUpdate> updates;
+};
+
+struct GroupActivityUpdate {
+    uint32_t ssrc = 0;
+};
+
+struct GroupActivitiesUpdate {
+    std::vector<GroupActivityUpdate> updates;
+};
+
+
+class BroadcastPartTask {
+public:
+    virtual ~BroadcastPartTask() = default;
+
+    virtual void cancel() = 0;
+};
+
+struct BroadcastPart {
+    struct VideoParams {
+    };
+
+    enum class Status {
+        Success,
+        NotReady,
+        ResyncNeeded
+    };
+
+    int64_t timestampMilliseconds = 0;
+    double responseTimestamp = 0;
+    Status status = Status::NotReady;
+    std::vector<uint8_t> data;
+};
+
+enum class GroupConnectionMode {
+    GroupConnectionModeNone,
+    GroupConnectionModeRtc,
+    GroupConnectionModeBroadcast
+};
+
+struct GroupNetworkState {
+    bool isConnected = false;
+    bool isTransitioningFromBroadcastToRtc = false;
+    GroupConnectionMode connectionMode = GroupConnectionMode::GroupConnectionModeRtc;
+};
+
+enum class VideoContentType {
+    None,
+    Screencast,
+    Generic
+};
+
+enum class VideoCodecName {
+    VP8,
+    VP9,
+    H264
+};
+
+class RequestMediaChannelDescriptionTask {
+public:
+    virtual ~RequestMediaChannelDescriptionTask() = default;
+
+    virtual void cancel() = 0;
+};
+
+struct MediaChannelDescription {
+    enum class Type {
+        Audio,
+        Video
+    };
+
+    Type type = Type::Audio;
+    uint32_t audioSsrc = 0;
+    int64_t userId = 0;
+    std::string videoInformation;
+};
+
+struct MediaSsrcGroup {
+    std::string semantics;
+    std::vector<uint32_t> ssrcs;
+};
+
+struct VideoChannelDescription {
+    enum class Quality {
+        Thumbnail,
+        Medium,
+        Full
+    };
+    uint32_t audioSsrc = 0;
+    int64_t userId = 0;
+    std::string endpointId;
+    std::vector<MediaSsrcGroup> ssrcGroups;
+    Quality minQuality = Quality::Thumbnail;
+    Quality maxQuality = Quality::Thumbnail;
+};
+
+struct GroupInstanceStats {
+    struct IncomingVideoStats {
+        int receivingQuality = 0;
+        int availableQuality = 0;
+    };
+
+    std::vector<std::pair<std::string, IncomingVideoStats>> incomingVideoStats;
+};
+
+struct GroupInstanceDescriptor {
+    std::shared_ptr<Threads> threads;
+    GroupConfig config;
+    std::string statsLogPath;
+    std::function<void(GroupNetworkState)> networkStateUpdated;
+    std::function<void(int)> signalBarsUpdated;
+    std::function<void(GroupLevelsUpdate const &)> audioLevelsUpdated;
+    std::function<void(uint32_t, const AudioFrame &)> onAudioFrame;
+    std::function<void(GroupActivitiesUpdate const &)> ssrcActivityUpdated;
+    std::string initialInputDeviceId;
+    std::string initialOutputDeviceId;
+    bool useDummyChannel{true};
+    bool disableIncomingChannels{false};
+    std::function<webrtc::scoped_refptr<webrtc::AudioDeviceModule>(webrtc::TaskQueueFactory*)> createAudioDeviceModule;
+    std::function<webrtc::scoped_refptr<WrappedAudioDeviceModule>(webrtc::TaskQueueFactory*)> createWrappedAudioDeviceModule;
+    std::shared_ptr<VideoCaptureInterface> videoCapture; // deprecated
+    std::function<webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface>()> getVideoSource;
+    std::function<std::shared_ptr<BroadcastPartTask>(std::function<void(int64_t)>)> requestCurrentTime;
+    std::function<std::shared_ptr<BroadcastPartTask>(int64_t, int64_t, std::function<void(BroadcastPart &&)>)> requestAudioBroadcastPart;
+    std::function<std::shared_ptr<BroadcastPartTask>(int64_t, int64_t, int32_t, VideoChannelDescription::Quality, std::function<void(BroadcastPart &&)>)> requestVideoBroadcastPart;
+    int outgoingAudioBitrateKbit{32};
+    bool disableOutgoingAudioProcessing{false};
+    bool disableAudioInput{false};
+    bool ios_enableSystemMute{false};
+    VideoContentType videoContentType{VideoContentType::None};
+    bool initialEnableNoiseSuppression{false};
+    std::vector<VideoCodecName> videoCodecPreferences;
+    std::function<std::shared_ptr<RequestMediaChannelDescriptionTask>(std::vector<uint32_t> const &, std::function<void(std::vector<MediaChannelDescription> &&)>)> requestMediaChannelDescriptions;
+    int minOutgoingVideoBitrateKbit{100};
+    std::function<void(bool)> onMutedSpeechActivityDetected;
+    std::function<std::vector<uint8_t>(std::vector<uint8_t> const &, int64_t, bool, int32_t)> e2eEncryptDecrypt;
+    bool isConference{false};
+};
+
+template <typename T>
+class ThreadLocalObject;
+
+class GroupInstanceInterface {
+protected:
+    GroupInstanceInterface() = default;
+
+public:
+    virtual ~GroupInstanceInterface() = default;
+
+    virtual void stop(std::function<void()> completion) = 0;
+
+    virtual void setConnectionMode(GroupConnectionMode connectionMode, bool keepBroadcastIfWasEnabled, bool isUnifiedBroadcast) = 0;
+
+    virtual void emitJoinPayload(std::function<void(GroupJoinPayload const &)> completion) = 0;
+    virtual void setJoinResponsePayload(std::string const &payload) = 0;
+    virtual void removeSsrcs(std::vector<uint32_t> ssrcs) = 0;
+    virtual void removeIncomingVideoSource(uint32_t ssrc) = 0;
+
+    virtual void setIsMuted(bool isMuted) = 0;
+    virtual void setIsNoiseSuppressionEnabled(bool isNoiseSuppressionEnabled) = 0;
+    virtual void setVideoCapture(std::shared_ptr<VideoCaptureInterface> videoCapture) = 0;
+    virtual void setVideoSource(std::function<webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface>()> getVideoSource) = 0;
+    virtual void setAudioOutputDevice(std::string id) = 0;
+    virtual void setAudioInputDevice(std::string id) = 0;
+    virtual void addExternalAudioSamples(std::vector<uint8_t> &&samples) = 0;
+
+    virtual void addOutgoingVideoOutput(std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) = 0;
+    virtual void addIncomingVideoOutput(std::string const &endpointId, std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) = 0;
+
+    virtual void setVolume(uint32_t ssrc, double volume) = 0;
+    virtual void setRequestedVideoChannels(std::vector<VideoChannelDescription> &&requestedVideoChannels) = 0;
+
+    virtual void getStats(std::function<void(GroupInstanceStats)> completion) = 0;
+    virtual void internal_addCustomNetworkEvent(bool isRemoteConnected) = 0;
+
+    struct AudioDevice {
+      enum class Type {Input, Output};
+      std::string name;
+      std::string guid;
+    };
+    static std::vector<GroupInstanceInterface::AudioDevice> getAudioDevices(AudioDevice::Type type);
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/GroupJoinPayload.h

@@ -0,0 +1,78 @@
+#ifndef TGCALLS_GROUP_JOIN_PAYLOAD_H
+#define TGCALLS_GROUP_JOIN_PAYLOAD_H
+
+#include <vector>
+#include <string>
+#include <stdint.h>
+
+namespace tgcalls {
+
+struct GroupJoinPayloadVideoSourceGroup {
+    std::vector<uint32_t> ssrcs;
+    std::string semantics;
+};
+
+struct GroupJoinPayloadVideoPayloadType {
+    struct FeedbackType {
+        std::string type;
+        std::string subtype;
+    };
+
+    uint32_t id = 0;
+    std::string name;
+    uint32_t clockrate = 0;
+    uint32_t channels = 0;
+    std::vector<FeedbackType> feedbackTypes;
+    std::vector<std::pair<std::string, std::string>> parameters;
+};
+
+struct GroupJoinTransportDescription {
+    struct Fingerprint {
+        std::string hash;
+        std::string setup;
+        std::string fingerprint;
+    };
+
+    struct Candidate {
+        std::string port;
+        std::string protocol;
+        std::string network;
+        std::string generation;
+        std::string id;
+        std::string component;
+        std::string foundation;
+        std::string priority;
+        std::string ip;
+        std::string type;
+
+        std::string tcpType;
+        std::string relAddr;
+        std::string relPort;
+    };
+
+    std::string ufrag;
+    std::string pwd;
+    std::vector<Fingerprint> fingerprints;
+    std::vector<Candidate> candidates;
+};
+
+struct GroupJoinVideoInformation {
+    uint32_t serverVideoBandwidthProbingSsrc = 0;
+    std::string endpointId;
+    std::vector<GroupJoinPayloadVideoPayloadType> payloadTypes;
+    std::vector<std::pair<uint32_t, std::string>> extensionMap;
+};
+
+struct GroupParticipantVideoInformation {
+    std::string endpointId;
+    std::vector<GroupJoinPayloadVideoSourceGroup> ssrcGroups;
+};
+
+struct GroupJoinPayload {
+    uint32_t audioSsrc = 0;
+    std::string json;
+};
+
+}
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/GroupJoinPayloadInternal.cpp

@@ -0,0 +1,373 @@
+#include "GroupJoinPayloadInternal.h"
+
+#include "third-party/json11.hpp"
+#include <sstream>
+
+namespace tgcalls {
+
+namespace {
+
+absl::optional<int32_t> parseInt(json11::Json::object const &object, std::string const &key) {
+    const auto value = object.find(key);
+    if (value == object.end() || !value->second.is_number()) {
+        return absl::nullopt;
+    }
+    return value->second.int_value();
+}
+
+absl::optional<std::string> parseString(json11::Json::object const &object, std::string const &key) {
+    const auto value = object.find(key);
+    if (value == object.end() || !value->second.is_string()) {
+        return absl::nullopt;
+    }
+    return value->second.string_value();
+}
+
+template <typename Out>
+void splitString(const std::string &s, char delim, Out result) {
+    std::istringstream iss(s);
+    std::string item;
+    while (std::getline(iss, item, delim)) {
+        *result++ = item;
+    }
+}
+
+std::vector<std::string> splitString(const std::string &s, char delim) {
+    std::vector<std::string> elems;
+    splitString(s, delim, std::back_inserter(elems));
+    return elems;
+}
+
+absl::optional<GroupJoinTransportDescription> parseTransportDescription(json11::Json::object const &object) {
+    GroupJoinTransportDescription result;
+
+    if (const auto pwd = parseString(object, "pwd")) {
+        result.pwd = pwd.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    if (const auto ufrag = parseString(object, "ufrag")) {
+        result.ufrag = ufrag.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    const auto fingerprints = object.find("fingerprints");
+    if (fingerprints == object.end() || !fingerprints->second.is_array()) {
+        return absl::nullopt;
+    }
+    for (const auto &fingerprint : fingerprints->second.array_items()) {
+        if (!fingerprint.is_object()) {
+            return absl::nullopt;
+        }
+
+        GroupJoinTransportDescription::Fingerprint parsedFingerprint;
+
+        if (const auto hash = parseString(fingerprint.object_items(), "hash")) {
+            parsedFingerprint.hash = hash.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto fingerprintValue = parseString(fingerprint.object_items(), "fingerprint")) {
+            parsedFingerprint.fingerprint = fingerprintValue.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto setup = parseString(fingerprint.object_items(), "setup")) {
+            parsedFingerprint.setup = setup.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        result.fingerprints.push_back(std::move(parsedFingerprint));
+    }
+
+    const auto candidates = object.find("candidates");
+    if (candidates == object.end() || !candidates->second.is_array()) {
+        return absl::nullopt;
+    }
+    for (const auto &candidate : candidates->second.array_items()) {
+        if (!candidate.is_object()) {
+            return absl::nullopt;
+        }
+
+        GroupJoinTransportDescription::Candidate parsedCandidate;
+
+        if (const auto port = parseString(candidate.object_items(), "port")) {
+            parsedCandidate.port = port.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto protocol = parseString(candidate.object_items(), "protocol")) {
+            parsedCandidate.protocol = protocol.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto network = parseString(candidate.object_items(), "network")) {
+            parsedCandidate.network = network.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto generation = parseString(candidate.object_items(), "generation")) {
+            parsedCandidate.generation = generation.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto id = parseString(candidate.object_items(), "id")) {
+            parsedCandidate.id = id.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto component = parseString(candidate.object_items(), "component")) {
+            parsedCandidate.component = component.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto foundation = parseString(candidate.object_items(), "foundation")) {
+            parsedCandidate.foundation = foundation.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto priority = parseString(candidate.object_items(), "priority")) {
+            parsedCandidate.priority = priority.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto ip = parseString(candidate.object_items(), "ip")) {
+            parsedCandidate.ip = ip.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto type = parseString(candidate.object_items(), "type")) {
+            parsedCandidate.type = type.value();
+        } else {
+            return absl::nullopt;
+        }
+
+        if (const auto tcpType = parseString(candidate.object_items(), "tcptype")) {
+            parsedCandidate.tcpType = tcpType.value();
+        }
+
+        if (const auto relAddr = parseString(candidate.object_items(), "rel-addr")) {
+            parsedCandidate.relAddr = relAddr.value();
+        }
+
+        if (const auto relPort = parseString(candidate.object_items(), "rel-port")) {
+            parsedCandidate.relPort = relPort.value();
+        }
+
+        result.candidates.push_back(std::move(parsedCandidate));
+    }
+
+    return result;
+}
+
+absl::optional<GroupJoinPayloadVideoPayloadType> parsePayloadType(json11::Json::object const &object) {
+    GroupJoinPayloadVideoPayloadType result;
+
+    if (const auto id = parseInt(object, "id")) {
+        result.id = (uint32_t)id.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    if (const auto name = parseString(object, "name")) {
+        result.name = name.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    if (const auto clockrate = parseInt(object, "clockrate")) {
+        result.clockrate = (uint32_t)clockrate.value();
+    } else {
+        result.clockrate = 0;
+    }
+
+    if (const auto channels = parseInt(object, "channels")) {
+        result.channels = (uint32_t)channels.value();
+    } else {
+        result.channels = 1;
+    }
+
+    const auto parameters = object.find("parameters");
+    if (parameters != object.end() && parameters->second.is_object()) {
+        for (const auto &parameter : parameters->second.object_items()) {
+            if (parameter.second.is_string()) {
+                result.parameters.push_back(std::make_pair(parameter.first, parameter.second.string_value()));
+            }
+        }
+    }
+
+    const auto rtcpFbs = object.find("rtcp-fbs");
+    if (rtcpFbs != object.end() && rtcpFbs->second.is_array()) {
+        for (const auto &item : rtcpFbs->second.array_items()) {
+            if (item.is_object()) {
+                const auto type = item.object_items().find("type");
+                if (type != item.object_items().end() && type->second.is_string()) {
+                    GroupJoinPayloadVideoPayloadType::FeedbackType parsedFeedbackType;
+
+                    const auto typeString = type->second.string_value();
+
+                    const auto subtype = item.object_items().find("subtype");
+                    if (subtype != item.object_items().end() && subtype->second.is_string()) {
+                        parsedFeedbackType.type = typeString;
+                        parsedFeedbackType.subtype = subtype->second.string_value();
+                    } else {
+                        auto components = splitString(typeString, ' ');
+                        if (components.size() == 1) {
+                            parsedFeedbackType.type = components[0];
+                        } else if (components.size() == 2) {
+                            parsedFeedbackType.type = components[0];
+                            parsedFeedbackType.subtype = components[1];
+                        } else {
+                            continue;
+                        }
+                    }
+
+                    result.feedbackTypes.push_back(std::move(parsedFeedbackType));
+                }
+            }
+        }
+    }
+
+    return result;
+}
+
+absl::optional<GroupJoinVideoInformation> parseVideoInformation(json11::Json::object const &object) {
+    GroupJoinVideoInformation result;
+
+    const auto serverSources = object.find("server_sources");
+    if (serverSources != object.end() && serverSources->second.is_array()) {
+        for (const auto &item : serverSources->second.array_items()) {
+            if (item.is_number()) {
+                int32_t value = item.int_value();
+                uint32_t unsignedValue = *(uint32_t *)&value;
+                result.serverVideoBandwidthProbingSsrc = unsignedValue;
+            }
+        }
+    }
+
+    const auto payloadTypes = object.find("payload-types");
+    if (payloadTypes != object.end() && payloadTypes->second.is_array()) {
+        for (const auto &payloadType : payloadTypes->second.array_items()) {
+            if (payloadType.is_object()) {
+                if (const auto parsedPayloadType = parsePayloadType(payloadType.object_items())) {
+                    result.payloadTypes.push_back(parsedPayloadType.value());
+                }
+            }
+        }
+    }
+
+    const auto rtpHdrexts = object.find("rtp-hdrexts");
+    if (rtpHdrexts != object.end() && rtpHdrexts->second.is_array()) {
+        for (const auto &rtpHdrext : rtpHdrexts->second.array_items()) {
+            if (rtpHdrext.is_object()) {
+                const auto id = rtpHdrext.object_items().find("id");
+                if (id == rtpHdrext.object_items().end() || !id->second.is_number()) {
+                    continue;
+                }
+
+                const auto uri = rtpHdrext.object_items().find("uri");
+                if (uri == rtpHdrext.object_items().end() || !uri->second.is_string()) {
+                    continue;
+                }
+
+                result.extensionMap.push_back(std::make_pair(id->second.int_value(), uri->second.string_value()));
+            }
+        }
+    }
+
+    const auto endpointId = object.find("endpoint");
+    if (endpointId != object.end() && endpointId->second.is_string()) {
+        result.endpointId = endpointId->second.string_value();
+    }
+
+    return result;
+}
+
+}
+
+std::string GroupJoinInternalPayload::serialize() {
+    json11::Json::object object;
+
+    int32_t signedSsrc = *(int32_t *)&audioSsrc;
+
+    object.insert(std::make_pair("ssrc", json11::Json(signedSsrc)));
+    object.insert(std::make_pair("ufrag", json11::Json(transport.ufrag)));
+    object.insert(std::make_pair("pwd", json11::Json(transport.pwd)));
+
+    json11::Json::array fingerprints;
+    for (const auto &fingerprint : transport.fingerprints) {
+        json11::Json::object fingerprintJson;
+
+        fingerprintJson.insert(std::make_pair("hash", json11::Json(fingerprint.hash)));
+        fingerprintJson.insert(std::make_pair("fingerprint", json11::Json(fingerprint.fingerprint)));
+        fingerprintJson.insert(std::make_pair("setup", json11::Json(fingerprint.setup)));
+
+        fingerprints.push_back(json11::Json(std::move(fingerprintJson)));
+    }
+    object.insert(std::make_pair("fingerprints", json11::Json(std::move(fingerprints))));
+
+    if (videoInformation) {
+        json11::Json::array ssrcGroups;
+        for (const auto &ssrcGroup : videoInformation->ssrcGroups) {
+            json11::Json::object ssrcGroupJson;
+
+            json11::Json::array ssrcGroupSources;
+            for (auto ssrc : ssrcGroup.ssrcs) {
+                int32_t signedValue = *(int32_t *)&ssrc;
+                ssrcGroupSources.push_back(json11::Json(signedValue));
+            }
+
+            ssrcGroupJson.insert(std::make_pair("sources", json11::Json(std::move(ssrcGroupSources))));
+            ssrcGroupJson.insert(std::make_pair("semantics", json11::Json(ssrcGroup.semantics)));
+
+            ssrcGroups.push_back(json11::Json(std::move(ssrcGroupJson)));
+        }
+        object.insert(std::make_pair("ssrc-groups", json11::Json(std::move(ssrcGroups))));
+    }
+
+    auto json = json11::Json(std::move(object));
+    return json.dump();
+}
+
+absl::optional<GroupJoinResponsePayload> GroupJoinResponsePayload::parse(std::string const &data) {
+    std::string parsingError;
+    auto json = json11::Json::parse(std::string(data.begin(), data.end()), parsingError);
+    if (json.type() != json11::Json::OBJECT) {
+        return absl::nullopt;
+    }
+
+    tgcalls::GroupJoinResponsePayload result;
+
+    const auto transport = json.object_items().find("transport");
+    if (transport == json.object_items().end() || !transport->second.is_object()) {
+        return absl::nullopt;
+    }
+    if (const auto parsedTransport = parseTransportDescription(transport->second.object_items())) {
+        result.transport = parsedTransport.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    const auto video = json.object_items().find("video");
+    if (video != json.object_items().end() && video->second.is_object()) {
+        result.videoInformation = parseVideoInformation(video->second.object_items());
+    }
+
+    return result;
+}
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/group/GroupJoinPayloadInternal.h

@@ -0,0 +1,32 @@
+#ifndef TGCALLS_GROUP_JOIN_PAYLOAD_INTERNAL_H
+#define TGCALLS_GROUP_JOIN_PAYLOAD_INTERNAL_H
+
+#include "GroupJoinPayload.h"
+
+#include <vector>
+#include <string>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+
+namespace tgcalls {
+
+struct GroupJoinResponsePayload {
+    GroupJoinTransportDescription transport;
+    absl::optional<GroupJoinVideoInformation> videoInformation;
+
+    static absl::optional<GroupJoinResponsePayload> parse(std::string const &data);
+};
+
+struct GroupJoinInternalPayload {
+    GroupJoinTransportDescription transport;
+
+    uint32_t audioSsrc = 0;
+    absl::optional<GroupParticipantVideoInformation> videoInformation;
+
+    std::string serialize();
+};
+
+}
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/GroupNetworkManager.cpp

@@ -0,0 +1,680 @@
+#include "group/GroupNetworkManager.h"
+
+#include "p2p/base/basic_packet_socket_factory.h"
+#include "p2p/client/basic_port_allocator.h"
+#include "p2p/base/p2p_transport_channel.h"
+#include "p2p/base/basic_async_resolver_factory.h"
+#include "api/packet_socket_factory.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "p2p/base/ice_credentials_iterator.h"
+#include "api/jsep_ice_candidate.h"
+#include "p2p/base/dtls_transport.h"
+#include "p2p/base/dtls_transport_factory.h"
+#include "pc/dtls_srtp_transport.h"
+#include "pc/dtls_transport.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "platform/PlatformInterface.h"
+#include "TurnCustomizerImpl.h"
+#include "SctpDataChannelProviderInterfaceImpl.h"
+#include "StaticThreads.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+
+namespace tgcalls {
+
+enum {
+    kRtcpExpectedVersion = 2,
+    kRtcpMinHeaderLength = 4,
+    kRtcpMinParseLength = 8,
+
+    kRtpExpectedVersion = 2,
+    kRtpMinParseLength = 12
+};
+
+static void updateHeaderWithVoiceActivity(rtc::CopyOnWriteBuffer *packet, const uint8_t* ptrRTPDataExtensionEnd, const uint8_t* ptr, bool voiceActivity, bool zeroAudioLevel) {
+    while (ptrRTPDataExtensionEnd - ptr > 0) {
+        //  0
+        //  0 1 2 3 4 5 6 7
+        // +-+-+-+-+-+-+-+-+
+        // |  ID   |  len  |
+        // +-+-+-+-+-+-+-+-+
+
+        // Note that 'len' is the header extension element length, which is the
+        // number of bytes - 1.
+        const int id = (*ptr & 0xf0) >> 4;
+        const int len = (*ptr & 0x0f);
+        ptr++;
+
+        if (id == 0) {
+            // Padding byte, skip ignoring len.
+            continue;
+        }
+
+        if (id == 15) {
+            RTC_LOG(LS_VERBOSE)
+            << "RTP extension header 15 encountered. Terminate parsing.";
+            return;
+        }
+
+        if (ptrRTPDataExtensionEnd - ptr < (len + 1)) {
+            RTC_LOG(LS_WARNING) << "Incorrect one-byte extension len: " << (len + 1)
+            << ", bytes left in buffer: "
+            << (ptrRTPDataExtensionEnd - ptr);
+            return;
+        }
+
+        if (id == 1) { // kAudioLevelUri
+            uint8_t audioLevel = ptr[0] & 0x7f;
+            if (zeroAudioLevel) {
+                if (audioLevel < 47) {
+                    audioLevel = 0;
+                } else if (audioLevel < 107) {
+                    audioLevel = 106;
+                } else {
+                    audioLevel = 127;
+                }
+            }
+            bool parsedVoiceActivity = (ptr[0] & 0x80) != 0;
+
+            if (parsedVoiceActivity != voiceActivity) {
+                ptrdiff_t byteOffset = ptr - packet->data();
+                uint8_t *mutableBytes = packet->MutableData();
+                uint8_t audioActivityBit = voiceActivity ? 0x80 : 0;
+                mutableBytes[byteOffset] = audioLevel | audioActivityBit;
+            }
+            return;
+        }
+
+        ptr += (len + 1);
+    }
+}
+
+#if 0 // Currently unused.
+static void readHeaderVoiceActivity(const uint8_t* ptrRTPDataExtensionEnd, const uint8_t* ptr, bool &didRead, uint8_t &audioLevel, bool &voiceActivity) {
+    while (ptrRTPDataExtensionEnd - ptr > 0) {
+        //  0
+        //  0 1 2 3 4 5 6 7
+        // +-+-+-+-+-+-+-+-+
+        // |  ID   |  len  |
+        // +-+-+-+-+-+-+-+-+
+
+        // Note that 'len' is the header extension element length, which is the
+        // number of bytes - 1.
+        const int id = (*ptr & 0xf0) >> 4;
+        const int len = (*ptr & 0x0f);
+        ptr++;
+
+        if (id == 0) {
+            // Padding byte, skip ignoring len.
+            continue;
+        }
+
+        if (id == 15) {
+            RTC_LOG(LS_VERBOSE)
+            << "RTP extension header 15 encountered. Terminate parsing.";
+            return;
+        }
+
+        if (ptrRTPDataExtensionEnd - ptr < (len + 1)) {
+            RTC_LOG(LS_WARNING) << "Incorrect one-byte extension len: " << (len + 1)
+            << ", bytes left in buffer: "
+            << (ptrRTPDataExtensionEnd - ptr);
+            return;
+        }
+
+        if (id == 1) { // kAudioLevelUri
+            didRead = true;
+            audioLevel = ptr[0] & 0x7f;
+            voiceActivity = (ptr[0] & 0x80) != 0;
+
+            return;
+        }
+
+        ptr += (len + 1);
+    }
+}
+#endif
+
+
+static void maybeUpdateRtpVoiceActivity(rtc::CopyOnWriteBuffer *packet, bool voiceActivity, bool zeroAudioLevel) {
+    const uint8_t *_ptrRTPDataBegin = packet->data();
+    const uint8_t *_ptrRTPDataEnd = packet->data() + packet->size();
+
+    const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
+    if (length < kRtpMinParseLength) {
+        return;
+    }
+
+    // Version
+    const uint8_t V = _ptrRTPDataBegin[0] >> 6;
+    // eXtension
+    const bool X = ((_ptrRTPDataBegin[0] & 0x10) == 0) ? false : true;
+    const uint8_t CC = _ptrRTPDataBegin[0] & 0x0f;
+
+    const uint8_t PT = _ptrRTPDataBegin[1] & 0x7f;
+
+    const uint8_t* ptr = &_ptrRTPDataBegin[4];
+
+    ptr += 4;
+
+    ptr += 4;
+
+    if (V != kRtpExpectedVersion) {
+        return;
+    }
+
+    const size_t CSRCocts = CC * 4;
+
+    if ((ptr + CSRCocts) > _ptrRTPDataEnd) {
+        return;
+    }
+
+    if (PT != 111) {
+        return;
+    }
+
+    for (uint8_t i = 0; i < CC; ++i) {
+        ptr += 4;
+    }
+
+    if (X) {
+      /* RTP header extension, RFC 3550.
+       0                   1                   2                   3
+       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      |      defined by profile       |           length              |
+      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      |                        header extension                       |
+      |                             ....                              |
+      */
+      const ptrdiff_t remain = _ptrRTPDataEnd - ptr;
+      if (remain < 4) {
+          return;
+      }
+
+      uint16_t definedByProfile = webrtc::ByteReader<uint16_t>::ReadBigEndian(ptr);
+      ptr += 2;
+
+      // in 32 bit words
+      size_t XLen = webrtc::ByteReader<uint16_t>::ReadBigEndian(ptr);
+      ptr += 2;
+      XLen *= 4;  // in bytes
+
+      if (static_cast<size_t>(remain) < (4 + XLen)) {
+          return;
+      }
+      static constexpr uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
+      if (definedByProfile == kRtpOneByteHeaderExtensionId) {
+          const uint8_t* ptrRTPDataExtensionEnd = ptr + XLen;
+          updateHeaderWithVoiceActivity(packet, ptrRTPDataExtensionEnd, ptr, voiceActivity, zeroAudioLevel);
+      }
+    }
+}
+
+#if 0 // Currently unused.
+static void maybeReadRtpVoiceActivity(rtc::CopyOnWriteBuffer *packet, bool &didRead, uint32_t &ssrc, uint8_t &audioLevel, bool &voiceActivity) {
+    const uint8_t *_ptrRTPDataBegin = packet->data();
+    const uint8_t *_ptrRTPDataEnd = packet->data() + packet->size();
+
+    const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
+    if (length < kRtpMinParseLength) {
+        return;
+    }
+
+    // Version
+    const uint8_t V = _ptrRTPDataBegin[0] >> 6;
+    // eXtension
+    const bool X = ((_ptrRTPDataBegin[0] & 0x10) == 0) ? false : true;
+    const uint8_t CC = _ptrRTPDataBegin[0] & 0x0f;
+
+    const uint8_t PT = _ptrRTPDataBegin[1] & 0x7f;
+
+    const uint8_t* ptr = &_ptrRTPDataBegin[4];
+
+    ptr += 4;
+
+    ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(ptr);
+    ptr += 4;
+
+    if (V != kRtpExpectedVersion) {
+        return;
+    }
+
+    const size_t CSRCocts = CC * 4;
+
+    if ((ptr + CSRCocts) > _ptrRTPDataEnd) {
+        return;
+    }
+
+    if (PT != 111) {
+        return;
+    }
+
+    for (uint8_t i = 0; i < CC; ++i) {
+        ptr += 4;
+    }
+
+    if (X) {
+      /* RTP header extension, RFC 3550.
+       0                   1                   2                   3
+       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      |      defined by profile       |           length              |
+      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+      |                        header extension                       |
+      |                             ....                              |
+      */
+      const ptrdiff_t remain = _ptrRTPDataEnd - ptr;
+      if (remain < 4) {
+          return;
+      }
+
+      uint16_t definedByProfile = webrtc::ByteReader<uint16_t>::ReadBigEndian(ptr);
+      ptr += 2;
+
+      // in 32 bit words
+      size_t XLen = webrtc::ByteReader<uint16_t>::ReadBigEndian(ptr);
+      ptr += 2;
+      XLen *= 4;  // in bytes
+
+      if (static_cast<size_t>(remain) < (4 + XLen)) {
+          return;
+      }
+      static constexpr uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
+      if (definedByProfile == kRtpOneByteHeaderExtensionId) {
+          const uint8_t* ptrRTPDataExtensionEnd = ptr + XLen;
+          readHeaderVoiceActivity(ptrRTPDataExtensionEnd, ptr, didRead, audioLevel, voiceActivity);
+      }
+    }
+}
+#endif
+
+class WrappedDtlsSrtpTransport : public webrtc::DtlsSrtpTransport {
+public:
+    bool _voiceActivity = false;
+
+public:
+    WrappedDtlsSrtpTransport(bool rtcp_mux_enabled, const webrtc::FieldTrialsView& fieldTrials, std::function<void(webrtc::RtpPacketReceived const &, bool)> &&processRtpPacket, bool zeroAudioLevel) :
+    webrtc::DtlsSrtpTransport(rtcp_mux_enabled, fieldTrials),
+    _processRtpPacket(std::move(processRtpPacket)),
+    _zeroAudioLevel(zeroAudioLevel) {
+    }
+
+    virtual ~WrappedDtlsSrtpTransport() {
+    }
+
+    bool SendRtpPacket(rtc::CopyOnWriteBuffer *packet, const rtc::PacketOptions& options, int flags) override {
+        maybeUpdateRtpVoiceActivity(packet, _voiceActivity, _zeroAudioLevel);
+        return webrtc::DtlsSrtpTransport::SendRtpPacket(packet, options, flags);
+    }
+    
+    void ProcessRtpPacket(webrtc::RtpPacketReceived const &packet, bool isUnresolved) override {
+        _processRtpPacket(packet, isUnresolved);
+    }
+
+private:
+    std::function<void(webrtc::RtpPacketReceived const &, bool)> _processRtpPacket;
+    bool _zeroAudioLevel;
+};
+
+webrtc::CryptoOptions GroupNetworkManager::getDefaulCryptoOptions() {
+    auto options = webrtc::CryptoOptions();
+    options.srtp.enable_aes128_sha1_80_crypto_cipher = false;
+    options.srtp.enable_gcm_crypto_suites = true;
+    return options;
+}
+
+GroupNetworkManager::GroupNetworkManager(
+    const webrtc::FieldTrialsView &fieldTrials,
+    std::function<void(const State &)> stateUpdated,
+    std::function<void(uint32_t, int)> unknownSsrcPacketReceived,
+    std::function<void(bool)> dataChannelStateUpdated,
+    std::function<void(std::string const &)> dataChannelMessageReceived,
+    std::function<void(uint32_t, uint8_t, bool)> audioActivityUpdated,
+    bool zeroAudioLevel,
+    std::function<void(uint32_t)> anyActivityUpdated,
+    std::shared_ptr<Threads> threads) :
+_threads(std::move(threads)),
+_stateUpdated(std::move(stateUpdated)),
+_unknownSsrcPacketReceived(std::move(unknownSsrcPacketReceived)),
+_dataChannelStateUpdated(dataChannelStateUpdated),
+_dataChannelMessageReceived(dataChannelMessageReceived),
+_audioActivityUpdated(audioActivityUpdated),
+_zeroAudioLevel(zeroAudioLevel),
+_anyActivityUpdated(anyActivityUpdated) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    _localIceParameters = PeerIceParameters(rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH), rtc::CreateRandomString(cricket::ICE_PWD_LENGTH), false);
+
+    _localCertificate = rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(rtc::KT_ECDSA), absl::nullopt);
+
+    _networkMonitorFactory = PlatformInterface::SharedInstance()->createNetworkMonitorFactory();
+
+    _socketFactory.reset(new rtc::BasicPacketSocketFactory(_threads->getNetworkThread()->socketserver()));
+    _networkManager = std::make_unique<rtc::BasicNetworkManager>(_networkMonitorFactory.get(), _threads->getNetworkThread()->socketserver());
+    _asyncResolverFactory = std::make_unique<webrtc::BasicAsyncDnsResolverFactory>();
+
+    _dtlsSrtpTransport = std::make_unique<WrappedDtlsSrtpTransport>(true, fieldTrials, [this](webrtc::RtpPacketReceived const &packet, bool isUnresolved) {
+        this->RtpPacketReceived_n(packet, isUnresolved);
+    }, _zeroAudioLevel);
+    _dtlsSrtpTransport->SetDtlsTransports(nullptr, nullptr);
+    _dtlsSrtpTransport->SetActiveResetSrtpParams(false);
+    _dtlsSrtpTransport->SubscribeReadyToSend(this, [this](bool value) {
+        this->DtlsReadyToSend(value);
+    });
+
+    resetDtlsSrtpTransport();
+}
+
+GroupNetworkManager::~GroupNetworkManager() {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    RTC_LOG(LS_INFO) << "GroupNetworkManager::~GroupNetworkManager()";
+
+    _dataChannelInterface.reset();
+    _dtlsSrtpTransport.reset();
+    _dtlsTransport.reset();
+    _transportChannel.reset();
+    _asyncResolverFactory.reset();
+    _portAllocator.reset();
+    _networkManager.reset();
+    _socketFactory.reset();
+}
+
+void GroupNetworkManager::resetDtlsSrtpTransport() {
+    std::unique_ptr<cricket::BasicPortAllocator> portAllocator = std::make_unique<cricket::BasicPortAllocator>(_networkManager.get(), _socketFactory.get(), _turnCustomizer.get(), nullptr);
+    
+    uint32_t flags = portAllocator->flags();
+    
+    flags |=
+        cricket::PORTALLOCATOR_ENABLE_IPV6 |
+        cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
+    
+    portAllocator->set_flags(flags);
+    portAllocator->Initialize();
+
+    portAllocator->SetConfiguration({}, {}, 2, webrtc::NO_PRUNE, _turnCustomizer.get());
+
+    webrtc::IceTransportInit iceTransportInit;
+    iceTransportInit.set_port_allocator(portAllocator.get());
+    iceTransportInit.set_async_dns_resolver_factory(_asyncResolverFactory.get());
+
+    auto transportChannel = cricket::P2PTransportChannel::Create("transport", 0, std::move(iceTransportInit));
+
+    cricket::IceConfig iceConfig;
+    iceConfig.continual_gathering_policy = cricket::GATHER_CONTINUALLY;
+    iceConfig.prioritize_most_likely_candidate_pairs = true;
+    iceConfig.regather_on_failed_networks_interval = 2000;
+    transportChannel->SetIceConfig(iceConfig);
+
+    cricket::IceParameters localIceParameters(
+        _localIceParameters.ufrag,
+        _localIceParameters.pwd,
+        false
+    );
+
+    transportChannel->SetIceParameters(localIceParameters);
+    const bool isOutgoing = false;
+    transportChannel->SetIceRole(isOutgoing ? cricket::ICEROLE_CONTROLLING : cricket::ICEROLE_CONTROLLED);
+    transportChannel->SetRemoteIceMode(cricket::ICEMODE_LITE);
+
+    transportChannel->SignalIceTransportStateChanged.connect(this, &GroupNetworkManager::transportStateChanged);
+    transportChannel->SignalReadPacket.connect(this, &GroupNetworkManager::transportPacketReceived);
+
+    webrtc::CryptoOptions cryptoOptions = GroupNetworkManager::getDefaulCryptoOptions();
+
+    auto dtlsTransport = std::make_unique<cricket::DtlsTransport>(transportChannel.get(), cryptoOptions, nullptr);
+
+    dtlsTransport->SignalWritableState.connect(
+        this, &GroupNetworkManager::OnTransportWritableState_n);
+    dtlsTransport->SignalReceivingState.connect(
+        this, &GroupNetworkManager::OnTransportReceivingState_n);
+
+    dtlsTransport->SetDtlsRole(rtc::SSLRole::SSL_SERVER);
+    dtlsTransport->SetLocalCertificate(_localCertificate);
+
+    _dtlsSrtpTransport->SetDtlsTransports(dtlsTransport.get(), nullptr);
+
+    _dtlsTransport = std::move(dtlsTransport);
+    _transportChannel = std::move(transportChannel);
+    _portAllocator = std::move(portAllocator);
+}
+
+void GroupNetworkManager::start() {
+    _transportChannel->MaybeStartGathering();
+
+    restartDataChannel();
+}
+
+void GroupNetworkManager::restartDataChannel() {
+    _dataChannelStateUpdated(false);
+
+    const auto weak = std::weak_ptr<GroupNetworkManager>(shared_from_this());
+    _dataChannelInterface.reset(new SctpDataChannelProviderInterfaceImpl(
+        _dtlsTransport.get(),
+        true,
+        [weak, threads = _threads](bool state) {
+            assert(threads->getNetworkThread()->IsCurrent());
+            const auto strong = weak.lock();
+            if (!strong) {
+                return;
+            }
+            strong->_dataChannelStateUpdated(state);
+        },
+        [weak, threads = _threads]() {
+            assert(threads->getNetworkThread()->IsCurrent());
+            const auto strong = weak.lock();
+            if (!strong) {
+                return;
+            }
+            strong->restartDataChannel();
+        },
+        [weak, threads = _threads](std::string const &message) {
+            assert(threads->getNetworkThread()->IsCurrent());
+            const auto strong = weak.lock();
+            if (!strong) {
+                return;
+            }
+            strong->_dataChannelMessageReceived(message);
+        },
+        _threads
+    ));
+
+    _dataChannelInterface->updateIsConnected(_isConnected);
+}
+
+void GroupNetworkManager::stop() {
+    _transportChannel->SignalIceTransportStateChanged.disconnect(this);
+    _transportChannel->SignalReadPacket.disconnect(this);
+
+    _dtlsTransport->SignalWritableState.disconnect(this);
+    _dtlsTransport->SignalReceivingState.disconnect(this);
+
+    _dataChannelInterface.reset();
+
+    _localIceParameters = PeerIceParameters(rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH), rtc::CreateRandomString(cricket::ICE_PWD_LENGTH), false);
+
+    _localCertificate = rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(rtc::KT_ECDSA), absl::nullopt);
+
+    resetDtlsSrtpTransport();
+}
+
+PeerIceParameters GroupNetworkManager::getLocalIceParameters() {
+    return _localIceParameters;
+}
+
+std::unique_ptr<rtc::SSLFingerprint> GroupNetworkManager::getLocalFingerprint() {
+    auto certificate = _localCertificate;
+    if (!certificate) {
+        return nullptr;
+    }
+    return rtc::SSLFingerprint::CreateFromCertificate(*certificate);
+}
+
+void GroupNetworkManager::setRemoteParams(PeerIceParameters const &remoteIceParameters, std::vector<cricket::Candidate> const &iceCandidates, rtc::SSLFingerprint *fingerprint) {
+    _remoteIceParameters = remoteIceParameters;
+
+    cricket::IceParameters parameters(
+        remoteIceParameters.ufrag,
+        remoteIceParameters.pwd,
+        true
+    );
+
+    _transportChannel->SetRemoteIceParameters(parameters);
+
+    for (const auto &candidate : iceCandidates) {
+        _transportChannel->AddRemoteCandidate(candidate);
+    }
+
+    if (fingerprint) {
+        _dtlsTransport->SetRemoteFingerprint(fingerprint->algorithm, fingerprint->digest.data(), fingerprint->digest.size());
+    }
+}
+
+void GroupNetworkManager::sendDataChannelMessage(std::string const &message) {
+    if (_dataChannelInterface) {
+        _dataChannelInterface->sendDataChannelMessage(message);
+    }
+}
+
+void GroupNetworkManager::setOutgoingVoiceActivity(bool isSpeech) {
+    if (_dtlsSrtpTransport) {
+        ((WrappedDtlsSrtpTransport *)_dtlsSrtpTransport.get())->_voiceActivity = isSpeech;
+    }
+}
+
+webrtc::RtpTransport *GroupNetworkManager::getRtpTransport() {
+    return _dtlsSrtpTransport.get();
+}
+
+void GroupNetworkManager::checkConnectionTimeout() {
+    const auto weak = std::weak_ptr<GroupNetworkManager>(shared_from_this());
+    _threads->getNetworkThread()->PostDelayedTask([weak]() {
+        auto strong = weak.lock();
+        if (!strong) {
+            return;
+        }
+
+        int64_t currentTimestamp = rtc::TimeMillis();
+        const int64_t maxTimeout = 20000;
+
+        if (strong->_lastNetworkActivityMs + maxTimeout < currentTimestamp) {
+            GroupNetworkManager::State emitState;
+            emitState.isReadyToSendData = false;
+            emitState.isFailed = true;
+            strong->_stateUpdated(emitState);
+        }
+
+        strong->checkConnectionTimeout();
+    }, webrtc::TimeDelta::Millis(1000));
+}
+
+void GroupNetworkManager::candidateGathered(cricket::IceTransportInternal *transport, const cricket::Candidate &candidate) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+}
+
+void GroupNetworkManager::candidateGatheringState(cricket::IceTransportInternal *transport) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+}
+
+void GroupNetworkManager::OnTransportWritableState_n(rtc::PacketTransportInternal *transport) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    UpdateAggregateStates_n();
+}
+void GroupNetworkManager::OnTransportReceivingState_n(rtc::PacketTransportInternal *transport) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    UpdateAggregateStates_n();
+}
+
+void GroupNetworkManager::DtlsReadyToSend(bool isReadyToSend) {
+    UpdateAggregateStates_n();
+
+    if (isReadyToSend) {
+        const auto weak = std::weak_ptr<GroupNetworkManager>(shared_from_this());
+        _threads->getNetworkThread()->PostTask([weak]() {
+            const auto strong = weak.lock();
+            if (!strong) {
+                return;
+            }
+            strong->UpdateAggregateStates_n();
+        });
+    }
+}
+
+void GroupNetworkManager::transportStateChanged(cricket::IceTransportInternal *transport) {
+    UpdateAggregateStates_n();
+}
+
+void GroupNetworkManager::transportReadyToSend(cricket::IceTransportInternal *transport) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+}
+
+void GroupNetworkManager::transportPacketReceived(rtc::PacketTransportInternal *transport, const char *bytes, size_t size, const int64_t &timestamp, int unused) {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    _lastNetworkActivityMs = rtc::TimeMillis();
+}
+
+void GroupNetworkManager::RtpPacketReceived_n(webrtc::RtpPacketReceived const &packet, bool isUnresolved) {
+    if (packet.HasExtension(webrtc::kRtpExtensionAudioLevel)) {
+        uint8_t audioLevel = 0;
+        bool isSpeech = false;
+
+        if (packet.GetExtension<webrtc::AudioLevel>(&isSpeech, &audioLevel)) {
+            if (_audioActivityUpdated) {
+                _audioActivityUpdated(packet.Ssrc(), audioLevel, isSpeech);
+            }
+        }
+    }
+    
+    if (_anyActivityUpdated) {
+        _anyActivityUpdated(packet.Ssrc());
+    }
+
+    if (isUnresolved && _unknownSsrcPacketReceived) {
+        uint32_t ssrc = packet.Ssrc();
+        int payloadType = packet.PayloadType();
+
+        _unknownSsrcPacketReceived(ssrc, payloadType);
+    }
+}
+
+void GroupNetworkManager::UpdateAggregateStates_n() {
+    assert(_threads->getNetworkThread()->IsCurrent());
+
+    auto state = _transportChannel->GetIceTransportState();
+    bool isConnected = false;
+    switch (state) {
+        case webrtc::IceTransportState::kConnected:
+        case webrtc::IceTransportState::kCompleted:
+            isConnected = true;
+            break;
+        default:
+            break;
+    }
+
+    if (!_dtlsSrtpTransport->IsWritable(false)) {
+        isConnected = false;
+    }
+
+    if (_isConnected != isConnected) {
+        _isConnected = isConnected;
+
+        GroupNetworkManager::State emitState;
+        emitState.isReadyToSendData = isConnected;
+        _stateUpdated(emitState);
+
+        if (_dataChannelInterface) {
+            _dataChannelInterface->updateIsConnected(isConnected);
+        }
+    }
+}
+
+void GroupNetworkManager::sctpReadyToSendData() {
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/group/GroupNetworkManager.h

@@ -0,0 +1,133 @@
+#ifndef TGCALLS_GROUP_NETWORK_MANAGER_H
+#define TGCALLS_GROUP_NETWORK_MANAGER_H
+
+#ifdef WEBRTC_WIN
+// Compiler errors in conflicting Windows headers if not included here.
+#include <winsock2.h>
+#endif // WEBRTC_WIN
+
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+#include "api/candidate.h"
+#include "media/base/media_channel.h"
+#include "pc/sctp_transport.h"
+#include "pc/sctp_data_channel.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+
+#include <functional>
+#include <memory>
+
+#include "Message.h"
+#include "ThreadLocalObject.h"
+
+namespace rtc {
+class BasicPacketSocketFactory;
+class BasicNetworkManager;
+class PacketTransportInternal;
+struct NetworkRoute;
+} // namespace rtc
+
+namespace cricket {
+class BasicPortAllocator;
+class P2PTransportChannel;
+class IceTransportInternal;
+class DtlsTransport;
+} // namespace cricket
+
+namespace webrtc {
+class BasicAsyncResolverFactory;
+class TurnCustomizer;
+class DtlsSrtpTransport;
+class RtpTransport;
+} // namespace webrtc
+
+namespace tgcalls {
+
+struct Message;
+class SctpDataChannelProviderInterfaceImpl;
+class Threads;
+
+class GroupNetworkManager : public sigslot::has_slots<>, public std::enable_shared_from_this<GroupNetworkManager> {
+public:
+    struct State {
+        bool isReadyToSendData = false;
+        bool isFailed = false;
+    };
+    
+    static webrtc::CryptoOptions getDefaulCryptoOptions();
+
+    GroupNetworkManager(
+        const webrtc::FieldTrialsView& fieldTrials,
+        std::function<void(const State &)> stateUpdated,
+        std::function<void(uint32_t, int)> unknownSsrcPacketReceived,
+        std::function<void(bool)> dataChannelStateUpdated,
+        std::function<void(std::string const &)> dataChannelMessageReceived,
+        std::function<void(uint32_t, uint8_t, bool)> audioActivityUpdated,
+        bool zeroAudioLevel,
+        std::function<void(uint32_t)> anyActivityUpdated,
+        std::shared_ptr<Threads> threads);
+    ~GroupNetworkManager();
+
+    void start();
+    void stop();
+
+    PeerIceParameters getLocalIceParameters();
+    std::unique_ptr<rtc::SSLFingerprint> getLocalFingerprint();
+    void setRemoteParams(PeerIceParameters const &remoteIceParameters, std::vector<cricket::Candidate> const &iceCandidates, rtc::SSLFingerprint *fingerprint);
+
+    void sendDataChannelMessage(std::string const &message);
+
+    void setOutgoingVoiceActivity(bool isSpeech);
+
+    webrtc::RtpTransport *getRtpTransport();
+
+private:
+    void resetDtlsSrtpTransport();
+    void restartDataChannel();
+    void checkConnectionTimeout();
+    void candidateGathered(cricket::IceTransportInternal *transport, const cricket::Candidate &candidate);
+    void candidateGatheringState(cricket::IceTransportInternal *transport);
+    void OnTransportWritableState_n(rtc::PacketTransportInternal *transport);
+    void OnTransportReceivingState_n(rtc::PacketTransportInternal *transport);
+    void transportStateChanged(cricket::IceTransportInternal *transport);
+    void transportReadyToSend(cricket::IceTransportInternal *transport);
+    void transportPacketReceived(rtc::PacketTransportInternal *transport, const char *bytes, size_t size, const int64_t &timestamp, int unused);
+    void DtlsReadyToSend(bool DtlsReadyToSend);
+    void UpdateAggregateStates_n();
+    void RtpPacketReceived_n(webrtc::RtpPacketReceived const &packet, bool isUnresolved);
+    void OnRtcpPacketReceived_n(rtc::CopyOnWriteBuffer *packet, int64_t packet_time_us);
+
+    void sctpReadyToSendData();
+
+    std::shared_ptr<Threads> _threads;
+    std::function<void(const GroupNetworkManager::State &)> _stateUpdated;
+    std::function<void(uint32_t, int)> _unknownSsrcPacketReceived;
+    std::function<void(bool)> _dataChannelStateUpdated;
+    std::function<void(std::string const &)> _dataChannelMessageReceived;
+    std::function<void(uint32_t, uint8_t, bool)> _audioActivityUpdated;
+    bool _zeroAudioLevel = false;
+    std::function<void(uint32_t)> _anyActivityUpdated;
+
+    std::unique_ptr<rtc::NetworkMonitorFactory> _networkMonitorFactory;
+    std::unique_ptr<rtc::BasicPacketSocketFactory> _socketFactory;
+    std::unique_ptr<rtc::BasicNetworkManager> _networkManager;
+    std::unique_ptr<webrtc::TurnCustomizer> _turnCustomizer;
+    std::unique_ptr<cricket::BasicPortAllocator> _portAllocator;
+    std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface> _asyncResolverFactory;
+    std::unique_ptr<cricket::P2PTransportChannel> _transportChannel;
+    std::unique_ptr<cricket::DtlsTransport> _dtlsTransport;
+    std::unique_ptr<webrtc::DtlsSrtpTransport> _dtlsSrtpTransport;
+
+    std::unique_ptr<SctpDataChannelProviderInterfaceImpl> _dataChannelInterface;
+
+    webrtc::scoped_refptr<rtc::RTCCertificate> _localCertificate;
+    PeerIceParameters _localIceParameters;
+    absl::optional<PeerIceParameters> _remoteIceParameters;
+
+    bool _isConnected = false;
+    int64_t _lastNetworkActivityMs = 0;
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/StreamingMediaContext.h

@@ -0,0 +1,53 @@
+#ifndef TGCALLS_STREAMING_MEDIA_CONTEXT_H
+#define TGCALLS_STREAMING_MEDIA_CONTEXT_H
+
+#include "GroupInstanceImpl.h"
+#include <stdint.h>
+#include "../StaticThreads.h"
+
+namespace tgcalls {
+
+class StreamingMediaContextPrivate;
+
+class StreamingMediaContext {
+public:
+    struct VideoChannel {
+        VideoChannelDescription::Quality quality = VideoChannelDescription::Quality::Thumbnail;
+        std::string endpoint;
+
+        VideoChannel(VideoChannelDescription::Quality quality_, std::string endpoint_) :
+        quality(quality_),
+        endpoint(endpoint_) {
+        }
+    };
+
+public:
+    struct StreamingMediaContextArguments {
+        std::shared_ptr<Threads> threads;
+        bool isUnifiedBroadcast = false;
+        std::function<std::shared_ptr<BroadcastPartTask>(std::function<void(int64_t)>)> requestCurrentTime;
+        std::function<std::shared_ptr<BroadcastPartTask>(int64_t, int64_t, std::function<void(BroadcastPart &&)>)> requestAudioBroadcastPart;
+        std::function<std::shared_ptr<BroadcastPartTask>(int64_t, int64_t, int32_t, VideoChannelDescription::Quality, std::function<void(BroadcastPart &&)>)> requestVideoBroadcastPart;
+        std::function<void(uint32_t, float, bool)> updateAudioLevel;
+    };
+
+public:
+    StreamingMediaContext(StreamingMediaContextArguments &&arguments);
+    ~StreamingMediaContext();
+    
+    StreamingMediaContext& operator=(const StreamingMediaContext&) = delete;
+    StreamingMediaContext& operator=(StreamingMediaContext&&) = delete;
+
+    void setActiveVideoChannels(std::vector<VideoChannel> const &videoChannels);
+    void setVolume(uint32_t ssrc, double volume);
+    void addVideoSink(std::string const &endpointId, std::weak_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+
+    void getAudio(int16_t *audio_samples, const size_t num_samples, const size_t num_channels, const uint32_t samples_per_sec);
+    
+private:
+    std::shared_ptr<StreamingMediaContextPrivate> _private;
+};
+
+}
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/group/VideoStreamingPart.cpp

@@ -0,0 +1,925 @@
+#include "VideoStreamingPart.h"
+
+#include "rtc_base/logging.h"
+#include "rtc_base/third_party/base64/base64.h"
+#include "api/video/i420_buffer.h"
+
+#include "AVIOContextImpl.h"
+#include "platform/PlatformInterface.h"
+
+#include <string>
+#include <set>
+#include <map>
+
+namespace tgcalls {
+
+namespace {
+
+class MediaDataPacket {
+public:
+    MediaDataPacket() : _packet(av_packet_alloc()) {
+    }
+
+    MediaDataPacket(MediaDataPacket &&other) : _packet(other._packet) {
+        other._packet = nullptr;
+    }
+
+    ~MediaDataPacket() {
+        if (_packet) {
+            av_packet_free(&_packet);
+        }
+    }
+
+    AVPacket *packet() {
+        return _packet;
+    }
+
+private:
+    AVPacket *_packet = nullptr;
+};
+
+class DecodableFrame {
+public:
+    DecodableFrame(MediaDataPacket packet, int64_t pts, int64_t dts):
+    _packet(std::move(packet)),
+    _pts(pts),
+    _dts(dts) {
+    }
+
+    ~DecodableFrame() {
+    }
+
+    MediaDataPacket &packet() {
+        return _packet;
+    }
+
+    int64_t pts() {
+        return _pts;
+    }
+
+    int64_t dts() {
+        return _dts;
+    }
+
+private:
+    MediaDataPacket _packet;
+    int64_t _pts = 0;
+    int64_t _dts = 0;
+};
+
+class Frame {
+public:
+    Frame() {
+        _frame = av_frame_alloc();
+    }
+
+    Frame(Frame &&other) {
+        _frame = other._frame;
+        other._frame = nullptr;
+    }
+
+    ~Frame() {
+        if (_frame) {
+            av_frame_free(&_frame);
+        }
+    }
+
+    AVFrame *frame() {
+        return _frame;
+    }
+
+    double pts(AVStream *stream, double &firstFramePts) {
+        int64_t framePts = _frame->pts;
+        double spf = av_q2d(stream->time_base);
+        double value = ((double)framePts) * spf;
+        
+        if (firstFramePts < 0.0) {
+            firstFramePts = value;
+        }
+        
+        return value - firstFramePts;
+    }
+
+private:
+    AVFrame *_frame = nullptr;
+};
+
+struct VideoStreamEvent {
+    int32_t offset = 0;
+    std::string endpointId;
+    int32_t rotation = 0;
+    int32_t extra = 0;
+};
+
+struct VideoStreamInfo {
+    std::string container;
+    int32_t activeMask = 0;
+    std::vector<VideoStreamEvent> events;
+};
+
+absl::optional<int32_t> readInt32(std::vector<uint8_t> const &data, int &offset) {
+    if (offset + 4 > data.size()) {
+        return absl::nullopt;
+    }
+
+    int32_t value = 0;
+    memcpy(&value, data.data() + offset, 4);
+    offset += 4;
+
+    return value;
+}
+
+absl::optional<uint8_t> readBytesAsInt32(std::vector<uint8_t> const &data, int &offset, int count) {
+    if (offset + count > data.size()) {
+        return absl::nullopt;
+    }
+
+    if (count == 0) {
+        return absl::nullopt;
+    }
+
+    if (count <= 4) {
+        int32_t value = 0;
+        memcpy(&value, data.data() + offset, count);
+        offset += count;
+        return value;
+    } else {
+        return absl::nullopt;
+    }
+}
+
+int32_t roundUp(int32_t numToRound, int32_t multiple) {
+    if (multiple == 0) {
+        return numToRound;
+    }
+
+    int32_t remainder = numToRound % multiple;
+    if (remainder == 0) {
+        return numToRound;
+    }
+
+    return numToRound + multiple - remainder;
+}
+
+absl::optional<std::string> readSerializedString(std::vector<uint8_t> const &data, int &offset) {
+    if (const auto tmp = readBytesAsInt32(data, offset, 1)) {
+        int paddingBytes = 0;
+        int length = 0;
+        if (tmp.value() == 254) {
+            if (const auto len = readBytesAsInt32(data, offset, 3)) {
+                length = len.value();
+                paddingBytes = roundUp(length, 4) - length;
+            } else {
+                return absl::nullopt;
+            }
+        }
+        else {
+            length = tmp.value();
+            paddingBytes = roundUp(length + 1, 4) - (length + 1);
+        }
+
+        if (offset + length > data.size()) {
+            return absl::nullopt;
+        }
+
+        std::string result(data.data() + offset, data.data() + offset + length);
+
+        offset += length;
+        offset += paddingBytes;
+
+        return result;
+    } else {
+        return absl::nullopt;
+    }
+}
+
+absl::optional<VideoStreamEvent> readVideoStreamEvent(std::vector<uint8_t> const &data, int &offset) {
+    VideoStreamEvent event;
+
+    if (const auto offsetValue = readInt32(data, offset)) {
+        event.offset = offsetValue.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    if (const auto endpointId = readSerializedString(data, offset)) {
+        event.endpointId = endpointId.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    if (const auto rotation = readInt32(data, offset)) {
+        event.rotation = rotation.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    if (const auto extra = readInt32(data, offset)) {
+        event.extra = extra.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    return event;
+}
+
+absl::optional<VideoStreamInfo> consumeVideoStreamInfo(std::vector<uint8_t> &data) {
+    int offset = 0;
+    if (const auto signature = readInt32(data, offset)) {
+        if (signature.value() != 0xa12e810d) {
+            return absl::nullopt;
+        }
+    } else {
+        return absl::nullopt;
+    }
+
+    VideoStreamInfo info;
+
+    if (const auto container = readSerializedString(data, offset)) {
+        info.container = container.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    if (const auto activeMask = readInt32(data, offset)) {
+        info.activeMask = activeMask.value();
+    } else {
+        return absl::nullopt;
+    }
+
+    if (const auto eventCount = readInt32(data, offset)) {
+        if (eventCount > 0) {
+            if (const auto event = readVideoStreamEvent(data, offset)) {
+                info.events.push_back(event.value());
+            } else {
+                return absl::nullopt;
+            }
+        } else {
+            return absl::nullopt;
+        }
+    } else {
+        return absl::nullopt;
+    }
+
+    data.erase(data.begin(), data.begin() + offset);
+
+    return info;
+}
+
+bool areCodecParametersEqual(AVCodecParameters const &lhs, AVCodecParameters const &rhs) {
+    if (lhs.codec_id != rhs.codec_id) {
+        return false;
+    }
+    if (lhs.extradata_size != rhs.extradata_size) {
+        return false;
+    }
+    if (lhs.extradata_size != 0) {
+        if (memcmp(lhs.extradata, rhs.extradata, lhs.extradata_size)) {
+            return false;
+        }
+    }
+    if (lhs.format != rhs.format) {
+        return false;
+    }
+    if (lhs.profile != rhs.profile) {
+        return false;
+    }
+    if (lhs.level != rhs.level) {
+        return false;
+    }
+    if (lhs.width != rhs.width) {
+        return false;
+    }
+    if (lhs.height != rhs.height) {
+        return false;
+    }
+    if (lhs.sample_aspect_ratio.num != rhs.sample_aspect_ratio.num) {
+        return false;
+    }
+    if (lhs.sample_aspect_ratio.den != rhs.sample_aspect_ratio.den) {
+        return false;
+    }
+    if (lhs.field_order != rhs.field_order) {
+        return false;
+    }
+    if (lhs.color_range != rhs.color_range) {
+        return false;
+    }
+    if (lhs.color_primaries != rhs.color_primaries) {
+        return false;
+    }
+    if (lhs.color_trc != rhs.color_trc) {
+        return false;
+    }
+    if (lhs.color_space != rhs.color_space) {
+        return false;
+    }
+    if (lhs.chroma_location != rhs.chroma_location) {
+        return false;
+    }
+    
+    return true;
+}
+
+class VideoStreamingDecoderState {
+public:
+    static std::unique_ptr<VideoStreamingDecoderState> create(
+        AVCodecParameters const *codecParameters,
+        AVRational pktTimebase
+    ) {
+        AVCodec const *codec = nullptr;
+        if (!codec) {
+            codec = avcodec_find_decoder(codecParameters->codec_id);
+        }
+        if (!codec) {
+            return nullptr;
+        }
+        AVCodecContext *codecContext = avcodec_alloc_context3(codec);
+        int ret = avcodec_parameters_to_context(codecContext, codecParameters);
+        if (ret < 0) {
+            avcodec_free_context(&codecContext);
+            return nullptr;
+        } else {
+            codecContext->pkt_timebase = pktTimebase;
+            
+            PlatformInterface::SharedInstance()->setupVideoDecoding(codecContext);
+            
+            ret = avcodec_open2(codecContext, codec, nullptr);
+            if (ret < 0) {
+                avcodec_free_context(&codecContext);
+                return nullptr;
+            }
+        }
+        
+        return std::make_unique<VideoStreamingDecoderState>(
+            codecContext,
+            codecParameters,
+            pktTimebase
+        );
+    }
+    
+public:
+    VideoStreamingDecoderState(
+        AVCodecContext *codecContext,
+        AVCodecParameters const *codecParameters,
+        AVRational pktTimebase
+    ) {
+        _codecContext = codecContext;
+        _codecParameters = avcodec_parameters_alloc();
+        avcodec_parameters_copy(_codecParameters, codecParameters);
+        _pktTimebase = pktTimebase;
+    }
+    
+    ~VideoStreamingDecoderState() {
+        if (_codecContext) {
+            #if LIBAVFORMAT_VERSION_MAJOR < 59
+            avcodec_close(_codecContext);
+            #endif
+            avcodec_free_context(&_codecContext);
+        }
+        if (_codecParameters) {
+            avcodec_parameters_free(&_codecParameters);
+        }
+    }
+    
+    bool supportsDecoding(
+        AVCodecParameters const *codecParameters,
+        AVRational pktTimebase
+    ) const {
+        if (!areCodecParametersEqual(*_codecParameters, *codecParameters)) {
+            return false;
+        }
+        if (_pktTimebase.num != pktTimebase.num) {
+            return false;
+        }
+        if (_pktTimebase.den != pktTimebase.den) {
+            return false;
+        }
+        return true;
+    }
+    
+    int sendFrame(std::shared_ptr<DecodableFrame> frame) {
+        if (frame) {
+            int status = avcodec_send_packet(_codecContext, frame->packet().packet());
+            return status;
+        } else {
+            int status = avcodec_send_packet(_codecContext, nullptr);
+            return status;
+        }
+    }
+    
+    int receiveFrame(Frame &frame) {
+        int status = avcodec_receive_frame(_codecContext, frame.frame());
+        return status;
+    }
+    
+    void reset() {
+        avcodec_flush_buffers(_codecContext);
+    }
+    
+private:
+    AVCodecContext *_codecContext = nullptr;
+    AVCodecParameters *_codecParameters = nullptr;
+    AVRational _pktTimebase;
+};
+
+}
+
+class VideoStreamingSharedStateInternal {
+public:
+    VideoStreamingSharedStateInternal() {
+    }
+    
+    ~VideoStreamingSharedStateInternal() {
+    }
+    
+    void updateDecoderState(
+        AVCodecParameters const *codecParameters,
+        AVRational pktTimebase
+    ) {
+        if (_decoderState && _decoderState->supportsDecoding(codecParameters, pktTimebase)) {
+            return;
+        }
+        
+        _decoderState.reset();
+        _decoderState = VideoStreamingDecoderState::create(codecParameters, pktTimebase);
+    }
+    
+    int sendFrame(std::shared_ptr<DecodableFrame> frame) {
+        if (!_decoderState) {
+            return AVERROR(EIO);
+        }
+        return _decoderState->sendFrame(frame);
+    }
+    
+    int receiveFrame(Frame &frame) {
+        if (!_decoderState) {
+            return AVERROR(EIO);
+        }
+        return _decoderState->receiveFrame(frame);
+    }
+    
+    void reset() {
+        if (!_decoderState) {
+            return;
+        }
+        _decoderState->reset();
+    }
+    
+private:
+    std::unique_ptr<VideoStreamingDecoderState> _decoderState;
+};
+
+VideoStreamingSharedState::VideoStreamingSharedState() {
+    _impl = new VideoStreamingSharedStateInternal();
+}
+
+VideoStreamingSharedState::~VideoStreamingSharedState() {
+    delete _impl;
+}
+
+class VideoStreamingPartInternal {
+public:
+    VideoStreamingPartInternal(std::string endpointId, webrtc::VideoRotation rotation, std::vector<uint8_t> &&fileData, std::string const &container) :
+    _endpointId(endpointId),
+    _rotation(rotation) {
+        _avIoContext = std::make_unique<AVIOContextImpl>(std::move(fileData));
+
+        int ret = 0;
+
+#if LIBAVFORMAT_VERSION_MAJOR >= 59
+        const
+#endif
+        AVInputFormat *inputFormat = av_find_input_format(container.c_str());
+        if (!inputFormat) {
+            _didReadToEnd = true;
+            return;
+        }
+
+        _inputFormatContext = avformat_alloc_context();
+        if (!_inputFormatContext) {
+            _didReadToEnd = true;
+            return;
+        }
+
+        _inputFormatContext->pb = _avIoContext->getContext();
+
+        if ((ret = avformat_open_input(&_inputFormatContext, "", inputFormat, nullptr)) < 0) {
+            _didReadToEnd = true;
+            return;
+        }
+
+        if ((ret = avformat_find_stream_info(_inputFormatContext, nullptr)) < 0) {
+            _didReadToEnd = true;
+
+            avformat_close_input(&_inputFormatContext);
+            _inputFormatContext = nullptr;
+            return;
+        }
+
+        AVCodecParameters *videoCodecParameters = nullptr;
+        AVStream *videoStream = nullptr;
+        for (int i = 0; i < _inputFormatContext->nb_streams; i++) {
+            AVStream *inStream = _inputFormatContext->streams[i];
+
+            AVCodecParameters *inCodecpar = inStream->codecpar;
+            if (inCodecpar->codec_type != AVMEDIA_TYPE_VIDEO) {
+                continue;
+            }
+            videoCodecParameters = inCodecpar;
+            videoStream = inStream;
+
+            break;
+        }
+
+        if (videoCodecParameters && videoStream) {
+            _videoCodecParameters = avcodec_parameters_alloc();
+            avcodec_parameters_copy(_videoCodecParameters, videoCodecParameters);
+            _videoStream = videoStream;
+            
+            /*const AVCodec *codec = avcodec_find_decoder(videoCodecParameters->codec_id);
+            if (codec) {
+                _codecContext = avcodec_alloc_context3(codec);
+                ret = avcodec_parameters_to_context(_codecContext, videoCodecParameters);
+                if (ret < 0) {
+                    _didReadToEnd = true;
+
+                    avcodec_free_context(&_codecContext);
+                    _codecContext = nullptr;
+                } else {
+                    _codecContext->pkt_timebase = videoStream->time_base;
+
+                    ret = avcodec_open2(_codecContext, codec, nullptr);
+                    if (ret < 0) {
+                        _didReadToEnd = true;
+
+                        avcodec_free_context(&_codecContext);
+                        _codecContext = nullptr;
+                    } else {
+                        _videoStream = videoStream;
+                    }
+                }
+            }*/
+        }
+    }
+
+    ~VideoStreamingPartInternal() {
+        if (_videoCodecParameters) {
+            avcodec_parameters_free(&_videoCodecParameters);
+        }
+        if (_inputFormatContext) {
+            avformat_close_input(&_inputFormatContext);
+        }
+    }
+
+    std::string endpointId() {
+        return _endpointId;
+    }
+
+    absl::optional<MediaDataPacket> readPacket() {
+        if (_didReadToEnd) {
+            return absl::nullopt;
+        }
+        if (!_inputFormatContext) {
+            return absl::nullopt;
+        }
+
+        MediaDataPacket packet;
+        int result = av_read_frame(_inputFormatContext, packet.packet());
+        if (result < 0) {
+            return absl::nullopt;
+        }
+
+        return packet;
+    }
+
+    std::shared_ptr<DecodableFrame> readNextDecodableFrame() {
+        while (true) {
+            absl::optional<MediaDataPacket> packet = readPacket();
+            if (packet) {
+                if (_videoStream && packet->packet()->stream_index == _videoStream->index) {
+                    return std::make_shared<DecodableFrame>(std::move(packet.value()), packet->packet()->pts, packet->packet()->dts);
+                }
+            } else {
+                return nullptr;
+            }
+        }
+    }
+
+    absl::optional<VideoStreamingPartFrame> convertCurrentFrame() {
+        auto platformFrameBuffer = PlatformInterface::SharedInstance()->createPlatformFrameFromData(_frame.frame());
+        if (platformFrameBuffer) {
+            auto videoFrame = webrtc::VideoFrame::Builder()
+                .set_video_frame_buffer(platformFrameBuffer)
+                .set_rotation(_rotation)
+                .build();
+
+            return VideoStreamingPartFrame(_endpointId, videoFrame, _frame.pts(_videoStream, _firstFramePts), _frameIndex);
+        } else {
+            webrtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = webrtc::I420Buffer::Copy(
+                _frame.frame()->width,
+                _frame.frame()->height,
+                _frame.frame()->data[0],
+                _frame.frame()->linesize[0],
+                _frame.frame()->data[1],
+                _frame.frame()->linesize[1],
+                _frame.frame()->data[2],
+                _frame.frame()->linesize[2]
+            );
+            if (i420Buffer) {
+                auto videoFrame = webrtc::VideoFrame::Builder()
+                    .set_video_frame_buffer(i420Buffer)
+                    .set_rotation(_rotation)
+                    .build();
+
+                return VideoStreamingPartFrame(_endpointId, videoFrame, _frame.pts(_videoStream, _firstFramePts), _frameIndex);
+            } else {
+                return absl::nullopt;
+            }
+        }
+    }
+
+    absl::optional<VideoStreamingPartFrame> getNextFrame(VideoStreamingSharedState const *sharedState) {
+        if (!_videoStream) {
+            return {};
+        }
+        if (!_videoCodecParameters) {
+            return {};
+        }
+        
+        sharedState->impl()->updateDecoderState(_videoCodecParameters, _videoStream->time_base);
+
+        while (true) {
+            if (_didReadToEnd) {
+                if (!_finalFrames.empty()) {
+                    auto frame = _finalFrames[0];
+                    _finalFrames.erase(_finalFrames.begin());
+                    return frame;
+                } else {
+                    break;
+                }
+            } else {
+                const auto frame = readNextDecodableFrame();
+                if (frame) {
+                    int sendStatus = sharedState->impl()->sendFrame(frame);
+                    if (sendStatus == 0) {
+                        int receiveStatus = sharedState->impl()->receiveFrame(_frame);
+                        if (receiveStatus == 0) {
+                            auto convertedFrame = convertCurrentFrame();
+                            if (convertedFrame) {
+                                _frameIndex++;
+                                return convertedFrame;
+                            }
+                        } else if (receiveStatus == AVERROR(EAGAIN)) {
+                            // more data needed
+                        } else {
+                            RTC_LOG(LS_ERROR) << "avcodec_receive_frame failed with result: " << receiveStatus;
+                            _didReadToEnd = true;
+                            break;
+                        }
+                    } else {
+                        RTC_LOG(LS_ERROR) << "avcodec_send_packet failed with result: " << sendStatus;
+                        _didReadToEnd = true;
+                        return {};
+                    }
+                } else {
+                    _didReadToEnd = true;
+                    int sendStatus = sharedState->impl()->sendFrame(nullptr);
+                    if (sendStatus == 0) {
+                        while (true) {
+                            int receiveStatus = sharedState->impl()->receiveFrame(_frame);
+                            if (receiveStatus == 0) {
+                                auto convertedFrame = convertCurrentFrame();
+                                if (convertedFrame) {
+                                    _frameIndex++;
+                                    _finalFrames.push_back(convertedFrame.value());
+                                }
+                            } else {
+                                if (receiveStatus != AVERROR_EOF) {
+                                    RTC_LOG(LS_ERROR) << "avcodec_receive_frame (drain) failed with result: " << receiveStatus;
+                                }
+                                break;
+                            }
+                        }
+                    } else {
+                        RTC_LOG(LS_ERROR) << "avcodec_send_packet (drain) failed with result: " << sendStatus;
+                    }
+                    sharedState->impl()->reset();
+                }
+            }
+        }
+
+        return {};
+    }
+
+private:
+    std::string _endpointId;
+    webrtc::VideoRotation _rotation = webrtc::VideoRotation::kVideoRotation_0;
+
+    std::unique_ptr<AVIOContextImpl> _avIoContext;
+
+    AVFormatContext *_inputFormatContext = nullptr;
+    AVStream *_videoStream = nullptr;
+    Frame _frame;
+    
+    AVCodecParameters *_videoCodecParameters = nullptr;
+
+    std::vector<VideoStreamingPartFrame> _finalFrames;
+
+    int _frameIndex = 0;
+    double _firstFramePts = -1.0;
+    bool _didReadToEnd = false;
+};
+
+class VideoStreamingPartState {
+public:
+    VideoStreamingPartState(std::vector<uint8_t> &&data, VideoStreamingPart::ContentType contentType) {
+        _videoStreamInfo = consumeVideoStreamInfo(data);
+        if (!_videoStreamInfo) {
+            return;
+        }
+
+        for (size_t i = 0; i < _videoStreamInfo->events.size(); i++) {
+            if (_videoStreamInfo->events[i].offset < 0) {
+                continue;
+            }
+            size_t endOffset = 0;
+            if (i == _videoStreamInfo->events.size() - 1) {
+                endOffset = data.size();
+            } else {
+                endOffset = _videoStreamInfo->events[i + 1].offset;
+            }
+            if (endOffset <= _videoStreamInfo->events[i].offset) {
+                continue;
+            }
+            if (endOffset > data.size()) {
+                continue;
+            }
+            std::vector<uint8_t> dataSlice(data.begin() + _videoStreamInfo->events[i].offset, data.begin() + endOffset);
+            webrtc::VideoRotation rotation = webrtc::VideoRotation::kVideoRotation_0;
+            switch (_videoStreamInfo->events[i].rotation) {
+                case 0: {
+                    rotation = webrtc::VideoRotation::kVideoRotation_0;
+                    break;
+                }
+                case 90: {
+                    rotation = webrtc::VideoRotation::kVideoRotation_90;
+                    break;
+                }
+                case 180: {
+                    rotation = webrtc::VideoRotation::kVideoRotation_180;
+                    break;
+                }
+                case 270: {
+                    rotation = webrtc::VideoRotation::kVideoRotation_270;
+                    break;
+                }
+                default: {
+                    break;
+                }
+            }
+
+            switch (contentType) {
+                case VideoStreamingPart::ContentType::Audio: {
+                    auto part = std::make_unique<AudioStreamingPart>(std::move(dataSlice), _videoStreamInfo->container, true);
+                    _parsedAudioParts.push_back(std::move(part));
+
+                    break;
+                }
+                case VideoStreamingPart::ContentType::Video: {
+                    auto part = std::make_unique<VideoStreamingPartInternal>(_videoStreamInfo->events[i].endpointId, rotation, std::move(dataSlice), _videoStreamInfo->container);
+                    _parsedVideoParts.push_back(std::move(part));
+
+                    break;
+                }
+                default: {
+                    break;
+                }
+            }
+        }
+    }
+
+    ~VideoStreamingPartState() {
+    }
+
+    absl::optional<VideoStreamingPartFrame> getFrameAtRelativeTimestamp(VideoStreamingSharedState const *sharedState, double timestamp) {
+        while (true) {
+            while (_availableFrames.size() >= 2) {
+                if (timestamp >= _availableFrames[1].pts) {
+                    _availableFrames.erase(_availableFrames.begin());
+                } else {
+                    break;
+                }
+            }
+            
+            if (_availableFrames.size() < 2) {
+                if (!_parsedVideoParts.empty()) {
+                    auto result = _parsedVideoParts[0]->getNextFrame(sharedState);
+                    if (result) {
+                        _availableFrames.push_back(result.value());
+                    } else {
+                        _parsedVideoParts.erase(_parsedVideoParts.begin());
+                    }
+                    continue;
+                }
+            }
+
+            if (!_availableFrames.empty()) {
+                for (size_t i = 1; i < _availableFrames.size(); i++) {
+                    if (timestamp < _availableFrames[i].pts) {
+                        return _availableFrames[i - 1];
+                    }
+                }
+                return _availableFrames[_availableFrames.size() - 1];
+            } else {
+                return absl::nullopt;
+            }
+        }
+    }
+
+    absl::optional<std::string> getActiveEndpointId() const {
+        if (!_parsedVideoParts.empty()) {
+            return _parsedVideoParts[0]->endpointId();
+        } else {
+            return absl::nullopt;
+        }
+    }
+    
+    bool hasRemainingFrames() const {
+        return !_parsedVideoParts.empty() || !_parsedAudioParts.empty();
+    }
+
+    int getAudioRemainingMilliseconds() {
+        while (!_parsedAudioParts.empty()) {
+            auto firstPartResult = _parsedAudioParts[0]->getRemainingMilliseconds();
+            if (firstPartResult <= 0) {
+                _parsedAudioParts.erase(_parsedAudioParts.begin());
+            } else {
+                return firstPartResult;
+            }
+        }
+        return 0;
+    }
+
+    std::vector<AudioStreamingPart::StreamingPartChannel> getAudio10msPerChannel(AudioStreamingPartPersistentDecoder &persistentDecoder) {
+        while (!_parsedAudioParts.empty()) {
+            auto firstPartResult = _parsedAudioParts[0]->get10msPerChannel(persistentDecoder);
+            if (firstPartResult.empty()) {
+                _parsedAudioParts.erase(_parsedAudioParts.begin());
+            } else {
+                return firstPartResult;
+            }
+        }
+        return {};
+    }
+
+private:
+    absl::optional<VideoStreamInfo> _videoStreamInfo;
+    std::vector<std::unique_ptr<VideoStreamingPartInternal>> _parsedVideoParts;
+    std::vector<VideoStreamingPartFrame> _availableFrames;
+
+    std::vector<std::unique_ptr<AudioStreamingPart>> _parsedAudioParts;
+};
+
+VideoStreamingPart::VideoStreamingPart(std::vector<uint8_t> &&data, VideoStreamingPart::ContentType contentType) {
+    if (!data.empty()) {
+        _state = new VideoStreamingPartState(std::move(data), contentType);
+    }
+}
+
+VideoStreamingPart::~VideoStreamingPart() {
+    if (_state) {
+        delete _state;
+    }
+}
+
+absl::optional<VideoStreamingPartFrame> VideoStreamingPart::getFrameAtRelativeTimestamp(VideoStreamingSharedState const *sharedState, double timestamp) {
+    return _state
+        ? _state->getFrameAtRelativeTimestamp(sharedState, timestamp)
+        : absl::nullopt;
+}
+
+absl::optional<std::string> VideoStreamingPart::getActiveEndpointId() const {
+    return _state
+        ? _state->getActiveEndpointId()
+        : absl::nullopt;
+}
+
+bool VideoStreamingPart::hasRemainingFrames() const {
+    return _state
+        ? _state->hasRemainingFrames()
+        : false;
+}
+
+int VideoStreamingPart::getAudioRemainingMilliseconds() {
+    return _state
+        ? _state->getAudioRemainingMilliseconds()
+        : 0;
+}
+std::vector<AudioStreamingPart::StreamingPartChannel> VideoStreamingPart::getAudio10msPerChannel(AudioStreamingPartPersistentDecoder &persistentDecoder) {
+    return _state
+        ? _state->getAudio10msPerChannel(persistentDecoder)
+        : std::vector<AudioStreamingPart::StreamingPartChannel>();
+}
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/group/VideoStreamingPart.h

@@ -0,0 +1,78 @@
+#ifndef TGCALLS_VIDEO_STREAMING_PART_H
+#define TGCALLS_VIDEO_STREAMING_PART_H
+
+#include "absl/types/optional.h"
+#include <vector>
+#include <stdint.h>
+
+#include "api/video/video_frame.h"
+#include "absl/types/optional.h"
+
+#include "AudioStreamingPart.h"
+#include "AudioStreamingPartInternal.h"
+
+namespace tgcalls {
+
+class VideoStreamingPartState;
+class VideoStreamingSharedStateInternal;
+
+struct VideoStreamingPartFrame {
+    std::string endpointId;
+    webrtc::VideoFrame frame;
+    double pts = 0;
+    int index = 0;
+
+    VideoStreamingPartFrame(std::string endpointId_, webrtc::VideoFrame const &frame_, double pts_, int index_) :
+    endpointId(endpointId_),
+    frame(frame_),
+    pts(pts_),
+    index(index_) {
+    }
+};
+
+class VideoStreamingSharedState {
+public:
+    VideoStreamingSharedState();
+    ~VideoStreamingSharedState();
+    
+    VideoStreamingSharedStateInternal *impl() const {
+        return _impl;
+    }
+    
+private:
+    VideoStreamingSharedStateInternal *_impl = nullptr;
+};
+
+class VideoStreamingPart {
+public:
+    enum class ContentType {
+        Audio,
+        Video
+    };
+    
+public:
+    explicit VideoStreamingPart(std::vector<uint8_t> &&data, VideoStreamingPart::ContentType contentType);
+    ~VideoStreamingPart();
+    
+    VideoStreamingPart(const VideoStreamingPart&) = delete;
+    VideoStreamingPart(VideoStreamingPart&& other) {
+        _state = other._state;
+        other._state = nullptr;
+    }
+    VideoStreamingPart& operator=(const VideoStreamingPart&) = delete;
+    VideoStreamingPart& operator=(VideoStreamingPart&&) = delete;
+
+    absl::optional<VideoStreamingPartFrame> getFrameAtRelativeTimestamp(VideoStreamingSharedState const *sharedState, double timestamp);
+    absl::optional<std::string> getActiveEndpointId() const;
+    bool hasRemainingFrames() const;
+    
+    int getAudioRemainingMilliseconds();
+    std::vector<AudioStreamingPart::StreamingPartChannel> getAudio10msPerChannel(AudioStreamingPartPersistentDecoder &persistentDecoder);
+    
+private:
+    VideoStreamingPartState *_state = nullptr;
+};
+
+}
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/platform/PlatformInterface.h

@@ -0,0 +1,363 @@
+#ifndef TGCALLS_PLATFORM_INTERFACE_H
+#define TGCALLS_PLATFORM_INTERFACE_H
+
+#include "rtc_base/thread.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/media_stream_interface.h"
+#include "rtc_base/network_monitor_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/ref_counted_object.h"
+#include <string>
+#include <map>
+
+struct AVFrame;
+struct AVCodecContext;
+
+namespace tgcalls {
+
+enum class VideoState;
+
+class VideoCapturerInterface;
+class PlatformContext;
+
+struct PlatformCaptureInfo {
+    bool shouldBeAdaptedToReceiverAspectRate = false;
+    int rotation = 0;
+};
+
+class WrappedAudioDeviceModule : public webrtc::AudioDeviceModule {
+public:
+    virtual void Stop() = 0;
+    virtual void setIsActive(bool isActive) = 0;
+};
+
+class DefaultWrappedAudioDeviceModule : public WrappedAudioDeviceModule {
+public:
+    DefaultWrappedAudioDeviceModule(webrtc::scoped_refptr<webrtc::AudioDeviceModule> impl) :
+    _impl(impl) {
+    }
+
+    virtual ~DefaultWrappedAudioDeviceModule() {
+    }
+
+    virtual void Stop() override {
+    }
+    
+    virtual void setIsActive(bool isActive) override {
+    }
+
+    virtual int32_t ActiveAudioLayer(AudioLayer *audioLayer) const override {
+        return _impl->ActiveAudioLayer(audioLayer);
+    }
+
+    virtual int32_t RegisterAudioCallback(webrtc::AudioTransport *audioCallback) override {
+        return _impl->RegisterAudioCallback(audioCallback);
+    }
+
+    virtual int32_t Init() override {
+        return _impl->Init();
+    }
+
+    virtual int32_t Terminate() override {
+        return _impl->Terminate();
+    }
+
+    virtual bool Initialized() const override {
+        return _impl->Initialized();
+    }
+
+    virtual int16_t PlayoutDevices() override {
+        return _impl->PlayoutDevices();
+    }
+
+    virtual int16_t RecordingDevices() override {
+        return _impl->RecordingDevices();
+    }
+
+    virtual int32_t PlayoutDeviceName(uint16_t index, char name[webrtc::kAdmMaxDeviceNameSize], char guid[webrtc::kAdmMaxGuidSize]) override {
+        return _impl->PlayoutDeviceName(index, name, guid);
+    }
+
+    virtual int32_t RecordingDeviceName(uint16_t index, char name[webrtc::kAdmMaxDeviceNameSize], char guid[webrtc::kAdmMaxGuidSize]) override {
+        return _impl->RecordingDeviceName(index, name, guid);
+    }
+
+    virtual int32_t SetPlayoutDevice(uint16_t index) override {
+        return _impl->SetPlayoutDevice(index);
+    }
+
+#ifdef TGCALLS_UWP_DESKTOP
+    virtual int32_t SetPlayoutDevice(std::string deviceId) override {
+        return _impl->SetPlayoutDevice(deviceId);
+    }
+#endif
+
+    virtual int32_t SetPlayoutDevice(WindowsDeviceType device) override {
+        return _impl->SetPlayoutDevice(device);
+    }
+
+    virtual int32_t SetRecordingDevice(uint16_t index) override {
+        return _impl->SetRecordingDevice(index);
+    }
+
+#ifdef TGCALLS_UWP_DESKTOP
+    virtual int32_t SetRecordingDevice(std::string deviceId) override {
+        return _impl->SetRecordingDevice(deviceId);
+    }
+#endif
+
+    virtual int32_t SetRecordingDevice(WindowsDeviceType device) override {
+        return _impl->SetRecordingDevice(device);
+    }
+
+    virtual int32_t PlayoutIsAvailable(bool *available) override {
+        return _impl->PlayoutIsAvailable(available);
+    }
+
+    virtual int32_t InitPlayout() override {
+        return _impl->InitPlayout();
+    }
+
+    virtual bool PlayoutIsInitialized() const override {
+        return _impl->PlayoutIsInitialized();
+    }
+
+    virtual int32_t RecordingIsAvailable(bool *available) override {
+        return _impl->RecordingIsAvailable(available);
+    }
+
+    virtual int32_t InitRecording() override {
+        return _impl->InitRecording();
+    }
+
+    virtual bool RecordingIsInitialized() const override {
+        return _impl->RecordingIsInitialized();
+    }
+
+    virtual int32_t StartPlayout() override {
+        return _impl->StartPlayout();
+    }
+
+    virtual int32_t StopPlayout() override {
+        return _impl->StopPlayout();
+    }
+
+    virtual bool Playing() const override {
+        return _impl->Playing();
+    }
+
+    virtual int32_t StartRecording() override {
+        return _impl->StartRecording();
+    }
+
+    virtual int32_t StopRecording() override {
+        return _impl->StopRecording();
+    }
+
+    virtual bool Recording() const override {
+        return _impl->Recording();
+    }
+
+    virtual int32_t InitSpeaker() override {
+        return _impl->InitSpeaker();
+    }
+
+    virtual bool SpeakerIsInitialized() const override {
+        return _impl->SpeakerIsInitialized();
+    }
+
+    virtual int32_t InitMicrophone() override {
+        return _impl->InitMicrophone();
+    }
+
+    virtual bool MicrophoneIsInitialized() const override {
+        return _impl->MicrophoneIsInitialized();
+    }
+
+    virtual int32_t SpeakerVolumeIsAvailable(bool *available) override {
+        return _impl->SpeakerVolumeIsAvailable(available);
+    }
+
+    virtual int32_t SetSpeakerVolume(uint32_t volume) override {
+        return _impl->SetSpeakerVolume(volume);
+    }
+
+    virtual int32_t SpeakerVolume(uint32_t* volume) const override {
+        return _impl->SpeakerVolume(volume);
+    }
+
+    virtual int32_t MaxSpeakerVolume(uint32_t *maxVolume) const override {
+        return _impl->MaxSpeakerVolume(maxVolume);
+    }
+
+    virtual int32_t MinSpeakerVolume(uint32_t *minVolume) const override {
+        return _impl->MinSpeakerVolume(minVolume);
+    }
+
+    virtual int32_t MicrophoneVolumeIsAvailable(bool *available) override {
+        return _impl->MicrophoneVolumeIsAvailable(available);
+    }
+
+    virtual int32_t SetMicrophoneVolume(uint32_t volume) override {
+        return _impl->SetMicrophoneVolume(volume);
+    }
+
+    virtual int32_t MicrophoneVolume(uint32_t *volume) const override {
+        return _impl->MicrophoneVolume(volume);
+    }
+
+    virtual int32_t MaxMicrophoneVolume(uint32_t *maxVolume) const override {
+        return _impl->MaxMicrophoneVolume(maxVolume);
+    }
+
+    virtual int32_t MinMicrophoneVolume(uint32_t *minVolume) const override {
+        return _impl->MinMicrophoneVolume(minVolume);
+    }
+
+    virtual int32_t SpeakerMuteIsAvailable(bool *available) override {
+        return _impl->SpeakerMuteIsAvailable(available);
+    }
+
+    virtual int32_t SetSpeakerMute(bool enable) override {
+        return _impl->SetSpeakerMute(enable);
+    }
+
+    virtual int32_t SpeakerMute(bool *enabled) const override {
+        return _impl->SpeakerMute(enabled);
+    }
+
+    virtual int32_t MicrophoneMuteIsAvailable(bool *available) override {
+        return _impl->MicrophoneMuteIsAvailable(available);
+    }
+
+    virtual int32_t SetMicrophoneMute(bool enable) override {
+        return _impl->SetMicrophoneMute(enable);
+    }
+
+    virtual int32_t MicrophoneMute(bool *enabled) const override {
+        return _impl->MicrophoneMute(enabled);
+    }
+
+    virtual int32_t StereoPlayoutIsAvailable(bool *available) const override {
+        return _impl->StereoPlayoutIsAvailable(available);
+    }
+
+    virtual int32_t SetStereoPlayout(bool enable) override {
+        return _impl->SetStereoPlayout(enable);
+    }
+
+    virtual int32_t StereoPlayout(bool *enabled) const override {
+        return _impl->StereoPlayout(enabled);
+    }
+
+    virtual int32_t StereoRecordingIsAvailable(bool *available) const override {
+        return _impl->StereoRecordingIsAvailable(available);
+    }
+
+    virtual int32_t SetStereoRecording(bool enable) override {
+        return _impl->SetStereoRecording(enable);
+    }
+
+    virtual int32_t StereoRecording(bool *enabled) const override {
+        return _impl->StereoRecording(enabled);
+    }
+
+    virtual int32_t PlayoutDelay(uint16_t* delayMS) const override {
+        return _impl->PlayoutDelay(delayMS);
+    }
+
+    virtual bool BuiltInAECIsAvailable() const override {
+        return _impl->BuiltInAECIsAvailable();
+    }
+
+    virtual bool BuiltInAGCIsAvailable() const override {
+        return _impl->BuiltInAGCIsAvailable();
+    }
+
+    virtual bool BuiltInNSIsAvailable() const override {
+        return _impl->BuiltInNSIsAvailable();
+    }
+
+    virtual int32_t EnableBuiltInAEC(bool enable) override {
+        return _impl->EnableBuiltInAEC(enable);
+    }
+
+    virtual int32_t EnableBuiltInAGC(bool enable) override {
+        return _impl->EnableBuiltInAGC(enable);
+    }
+
+    virtual int32_t EnableBuiltInNS(bool enable) override {
+        return _impl->EnableBuiltInNS(enable);
+    }
+
+    virtual int32_t GetPlayoutUnderrunCount() const override {
+        return _impl->GetPlayoutUnderrunCount();
+    }
+
+#if defined(WEBRTC_IOS)
+    virtual int GetPlayoutAudioParameters(webrtc::AudioParameters *params) const override {
+        return _impl->GetPlayoutAudioParameters(params);
+    }
+    virtual int GetRecordAudioParameters(webrtc::AudioParameters *params) const override {
+        return _impl->GetRecordAudioParameters(params);
+    }
+#endif  // WEBRTC_IOS
+
+    webrtc::scoped_refptr<webrtc::AudioDeviceModule> WrappedInstance() const {
+        return _impl;
+    }
+
+private:
+    webrtc::scoped_refptr<webrtc::AudioDeviceModule> _impl;
+};
+
+class PlatformVideoFrame {
+public:
+    PlatformVideoFrame() {
+    }
+    
+    virtual ~PlatformVideoFrame() = default;
+};
+
+class PlatformInterface {
+public:
+	static PlatformInterface *SharedInstance();
+	virtual ~PlatformInterface() = default;
+
+	virtual void configurePlatformAudio(int numChannels = 1) {
+	}
+
+    virtual std::unique_ptr<rtc::NetworkMonitorFactory> createNetworkMonitorFactory() {
+        return nullptr;
+    }
+    
+	virtual std::unique_ptr<webrtc::VideoEncoderFactory> makeVideoEncoderFactory(bool preferHardwareEncoding = false, bool isScreencast = false) = 0;
+	virtual std::unique_ptr<webrtc::VideoDecoderFactory> makeVideoDecoderFactory() = 0;
+	virtual bool supportsEncoding(const std::string &codecName) = 0;
+	virtual webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface> makeVideoSource(rtc::Thread *signalingThread, rtc::Thread *workerThread) = 0;
+    virtual void adaptVideoSource(webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface> videoSource, int width, int height, int fps) = 0;
+	virtual std::unique_ptr<VideoCapturerInterface> makeVideoCapturer(webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source, std::string deviceId, std::function<void(VideoState)> stateUpdated, std::function<void(PlatformCaptureInfo)> captureInfoUpdated, std::shared_ptr<PlatformContext> platformContext, std::pair<int, int> &outResolution) = 0;
+    virtual webrtc::scoped_refptr<WrappedAudioDeviceModule> wrapAudioDeviceModule(webrtc::scoped_refptr<webrtc::AudioDeviceModule> module) {
+        return rtc::make_ref_counted<DefaultWrappedAudioDeviceModule>(module);
+    }
+    virtual void setupVideoDecoding(AVCodecContext *codecContext) {
+    }
+    virtual webrtc::scoped_refptr<webrtc::VideoFrameBuffer> createPlatformFrameFromData(AVFrame const *frame) {
+        return nullptr;
+    }
+
+public:
+    bool preferX264 = false;
+};
+
+std::unique_ptr<PlatformInterface> CreatePlatformInterface();
+
+inline PlatformInterface *PlatformInterface::SharedInstance() {
+	static const auto result = CreatePlatformInterface();
+	return result.get();
+}
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/platform/android/AndroidContext.cpp

@@ -0,0 +1,32 @@
+#include "AndroidContext.h"
+
+#include "sdk/android/native_api/jni/jvm.h"
+
+namespace tgcalls {
+
+AndroidContext::AndroidContext(JNIEnv *env) {
+    VideoCameraCapturerClass = (jclass) env->NewGlobalRef(env->FindClass("org/telegram/messenger/voip/VideoCameraCapturer"));
+    jmethodID initMethodId = env->GetMethodID(VideoCameraCapturerClass, "<init>", "()V");
+    javaCapturer = env->NewGlobalRef(env->NewObject(VideoCameraCapturerClass, initMethodId));
+}
+
+AndroidContext::~AndroidContext() {
+    JNIEnv *env = webrtc::AttachCurrentThreadIfNeeded();
+
+    jmethodID onDestroyMethodId = env->GetMethodID(VideoCameraCapturerClass, "onDestroy", "()V");
+    env->CallVoidMethod(javaCapturer, onDestroyMethodId);
+    env->DeleteGlobalRef(javaCapturer);
+    javaCapturer = nullptr;
+
+    env->DeleteGlobalRef(VideoCameraCapturerClass);
+}
+
+jobject AndroidContext::getJavaCapturer() {
+    return javaCapturer;
+}
+
+jclass AndroidContext::getJavaCapturerClass() {
+    return VideoCameraCapturerClass;
+}
+
+}  // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/platform/android/AndroidContext.h

@@ -0,0 +1,26 @@
+#ifndef TGCALLS_ANDROID_CONTEXT_H
+#define TGCALLS_ANDROID_CONTEXT_H
+
+#include "PlatformContext.h"
+
+#include <jni.h>
+
+namespace tgcalls {
+
+class AndroidContext final : public PlatformContext {
+public:
+    AndroidContext(JNIEnv *env);
+    ~AndroidContext() override;
+
+    jobject getJavaCapturer();
+    jclass getJavaCapturerClass();
+
+private:
+    jclass VideoCameraCapturerClass = nullptr;
+    jobject javaCapturer = nullptr;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/platform/android/AndroidInterface.cpp

@@ -0,0 +1,95 @@
+#include "AndroidInterface.h"
+
+#include <rtc_base/ssl_adapter.h>
+#include <modules/utility/include/jvm_android.h>
+#include <sdk/android/src/jni/android_video_track_source.h>
+#include <media/base/media_constants.h>
+
+#include "VideoCapturerInterfaceImpl.h"
+
+#include "sdk/android/native_api/base/init.h"
+#include "sdk/android/native_api/codecs/wrapper.h"
+#include "sdk/android/native_api/jni/class_loader.h"
+#include "sdk/android/native_api/jni/jvm.h"
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
+#include "sdk/android/native_api/video/video_source.h"
+#include "api/video_codecs/builtin_video_encoder_factory.h"
+#include "api/video_codecs/builtin_video_decoder_factory.h"
+#include "api/video_track_source_proxy.h"
+
+
+namespace tgcalls {
+
+void AndroidInterface::configurePlatformAudio() {
+
+}
+
+std::unique_ptr<webrtc::VideoEncoderFactory> AndroidInterface::makeVideoEncoderFactory() {
+    JNIEnv *env = webrtc::AttachCurrentThreadIfNeeded();
+    webrtc::ScopedJavaLocalRef<jclass> factory_class =
+            webrtc::GetClass(env, "org/webrtc/DefaultVideoEncoderFactory");
+    jmethodID factory_constructor = env->GetMethodID(
+            factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;ZZ)V");
+    webrtc::ScopedJavaLocalRef<jobject> factory_object(
+            env, env->NewObject(factory_class.obj(), factory_constructor,
+                                nullptr /* shared_context */,
+                                false /* enable_intel_vp8_encoder */,
+                                true /* enable_h264_high_profile */));
+    return webrtc::JavaToNativeVideoEncoderFactory(env, factory_object.obj());
+}
+
+std::unique_ptr<webrtc::VideoDecoderFactory> AndroidInterface::makeVideoDecoderFactory() {
+    JNIEnv *env = webrtc::AttachCurrentThreadIfNeeded();
+    webrtc::ScopedJavaLocalRef<jclass> factory_class =
+            webrtc::GetClass(env, "org/webrtc/DefaultVideoDecoderFactory");
+    jmethodID factory_constructor = env->GetMethodID(
+            factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;)V");
+    webrtc::ScopedJavaLocalRef<jobject> factory_object(
+            env, env->NewObject(factory_class.obj(), factory_constructor,
+                                nullptr /* shared_context */));
+    return webrtc::JavaToNativeVideoDecoderFactory(env, factory_object.obj());
+}
+
+void AndroidInterface::adaptVideoSource(rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> videoSource, int width, int height, int fps) {
+
+}
+
+rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> AndroidInterface::makeVideoSource(rtc::Thread *signalingThread, rtc::Thread *workerThread) {
+    JNIEnv *env = webrtc::AttachCurrentThreadIfNeeded();
+    _source = webrtc::CreateJavaVideoSource(env, signalingThread, false, false);
+    return webrtc::VideoTrackSourceProxy::Create(signalingThread, workerThread, _source);
+}
+
+bool AndroidInterface::supportsEncoding(const std::string &codecName) {
+    if (hardwareVideoEncoderFactory == nullptr) {
+        JNIEnv *env = webrtc::AttachCurrentThreadIfNeeded();
+        webrtc::ScopedJavaLocalRef<jclass> factory_class =
+                webrtc::GetClass(env, "org/webrtc/HardwareVideoEncoderFactory");
+        jmethodID factory_constructor = env->GetMethodID(
+                factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;ZZ)V");
+        webrtc::ScopedJavaLocalRef<jobject> factory_object(
+                env, env->NewObject(factory_class.obj(), factory_constructor,
+                                    nullptr,
+                                    false,
+                                    true));
+        hardwareVideoEncoderFactory = webrtc::JavaToNativeVideoEncoderFactory(env, factory_object.obj());
+    }
+    auto formats = hardwareVideoEncoderFactory->GetSupportedFormats();
+    for (auto format : formats) {
+        if (format.name == codecName) {
+            return true;
+        }
+    }
+    return codecName == cricket::kVp8CodecName;
+}
+
+std::unique_ptr<VideoCapturerInterface> AndroidInterface::makeVideoCapturer(rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source, std::string deviceId, std::function<void(VideoState)> stateUpdated, std::shared_ptr<PlatformContext> platformContext) {
+    return std::make_unique<VideoCapturerInterfaceImpl>(_source, deviceId, stateUpdated, platformContext);
+}
+
+
+std::unique_ptr<PlatformInterface> CreatePlatformInterface() {
+	return std::make_unique<AndroidInterface>();
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/platform/android/AndroidInterface.h

@@ -0,0 +1,29 @@
+#ifndef TGCALLS_ANDROID_INTERFACE_H
+#define TGCALLS_ANDROID_INTERFACE_H
+
+#include "sdk/android/native_api/video/video_source.h"
+#include "platform/PlatformInterface.h"
+#include "VideoCapturerInterface.h"
+
+namespace tgcalls {
+
+class AndroidInterface : public PlatformInterface {
+public:
+    void configurePlatformAudio() override;
+	std::unique_ptr<webrtc::VideoEncoderFactory> makeVideoEncoderFactory() override;
+	std::unique_ptr<webrtc::VideoDecoderFactory> makeVideoDecoderFactory() override;
+	bool supportsEncoding(const std::string &codecName) override;
+	rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> makeVideoSource(rtc::Thread *signalingThread, rtc::Thread *workerThread) override;
+    void adaptVideoSource(rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> videoSource, int width, int height, int fps) override;
+	std::unique_ptr<VideoCapturerInterface> makeVideoCapturer(rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source, std::string deviceId, std::function<void(VideoState)> stateUpdated, std::shared_ptr<PlatformContext> platformContext, std::pair<int, int> &outResolution) override;
+
+private:
+	rtc::scoped_refptr<webrtc::JavaVideoTrackSourceInterface> _source;
+	std::unique_ptr<webrtc::VideoEncoderFactory> hardwareVideoEncoderFactory;
+	std::unique_ptr<webrtc::VideoEncoderFactory> softwareVideoEncoderFactory;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/platform/android/VideoCameraCapturer.cpp

@@ -0,0 +1,63 @@
+#include "VideoCameraCapturer.h"
+
+#include <stdint.h>
+#include <memory>
+#include <algorithm>
+
+#include "AndroidInterface.h"
+#include "AndroidContext.h"
+#include "sdk/android/native_api/jni/jvm.h"
+
+namespace tgcalls {
+
+VideoCameraCapturer::VideoCameraCapturer(rtc::scoped_refptr<webrtc::JavaVideoTrackSourceInterface> source, std::string deviceId, std::function<void(VideoState)> stateUpdated, std::shared_ptr<PlatformContext> platformContext) : _source(source), _stateUpdated(stateUpdated), _platformContext(platformContext) {
+    AndroidContext *context = (AndroidContext *) platformContext.get();
+    JNIEnv *env = webrtc::AttachCurrentThreadIfNeeded();
+    jmethodID methodId = env->GetMethodID(context->getJavaCapturerClass(), "init", "(JZ)V");
+    const auto useFrontCamera = deviceId.empty() || (deviceId == "default") || (deviceId == "front");
+    env->CallVoidMethod(context->getJavaCapturer(), methodId, reinterpret_cast<intptr_t>(this), useFrontCamera);
+}
+
+void VideoCameraCapturer::setState(VideoState state) {
+    _state = state;
+    if (_stateUpdated) {
+        _stateUpdated(_state);
+    }
+    JNIEnv *env = webrtc::AttachCurrentThreadIfNeeded();
+    AndroidContext *context = (AndroidContext *) _platformContext.get();
+    jmethodID methodId = env->GetMethodID(context->getJavaCapturerClass(), "onStateChanged", "(JI)V");
+    env->CallVoidMethod(context->getJavaCapturer(), methodId, (jlong) (intptr_t) this, (jint) state);
+}
+
+void VideoCameraCapturer::setPreferredCaptureAspectRatio(float aspectRatio) {
+    _aspectRatio = aspectRatio;
+    JNIEnv *env = webrtc::AttachCurrentThreadIfNeeded();
+    AndroidContext *context = (AndroidContext *) _platformContext.get();
+    jmethodID methodId = env->GetMethodID(context->getJavaCapturerClass(), "onAspectRatioRequested", "(F)V");
+    env->CallVoidMethod(context->getJavaCapturer(), methodId, (jfloat) aspectRatio);
+}
+
+void VideoCameraCapturer::setUncroppedSink(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+    if (_uncroppedSink != nullptr) {
+        _source->RemoveSink(_uncroppedSink.get());
+    }
+    if (sink != nullptr) {
+        _source->AddOrUpdateSink(sink.get(), rtc::VideoSinkWants());
+    }
+    _uncroppedSink = sink;
+}
+
+webrtc::ScopedJavaLocalRef<jobject> VideoCameraCapturer::GetJavaVideoCapturerObserver(JNIEnv *env) {
+    return _source->GetJavaVideoCapturerObserver(env);
+}
+
+}  // namespace tgcalls
+
+extern "C" {
+
+JNIEXPORT jobject Java_org_telegram_messenger_voip_VideoCameraCapturer_nativeGetJavaVideoCapturerObserver(JNIEnv *env, jclass clazz, jlong ptr) {
+    tgcalls::VideoCameraCapturer *capturer = (tgcalls::VideoCameraCapturer *) (intptr_t) ptr;
+    return capturer->GetJavaVideoCapturerObserver(env).Release();
+}
+
+}

Telegram/ThirdParty/tgcalls/tgcalls/platform/android/VideoCameraCapturer.h

@@ -0,0 +1,45 @@
+#ifndef TGCALLS_VIDEO_CAMERA_CAPTURER_H
+#define TGCALLS_VIDEO_CAMERA_CAPTURER_H
+
+#include "api/scoped_refptr.h"
+#include "api/media_stream_interface.h"
+#include "modules/video_capture/video_capture.h"
+#include "sdk/android/native_api/jni/scoped_java_ref.h"
+#include "sdk/android/native_api/video/video_source.h"
+#include "VideoCaptureInterface.h"
+
+#include <memory>
+#include <vector>
+#include <stddef.h>
+#include <jni.h>
+
+namespace tgcalls {
+
+class VideoCameraCapturer;
+
+class VideoCameraCapturer {
+
+public:
+	VideoCameraCapturer(rtc::scoped_refptr<webrtc::JavaVideoTrackSourceInterface> source, std::string deviceId, std::function<void(VideoState)> stateUpdated, std::shared_ptr<PlatformContext> platformContext);
+
+	void setState(VideoState state);
+	void setPreferredCaptureAspectRatio(float aspectRatio);
+	void setUncroppedSink(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink);
+
+    webrtc::ScopedJavaLocalRef<jobject> GetJavaVideoCapturerObserver(JNIEnv* env);
+
+private:
+	rtc::scoped_refptr<webrtc::JavaVideoTrackSourceInterface> _source;
+
+	std::function<void(VideoState)> _stateUpdated;
+	VideoState _state;
+
+	std::shared_ptr<PlatformContext> _platformContext;
+
+	float _aspectRatio;
+	std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> _uncroppedSink;
+};
+
+}  // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/platform/android/VideoCapturerInterfaceImpl.cpp

@@ -0,0 +1,23 @@
+#include "VideoCapturerInterfaceImpl.h"
+
+#include "VideoCameraCapturer.h"
+
+namespace tgcalls {
+
+VideoCapturerInterfaceImpl::VideoCapturerInterfaceImpl(rtc::scoped_refptr<webrtc::JavaVideoTrackSourceInterface> source, std::string deviceId, std::function<void(VideoState)> stateUpdated, std::shared_ptr<PlatformContext> platformContext) {
+	_capturer = std::unique_ptr<VideoCameraCapturer>(new VideoCameraCapturer(source, deviceId, stateUpdated, platformContext));
+}
+
+void VideoCapturerInterfaceImpl::setState(VideoState state) {
+	_capturer->setState(state);
+}
+
+void VideoCapturerInterfaceImpl::setPreferredCaptureAspectRatio(float aspectRatio) {
+	_capturer->setPreferredCaptureAspectRatio(aspectRatio);
+}
+
+void VideoCapturerInterfaceImpl::setUncroppedOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) {
+	_capturer->setUncroppedSink(sink);
+}
+
+} // namespace tgcalls

Telegram/ThirdParty/tgcalls/tgcalls/platform/android/VideoCapturerInterfaceImpl.h

@@ -0,0 +1,25 @@
+#ifndef TGCALLS_VIDEO_CAPTURER_INTERFACE_IMPL_H
+#define TGCALLS_VIDEO_CAPTURER_INTERFACE_IMPL_H
+
+#include "VideoCapturerInterface.h"
+#include "VideoCameraCapturer.h"
+#include "api/media_stream_interface.h"
+
+namespace tgcalls {
+
+class VideoCapturerInterfaceImpl final : public VideoCapturerInterface {
+public:
+	VideoCapturerInterfaceImpl(rtc::scoped_refptr<webrtc::JavaVideoTrackSourceInterface> source, std::string deviceId, std::function<void(VideoState)> stateUpdated, std::shared_ptr<PlatformContext> platformContext);
+
+	void setState(VideoState state) override;
+	void setPreferredCaptureAspectRatio(float aspectRatio) override;
+	void setUncroppedOutput(std::shared_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> sink) override;
+
+private:
+	std::unique_ptr<VideoCameraCapturer> _capturer;
+
+};
+
+} // namespace tgcalls
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/platform/darwin/AudioDeviceModuleIOS.h

@@ -0,0 +1,39 @@
+#ifndef TGCALLS_AUDIO_DEVICE_MODULE_IOS
+#define TGCALLS_AUDIO_DEVICE_MODULE_IOS
+
+#include "platform/PlatformInterface.h"
+
+namespace tgcalls {
+
+class AudioDeviceModuleIOS : public DefaultWrappedAudioDeviceModule {
+public:
+    AudioDeviceModuleIOS(webrtc::scoped_refptr<webrtc::AudioDeviceModule> impl) :
+    DefaultWrappedAudioDeviceModule(impl) {
+    }
+
+    virtual ~AudioDeviceModuleIOS() {
+    }
+
+    virtual int32_t StopPlayout() override {
+        return 0;
+    }
+
+    virtual int32_t StopRecording() override {
+        return 0;
+    }
+
+    virtual int32_t Terminate() override {
+        return 0;
+    }
+
+    virtual void Stop() override {
+        WrappedInstance()->StopPlayout();
+        WrappedInstance()->StopRecording();
+        WrappedInstance()->Terminate();
+    }
+};
+
+}
+
+#endif
+

Telegram/ThirdParty/tgcalls/tgcalls/platform/darwin/AudioDeviceModuleMacos.h

@@ -0,0 +1,31 @@
+
+#ifndef TGCALLS_AUDIO_DEVICE_MODULE_MACOS
+#define TGCALLS_AUDIO_DEVICE_MODULE_MACOS
+
+#include "platform/PlatformInterface.h"
+
+namespace tgcalls {
+
+class AudioDeviceModuleMacos : public DefaultWrappedAudioDeviceModule {
+public:
+    AudioDeviceModuleMacos(webrtc::scoped_refptr<webrtc::AudioDeviceModule> impl) :
+    DefaultWrappedAudioDeviceModule(impl) {
+    }
+
+    virtual ~AudioDeviceModuleMacos() {
+    }
+    virtual int32_t SetStereoPlayout(bool enable) override {
+        return WrappedInstance()->SetStereoPlayout(enable);
+    }
+    
+
+    virtual void Stop() override {
+        WrappedInstance()->StopPlayout();
+        WrappedInstance()->StopRecording();
+        WrappedInstance()->Terminate();
+    }
+};
+
+}
+
+#endif

Telegram/ThirdParty/tgcalls/tgcalls/platform/darwin/CustomExternalCapturer.h

@@ -0,0 +1,21 @@
+#ifndef TGCALLS_CUSTOM_EXTERNAL_CAPTURER_H
+#define TGCALLS_CUSTOM_EXTERNAL_CAPTURER_H
+#ifdef WEBRTC_IOS
+#import <Foundation/Foundation.h>
+#import <AVFoundation/AVFoundation.h>
+
+#include <memory>
+#include "api/scoped_refptr.h"
+#include "api/media_stream_interface.h"
+#import "base/RTCVideoFrame.h"
+#include "Instance.h"
+
+@interface CustomExternalCapturer : NSObject
+
+- (instancetype)initWithSource:(webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)source;
+
++ (void)passPixelBuffer:(CVPixelBufferRef)pixelBuffer sampleBufferReference:(CMSampleBufferRef)sampleBufferReference rotation:(RTCVideoRotation)rotation toSource:(webrtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)source croppingBuffer:(std::vector<uint8_t> &)croppingBuffer;
+
+@end
+#endif // WEBRTC_IOS
+#endif