delegate;
+
+/**
+ * 1.4 Set the queue that drives the `TRTCCloudDelegate` event callback
+ *
+ * If you do not specify a `delegateQueue`, the SDK will use `MainQueue` as the queue for driving {@link TRTCCloudDelegate} event callbacks by default.
+ * In other words, if you do not set the `delegateQueue` attribute, all callback functions in {@link TRTCCloudDelegate} will be driven by `MainQueue`.
+ * @note If you specify a `delegateQueue`, please do not manipulate the UI in the {@link TRTCCloudDelegate} callback function; otherwise, thread safety issues will occur.
+ */
+@property(nonatomic, strong) dispatch_queue_t delegateQueue;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Room APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Room APIs
+/// @{
+
+/**
+ * 2.1 Enter room
+ *
+ * All TRTC users need to enter a room before they can "publish" or "subscribe to" audio/video streams. "Publishing" refers to pushing their own streams to the cloud, and "subscribing to" refers to pulling the streams of other users in the room from
+ * the cloud. When calling this API, you need to specify your application scenario ({@link TRTCAppScene}) to get the best audio/video transfer experience. We provide the following four scenarios for your choice:
+ * - {@link TRTCAppSceneVideoCall}:
+ * Video call scenario. Use cases: [one-to-one video call], [video conferencing with up to 300 participants], [online medical diagnosis], [small class], [video interview], etc.
+ * In this scenario, each room supports up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
+ * - {@link TRTCAppSceneAudioCall}:
+ * Audio call scenario. Use cases: [one-to-one audio call], [audio conferencing with up to 300 participants], [audio chat], [online Werewolf], etc.
+ * In this scenario, each room supports up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
+ * - {@link TRTCAppSceneLIVE}:
+ * Live streaming scenario. Use cases: [low-latency video live streaming], [interactive classroom for up to 100,000 participants], [live video competition], [video dating room], [remote training], [large-scale conferencing], etc.
+ * In this scenario, each room supports up to 100,000 concurrent online users, but you should specify the user roles: anchor ({@link TRTCRoleAnchor }) or audience ({@link TRTCRoleAudience }).
+ * - {@link TRTCAppSceneVoiceChatRoom}:
+ * Audio chat room scenario. Use cases: [Clubhouse], [online karaoke room], [music live room], [FM radio], etc.
+ * In this scenario, each room supports up to 100,000 concurrent online users, but you should specify the user roles: anchor ({@link TRTCRoleAnchor }) or audience ({@link TRTCRoleAudience }).
+ * After calling this API, you will receive the `onEnterRoom(result)` callback from {@link TRTCCloudDelegate}:
+ * - If room entry succeeded, the `result` parameter will be a positive number (`result` > 0), indicating the time in milliseconds (ms) between function call and room entry.
+ * - If room entry failed, the `result` parameter will be a negative number (`result` < 0), indicating the [error code](https://cloud.tencent.com/document/product/647/32257) for room entry failure.
+ * @param param Room entry parameter, which is used to specify the user's identity, role, authentication credentials, and other information. For more information, please see {@link TRTCParams}.
+ * @param scene Application scenario, which is used to specify the use case. The same {@link TRTCAppScene} should be configured for all users in the same room.
+ * @note
+ * 1. If `scene` is specified as {@link TRTCAppSceneLIVE} or {@link TRTCAppSceneVoiceChatRoom}, you must use the `role` field in {@link TRTCParams} to specify the role of the current user in the room.
+ * 2. The same `scene` should be configured for all users in the same room.
+ * 3. Please try to ensure that {@link enterRoom} and {@link exitRoom} are used in pair; that is, please make sure that "the previous room is exited before the next room is entered"; otherwise, many issues may occur.
+ */
+- (void)enterRoom:(TRTCParams *)param appScene:(TRTCAppScene)scene;
+
+/**
+ * 2.2 Exit room
+ *
+ * Calling this API will allow the user to leave the current audio or video room and release the camera, mic, speaker, and other device resources.
+ * After resources are released, the SDK will use the `onExitRoom()` callback in {@link TRTCCloudDelegate} to notify you.
+ * If you need to call {@link enterRoom} again or switch to the SDK of another provider, we recommend you wait until you receive the `onExitRoom()` callback, so as to avoid the problem of the camera or mic being occupied.
+ */
+- (void)exitRoom;
+
+/**
+ * 2.3 Switch role
+ *
+ * This API is used to switch the user role between "anchor" and "audience".
+ * As video live rooms and audio chat rooms need to support an audience of up to 100,000 concurrent online users, the rule "only anchors can publish their audio/video streams" has been set. Therefore, when some users want to publish their streams (so
+ * that they can interact with anchors), they need to switch their role to "anchor" first. You can use the `role` field in {@link TRTCParams} during room entry to specify the user role in advance or use the `switchRole` API to switch roles after room
+ * entry.
+ * @param role Role, which is "anchor" by default:
+ * - {@link TRTCRoleAnchor}: anchor, who can publish their audio/video streams. Up to 50 anchors are allowed to publish streams at the same time in one room.
+ * - {@link TRTCRoleAudience}: audience, who cannot publish their audio/video streams, but can only watch streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through {@link
+ * switchRole}. One room supports an audience of up to 100,000 concurrent online users.
+ * @note
+ * 1. This API is only applicable to two scenarios: live streaming ({@link TRTCAppSceneLIVE}) and audio chat room ({@link TRTCAppSceneVoiceChatRoom}).
+ * 2. If the `scene` you specify in {@link enterRoom} is {@link TRTCAppSceneVideoCall} or {@link TRTCAppSceneAudioCall}, please do not call this API.
+ */
+- (void)switchRole:(TRTCRoleType)role;
+
+/**
+ * 2.4 Switch room
+ *
+ * This API is used to quickly switch a user from one room to another.
+ * - If the user's role is "audience", calling this API is equivalent to `exitRoom` (current room) + `enterRoom` (new room).
+ * - If the user's role is "anchor", the API will retain the current audio/video publishing status while switching the room; therefore, during the room switch, camera preview and sound capturing will not be interrupted.
+ * This API is suitable for the online education scenario where the supervising teacher can perform fast room switch across multiple rooms. In this scenario, using `switchRoom` can get better smoothness and use less code than `exitRoom + enterRoom`.
+ * The API call result will be called back through `onSwitchRoom(errCode, errMsg)` in {@link TRTCCloudDelegate}.
+ * @param config Room parameter. For more information, please see {@link TRTCSwitchRoomConfig}.
+ * @note Due to the requirement for compatibility with legacy versions of the SDK, the `config` parameter contains both `roomId` and `strRoomId` parameters. You should pay special attention as detailed below when specifying these two parameters:
+ * 1. If you decide to use `strRoomId`, then set `roomId` to 0. If both are specified, `roomId` will be used.
+ * 2. All rooms need to use either `strRoomId` or `roomId` at the same time. They cannot be mixed; otherwise, there will be many unexpected bugs.
+ */
+- (void)switchRoom:(TRTCSwitchRoomConfig *)config;
+
+/**
+ * 2.5 Request cross-room call
+ *
+ * By default, only users in the same room can make audio/video calls with each other, and the audio/video streams in different rooms are isolated from each other.
+ * However, you can publish the audio/video streams of an anchor in another room to the current room by calling this API. At the same time, this API will also publish the local audio/video streams to the target anchor's room.
+ * In other words, you can use this API to share the audio/video streams of two anchors in two different rooms, so that the audience in each room can watch the streams of these two anchors. This feature can be used to implement anchor competition.
+ * The result of requesting cross-room call will be returned through the `onConnectOtherRoom()` callback in {@link TRTCCloudDelegate}.
+ * For example, after anchor A in room "101" uses `connectOtherRoom()` to successfully call anchor B in room "102":
+ * - All users in room "101" will receive the `onRemoteUserEnterRoom(B)` and `onUserVideoAvailable(B,YES)` event callbacks of anchor B; that is, all users in room "101" can subscribe to the audio/video streams of anchor B.
+ * - All users in room "102" will receive the `onRemoteUserEnterRoom(A)` and `onUserVideoAvailable(A,YES)` event callbacks of anchor A; that is, all users in room "102" can subscribe to the audio/video streams of anchor A.
+ *
+ * Room 101 Room 102
+ * --------------------- ---------------------
+ * Before cross-room call: | Anchor: A | | Anchor: B |
+ * | Users : U, V, W | | Users: X, Y, Z |
+ * --------------------- ---------------------
+ *
+ * Room 101 Room 102
+ * --------------------- ---------------------
+ * After cross-room call: | Anchors: A and B | | Anchors: B and A |
+ * | Users : U, V, W | | Users : X, Y, Z |
+ * --------------------- ---------------------
+ *
+ * For compatibility with subsequent extended fields for cross-room call, parameters in JSON format are used currently.
+ * Case 1: numeric room ID
+ * If anchor A in room "101" wants to co-anchor with anchor B in room "102", then anchor A needs to pass in {"roomId": 102, "userId": "userB"} when calling this API.
+ * Below is the sample code:
+ *
+ * NSMutableDictionaryjsonDict = [[NSMutableDictionary alloc] init];
+ * [jsonDict setObject:@(102) forKey:@"roomId"];
+ * [jsonDict setObject:@"userB" forKey:@"userId"];
+ * NSData* jsonData = [NSJSONSerialization dataWithJSONObject:jsonDict options:NSJSONWritingPrettyPrinted error:nil];
+ * NSString* jsonString = [[NSString alloc] initWithData:jsonData encoding:NSUTF8StringEncoding];
+ * [trtc connectOtherRoom:jsonString];
+ *
+ *
+ * Case 2: string room ID
+ * If you use a string room ID, please be sure to replace the `roomId` in JSON with `strRoomId`, such as {"strRoomId": "102", "userId": "userB"}
+ * Below is the sample code:
+ *
+ * NSMutableDictionaryjsonDict = [[NSMutableDictionary alloc] init];
+ * [jsonDict setObject:@"102" forKey:@"strRoomId"];
+ * [jsonDict setObject:@"userB" forKey:@"userId"];
+ * NSData* jsonData = [NSJSONSerialization dataWithJSONObject:jsonDict options:NSJSONWritingPrettyPrinted error:nil];
+ * NSString* jsonString = [[NSString alloc] initWithData:jsonData encoding:NSUTF8StringEncoding];
+ * [trtc connectOtherRoom:jsonString];
+ *
+ *
+ * @param param You need to pass in a string parameter in JSON format: `roomId` represents the room ID in numeric format, `strRoomId` represents the room ID in string format, and `userId` represents the user ID of the target anchor.
+ */
+- (void)connectOtherRoom:(NSString *)param;
+
+/**
+ * 2.6 Exit cross-room call
+ *
+ * The result will be returned through the `onDisconnectOtherRoom()` callback in {@link TRTCCloudDelegate}.
+ */
+- (void)disconnectOtherRoom;
+
+/**
+ * 2.7 Set subscription mode (which must be set before room entry for it to take effect)
+ *
+ * You can switch between the "automatic subscription" and "manual subscription" modes through this API:
+ * - Automatic subscription: this is the default mode, where the user will immediately receive the audio/video streams in the room after room entry, so that the audio will be automatically played back, and the video will be automatically decoded (you
+ * still need to bind the rendering control through the `startRemoteView` API).
+ * - Manual subscription: after room entry, the user needs to manually call the {@startRemoteView} API to start subscribing to and decoding the video stream and call the `{@muteRemoteAudio} (NO)` API to start playing back the audio stream.
+ * In most scenarios, users will subscribe to the audio/video streams of all anchors in the room after room entry. Therefore, TRTC adopts the automatic subscription mode by default in order to achieve the best "instant streaming experience".
+ * In your application scenario, if there are many audio/video streams being published at the same time in each room, and each user only wants to subscribe to 1–2 streams of them, we recommend you use the "manual subscription" mode to reduce the
+ * traffic costs.
+ * @param autoRecvAudio YES: automatic subscription to audio; NO: manual subscription to audio by calling `muteRemoteAudio(NO)`. Default value: YES
+ * @param autoRecvVideo YES: automatic subscription to video; NO: manual subscription to video by calling `startRemoteView`. Default value: YES
+ * @note
+ * 1. The configuration takes effect only if this API is called before room entry (enterRoom).
+ * 2. In the automatic subscription mode, if the user does not call {@startRemoteView} to subscribe to the video stream after room entry, the SDK will automatically stop subscribing to the video stream in order to reduce the traffic consumption.
+ */
+- (void)setDefaultStreamRecvMode:(BOOL)autoRecvAudio video:(BOOL)autoRecvVideo;
+
+/**
+ * 2.8 Create room subinstance (for concurrent multi-room listen/watch)
+ *
+ * `TRTCCloud` was originally designed to work in the singleton mode, which limited the ability to watch concurrently in multiple rooms.
+ * By calling this API, you can create multiple `TRTCCloud` instances, so that you can enter multiple different rooms at the same time to listen/watch audio/video streams.
+ * However, it should be noted that because there are still only one camera and one mic available, you can exist as an "anchor" in only one `TRTCCloud` instance at any time; that is, you can only publish your audio/video streams in one `TRTCCloud`
+ * instance at any time. This feature is mainly used in the "super small class" use case in the online education scenario to break the limit that "only up to 50 users can publish their audio/video streams simultaneously in one TRTC room". Below is
+ * the sample code: TRTCCloud *mainCloud = [TRTCCloud sharedInstance]; [mainCloud enterRoom:params1 appScene:TRTCAppSceneLIVE)];
+ * //...
+ * //Switch the role from "anchor" to "audience" in your own room
+ * [mainCloud switchRole:TRTCRoleAudience];
+ * [mainCloud muteLocalVideo:YES];
+ * [mainCloud muteLocalAudio:YES];
+ * //...
+ * //Use subcloud to enter another room and switch the role from "audience" to "anchor"
+ * TRTCCloud *subCloud = [mainCloud createSubCloud];
+ * [subCloud enterRoom:params2 appScene:TRTCAppSceneLIVE)];
+ * [subCloud switchRole:TRTCRoleAnchor];
+ * [subCloud muteLocalVideo:NO];
+ * [subCloud muteLocalAudio:NO];
+ * //...
+ * //Exit from new room and release it.
+ * [subCloud exitRoom];
+ * [mainCloud destroySubCloud:subCloud];
+ *
+ *
+ * @note
+ * - The same user can enter multiple rooms with different `roomId` values by using the same `userId`.
+ * - Two devices cannot use the same `userId` to enter the same room with a specified `roomId`.
+ * - The same user can push a stream in only one `TRTCCloud` instance at any time. If streams are pushed simultaneously in different rooms, a status mess will be caused in the cloud, leading to various bugs.
+ * - The `TRTCCloud` instance created by the `createSubCloud` API cannot call APIs related to the local audio/video in the subinstance, except `switchRole`, `muteLocalVideo`, and `muteLocalAudio`. To use APIs such as the beauty filter, please use the
+ * original `TRTCCloud` instance object.
+ * @return `TRTCCloud` subinstance
+ */
+- (TRTCCloud *)createSubCloud;
+
+/**
+ * 2.9 Terminate room subinstance
+ *
+ * @param subCloud
+ */
+- (void)destroySubCloud:(TRTCCloud *)subCloud;
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// CDN APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * 3.1 Start publishing audio/video streams to Tencent Cloud CSS CDN
+ *
+ * This API sends a command to the TRTC server, requesting it to relay the current user's audio/video streams to CSS CDN.
+ * You can set the `StreamId` of the live stream through the `streamId` parameter, so as to specify the playback address of the user's audio/video streams on CSS CDN.
+ * For example, if you specify the current user's live stream ID as `user_stream_001` through this API, then the corresponding CDN playback address is:
+ * "http://yourdomain/live/user_stream_001.flv", where `yourdomain` is your playback domain name with an ICP filing.
+ * You can configure your playback domain name in the [CSS console](https://console.cloud.tencent.com/live). Tencent Cloud does not provide a default playback domain name.
+ *
+ * TRTCCloud *trtcCloud = [TRTCCloud sharedInstance];
+ * [trtcCloud enterRoom:params appScene:TRTCAppSceneLIVE];
+ * [trtcCloud startLocalPreview:frontCamera view:localView];
+ * [trtcCloud startLocalAudio];
+ * [trtcCloud startPublishing: @"user_stream_001" type:TRTCVideoStreamTypeBig];
+ *
+ *
+ * You can also specify the `streamId` when setting the `TRTCParams` parameter of `enterRoom`, which is the recommended approach.
+ * @param streamId Custom stream ID.
+ * @param streamType Only `TRTCVideoStreamTypeBig` and `TRTCVideoStreamTypeSub` are supported.
+ * @note You need to enable the "Enable Relayed Push" option on the "Function Configuration" page in the [TRTC console](https://console.cloud.tencent.com/trtc/) in advance.
+ * - If you select "Specified stream for relayed push", you can use this API to push the corresponding audio/video stream to Tencent Cloud CDN and specify the entered stream ID.
+ * - If you select "Global auto-relayed push", you can use this API to adjust the default stream ID.
+ */
+- (void)startPublishing:(NSString *)streamId type:(TRTCVideoStreamType)streamType;
+
+/**
+ * 3.2 Stop publishing audio/video streams to Tencent Cloud CSS CDN
+ */
+- (void)stopPublishing;
+
+/**
+ * 3.3 Start publishing audio/video streams to non-Tencent Cloud CDN
+ *
+ * This API is similar to the `startPublishing` API. The difference is that `startPublishing` can only publish audio/video streams to Tencent Cloud CDN, while this API can relay streams to live streaming CDN services of other cloud providers.
+ * @param param CDN relaying parameter. For more information, please see {@link TRTCPublishCDNParam}
+ * @note
+ * - Using the `startPublishing` API to publish audio/video streams to Tencent Cloud CSS CDN does not incur additional fees.
+ * - Using the `startPublishCDNStream` API to publish audio/video streams to non-Tencent Cloud CDN incurs additional relaying bandwidth fees.
+ */
+- (void)startPublishCDNStream:(TRTCPublishCDNParam *)param;
+
+/**
+ * 3.4 Stop publishing audio/video streams to non-Tencent Cloud CDN
+ */
+- (void)stopPublishCDNStream;
+
+/**
+ * 3.5 Set the layout and transcoding parameters of On-Cloud MixTranscoding
+ *
+ * In a live room, there may be multiple anchors publishing their audio/video streams at the same time, but for audience on CSS CDN, they only need to watch one video stream in HTTP-FLV or HLS format.
+ * When you call this API, the SDK will send a command to the TRTC mixtranscoding server to combine multiple audio/video streams in the room into one stream.
+ * You can use the {@link TRTCTranscodingConfig} parameter to set the layout of each channel of image. You can also set the encoding parameters of the mixed audio/video streams.
+ * For more information, please see [On-Cloud MixTranscoding](https://cloud.tencent.com/document/product/647/16827).
+ *
+ * **Image 1** => decoding ====> \\
+ * \\
+ * **Image 2** => decoding => image mixing => encoding => **mixed image**
+ * //
+ * **Image 3** => decoding ====> //
+ *
+ * **Audio 1** => decoding ====> \\
+ * \\
+ * **Audio 2** => decoding => audio mixing => encoding => **mixed audio**
+ * //
+ * **Audio 3** => decoding ====> //
+ *
+ * @param config If `config` is not empty, On-Cloud MixTranscoding will be started; otherwise, it will be stopped. For more information, please see {@link TRTCTranscodingConfig}.
+ * @note Notes on On-Cloud MixTranscoding:
+ * - Mixed-stream transcoding is a chargeable function, calling the interface will incur cloud-based mixed-stream transcoding fees, see https://intl.cloud.tencent.com/document/product/647/38929.
+ * - If the user calling this API does not set `streamId` in the `config` parameter, TRTC will mix the multiple channels of images in the room into the audio/video streams corresponding to the current user, i.e., A + B => A.
+ * - If the user calling this API sets `streamId` in the `config` parameter, TRTC will mix the multiple channels of images in the room into the specified `streamId`, i.e., A + B => streamId.
+ * - Please note that if you are still in the room but do not need mixtranscoding anymore, be sure to call this API again and leave `config` empty to cancel it; otherwise, additional fees may be incurred.
+ * - Please rest assured that TRTC will automatically cancel the mixtranscoding status upon room exit.
+ */
+- (void)setMixTranscodingConfig:(TRTCTranscodingConfig *)config;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Video APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Video APIs
+/// @{
+
+/**
+ * 4.1 Enable the preview image of local camera (mobile)
+ *
+ * If this API is called before `enterRoom`, the SDK will only enable the camera and wait until `enterRoom` is called before starting push.
+ * If it is called after `enterRoom`, the SDK will enable the camera and automatically start pushing the video stream.
+ * When the first camera video frame starts to be rendered, you will receive the `onCameraDidReady` callback in {@link TRTCCloudDelegate}.
+ * @param frontCamera YES: front camera; NO: rear camera
+ * @param view Control that carries the video image
+ * @note If you want to preview the camera image and adjust the beauty filter parameters through `BeautyManager` before going live, you can:
+ * - Scheme 1. Call `startLocalPreview` before calling `enterRoom`
+ * - Scheme 2. Call `startLocalPreview` and `muteLocalVideo(YES)` after calling `enterRoom`
+ */
+#if TARGET_OS_IPHONE
+- (void)startLocalPreview:(BOOL)frontCamera view:(TXView *)view;
+#endif
+
+/**
+ * 4.2 Enable the preview image of local camera (desktop)
+ *
+ * Before this API is called, `setCurrentCameraDevice` can be called first to select whether to use the macOS device's built-in camera or an external camera.
+ * If this API is called before `enterRoom`, the SDK will only enable the camera and wait until `enterRoom` is called before starting push.
+ * If it is called after `enterRoom`, the SDK will enable the camera and automatically start pushing the video stream.
+ * When the first camera video frame starts to be rendered, you will receive the `onCameraDidReady` callback in {@link TRTCCloudDelegate}.
+ * @param view Control that carries the video image
+ * @note If you want to preview the camera image and adjust the beauty filter parameters through `BeautyManager` before going live, you can:
+ * - Scheme 1. Call `startLocalPreview` before calling `enterRoom`
+ * - Scheme 2. Call `startLocalPreview` and `muteLocalVideo(YES)` after calling `enterRoom`
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)startLocalPreview:(TXView *)view;
+#endif
+
+/**
+ * 4.3 Update the preview image of local camera
+ */
+- (void)updateLocalView:(TXView *)view;
+
+/**
+ * 4.4 Stop camera preview
+ */
+- (void)stopLocalPreview;
+
+/**
+ * 4.5 Pause/Resume publishing local video stream
+ *
+ * This API can pause (or resume) publishing the local video image. After the pause, other users in the same room will not be able to see the local image.
+ * This API is equivalent to the two APIs of `startLocalPreview/stopLocalPreview` when TRTCVideoStreamTypeBig is specified, but has higher performance and response speed.
+ * The `startLocalPreview/stopLocalPreview` APIs need to enable/disable the camera, which are hardware device-related operations, so they are very time-consuming.
+ * In contrast, `muteLocalVideo` only needs to pause or allow the data stream at the software level, so it is more efficient and more suitable for scenarios where frequent enabling/disabling are needed.
+ * After local video publishing is paused, other members in the same room will receive the `onUserVideoAvailable(userId, NO)` callback notification.
+ * After local video publishing is resumed, other members in the same room will receive the `onUserVideoAvailable(userId, YES)` callback notification.
+ * @param streamType Specify for which video stream to pause (or resume). Only {@link TRTCVideoStreamTypeBig} and {@link TRTCVideoStreamTypeSub} are supported
+ * @param mute YES: pause; NO: resume
+ */
+- (void)muteLocalVideo:(TRTCVideoStreamType)streamType mute:(BOOL)mute;
+
+/**
+ * 4.6 Set placeholder image during local video pause
+ *
+ * When you call `muteLocalVideo(YES)` to pause the local video image, you can set a placeholder image by calling this API. Then, other users in the room will see this image instead of a black screen.
+ * @param image Placeholder image. A null value means that no more video stream data will be sent after `muteLocalVideo`. The default value is null.
+ * @param fps Frame rate of the placeholder image. Minimum value: 5. Maximum value: 10. Default value: 5
+ */
+- (void)setVideoMuteImage:(TXImage *)image fps:(NSInteger)fps;
+
+/**
+ * 4.7 Subscribe to remote user's video stream and bind video rendering control
+ *
+ * Calling this API allows the SDK to pull the video stream of the specified `userId` and render it to the rendering control specified by the `view` parameter. You can set the display mode of the video image through {@link setRemoteRenderParams}.
+ * - If you already know the `userId` of a user who has a video stream in the room, you can directly call `startRemoteView` to subscribe to the user's video image.
+ * - If you don't know which users in the room are publishing video streams, you can wait for the notification from {@link onUserVideoAvailable} after `enterRoom`.
+ *
+ * Calling this API only starts pulling the video stream, and the image needs to be loaded and buffered at this time. After the buffering is completed, you will receive a notification from {@link onFirstVideoFrame}.
+ * @param userId ID of the specified remote user
+ * @param streamType Video stream type of the `userId` specified for watching:
+ * - HD big image: {@link TRTCVideoStreamTypeBig}
+ * - Smooth small image: {@link TRTCVideoStreamTypeSmall} (the remote user should enable dual-channel encoding through {@link enableEncSmallVideoStream} for this parameter to take effect)
+ * - Substream image (usually used for screen sharing): {@link TRTCVideoStreamTypeSub}
+ *
+ * @param view Rendering control that carries the video image
+ * @note The following requires your attention:
+ * 1. The SDK supports watching the big image and substream image or small image and substream image of a `userId` at the same time, but does not support watching the big image and small image at the same time.
+ * 2. Only when the specified `userId` enables dual-channel encoding through {@link enableEncSmallVideoStream} can the user's small image be viewed.
+ * 3. If the small image of the specified `userId` does not exist, the SDK will switch to the big image of the user by default.
+ */
+- (void)startRemoteView:(NSString *)userId streamType:(TRTCVideoStreamType)streamType view:(TXView *)view;
+
+/**
+ * 4.8 Update remote user's video rendering control
+ *
+ * This API can be used to update the rendering control of the remote video image. It is often used in interactive scenarios where the display area needs to be switched.
+ * @param view Control that carries the video image
+ * @param streamType Type of the stream for which to set the preview window (only {@link TRTCVideoStreamTypeBig} and {@link TRTCVideoStreamTypeSub} are supported)
+ * @param userId ID of the specified remote user
+ */
+- (void)updateRemoteView:(TXView *)view streamType:(TRTCVideoStreamType)streamType forUser:(NSString *)userId;
+
+/**
+ * 4.9 Stop subscribing to remote user's video stream and release rendering control
+ *
+ * Calling this API will cause the SDK to stop receiving the user's video stream and release the decoding and rendering resources for the stream.
+ * @param userId ID of the specified remote user
+ * @param streamType Video stream type of the `userId` specified for watching:
+ * - HD big image: {@link TRTCVideoStreamTypeBig}
+ * - Smooth small image: {@link TRTCVideoStreamTypeSmall}
+ * - Substream image (usually used for screen sharing): {@link TRTCVideoStreamTypeSub}
+ */
+- (void)stopRemoteView:(NSString *)userId streamType:(TRTCVideoStreamType)streamType;
+
+/**
+ * 4.10 Stop subscribing to all remote users' video streams and release all rendering resources
+ *
+ * Calling this API will cause the SDK to stop receiving all remote video streams and release all decoding and rendering resources.
+ * @note If a substream image (screen sharing) is being displayed, it will also be stopped.
+ */
+- (void)stopAllRemoteView;
+
+/**
+ * 4.11 Pause/Resume subscribing to remote user's video stream
+ *
+ * This API only pauses/resumes receiving the specified user's video stream but does not release displaying resources; therefore, the video image will freeze at the last frame before it is called.
+ * @param userId ID of the specified remote user
+ * @param streamType Specify for which video stream to pause (or resume). Only {@link TRTCVideoStreamTypeBig} and {@link TRTCVideoStreamTypeSub} are supported
+ * @param mute Whether to pause receiving
+ * @note This API can be called before room entry (enterRoom), and the pause status will be reset after room exit (exitRoom).
+ */
+- (void)muteRemoteVideoStream:(NSString *)userId streamType:(TRTCVideoStreamType)streamType mute:(BOOL)mute;
+
+/**
+ * 4.12 Pause/Resume subscribing to all remote users' video streams
+ *
+ * This API only pauses/resumes receiving all users' video streams but does not release displaying resources; therefore, the video image will freeze at the last frame before it is called.
+ * @param mute Whether to pause receiving
+ * @note This API can be called before room entry (enterRoom), and the pause status will be reset after room exit (exitRoom).
+ */
+- (void)muteAllRemoteVideoStreams:(BOOL)mute;
+
+/**
+ * 4.13 Set the encoding parameters of video encoder
+ *
+ * This setting can determine the quality of image viewed by remote users, which is also the image quality of on-cloud recording files.
+ * @param param It is used to set relevant parameters for the video encoder. For more information, please see {@link TRTCVideoEncParam}.
+ */
+- (void)setVideoEncoderParam:(TRTCVideoEncParam *)param;
+
+/**
+ * 4.14 Set network quality control parameters
+ *
+ * This setting determines the quality control policy in a poor network environment, such as "image quality preferred" or "smoothness preferred".
+ * @param param It is used to set relevant parameters for network quality control. For details, please refer to {@link TRTCNetworkQosParam}.
+ */
+- (void)setNetworkQosParam:(TRTCNetworkQosParam *)param;
+
+/**
+ * 4.15 Set the rendering parameters of local video image
+ *
+ * The parameters that can be set include video image rotation angle, fill mode, and mirror mode.
+ * @param params Video image rendering parameters. For more information, please see {@link TRTCRenderParams}.
+ */
+- (void)setLocalRenderParams:(TRTCRenderParams *)params;
+
+/**
+ * 4.16 Set the rendering mode of remote video image
+ *
+ * The parameters that can be set include video image rotation angle, fill mode, and mirror mode.
+ * @param userId ID of the specified remote user
+ * @param streamType It can be set to the primary stream image (TRTCVideoStreamTypeBig) or substream image (TRTCVideoStreamTypeSub).
+ * @param params Video image rendering parameters. For more information, please see {@link TRTCRenderParams}.
+ */
+- (void)setRemoteRenderParams:(NSString *)userId streamType:(TRTCVideoStreamType)streamType params:(TRTCRenderParams *)params;
+
+/**
+ * 4.17 Set the direction of image output by video encoder
+ *
+ * This setting does not affect the preview direction of the local video image, but affects the direction of the image viewed by other users in the room (and on-cloud recording files).
+ * When a phone or tablet is rotated upside down, as the capturing direction of the camera does not change, the video image viewed by other users in the room will become upside-down.
+ * In this case, you can call this API to rotate the image encoded by the SDK 180 degrees, so that other users in the room can view the image in the normal direction.
+ * If you want to achieve the aforementioned user-friendly interactive experience, we recommend you directly call {@link setGSensorMode} to implement smarter direction adaptation, with no need to call this API manually.
+ * @param rotation Currently, rotation angles of 0 and 180 degrees are supported. Default value: TRTCVideoRotation_0 (no rotation)
+ */
+- (void)setVideoEncoderRotation:(TRTCVideoRotation)rotation;
+
+/**
+ * 4.18 Set the mirror mode of image output by encoder
+ *
+ * This setting does not affect the mirror mode of the local video image, but affects the mirror mode of the image viewed by other users in the room (and on-cloud recording files).
+ * @param mirror Whether to enable remote mirror mode. YES: yes; NO: no. Default value: NO
+ */
+- (void)setVideoEncoderMirror:(BOOL)mirror;
+
+/**
+ * 4.19 Set the adaptation mode of G-sensor
+ *
+ * You can achieve the following user-friendly interactive experience through this API:
+ * When a phone or tablet is rotated upside down, as the capturing direction of the camera does not change, the video image viewed by other users in the room will become upside-down.
+ * In this case, you can call this API to let the SDK automatically adjust the rotation direction of the local video image and the image output by the encoder according to the direction of the device's gyroscope, so that remote viewers can see the
+ * image in the normal direction.
+ * @param mode G-sensor mode. For more information, please see {@link TRTCGSensorMode}. Default value: TRTCGSensorMode_UIAutoLayout
+ */
+- (void)setGSensorMode:(TRTCGSensorMode)mode;
+
+/**
+ * 4.20 Enable dual-channel encoding mode with big and small images
+ *
+ * In this mode, the current user's encoder will output two channels of video streams, i.e., **HD big image** and **Smooth small image**, at the same time (only one channel of audio stream will be output though).
+ * In this way, other users in the room can choose to subscribe to the **HD big image** or **Smooth small image** according to their own network conditions or screen size.
+ * @note Dual-channel encoding will consume more CPU resources and network bandwidth; therefore, this feature can be enabled on macOS, Windows, or high-spec tablets, but is not recommended for phones.
+ * @param enable Whether to enable small image encoding. Default value: NO
+ * @param smallVideoEncParam Video parameters of small image stream
+ * @return 0: success; -1: the current big image has been set to a lower quality, and it is not necessary to enable dual-channel encoding
+ */
+- (int)enableEncSmallVideoStream:(BOOL)enable withQuality:(TRTCVideoEncParam *)smallVideoEncParam;
+
+/**
+ * 4.21 Switch the big/small image of specified remote user
+ *
+ * After an anchor in a room enables dual-channel encoding, the video image that other users in the room subscribe to through {@link startRemoteView} will be **HD big image** by default.
+ * You can use this API to select whether the image subscribed to is the big image or small image. The API can take effect before or after {@link startRemoteView} is called.
+ * @note To implement this feature, the target user must have enabled the dual-channel encoding mode through {@link enableEncSmallVideoStream}; otherwise, this API will not work.
+ * @param userId ID of the specified remote user
+ * @param streamType Video stream type, i.e., big image or small image. Default value: big image
+ */
+- (void)setRemoteVideoStreamType:(NSString *)userId type:(TRTCVideoStreamType)streamType;
+
+/**
+ * 4.22 Screencapture video
+ *
+ * You can use this API to screencapture the local video image or the primary stream image and substream (screen sharing) image of a remote user.
+ * @param userId User ID. A null value indicates to screencapture the local video.
+ * @param streamType Video stream type, which can be the primary stream image ({@link TRTCVideoStreamTypeBig}, generally for camera) or substream image ({@link TRTCVideoStreamTypeSub}, generally for screen sharing)
+ * @param sourceType Video image source, which can be the video stream image ({@link TRTCSnapshotSourceTypeStream}, generally in higher definition) or the video rendering image ({@link TRTCSnapshotSourceTypeView})
+ * @note On Windows, only video image from the {@link TRTCSnapshotSourceTypeStream} source can be screencaptured currently.
+ */
+- (void)snapshotVideo:(NSString *)userId type:(TRTCVideoStreamType)streamType sourceType:(TRTCSnapshotSourceType)sourceType completionBlock:(void (^)(TXImage *image))completionBlock;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Audio APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Audio APIs
+/// @{
+
+/**
+ * 5.1 Enable local audio capturing and publishing
+ *
+ * The SDK does not enable the mic by default. When a user wants to publish the local audio, the user needs to call this API to enable mic capturing and encode and publish the audio to the current room.
+ * After local audio capturing and publishing is enabled, other users in the room will receive the {@link onUserAudioAvailable}(userId, YES) notification.
+ * @param quality Sound quality
+ * - {@link TRTCAudioQualitySpeech} - Smooth: sample rate: 16 kHz; mono channel; audio bitrate: 16 Kbps. This is suitable for audio call scenarios, such as online meeting and audio call.
+ * - {@link TRTCAudioQualityDefault} - Default: sample rate: 48 kHz; mono channel; audio bitrate: 50 Kbps. This is the default sound quality of the SDK and recommended if there are no special requirements.
+ * - {@link TRTCAudioQualityMusic} - HD: sample rate: 48 kHz; dual channel + full band; audio bitrate: 128 Kbps. This is suitable for scenarios where Hi-Fi music transfer is required, such as online karaoke and music live streaming.
+ * @note This API will check the mic permission. If the current application does not have permission to use the mic, the SDK will automatically ask the user to grant the mic permission.
+ */
+- (void)startLocalAudio:(TRTCAudioQuality)quality;
+
+/**
+ * 5.2 Stop local audio capturing and publishing
+ *
+ * After local audio capturing and publishing is stopped, other users in the room will receive the {@link onUserAudioAvailable}(userId, NO) notification.
+ */
+- (void)stopLocalAudio;
+
+/**
+ * 5.3 Pause/Resume publishing local audio stream
+ *
+ * After local audio publishing is paused, other users in the room will receive the {@link onUserAudioAvailable}(userId, NO) notification.
+ * After local audio publishing is resumed, other users in the room will receive the {@link onUserAudioAvailable}(userId, YES) notification.
+ * Different from {@link stopLocalAudio}, `muteLocalAudio(YES)` does not release the mic permission; instead, it continues to send mute packets with extremely low bitrate.
+ * This is very suitable for scenarios that require on-cloud recording, as video file formats such as MP4 have a high requirement for audio continuity, while an MP4 recording file cannot be played back smoothly if {@link stopLocalAudio} is used.
+ * Therefore, `muteLocalAudio` instead of `stopLocalAudio` is recommended in scenarios where the requirement for recording file quality is high.
+ * @param mute YES: mute; NO: unmute
+ */
+- (void)muteLocalAudio:(BOOL)mute;
+
+/**
+ * 5.4 Pause/Resume playing back remote audio stream
+ *
+ * When you mute the remote audio of a specified user, the SDK will stop playing back the user's audio and pulling the user's audio data.
+ * @param userId ID of the specified remote user
+ * @param mute YES: mute; NO: unmute
+ * @note This API works when called either before or after room entry (enterRoom), and the mute status will be reset to `NO` after room exit (exitRoom).
+ */
+- (void)muteRemoteAudio:(NSString *)userId mute:(BOOL)mute;
+
+/**
+ * 5.5 Pause/Resume playing back all remote users' audio streams
+ *
+ * When you mute the audio of all remote users, the SDK will stop playing back all their audio streams and pulling all their audio data.
+ * @param mute YES: mute; NO: unmute
+ * @note This API works when called either before or after room entry (enterRoom), and the mute status will be reset to `NO` after room exit (exitRoom).
+ */
+- (void)muteAllRemoteAudio:(BOOL)mute;
+
+/**
+ * 5.6 Set audio route
+ *
+ * Setting "audio route" is to determine whether the sound is played back from the speaker or receiver of a mobile device; therefore, this API is only applicable to mobile devices such as phones.
+ * Generally, a phone has two speakers: one is the receiver at the top, and the other is the stereo speaker at the bottom.
+ * If audio route is set to the receiver, the volume is relatively low, and the sound can be heard clearly only when the phone is put near the ear. This mode has a high level of privacy and is suitable for answering calls.
+ * If audio route is set to the speaker, the volume is relatively high, so there is no need to put the phone near the ear. Therefore, this mode can implement the "hands-free" feature.
+ * @param route Audio route, i.e., whether the audio is output by speaker or receiver. Default value: TRTCAudioModeSpeakerphone
+ */
+- (void)setAudioRoute:(TRTCAudioRoute)route;
+
+/**
+ * 5.7 Set the audio playback volume of remote user
+ *
+ * You can mute the audio of a remote user through `setRemoteAudioVolume(userId, 0)`.
+ * @param userId ID of the specified remote user
+ * @param volume Volume. 100 is the original volume. Value range: [0,150]. Default value: 100
+ * @note If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
+ */
+- (void)setRemoteAudioVolume:(NSString *)userId volume:(int)volume;
+
+/**
+ * 5.8 Set the capturing volume of local audio
+ *
+ * @param volume Volume. 100 is the original volume. Value range: [0,150]. Default value: 100
+ * @note If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
+ */
+- (void)setAudioCaptureVolume:(NSInteger)volume;
+
+/**
+ * 5.9 Get the capturing volume of local audio
+ */
+- (NSInteger)getAudioCaptureVolume;
+
+/**
+ * 5.10 Set the playback volume of remote audio
+ *
+ * This API controls the volume of the sound ultimately delivered by the SDK to the system for playback. It affects the volume of the recorded local audio file but not the volume of in-ear monitoring.
+ * @param volume Volume. 100 is the original volume. Value range: [0,150]. Default value: 100
+ * @note If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
+ */
+- (void)setAudioPlayoutVolume:(NSInteger)volume;
+
+/**
+ * 5.11 Get the playback volume of remote audio
+ */
+- (NSInteger)getAudioPlayoutVolume;
+
+/**
+ * 5.12 Enable volume reminder
+ *
+ * After this feature is enabled, the SDK will return the remote audio volume in the {@link onUserVoiceVolume} callback of {@link TRTCCloudDelegate}.
+ * @note To enable this feature, call this API before calling `startLocalAudio`.
+ * @param interval Set the interval in ms for triggering the `onUserVoiceVolume` callback. The minimum interval is 100 ms. If the value is smaller than or equal to 0, the callback will be disabled. We recommend you set this parameter to 300 ms.
+ */
+- (void)enableAudioVolumeEvaluation:(NSUInteger)interval;
+
+/**
+ * 5.13 Start audio recording
+ *
+ * After you call this API, the SDK will selectively record local and remote audio streams (such as local audio, remote audio, background music, and sound effects) into a local file.
+ * This API works when called either before or after room entry. If a recording task has not been stopped through `stopAudioRecording` before room exit, it will be automatically stopped after room exit.
+ * @param param Recording parameter. For more information, please see {@link TRTCAudioRecordingParams}
+ * @return 0: success; -1: audio recording has been started; -2: failed to create file or directory; -3: the audio format of the specified file extension is not supported
+ */
+- (int)startAudioRecording:(TRTCAudioRecordingParams *)param;
+
+/**
+ * 5.14 Stop audio recording
+ *
+ * If a recording task has not been stopped through this API before room exit, it will be automatically stopped after room exit.
+ */
+- (void)stopAudioRecording;
+
+/**
+ * 5.15 Start local media recording
+ *
+ * This API records the audio/video content during live streaming into a local file.
+ * @param params Recording parameter. For more information, please see {@link TRTCLocalRecordingParams}
+ */
+- (void)startLocalRecording:(TRTCLocalRecordingParams *)params;
+
+/**
+ * 5.16 Stop local media recording
+ *
+ * If a recording task has not been stopped through this API before room exit, it will be automatically stopped after room exit.
+ */
+- (void)stopLocalRecording;
+
+/**
+ * 5.18 Set the parallel strategy of remote audio streams
+ *
+ * For room with many speakers.
+ * @param params Audio parallel parameter. For more information, please see {@link TRTCAudioParallelParams}
+ */
+- (void)setRemoteAudioParallelParams:(TRTCAudioParallelParams *)params;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Device management APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Device management APIs
+/// @{
+
+/**
+ * 6.1 Get device management class (TXDeviceManager)
+ */
+- (TXDeviceManager *)getDeviceManager;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Beauty filter and watermark APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Beauty filter and watermark APIs
+/// @{
+
+/**
+ * 7.1 Get beauty filter management class (TXBeautyManager)
+ *
+ * You can use the following features with beauty filter management:
+ * - Set beauty effects such as "skin smoothing", "brightening", and "rosy skin".
+ * - Set face adjustment effects such as "eye enlarging", "face slimming", "chin slimming", "chin lengthening/shortening", "face shortening", "nose narrowing", "eye brightening", "teeth whitening", "eye bag removal", "wrinkle removal", and "smile
+ * line removal".
+ * - Set face adjustment effects such as "hairline", "eye distance", "eye corners", "mouth shape", "nose wing", "nose position", "lip thickness", and "face shape".
+ * - Set makeup effects such as "eye shadow" and "blush".
+ * - Set animated effects such as animated sticker and facial pendant.
+ */
+- (TXBeautyManager *)getBeautyManager;
+
+/**
+ * 7.2 Add watermark
+ *
+ * The watermark position is determined by the `rect` parameter, which is a quadruple in the format of (x, y, width, height).
+ * - x: X coordinate of watermark, which is a floating-point number between 0 and 1.
+ * - y: Y coordinate of watermark, which is a floating-point number between 0 and 1.
+ * - width: width of watermark, which is a floating-point number between 0 and 1.
+ * - height: it does not need to be set. The SDK will automatically calculate it according to the watermark image's aspect ratio.
+ *
+ * Sample parameter:
+ * If the encoding resolution of the current video is 540x960, and the `rect` parameter is set to (0.1, 0.1, 0.2, 0.0),
+ * then the coordinates of the top-left point of the watermark will be (540 * 0.1, 960 * 0.1), i.e., (54, 96), the watermark width will be 540 * 0.2 = 108 px, and the watermark height will be calculated automatically by the SDK based on the watermark
+ * image's aspect ratio.
+ *
+ * @param image Watermark image, **which must be a PNG image with transparent background**
+ * @param streamType Specify for which image to set the watermark. For more information, please see {@link TRTCVideoStreamType}.
+ * @param rect Unified coordinates of the watermark relative to the encoded resolution. Value range of `x`, `y`, `width`, and `height`: 0–1.
+ * @note If you want to set watermarks for both the primary image (generally for the camera) and the substream image (generally for screen sharing), you need to call this API twice with `streamType` set to different values.
+ */
+- (void)setWatermark:(TXImage *)image streamType:(TRTCVideoStreamType)streamType rect:(CGRect)rect;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Background music and sound effect APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Background music and sound effect APIs
+/// @{
+
+/**
+ * 8.1 Get sound effect management class (TXAudioEffectManager)
+ *
+ * `TXAudioEffectManager` is a sound effect management API, through which you can implement the following features:
+ * - Background music: both online music and local music can be played back with various features such as speed adjustment, pitch adjustment, original voice, accompaniment, and loop.
+ * - In-ear monitoring: the sound captured by the mic is played back in the headphones in real time, which is generally used for music live streaming.
+ * - Reverb effect: karaoke room, small room, big hall, deep, resonant, and other effects.
+ * - Voice changing effect: young girl, middle-aged man, heavy metal, and other effects.
+ * - Short sound effect: short sound effect files such as applause and laughter are supported (for files less than 10 seconds in length, please set the `isShortFile` parameter to `YES`).
+ */
+- (TXAudioEffectManager *)getAudioEffectManager;
+
+/**
+ * 8.2 Enable system audio capturing (for desktop systems only)
+ *
+ * This API captures audio data from the sound card of a macOS computer and mixes it into the current audio data stream of the SDK, so that other users in the room can also hear the sound played back on the current macOS system.
+ * In use cases such as video teaching or music live streaming, the teacher can use this feature to let the SDK capture the sound in the video played back by the teacher, so that students in the same room can also hear the sound in the video.
+ * @note
+ * 1. This feature needs to install a virtual audio device plugin on the user's macOS system. After the installation is completed, the SDK will capture sound from the installed virtual device.
+ * 2. The SDK will automatically download the appropriate plugin from the internet for installation, but the download may be slow. If you want to speed up this process, you can package the virtual audio plugin file into the `Resources` directory of
+ * your app bundle.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)startSystemAudioLoopback;
+#endif
+
+/**
+ * 8.3 Stop system audio capturing (for desktop systems only)
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)stopSystemAudioLoopback;
+#endif
+
+/**
+ * 8.4 Set the volume of system audio capturing
+ *
+ * @param volume Set volume. Value range: [0, 150]. Default value: 100
+ */
+- (void)setSystemAudioLoopbackVolume:(uint32_t)volume;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Screen sharing APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Screen sharing APIs
+/// @{
+
+/**
+ * 9.1 Start in-app screen sharing (for iOS 13.0 and above only)
+ *
+ * This API captures the real-time screen content of the current application and shares it with other users in the same room. It is applicable to iOS 13.0 and above.
+ * If you want to capture the screen content of the entire iOS system (instead of the current application), we recommend you use {@link startScreenCaptureByReplaykit}.
+ * Video encoding parameters recommended for screen sharing on iPhone ({@link TRTCVideoEncParam}):
+ * - Resolution (videoResolution): 1280x720
+ * - Frame rate (videoFps): 10 fps
+ * - Bitrate (videoBitrate): 1600 Kbps
+ * - Resolution adaption (enableAdjustRes): NO
+ *
+ * @param streamType Channel used for screen sharing, which can be the primary stream ({@link TRTCVideoStreamTypeBig}) or substream ({@link TRTCVideoStreamTypeSub}).
+ * @param encParams Video encoding parameters for screen sharing. We recommend you use the above configuration.
+ * If you set `encParams` to `nil`, the SDK will use the video encoding parameters you set before calling the `startScreenCapture` API.
+ */
+#if TARGET_OS_IPHONE
+- (void)startScreenCaptureInApp:(TRTCVideoStreamType)streamType encParam:(TRTCVideoEncParam *)encParams API_AVAILABLE(ios(13.0));
+#endif
+
+/**
+ * 9.1 Start system-level screen sharing (for iOS 11.0 and above only)
+ *
+ * This API supports capturing the screen of the entire iOS system, which can implement system-wide screen sharing similar to VooV Meeting.
+ * However, the integration steps are slightly more complicated than those of {@link startScreenCaptureInApp}. You need to implement a ReplayKit extension module for your application.
+ * For more information, please see [iOS](https://cloud.tencent.com/document/product/647/45750)
+ * Video encoding parameters recommended for screen sharing on iPhone ({@link TRTCVideoEncParam}):
+ * - Resolution (videoResolution): 1280x720
+ * - Frame rate (videoFps): 10 fps
+ * - Bitrate (videoBitrate): 1600 Kbps
+ * - Resolution adaption (enableAdjustRes): NO
+ *
+ * @param streamType Channel used for screen sharing, which can be the primary stream ({@link TRTCVideoStreamTypeBig}) or substream ({@link TRTCVideoStreamTypeSub}).
+ * @param encParams Video encoding parameters for screen sharing. We recommend you use the above configuration.
+ * If you set `encParams` to `nil`, the SDK will use the video encoding parameters you set before calling the `startScreenCapture` API.
+ * @param appGroup Specify the `Application Group Identifier` shared by your application and the screen sharing process. You can specify this parameter as `nil`, but we recommend you set it as instructed in the documentation for higher reliability.
+ */
+#if TARGET_OS_IPHONE
+- (void)startScreenCaptureByReplaykit:(TRTCVideoStreamType)streamType encParam:(TRTCVideoEncParam *)encParams appGroup:(NSString *)appGroup API_AVAILABLE(ios(11.0));
+#endif
+
+/**
+ * 9.1 Start desktop screen sharing (for desktop systems only)
+ *
+ * This API can capture the screen content of the entire macOS system or a specified application and share it with other users in the same room.
+ * @param view Parent control of the rendering control, which can be set to a null value, indicating not to display the preview of the shared screen.
+ * @param streamType Channel used for screen sharing, which can be the primary stream ({@link TRTCVideoStreamTypeBig}) or substream ({@link TRTCVideoStreamTypeSub}).
+ * @param encParam Image encoding parameters used for screen sharing, which can be set to `nil`, indicating to let the SDK choose the optimal encoding parameters (such as resolution and bitrate).
+ *
+ * @note
+ * 1. A user can publish at most one primary stream ({@link TRTCVideoStreamTypeBig}) and one substream ({@link TRTCVideoStreamTypeSub}) at the same time.
+ * 2. By default, screen sharing uses the substream image. If you want to use the primary stream for screen sharing, you need to stop camera capturing (through {@link stopLocalPreview}) in advance to avoid conflicts.
+ * 3. Only one user can use the substream for screen sharing in the same room at any time; that is, only one user is allowed to enable the substream in the same room at any time.
+ * 4. When there is already a user in the room using the substream for screen sharing, calling this API will return the `onError(ERR_SERVER_CENTER_ANOTHER_USER_PUSH_SUB_VIDEO)` callback from {@link TRTCCloudDelegate}.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)startScreenCapture:(NSView *)view streamType:(TRTCVideoStreamType)streamType encParam:(TRTCVideoEncParam *)encParam;
+#endif
+
+/**
+ * 9.2 Stop screen sharing
+ */
+- (int)stopScreenCapture API_AVAILABLE(ios(11.0));
+
+/**
+ * 9.3 Pause screen sharing
+ */
+- (int)pauseScreenCapture API_AVAILABLE(ios(11.0));
+
+/**
+ * 9.4 Resume screen sharing
+ */
+- (int)resumeScreenCapture API_AVAILABLE(ios(11.0));
+
+/**
+ * 9.5 Enumerate shareable screens and windows (for macOS only)
+ *
+ * When you integrate the screen sharing feature of a desktop system, you generally need to display a UI for selecting the sharing target, so that users can use the UI to choose whether to share the entire screen or a certain window.
+ * Through this API, you can query the IDs, names, and thumbnails of sharable windows on the current system. We provide a default UI implementation in the demo for your reference.
+ * @note The returned list contains the screen and the application windows. The screen is the first element in the list. If the user has multiple displays, then each display is a sharing target.
+ * @param thumbnailSize Specify the thumbnail size of the window to be obtained. The thumbnail can be drawn on the window selection UI.
+ * @param iconSize Specify the icon size of the window to be obtained.
+ * @return List of windows (including the screen)
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (NSArray *)getScreenCaptureSourcesWithThumbnailSize:(CGSize)thumbnailSize iconSize:(CGSize)iconSize;
+#endif
+
+/**
+ * 9.6 Select the screen or window to share (for macOS only)
+ *
+ * After you get the sharable screen and windows through `getScreenCaptureSources`, you can call this API to select the target screen or window you want to share.
+ * During the screen sharing process, you can also call this API at any time to switch the sharing target.
+ * @param screenSource Specify sharing source
+ * @param rect Specify the area to be captured (set this parameter to `CGRectZero`: when the sharing target is a window, the entire window will be shared, and when the sharing target is the desktop, the entire desktop will be shared)
+ * @param capturesCursor Whether to capture mouse cursor
+ * @param highlight Whether to highlight the window being shared
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)selectScreenCaptureTarget:(TRTCScreenCaptureSourceInfo *)screenSource rect:(CGRect)rect capturesCursor:(BOOL)capturesCursor highlight:(BOOL)highlight;
+#endif
+
+/**
+ * 9.7 Set the video encoding parameters of screen sharing (i.e., substream) (for desktop and mobile systems)
+ *
+ * This API can set the image quality of screen sharing (i.e., the substream) viewed by remote users, which is also the image quality of screen sharing in on-cloud recording files.
+ * Please note the differences between the following two APIs:
+ * - {@link setVideoEncoderParam} is used to set the video encoding parameters of the primary stream image ({@link TRTCVideoStreamTypeBig}, generally for camera).
+ * - {@link setSubStreamEncoderParam} is used to set the video encoding parameters of the substream image ({@link TRTCVideoStreamTypeSub}, generally for screen sharing).
+ *
+ * @param param Substream encoding parameters. For more information, please see {@link TRTCVideoEncParam}.
+ * @note Even if you use the primary stream to transfer screen sharing data (set `type=TRTCVideoStreamTypeBig` when calling `startScreenCapture`), you still need to call the {@link setSubStreamEncoderParam} API instead of the {@link
+ * setVideoEncoderParam} API to set the screen sharing encoding parameters.
+ */
+- (void)setSubStreamEncoderParam:(TRTCVideoEncParam *)param;
+
+/**
+ * 9.8 Set the audio mixing volume of screen sharing (for desktop systems only)
+ *
+ * The greater the value, the larger the ratio of the screen sharing volume to the mic volume. We recommend you not set a high value for this parameter as a high volume will cover the mic sound.
+ * @param volume Set audio mixing volume. Value range: 0–100
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)setSubStreamMixVolume:(NSInteger)volume;
+#endif
+
+/**
+ * 9.9 Add specified windows to the exclusion list of screen sharing (for desktop systems only)
+ *
+ * The excluded windows will not be shared. This feature is generally used to add a certain application's window to the exclusion list to avoid privacy issues.
+ * You can set the filtered windows before starting screen sharing or dynamically add the filtered windows during screen sharing.
+ * @param window Window not to be shared
+ * @note
+ * 1. This API takes effect only if the `type` in {@link TRTCScreenCaptureSourceInfo} is specified as {@link TRTCScreenCaptureSourceTypeScreen}; that is, the feature of excluding specified windows works only when the entire screen is shared.
+ * 2. The windows added to the exclusion list through this API will be automatically cleared by the SDK after room exit.
+ * 3. On macOS, please pass in the window ID (CGWindowID), which can be obtained through the `sourceId` member in {@link TRTCScreenCaptureSourceInfo}.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)addExcludedShareWindow:(NSInteger)windowID;
+#endif
+
+/**
+ * 9.10 Remove specified windows from the exclusion list of screen sharing (for desktop systems only)
+ *
+ * @param windowID
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)removeExcludedShareWindow:(NSInteger)windowID;
+#endif
+
+/**
+ * 9.11 Remove all windows from the exclusion list of screen sharing (for desktop systems only)
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)removeAllExcludedShareWindows;
+#endif
+
+/**
+ * 9.12 Add specified windows to the inclusion list of screen sharing (for desktop systems only)
+ *
+ * This API takes effect only if the `type` in {@link TRTCScreenCaptureSourceInfo} is specified as {@link TRTCScreenCaptureSourceTypeWindow}; that is, the feature of additionally including specified windows works only when a window is shared.
+ * You can call it before or after {@link startScreenCapture}.
+ * @param windowID Window to be shared (which is a window handle `HWND` on Windows)
+ * @note The windows added to the inclusion list by this method will be automatically cleared by the SDK after room exit.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)addIncludedShareWindow:(NSInteger)windowID;
+#endif
+
+/**
+ * 9.13 Remove specified windows from the inclusion list of screen sharing (for desktop systems only)
+ *
+ * This API takes effect only if the `type` in {@link TRTCScreenCaptureSourceInfo} is specified as {@link TRTCScreenCaptureSourceTypeWindow}.
+ * That is, the feature of additionally including specified windows works only when a window is shared.
+ * @param windowID Window to be shared (window ID on macOS or HWND on Windows)
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)removeIncludedShareWindow:(NSInteger)windowID;
+#endif
+
+/**
+ * 9.14 Remove all windows from the inclusion list of screen sharing (for desktop systems only)
+ *
+ * This API takes effect only if the `type` in {@link TRTCScreenCaptureSourceInfo} is specified as {@link TRTCScreenCaptureSourceTypeWindow}.
+ * That is, the feature of additionally including specified windows works only when a window is shared.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)removeAllIncludedShareWindows;
+#endif
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Custom capturing and rendering APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Custom capturing and rendering APIs
+/// @{
+
+/**
+ * 10.1 Enable/Disable custom video capturing mode
+ *
+ * After this mode is enabled, the SDK will not run the original video capturing process (i.e., stopping camera data capturing and beauty filter operations) and will retain only the video encoding and sending capabilities.
+ * You need to use {@link sendCustomVideoData} to continuously insert the captured video image into the SDK.
+ * @param streamType Specify video stream type ({@link TRTCVideoStreamTypeBig}: HD big image; {@link TRTCVideoStreamTypeSub}: substream image).
+ * @param enable Whether to enable. Default value: NO
+ */
+- (void)enableCustomVideoCapture:(TRTCVideoStreamType)streamType enable:(BOOL)enable;
+
+/**
+ * 10.2 Deliver captured video frames to SDK
+ *
+ * You can use this API to deliver video frames you capture to the SDK, and the SDK will encode and transfer them through its own network module.
+ * We recommend you enter the following information for the {@link TRTCVideoFrame} parameter (other fields can be left empty):
+ * - pixelFormat: {@link TRTCVideoPixelFormat_NV12} is recommended.
+ * - bufferType: {@link TRTCVideoBufferType_PixelBuffer} is recommended.
+ * - pixelBuffer: common video data format on iOS/macOS.
+ * - data: raw video data format, which is used if `bufferType` is `NSData`.
+ * - timestamp (ms): Set it to the timestamp when video frames are captured, which you can obtain by calling {@link generateCustomPTS} after getting a video frame.
+ * - width: video image length, which needs to be set if `bufferType` is `NSData`.
+ * - height: video image width, which needs to be set if `bufferType` is `NSData`.
+ *
+ * For more information, please see [Custom Capturing and Rendering](https://cloud.tencent.com/document/product/647/34066).
+ * @param streamType Specify video stream type ({@link TRTCVideoStreamTypeBig}: HD big image; {@link TRTCVideoStreamTypeSub}: substream image).
+ * @param frame Video data, which can be in PixelBuffer NV12, BGRA, or I420 format.
+ * @note
+ * 1. We recommend you call the {@link generateCustomPTS} API to get the `timestamp` value of a video frame immediately after capturing it, so as to achieve the best audio/video sync effect.
+ * 2. The video frame rate eventually encoded by the SDK is not determined by the frequency at which you call this API, but by the FPS you set in {@link setVideoEncoderParam}.
+ * 3. Please try to keep the calling interval of this API even; otherwise, problems will occur, such as unstable output frame rate of the encoder or out-of-sync audio/video.
+ */
+- (void)sendCustomVideoData:(TRTCVideoStreamType)streamType frame:(TRTCVideoFrame *)frame;
+
+/**
+ * 10.3 Enable custom audio capturing mode
+ *
+ * After this mode is enabled, the SDK will not run the original audio capturing process (i.e., stopping mic data capturing) and will retain only the audio encoding and sending capabilities.
+ * You need to use {@link sendCustomAudioData} to continuously insert the captured audio data into the SDK.
+ * @param enable Whether to enable. Default value: NO
+ * @note As acoustic echo cancellation (AEC) requires strict control over the audio capturing and playback time, after custom audio capturing is enabled, AEC may fail.
+ */
+- (void)enableCustomAudioCapture:(BOOL)enable;
+
+/**
+ * 10.4 Deliver captured audio data to SDK
+ *
+ * We recommend you enter the following information for the {@link TRTCAudioFrame} parameter (other fields can be left empty):
+ * - audioFormat: audio data format, which can only be `TRTCAudioFrameFormatPCM`.
+ * - data: audio frame buffer. Audio frame data must be in PCM format, and it supports a frame length of 5–100 ms (20 ms is recommended). Length calculation method: **for example, if the sample rate is 48000, then the frame length for mono channel
+ * will be `48000 * 0.02s * 1 * 16 bit = 15360 bit = 1920 bytes`.**
+ * - sampleRate: sample rate. Valid values: 16000, 24000, 32000, 44100, 48000.
+ * - channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel.
+ * - timestamp (ms): Set it to the timestamp when audio frames are captured, which you can obtain by calling {@link generateCustomPTS} after getting a audio frame.
+ *
+ * For more information, please see [Custom Capturing and Rendering](https://cloud.tencent.com/document/product/647/34066).
+ * @param frame Audio data
+ * @note Please call this API accurately at intervals of the frame length; otherwise, sound lag may occur due to uneven data delivery intervals.
+ */
+- (void)sendCustomAudioData:(TRTCAudioFrame *)frame;
+
+/**
+ * 10.5 Enable/Disable custom audio track
+ *
+ * After this feature is enabled, you can mix a custom audio track into the SDK through this API. With two boolean parameters, you can control whether to play back this track remotely or locally.
+ * @param enablePublish Whether the mixed audio track should be played back remotely. Default value: NO
+ * @param enablePlayout Whether the mixed audio track should be played back locally. Default value: NO
+ * @note If you specify both `enablePublish` and `enablePlayout` as `NO`, the custom audio track will be completely closed.
+ */
+- (void)enableMixExternalAudioFrame:(BOOL)enablePublish playout:(BOOL)enablePlayout;
+
+/**
+ * 10.6 Mix custom audio track into SDK
+ *
+ * Before you use this API to mix custom PCM audio into the SDK, you need to first enable custom audio tracks through {@link enableMixExternalAudioFrame}.
+ * You are expected to feed audio data into the SDK at an even pace, but we understand that it can be challenging to call an API at absolutely regular intervals.
+ * Given this, we have provided a buffer pool in the SDK, which can cache the audio data you pass in to reduce the fluctuations in intervals between API calls.
+ * The value returned by this API indicates the size (ms) of the buffer pool. For example, if `50` is returned, it indicates that the buffer pool has 50 ms of audio data. As long as you call this API again within 50 ms, the SDK can make sure that
+ * continuous audio data is mixed. If the value returned is `100` or greater, you can wait after an audio frame is played to call the API again. If the value returned is smaller than `100`, then there isn’t enough data in the buffer pool, and you
+ * should feed more audio data into the SDK until the data in the buffer pool is above the safety level. Fill the fields in {@link TRTCAudioFrame} as follows (other fields are not required).
+ * - `data`: audio frame buffer. Audio frames must be in PCM format. Each frame can be 5-100 ms (20 ms is recommended) in duration. Assume that the sample rate is 48000, and sound channels mono-channel. Then the **frame size would be 48000 x 0.02s x
+ * 1 x 16 bit = 15360 bit = 1920 bytes**.
+ * - `sampleRate`: sample rate. Valid values: 16000, 24000, 32000, 44100, 48000
+ * - `channel`: number of sound channels (if dual-channel is used, data is interleaved). Valid values: `1` (mono-channel); `2` (dual channel)
+ * - `timestamp`: timestamp (ms). Set it to the timestamp when audio frames are captured, which you can obtain by calling {@link generateCustomPTS} after getting an audio frame.
+ *
+ * @param frame Audio data
+ * @return If the value returned is `0` or greater, the value represents the current size of the buffer pool; if the value returned is smaller than `0`, it means that an error occurred. `-1` indicates that you didn’t call {@link
+ * enableMixExternalAudioFrame} to enable custom audio tracks.
+ */
+- (int)mixExternalAudioFrame:(TRTCAudioFrame *)frame;
+
+/**
+ * 10.7 Set the publish volume and playback volume of mixed custom audio track
+ *
+ * @param publishVolume set the publish volume，from 0 to 100, -1 means no change
+ * @param playoutVolume set the play volume，from 0 to 100, -1 means no change
+ */
+- (void)setMixExternalAudioVolume:(NSInteger)publishVolume playoutVolume:(NSInteger)playoutVolume;
+
+/**
+ * 10.8 Generate custom capturing timestamp
+ *
+ * This API is only suitable for the custom capturing mode and is used to solve the problem of out-of-sync audio/video caused by the inconsistency between the capturing time and delivery time of audio/video frames.
+ * When you call APIs such as {@link sendCustomVideoData} or {@link sendCustomAudioData} for custom video or audio capturing, please use this API as instructed below:
+ * 1. First, when a video or audio frame is captured, call this API to get the corresponding PTS timestamp.
+ * 2. Then, send the video or audio frame to the preprocessing module you use (such as a third-party beauty filter or sound effect component).
+ * 3. When you actually call {@link sendCustomVideoData} or {@link sendCustomAudioData} for delivery, assign the PTS timestamp recorded when the frame was captured to the `timestamp` field in {@link TRTCVideoFrame} or {@link TRTCAudioFrame}.
+ *
+ * @return Timestamp in ms
+ */
++ (uint64_t)generateCustomPTS;
+
+/**
+ * 10.9 Set video data callback for third-party beauty filters
+ *
+ * After this callback is set, the SDK will call back the captured video frames through the `delegate` you set and use them for further processing by a third-party beauty filter component. Then, the SDK will encode and send the processed video
+ * frames.
+ * @param delegate Custom preprocessing callback. For more information, please see {@link TRTCVideoFrameDelegate}
+ * @param pixelFormat Specify the format of the pixel called back. Currently, only {@link TRTCVideoPixelFormat_Texture_2D} is supported
+ * @param bufferType Specify the format of the data called back. Currently, only {@link TRTCVideoBufferType_Texture} is supported
+ * @return 0: success; values smaller than 0: error
+ */
+- (int)setLocalVideoProcessDelegete:(id)delegate pixelFormat:(TRTCVideoPixelFormat)pixelFormat bufferType:(TRTCVideoBufferType)bufferType;
+
+/**
+ * 10.10 Set the callback of custom rendering for local video
+ *
+ * After this callback is set, the SDK will skip its own rendering process and call back the captured data. Therefore, you need to complete image rendering on your own.
+ * - `pixelFormat` specifies the format of the called back data, such as NV12, I420, and 32BGRA.
+ * - `bufferType` specifies the buffer type. `PixelBuffer` has the highest efficiency, while `NSData` makes the SDK perform a memory conversion internally, which will result in extra performance loss.
+ *
+ * For more information, please see [Custom Capturing and Rendering](https://cloud.tencent.com/document/product/647/34066).
+ * @param delegate Callback for custom rendering
+ * @param pixelFormat Specify the format of the pixel called back
+ * @param bufferType PixelBuffer: this can be directly converted to `UIImage` by using `imageWithCVImageBuffer`; NSData: this is memory-mapped video data.
+ * @return 0: success; values smaller than 0: error
+ */
+- (int)setLocalVideoRenderDelegate:(id)delegate pixelFormat:(TRTCVideoPixelFormat)pixelFormat bufferType:(TRTCVideoBufferType)bufferType;
+
+/**
+ * 10.11 Set the callback of custom rendering for remote video
+ *
+ * After this callback is set, the SDK will skip its own rendering process and call back the captured data. Therefore, you need to complete image rendering on your own.
+ * - `pixelFormat` specifies the format of the called back data, such as NV12, I420, and 32BGRA.
+ * - `bufferType` specifies the buffer type. `PixelBuffer` has the highest efficiency, while `NSData` makes the SDK perform a memory conversion internally, which will result in extra performance loss.
+ *
+ * For more information, please see [Custom Capturing and Rendering](https://cloud.tencent.com/document/product/647/34066).
+ * @note Before this API is called, `startRemoteView(nil)` needs to be called to get the video stream of the remote user (`view` can be set to `nil` for this end); otherwise, there will be no data called back.
+ * @param userId ID of the specified remote user
+ * @param delegate Callback for custom rendering
+ * @param pixelFormat Specify the format of the pixel called back
+ * @param bufferType PixelBuffer: this can be directly converted to `UIImage` by using `imageWithCVImageBuffer`; NSData: this is memory-mapped video data.
+ * @return 0: success; values smaller than 0: error
+ */
+- (int)setRemoteVideoRenderDelegate:(NSString *)userId delegate:(id)delegate pixelFormat:(TRTCVideoPixelFormat)pixelFormat bufferType:(TRTCVideoBufferType)bufferType;
+
+/**
+ * 10.12 Set custom audio data callback
+ *
+ * After this callback is set, the SDK will internally call back the audio data (in PCM format), including:
+ * - {@link onCapturedRawAudioFrame}: callback of the original audio data captured by the local mic
+ * - {@link onLocalProcessedAudioFrame}: callback of the audio data captured by the local mic and preprocessed by the audio module
+ * - {@link onRemoteUserAudioFrame}: audio data from each remote user before audio mixing
+ * - {@link onMixedPlayAudioFrame}: callback of the audio data that will be played back by the system after audio streams are mixed
+ *
+ * @note Setting the callback to null indicates to stop the custom audio callback, while setting it to a non-null value indicates to start the custom audio callback.
+ */
+- (void)setAudioFrameDelegate:(id)delegate;
+
+/**
+ * 10.13 Set the callback format of original audio frames captured by local mic
+ *
+ * This API is used to set the `AudioFrame` format called back by {@link onCapturedRawAudioFrame}:
+ * - sampleRate: sample rate. Valid values: 16000, 32000, 44100, 48000
+ * - channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel
+ * - samplesPerCall: number of sample points, which defines the frame length of the callback data. The frame length must be an integer multiple of 10 ms.
+ *
+ * If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows: number of sample points = number of milliseconds * sample rate / 1000
+ * For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as 960 = 20 * 48000 / 1000
+ * Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows: number of bytes = number of sample points * number of channels * 2 (bit
+ * width) For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is 3840 = 960 * 2 * 2
+ * @param format Audio data callback format
+ * @return 0: success; values smaller than 0: error
+ */
+- (int)setCapturedRawAudioFrameDelegateFormat:(TRTCAudioFrameDelegateFormat *)format;
+
+/**
+ * 10.14 Set the callback format of preprocessed local audio frames
+ *
+ * This API is used to set the `AudioFrame` format called back by {@link onLocalProcessedAudioFrame}:
+ * - sampleRate: sample rate. Valid values: 16000, 32000, 44100, 48000
+ * - channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel
+ * - samplesPerCall: number of sample points, which defines the frame length of the callback data. The frame length must be an integer multiple of 10 ms.
+ *
+ * If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows: number of sample points = number of milliseconds * sample rate / 1000
+ * For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as 960 = 20 * 48000 / 1000
+ * Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows: number of bytes = number of sample points * number of channels * 2 (bit
+ * width) For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is 3840 = 960 * 2 * 2
+ * @param format Audio data callback format
+ * @return 0: success; values smaller than 0: error
+ */
+- (int)setLocalProcessedAudioFrameDelegateFormat:(TRTCAudioFrameDelegateFormat *)format;
+
+/**
+ * 10.15 Set the callback format of audio frames to be played back by system
+ *
+ * This API is used to set the `AudioFrame` format called back by {@link onMixedPlayAudioFrame}:
+ * - sampleRate: sample rate. Valid values: 16000, 32000, 44100, 48000
+ * - channel: number of channels (if stereo is used, data is interwoven). Valid values: 1: mono channel; 2: dual channel
+ * - samplesPerCall: number of sample points, which defines the frame length of the callback data. The frame length must be an integer multiple of 10 ms.
+ *
+ * If you want to calculate the callback frame length in milliseconds, the formula for converting the number of milliseconds into the number of sample points is as follows: number of sample points = number of milliseconds * sample rate / 1000
+ * For example, if you want to call back the data of 20 ms frame length with 48000 sample rate, then the number of sample points should be entered as 960 = 20 * 48000 / 1000
+ * Note that the frame length of the final callback is in bytes, and the calculation formula for converting the number of sample points into the number of bytes is as follows: number of bytes = number of sample points * number of channels * 2 (bit
+ * width) For example, if the parameters are 48000 sample rate, dual channel, 20 ms frame length, and 960 sample points, then the number of bytes is 3840 = 960 * 2 * 2
+ * @param format Audio data callback format
+ * @return 0: success; values smaller than 0: error
+ */
+- (int)setMixedPlayAudioFrameDelegateFormat:(TRTCAudioFrameDelegateFormat *)format;
+
+/**
+ * 10.16 Enabling custom audio playback
+ *
+ * You can use this API to enable custom audio playback if you want to connect to an external audio device or control the audio playback logic by yourself.
+ * After you enable custom audio playback, the SDK will stop using its audio API to play back audio. You need to call {@link getCustomAudioRenderingFrame} to get audio frames and play them by yourself.
+ * @param enable Whether to enable custom audio playback. It’s disabled by default.
+ * @note The parameter must be set before room entry to take effect.
+ */
+- (void)enableCustomAudioRendering:(BOOL)enable;
+
+/**
+ * 10.17 Getting playable audio data
+ *
+ * Before calling this API, you need to first enable custom audio playback using {@link enableCustomAudioRendering}.
+ * Fill the fields in {@link TRTCAudioFrame} as follows (other fields are not required):
+ * - `sampleRate`: sample rate (required). Valid values: 16000, 24000, 32000, 44100, 48000
+ * - `channel`: number of sound channels (required). `1`: mono-channel; `2`: dual-channel; if dual-channel is used, data is interleaved.
+ * - `data`: the buffer used to get audio data. You need to allocate memory for the buffer based on the duration of an audio frame.
+ * The PCM data obtained can have a frame duration of 10 ms or 20 ms. 20 ms is recommended.
+ * Assume that the sample rate is 48000, and sound channels mono-channel. The buffer size for a 20 ms audio frame would be 48000 x 0.02s x 1 x 16 bit = 15360 bit = 1920 bytes.
+ *
+ * @param audioFrame Audio frames
+ * @note
+ * 1. You must set `sampleRate` and `channel` in `audioFrame`, and allocate memory for one frame of audio in advance.
+ * 2. The SDK will fill the data automatically based on `sampleRate` and `channel`.
+ * 3. We recommend that you use the system’s audio playback thread to drive the calling of this API, so that it is called each time the playback of an audio frame is complete.
+ *
+ */
+- (void)getCustomAudioRenderingFrame:(TRTCAudioFrame *)audioFrame;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Custom message sending APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Custom message sending APIs
+/// @{
+
+/**
+ * 11.1 Use UDP channel to send custom message to all users in room
+ *
+ * This API allows you to use TRTC's UDP channel to broadcast custom data to other users in the current room for signaling transfer.
+ * The UDP channel in TRTC was originally designed to transfer audio/video data. This API works by disguising the signaling data you want to send as audio/video data packets and sending them together with the audio/video data to be sent.
+ * Other users in the room can receive the message through the `onRecvCustomCmdMsg` callback in {@link TRTCCloudDelegate}.
+ * @param cmdID Message ID. Value range: 1–10
+ * @param data Message to be sent. The maximum length of one single message is 1 KB.
+ * @param reliable Whether reliable sending is enabled. Reliable sending can achieve a higher success rate but with a longer reception delay than unreliable sending.
+ * @param ordered Whether orderly sending is enabled, i.e., whether the data packets should be received in the same order in which they are sent; if so, a certain delay will be caused.
+ * @return YES: sent the message successfully; NO: failed to send the message.
+ * @note
+ * 1. Up to 30 messages can be sent per second to all users in the room (this is not supported for web and mini program currently).
+ * 2. A packet can contain up to 1 KB of data; if the threshold is exceeded, the packet is very likely to be discarded by the intermediate router or server.
+ * 3. A client can send up to 8 KB of data in total per second.
+ * 4. `reliable` and `ordered` must be set to the same value (`YES` or `NO`) and cannot be set to different values currently.
+ * 5. We strongly recommend you set different `cmdID` values for messages of different types. This can reduce message delay when orderly sending is required.
+ */
+- (BOOL)sendCustomCmdMsg:(NSInteger)cmdID data:(NSData *)data reliable:(BOOL)reliable ordered:(BOOL)ordered;
+
+/**
+ * 11.2 Use SEI channel to send custom message to all users in room
+ *
+ * This API allows you to use TRTC's SEI channel to broadcast custom data to other users in the current room for signaling transfer.
+ * The header of a video frame has a header data block called SEI. This API works by embedding the custom signaling data you want to send in the SEI block and sending it together with the video frame.
+ * Therefore, the SEI channel has a better compatibility than {@link sendCustomCmdMsg} as the signaling data can be transferred to the CSS CDN along with the video frame.
+ * However, because the data block of the video frame header cannot be too large, we recommend you limit the size of the signaling data to only a few bytes when using this API.
+ * The most common use is to embed the custom timestamp into video frames through this API so as to implement a perfect alignment between the message and video image (such as between the teaching material and video signal in the education scenario).
+ * Other users in the room can receive the message through the `onRecvSEIMsg` callback in {@link TRTCCloudDelegate}.
+ * @param data Data to be sent, which can be up to 1 KB (1,000 bytes)
+ * @param repeatCount Data sending count
+ * @return YES: the message is allowed and will be sent with subsequent video frames; NO: the message is not allowed to be sent
+ * @note This API has the following restrictions:
+ * 1. The data will not be instantly sent after this API is called; instead, it will be inserted into the next video frame after the API call.
+ * 2. Up to 30 messages can be sent per second to all users in the room (this limit is shared with `sendCustomCmdMsg`).
+ * 3. Each packet can be up to 1 KB (this limit is shared with `sendCustomCmdMsg`). If a large amount of data is sent, the video bitrate will increase, which may reduce the video quality or even cause lagging.
+ * 4. Each client can send up to 8 KB of data in total per second (this limit is shared with `sendCustomCmdMsg`).
+ * 5. If multiple times of sending is required (i.e., `repeatCount` > 1), the data will be inserted into subsequent `repeatCount` video frames in a row for sending, which will increase the video bitrate.
+ * 6. If `repeatCount` is greater than 1, the data will be sent for multiple times, and the same message may be received multiple times in the `onRecvSEIMsg` callback; therefore, deduplication is required.
+ */
+- (BOOL)sendSEIMsg:(NSData *)data repeatCount:(int)repeatCount;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Network test APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Network test APIs
+/// @{
+
+/**
+ * 12.1 Start network speed test (used before room entry)
+ *
+ * @param params speed test options
+ * @return interface call result, <0: failure
+ * @note
+ * 1. The speed measurement process will incur a small amount of basic service fees, See [Purchase Guide > Base Services](https://intl.cloud.tencent.com/document/product/647/34610?lang=en&pg=#basic-services).
+ * 2. Please perform the Network speed test before room entry, because if performed after room entry, the test will affect the normal audio/video transfer, and its result will be inaccurate due to interference in the room.
+ * 3. Only one network speed test task is allowed to run at the same time.
+ */
+- (int)startSpeedTest:(TRTCSpeedTestParams *)params;
+
+/**
+ * 12.2 Stop network speed test
+ */
+- (void)stopSpeedTest;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Debugging APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Debugging APIs
+/// @{
+
+/**
+ * 13.1 Get SDK version information
+ */
++ (NSString *)getSDKVersion;
+
+/**
+ * 13.2 Set log output level
+ *
+ * @param level For more information, please see {@link TRTCLogLevel}. Default value: {@link TRTCLogLevelNone}
+ */
++ (void)setLogLevel:(TRTCLogLevel)level;
+
+/**
+ * 13.3 Enable/Disable console log printing
+ *
+ * @param enabled Specify whether to enable it, which is disabled by default
+ */
++ (void)setConsoleEnabled:(BOOL)enabled;
+
+/**
+ * 13.4 Enable/Disable local log compression
+ *
+ * If compression is enabled, the log size will significantly reduce, but logs can be read only after being decompressed by the Python script provided by Tencent Cloud.
+ * If compression is disabled, logs will be stored in plaintext and can be read directly in Notepad, but will take up more storage capacity.
+ * @param enabled Specify whether to enable it, which is enabled by default
+ */
++ (void)setLogCompressEnabled:(BOOL)enabled;
+
+/**
+ * 13.5 Set local log storage path
+ *
+ * You can use this API to change the default storage path of the SDK's local logs, which is as follows:
+ * - Windows: C:/Users/[username]/AppData/Roaming/liteav/log, i.e., under `%appdata%/liteav/log`.
+ * - iOS or macOS: under `sandbox Documents/log`.
+ * - Android: under `/app directory/files/log/liteav/`.
+ * @note Please be sure to call this API before all other APIs and make sure that the directory you specify exists and your application has read/write permissions of the directory.
+ * @param path Log storage path
+ */
++ (void)setLogDirPath:(NSString *)path;
+
+/**
+ * 13.6 Set log callback
+ */
++ (void)setLogDelegate:(id)logDelegate;
+
+/**
+ * 13.7 Display dashboard
+ *
+ * "Dashboard" is a semi-transparent floating layer for debugging information on top of the video rendering control. It is used to display audio/video information and event information to facilitate integration and debugging.
+ * @param showType 0: does not display; 1: displays lite edition (only with audio/video information); 2: displays full edition (with audio/video information and event information).
+ */
+- (void)showDebugView:(NSInteger)showType;
+
+/**
+ * 13.8 Set dashboard margin
+ *
+ * This API is used to adjust the position of the dashboard in the video rendering control. It must be called before `showDebugView` for it to take effect.
+ * @param userId User ID
+ * @param margin Inner margin of the dashboard. It should be noted that this is based on the percentage of `parentView`. Value range: 0–1
+ */
+- (void)setDebugViewMargin:(NSString *)userId margin:(TXEdgeInsets)margin;
+
+/**
+ * 13.9 Call experimental APIs
+ */
+- (void)callExperimentalAPI:(NSString *)jsonStr;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Disused APIs (the corresponding new APIs are recommended)
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Disused APIs (the corresponding new APIs are recommended)
+/// @{
+
+/**
+ * Set mic volume
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link setAudioCaptureVolume} instead.
+ */
+- (void)setMicVolumeOnMixing:(NSInteger)volume __attribute__((deprecated("use setAudioCaptureVolume instead")));
+
+/**
+ * Set the strength of beauty, brightening, and rosy skin filters
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link getBeautyManager} instead.
+ */
+- (void)setBeautyStyle:(TRTCBeautyStyle)beautyStyle beautyLevel:(NSInteger)beautyLevel whitenessLevel:(NSInteger)whitenessLevel ruddinessLevel:(NSInteger)ruddinessLevel __attribute__((deprecated("use getBeautyManager instead")));
+
+/**
+ * Set the strength of eye enlarging filter
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link getBeautyManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setEyeScaleLevel:(float)eyeScaleLevel __attribute__((deprecated("use getBeautyManager instead")));
+#endif
+
+/**
+ * Set the strength of face slimming filter
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link getBeautyManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setFaceScaleLevel:(float)faceScaleLevel __attribute__((deprecated("use getBeautyManager instead")));
+#endif
+
+/**
+ * Set the strength of chin slimming filter
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link getBeautyManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setFaceVLevel:(float)faceVLevel __attribute__((deprecated("use getBeautyManager instead")));
+#endif
+
+/**
+ * Set the strength of chin lengthening/shortening filter
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link getBeautyManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setChinLevel:(float)chinLevel __attribute__((deprecated("use getBeautyManager instead")));
+#endif
+
+/**
+ * Set the strength of face shortening filter
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link getBeautyManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setFaceShortLevel:(float)faceShortlevel __attribute__((deprecated("use getBeautyManager instead")));
+#endif
+
+/**
+ * Set the strength of nose slimming filter
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link getBeautyManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setNoseSlimLevel:(float)noseSlimLevel __attribute__((deprecated("use getBeautyManager instead")));
+#endif
+
+/**
+ * Set animated sticker
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link getBeautyManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)selectMotionTmpl:(NSString *)tmplPath __attribute__((deprecated("use getBeautyManager instead")));
+#endif
+
+/**
+ * Mute animated sticker
+ *
+ * @deprecated This API is not recommended after v6.9. Please use {@link getBeautyManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setMotionMute:(BOOL)motionMute __attribute__((deprecated("use getBeautyManager instead")));
+#endif
+
+/**
+ * Start screen sharing
+ *
+ * @deprecated This API is not recommended after v7.2. Please use `startScreenCapture:streamType:encParam:` instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)startScreenCapture:(NSView *)view __attribute__((deprecated("use startScreenCapture:streamType:encParam: instead")));
+#endif
+
+/**
+ * Set color filter
+ *
+ * @deprecated This API is not recommended after v7.2. Please use {@link getBeautyManager} instead.
+ */
+- (void)setFilter:(TXImage *)image __attribute__((deprecated("use getBeautyManager instead")));
+
+/**
+ * Set the strength of color filter
+ *
+ * @deprecated This API is not recommended after v7.2. Please use {@link getBeautyManager} instead.
+ */
+- (void)setFilterConcentration:(float)concentration __attribute__((deprecated("use getBeautyManager instead")));
+
+/**
+ * Set green screen video
+ *
+ * @deprecated This API is not recommended after v7.2. Please use {@link getBeautyManager} instead.
+ */
+- (void)setGreenScreenFile:(NSURL *)file __attribute__((deprecated("use getBeautyManager instead")));
+
+/**
+ * Start background music
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link getAudioEffectManager} instead.
+ */
+- (void)playBGM:(NSString *)path
+ withBeginNotify:(void (^)(NSInteger errCode))beginNotify
+ withProgressNotify:(void (^)(NSInteger progressMS, NSInteger durationMS))progressNotify
+ andCompleteNotify:(void (^)(NSInteger errCode))completeNotify __attribute__((deprecated("use getAudioEffectManager instead")));
+
+/**
+ * Stop background music
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link getAudioEffectManager} instead.
+ */
+- (void)stopBGM __attribute__((deprecated("use getAudioEffectManager instead")));
+
+/**
+ * Stop background music
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link getAudioEffectManager} instead.
+ */
+- (void)pauseBGM __attribute__((deprecated("use getAudioEffectManager instead")));
+
+/**
+ * Stop background music
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link getAudioEffectManager} instead.
+ */
+- (void)resumeBGM __attribute__((deprecated("use getAudioEffectManager instead")));
+
+/**
+ * Get the total length of background music in ms
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#getMusicDurationInMS} instead.
+ */
+- (NSInteger)getBGMDuration:(NSString *)path __attribute__((deprecated("use TXAudioEffectManager#getMusicDurationInMS instead")));
+
+/**
+ * Set background music playback progress
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#seekMusicToPosInMS} instead.
+ */
+- (int)setBGMPosition:(NSInteger)pos __attribute__((deprecated("use TXAudioEffectManager#seekMusicToPosInMS instead")));
+
+/**
+ * Set background music volume
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#setMusicVolume} instead.
+ */
+- (void)setBGMVolume:(NSInteger)volume __attribute__((deprecated("use TXAudioEffectManager#setMusicVolume instead")));
+
+/**
+ * Set the local playback volume of background music
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#setMusicPlayoutVolume} instead.
+ */
+- (void)setBGMPlayoutVolume:(NSInteger)volume __attribute__((deprecated("use TXAudioEffectManager#setMusicPlayoutVolume instead")));
+
+/**
+ * Set the remote playback volume of background music
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#setBGMPublishVolume} instead.
+ */
+- (void)setBGMPublishVolume:(NSInteger)volume __attribute__((deprecated("use TXAudioEffectManager#setBGMPublishVolume instead")));
+
+/**
+ * Set reverb effect
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#setVoiceReverbType} instead.
+ */
+- (void)setReverbType:(TRTCReverbType)reverbType __attribute__((deprecated("use TXAudioEffectManager#setVoiceReverbType instead")));
+
+/**
+ * Set voice changing type
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#setVoiceChangerType} instead.
+ */
+- (void)setVoiceChangerType:(TRTCVoiceChangerType)voiceChangerType __attribute__((deprecated("use TXAudioEffectManager#setVoiceChangerType instead")));
+
+/**
+ * Play sound effect
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#startPlayMusic} instead.
+ */
+- (void)playAudioEffect:(TRTCAudioEffectParam *)effect __attribute__((deprecated("use TXAudioEffectManager#startPlayMusic instead")));
+
+/**
+ * Set sound effect volume
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#setMusicPublishVolume} and {@link TXAudioEffectManager#setMusicPlayoutVolume} instead.
+ */
+- (void)setAudioEffectVolume:(int)effectId volume:(int)volume __attribute__((deprecated("use setMusicPublishVolume/setMusicPlayoutVolume instead")));
+
+/**
+ * Stop sound effect
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#stopPlayMusic} instead.
+ */
+- (void)stopAudioEffect:(int)effectId __attribute__((deprecated("use TXAudioEffectManager#stopPlayMusic instead")));
+
+/**
+ * Stop all sound effects
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#stopPlayMusic} instead.
+ */
+- (void)stopAllAudioEffects __attribute__((deprecated("use TXAudioEffectManager#stopPlayMusic instead")));
+
+/**
+ * Set the volume of all sound effects
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#setMusicPublishVolume} and {@link TXAudioEffectManager#setMusicPlayoutVolume} instead.
+ */
+- (void)setAllAudioEffectsVolume:(int)volume __attribute__((deprecated("use setMusicPublishVolume/setMusicPlayoutVolume instead")));
+
+/**
+ * Pause sound effect
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#pauseAudioEffect} instead.
+ */
+- (void)pauseAudioEffect:(int)effectId __attribute__((deprecated("use TXAudioEffectManager#pauseAudioEffect instead")));
+
+/**
+ * Pause sound effect
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#resumePlayMusic} instead.
+ */
+- (void)resumeAudioEffect:(int)effectId __attribute__((deprecated("use TXAudioEffectManager#resumePlayMusic instead")));
+
+/**
+ * Enable or disable in-ear monitoring
+ *
+ * @deprecated This API is not recommended after v7.3. Please use {@link TXAudioEffectManager#setVoiceEarMonitor} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)enableAudioEarMonitoring:(BOOL)enable __attribute__((deprecated("use TXAudioEffectManager#setVoiceEarMonitor instead")));
+#endif
+
+/**
+ * Start displaying remote video image
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link startRemoteView:streamType:view} instead.
+ */
+- (void)startRemoteView:(NSString *)userId view:(TXView *)view __attribute__((deprecated("use startRemoteView:streamType:view: instead")));
+
+/**
+ * Stop displaying remote video image and pulling the video data stream of remote user
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link stopRemoteView:streamType:} instead.
+ */
+- (void)stopRemoteView:(NSString *)userId __attribute__((deprecated("use stopRemoteView:streamType: instead")));
+
+/**
+ * Set the rendering mode of remote image
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link setRemoteRenderParams:streamType:params:} instead.
+ */
+- (void)setRemoteViewFillMode:(NSString *)userId mode:(TRTCVideoFillMode)mode __attribute__((deprecated("use setRemoteRenderParams:streamType:params: instead")));
+
+/**
+ * Set the clockwise rotation angle of remote image
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link setRemoteRenderParams:streamType:params:} instead.
+ */
+- (void)setRemoteViewRotation:(NSString *)userId rotation:(TRTCVideoRotation)rotation __attribute__((deprecated("use setRemoteRenderParams:streamType:params: instead")));
+
+/**
+ * Set the rendering mode of local image
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link setLocalRenderParams} instead.
+ */
+- (void)setLocalViewFillMode:(TRTCVideoFillMode)mode __attribute__((deprecated("use setLocalRenderParams instead")));
+
+/**
+ * Set the clockwise rotation angle of local image
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link setLocalRenderParams} instead.
+ */
+- (void)setLocalViewRotation:(TRTCVideoRotation)rotation __attribute__((deprecated("use setLocalRenderParams instead")));
+
+/**
+ * Set the mirror mode of local camera's preview image
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link setLocalRenderParams} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setLocalViewMirror:(TRTCLocalVideoMirrorType)mirror __attribute__((deprecated("use setLocalRenderParams: instead")));
+#elif TARGET_OS_MAC
+- (void)setLocalViewMirror:(BOOL)mirror __attribute__((deprecated("use setLocalRenderParams: instead")));
+#endif
+
+/**
+ * Start displaying the substream image of remote user
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link startRemoteView:streamType:view} instead.
+ */
+- (void)startRemoteSubStreamView:(NSString *)userId view:(TXView *)view __attribute__((deprecated("use startRemoteView:type:view: instead")));
+
+/**
+ * Stop displaying the substream image of remote user
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link stopRemoteView:streamType:} instead.
+ */
+- (void)stopRemoteSubStreamView:(NSString *)userId __attribute__((deprecated("use stopRemoteView:streamType: instead")));
+
+/**
+ * Set the fill mode of substream image
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link setRemoteRenderParams:streamType:params:} instead.
+ */
+- (void)setRemoteSubStreamViewFillMode:(NSString *)userId mode:(TRTCVideoFillMode)mode __attribute__((deprecated("use setRemoteRenderParams:streamType:params: instead")));
+
+/**
+ * Set the clockwise rotation angle of substream image
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link setRemoteRenderParams:streamType:params:} instead.
+ */
+- (void)setRemoteSubStreamViewRotation:(NSString *)userId rotation:(TRTCVideoRotation)rotation __attribute__((deprecated("use setRemoteRenderParams:streamType:params: instead")));
+
+/**
+ * Specify whether to view the big or small image
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link startRemoteView:streamType:view:} instead.
+ */
+- (void)setPriorRemoteVideoStreamType:(TRTCVideoStreamType)streamType __attribute__((deprecated("use startRemoteView:streamType:view: instead")));
+
+/**
+ * Set sound quality
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link startLocalAudio:}(quality) instead.
+ */
+- (void)setAudioQuality:(TRTCAudioQuality)quality __attribute__((deprecated("use startLocalAudio(quality) instead")));
+
+/**
+ * Set sound quality
+ *
+ * @deprecated This API is not recommended after v8.0. Please use {@link startLocalAudio:}(quality) instead.
+ */
+- (void)startLocalAudio __attribute__((deprecated("use startLocalAudio(quality) instead")));
+
+/**
+ * Switch camera
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `switchCamera` API in {@link TXDeviceManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)switchCamera __attribute__((deprecated("use TXDeviceManager#switchCamera instead")));
+#endif
+
+/**
+ * Query whether the current camera supports zoom
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `isCameraZoomSupported` API in {@link TXDeviceManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (BOOL)isCameraZoomSupported __attribute__((deprecated("use TXDeviceManager#isCameraZoomSupported instead")));
+#endif
+
+/**
+ * Set camera zoom ratio (focal length)
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `setCameraZoomRatio` API in {@link TXDeviceManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setZoom:(CGFloat)distance __attribute__((deprecated("use TXDeviceManager#setCameraZoomRatio instead")));
+#endif
+
+/**
+ * Query whether the device supports flash
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `isCameraTorchSupported` API in {@link TXDeviceManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (BOOL)isCameraTorchSupported __attribute__((deprecated("use TXDeviceManager#isCameraTorchSupported instead")));
+#endif
+
+/**
+ * Enable/Disable flash
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `enableCameraTorch` API in {@link TXDeviceManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (BOOL)enbaleTorch:(BOOL)enable __attribute__((deprecated("use TXDeviceManager#enableCameraTorch instead")));
+#endif
+
+/**
+ * Query whether the camera supports setting focus
+ *
+ * @deprecated This API is not recommended after v8.0.
+ */
+#if TARGET_OS_IPHONE
+- (BOOL)isCameraFocusPositionInPreviewSupported __attribute__((deprecated));
+#endif
+
+/**
+ * Set the focal position of camera
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `setCameraFocusPosition` API in {@link TXDeviceManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)setFocusPosition:(CGPoint)touchPoint __attribute__((deprecated("use TXDeviceManager#setCameraFocusPosition instead")));
+#endif
+
+/**
+ * Query whether the device supports the automatic recognition of face position
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `isAutoFocusEnabled` API in {@link TXDeviceManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (BOOL)isCameraAutoFocusFaceModeSupported __attribute__((deprecated("use TXDeviceManager#isAutoFocusEnabled instead")));
+#endif
+
+/**
+ * Enable/Disable face auto focus
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `enableCameraAutoFocus` API in {@link TXDeviceManager} instead.
+ */
+#if TARGET_OS_IPHONE
+- (void)enableAutoFaceFoucs:(BOOL)enable __attribute__((deprecated("use TXDeviceManager#enableCameraAutoFocus instead")));
+#endif
+
+/**
+ * Start camera test
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `startCameraDeviceTest` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)startCameraDeviceTestInView:(NSView *)view __attribute__((deprecated("use TXDeviceManager#startCameraDeviceTest instead")));
+#endif
+
+/**
+ * Start camera test
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `stopCameraDeviceTest` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)stopCameraDeviceTest __attribute__((deprecated("use TXDeviceManager#stopCameraDeviceTest instead")));
+#endif
+
+/**
+ * Start mic test
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `startMicDeviceTest` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)startMicDeviceTest:(NSInteger)interval testEcho:(void (^)(NSInteger volume))testEcho __attribute__((deprecated("use TXDeviceManager#startMicDeviceTest instead")));
+#endif
+
+/**
+ * Start mic test
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `stopMicDeviceTest` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)stopMicDeviceTest __attribute__((deprecated("use TXDeviceManager#stopMicDeviceTest instead")));
+#endif
+
+/**
+ * Start speaker test
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `startSpeakerDeviceTest` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)startSpeakerDeviceTest:(NSString *)audioFilePath onVolumeChanged:(void (^)(NSInteger volume, BOOL isLastFrame))volumeBlock __attribute__((deprecated("use TXDeviceManager#startSpeakerDeviceTest instead")));
+#endif
+
+/**
+ * Stop speaker test
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `stopSpeakerDeviceTest` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)stopSpeakerDeviceTest __attribute__((deprecated("use TXDeviceManager#stopSpeakerDeviceTest instead")));
+#endif
+
+/**
+ * Get the list of mics
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getDevicesList` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (NSArray *)getMicDevicesList __attribute__((deprecated("use TXDeviceManager#getDevicesList instead")));
+#endif
+
+/**
+ * Get the current mic device
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getCurrentDevice` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (TRTCMediaDeviceInfo *)getCurrentMicDevice __attribute__((deprecated("use TXDeviceManager#getCurrentDevice instead")));
+#endif
+
+/**
+ * Select the currently used mic
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `setCurrentDevice` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (int)setCurrentMicDevice:(NSString *)deviceId __attribute__((deprecated("use TXDeviceManager#setCurrentDevice instead")));
+#endif
+
+/**
+ * Get the current mic volume
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getCurrentDeviceVolume` API in {@link TXDeviceManager} instead.
+ */
+#if TARGET_OS_MAC
+- (float)getCurrentMicDeviceVolume __attribute__((deprecated("use TXDeviceManager#getCurrentDeviceVolume instead")));
+#endif
+
+/**
+ * Set the current mic volume
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `setCurrentDeviceVolume` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)setCurrentMicDeviceVolume:(NSInteger)volume __attribute__((deprecated("use TXDeviceManager#setCurrentDeviceVolume instead")));
+#endif
+
+/**
+ * Get the mute status of the current system mic
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getCurrentDeviceMute` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (BOOL)getCurrentMicDeviceMute __attribute__((deprecated("use TXDeviceManager#getCurrentDeviceMute instead")));
+#endif
+
+/**
+ * Set the mute status of the current system mic
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `setCurrentDeviceMute` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)setCurrentMicDeviceMute:(BOOL)mute __attribute__((deprecated("use TXDeviceManager#setCurrentDeviceMute instead")));
+#endif
+
+/**
+ * Get the list of speakers
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getDevicesList` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (NSArray *)getSpeakerDevicesList __attribute__((deprecated("use TXDeviceManager#getDevicesList instead")));
+#endif
+
+/**
+ * Get the currently used speaker
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getCurrentDevice` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (TRTCMediaDeviceInfo *)getCurrentSpeakerDevice __attribute__((deprecated("use TXDeviceManager#getCurrentDevice instead")));
+#endif
+
+/**
+ * Set the speaker to use
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `setCurrentDevice` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (int)setCurrentSpeakerDevice:(NSString *)deviceId __attribute__((deprecated("use TXDeviceManager#setCurrentDevice instead")));
+#endif
+
+/**
+ * Get the current speaker volume
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getCurrentDeviceVolume` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (float)getCurrentSpeakerDeviceVolume __attribute__((deprecated("use TXDeviceManager#getCurrentDeviceVolume instead")));
+#endif
+
+/**
+ * Set the current speaker volume
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `setCurrentDeviceVolume` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (int)setCurrentSpeakerDeviceVolume:(NSInteger)volume __attribute__((deprecated("use TXDeviceManager#setCurrentDeviceVolume instead")));
+#endif
+
+/**
+ * Get the mute status of the current system speaker
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getCurrentDeviceMute` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (BOOL)getCurrentSpeakerDeviceMute __attribute__((deprecated("use TXDeviceManager#getCurrentDeviceMute instead")));
+#endif
+
+/**
+ * Set whether to mute the current system speaker
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `setCurrentDeviceMute` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)setCurrentSpeakerDeviceMute:(BOOL)mute __attribute__((deprecated("use TXDeviceManager#setCurrentDeviceMute instead")));
+#endif
+
+/**
+ * Get the list of cameras
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getDevicesList` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (NSArray *)getCameraDevicesList __attribute__((deprecated("use TXDeviceManager#getDevicesList instead")));
+#endif
+
+/**
+ * Get the currently used camera
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `getCurrentDevice` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (TRTCMediaDeviceInfo *)getCurrentCameraDevice __attribute__((deprecated("use TXDeviceManager#getCurrentDevice instead")));
+#endif
+
+/**
+ * Set the camera to be used currently
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `setCurrentDevice` API in {@link TXDeviceManager} instead.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (int)setCurrentCameraDevice:(NSString *)deviceId __attribute__((deprecated("use TXDeviceManager#setCurrentDevice instead")));
+#endif
+
+/**
+ * Setting the system volume type (for mobile OS)
+ *
+ * @deprecated This API is not recommended after v8.0. Please use the `startLocalAudio(quality)` instead, which param `quality` is used to decide audio quality.
+ */
+- (void)setSystemVolumeType:(TRTCSystemVolumeType)type __attribute__((deprecated("use startLocalAudio:quality instead")));
+
+/**
+ * Screencapture video
+ *
+ * @deprecated This API is not recommended after v8.2. Please use `snapshotVideo:type:sourceType:completionBlock` instead.
+ */
+- (void)snapshotVideo:(NSString *)userId type:(TRTCVideoStreamType)streamType completionBlock:(void (^)(TXImage *image))completionBlock __attribute__((deprecated("use snapshotVideo:type:sourceType:completionBlock instead")));
+
+/**
+ * Enable custom video capturing mode
+ *
+ * @deprecated This API is not recommended after v8.5. Please use `enableCustomVideoCapture(streamType,enable)` instead.
+ */
+- (void)enableCustomVideoCapture:(BOOL)enable __attribute__((deprecated("use enableCustomVideoCapture:enable instead")));
+
+/**
+ * Deliver captured video data to SDK
+ *
+ * @deprecated This API is not recommended after v8.5. Please use `sendCustomVideoData(streamType, TRTCVideoFrame)` instead.
+ */
+- (void)sendCustomVideoData:(TRTCVideoFrame *)frame __attribute__((deprecated("use sendCustomVideoData:frame instead")));
+
+/**
+ * tart in-app screen sharing (for iOS 13.0 and above only)
+ *
+ * @deprecated This API is not recommended after v8.6. Please use `startScreenCaptureInApp:encParam:` instead.
+ */
+- (void)startScreenCaptureInApp:(TRTCVideoEncParam *)encParams __attribute__((deprecated("use startScreenCaptureInApp:encParam: instead")));
+
+/**
+ * Start system-level screen sharing (for iOS 11.0 and above only)
+ *
+ * @deprecated This API is not recommended after v8.6. Please use `startScreenCaptureByReplaykit:encParam:appGroup:` instead.
+ */
+- (void)startScreenCaptureByReplaykit:(TRTCVideoEncParam *)encParams appGroup:(NSString *)appGroup __attribute__((deprecated("use startScreenCaptureByReplaykit:encParam:appGroup: instead")));
+
+/**
+ * Pause/Resume publishing local video stream
+ *
+ * @deprecated This API is not recommended after v8.9. Please use `muteLocalVideo(streamType, mute)` instead.
+ */
+- (void)muteLocalVideo:(BOOL)mute __attribute__((deprecated("use muteLocalVideo:streamType:mute: instead")));
+
+/**
+ * Pause/Resume subscribing to remote user's video stream
+ *
+ * @deprecated This API is not recommended after v8.9. Please use `muteRemoteVideoStream(userId, streamType, mute)` instead.
+ */
+- (void)muteRemoteVideoStream:(NSString *)userId mute:(BOOL)mute __attribute__((deprecated("use muteRemoteVideoStream:userid,streamType:mute: instead")));
+
+/**
+ * Start network speed test (used before room entry)
+ *
+ * @deprecated This API is not recommended after v9.2. Please use `startSpeedTest(params)` instead.
+ */
+- (void)startSpeedTest:(uint32_t)sdkAppId
+ userId:(NSString *)userId
+ userSig:(NSString *)userSig
+ completion:(void (^)(TRTCSpeedTestResult *result, NSInteger completedCount, NSInteger totalCount))completion __attribute__((deprecated("use startSpeedTest: instead")));
+
+///@}
+@end
+///@}
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TRTCCloudDef.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TRTCCloudDef.h
new file mode 100644
index 0000000..dad27a3
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TRTCCloudDef.h
@@ -0,0 +1,1486 @@
+/**
+ * Module: TRTC key class definition
+ * Description: definitions of enumerated and constant values such as resolution and quality level
+ */
+/// @defgroup TRTCCloudDef_ios TRTCCloudDef
+/// Tencent Cloud TRTC Key Type Definition
+/// @{
+#import
+#import "TXDeviceManager.h"
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Rendering control
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * [VIEW] Rendering control that renders the video image
+ *
+ * There are many APIs in TRTC that need to manipulate the video image, for which you should specify the video rendering control.
+ * - On iOS, you can directly use `UIView` as the video rendering control, and the SDK will draw the video image on the `UIView` you provide.
+ * - On macOS, you can directly use `NSView` as the video rendering control, and the SDK will draw the video image on the `NSView` you provide.
+ * Below is the sample code:
+ *
+ * UIView *videoView = [[UIView alloc] initWithFrame:CGRectMake(0, 0, 360, 640)];
+ * [self.view addSubview:videoView];
+ * [trtcCloud startLocalPreview:YES view:_localView];
+ *
+ */
+#if TARGET_OS_IPHONE || TARGET_OS_SIMULATOR
+#import
+typedef UIView TXView;
+typedef UIImage TXImage;
+typedef UIEdgeInsets TXEdgeInsets;
+#elif TARGET_OS_MAC
+#import
+typedef NSView TXView;
+typedef NSImage TXImage;
+typedef NSEdgeInsets TXEdgeInsets;
+#endif
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Definitions of video enumerated values
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * 1.1 Video resolution
+ *
+ * Here, only the landscape resolution (e.g., 640x360) is defined. If the portrait resolution (e.g., 360x640) needs to be used, `Portrait` must be selected for `TRTCVideoResolutionMode`.
+ */
+typedef NS_ENUM(NSInteger, TRTCVideoResolution) {
+
+ /// Aspect ratio: 1:1; resolution: 120x120; recommended bitrate (VideoCall): 80 Kbps; recommended bitrate (LIVE): 120 Kbps.
+ TRTCVideoResolution_120_120 = 1,
+
+ /// Aspect ratio: 1:1; resolution: 160x160; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
+ TRTCVideoResolution_160_160 = 3,
+
+ /// Aspect ratio: 1:1; resolution: 270x270; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
+ TRTCVideoResolution_270_270 = 5,
+
+ /// Aspect ratio: 1:1; resolution: 480x480; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 500 Kbps.
+ TRTCVideoResolution_480_480 = 7,
+
+ /// Aspect ratio: 4:3; resolution: 160x120; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
+ TRTCVideoResolution_160_120 = 50,
+
+ /// Aspect ratio: 4:3; resolution: 240x180; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
+ TRTCVideoResolution_240_180 = 52,
+
+ /// Aspect ratio: 4:3; resolution: 280x210; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
+ TRTCVideoResolution_280_210 = 54,
+
+ /// Aspect ratio: 4:3; resolution: 320x240; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 375 Kbps.
+ TRTCVideoResolution_320_240 = 56,
+
+ /// Aspect ratio: 4:3; resolution: 400x300; recommended bitrate (VideoCall): 300 Kbps; recommended bitrate (LIVE): 450 Kbps.
+ TRTCVideoResolution_400_300 = 58,
+
+ /// Aspect ratio: 4:3; resolution: 480x360; recommended bitrate (VideoCall): 400 Kbps; recommended bitrate (LIVE): 600 Kbps.
+ TRTCVideoResolution_480_360 = 60,
+
+ /// Aspect ratio: 4:3; resolution: 640x480; recommended bitrate (VideoCall): 600 Kbps; recommended bitrate (LIVE): 900 Kbps.
+ TRTCVideoResolution_640_480 = 62,
+
+ /// Aspect ratio: 4:3; resolution: 960x720; recommended bitrate (VideoCall): 1000 Kbps; recommended bitrate (LIVE): 1500 Kbps.
+ TRTCVideoResolution_960_720 = 64,
+
+ /// Aspect ratio: 16:9; resolution: 160x90; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
+ TRTCVideoResolution_160_90 = 100,
+
+ /// Aspect ratio: 16:9; resolution: 256x144; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
+ TRTCVideoResolution_256_144 = 102,
+
+ /// Aspect ratio: 16:9; resolution: 320x180; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 400 Kbps.
+ TRTCVideoResolution_320_180 = 104,
+
+ /// Aspect ratio: 16:9; resolution: 480x270; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 550 Kbps.
+ TRTCVideoResolution_480_270 = 106,
+
+ /// Aspect ratio: 16:9; resolution: 640x360; recommended bitrate (VideoCall): 500 Kbps; recommended bitrate (LIVE): 900 Kbps.
+ TRTCVideoResolution_640_360 = 108,
+
+ /// Aspect ratio: 16:9; resolution: 960x540; recommended bitrate (VideoCall): 850 Kbps; recommended bitrate (LIVE): 1300 Kbps.
+ TRTCVideoResolution_960_540 = 110,
+
+ /// Aspect ratio: 16:9; resolution: 1280x720; recommended bitrate (VideoCall): 1200 Kbps; recommended bitrate (LIVE): 1800 Kbps.
+ TRTCVideoResolution_1280_720 = 112,
+
+ /// Aspect ratio: 16:9; resolution: 1920x1080; recommended bitrate (VideoCall): 2000 Kbps; recommended bitrate (LIVE): 3000 Kbps.
+ TRTCVideoResolution_1920_1080 = 114,
+
+};
+
+/**
+ * 1.2 Video aspect ratio mode
+ *
+ * Only the landscape resolution (e.g., 640x360) is defined in `TRTCVideoResolution`. If the portrait resolution (e.g., 360x640) needs to be used, `Portrait` must be selected for `TRTCVideoResolutionMode`.
+ */
+typedef NS_ENUM(NSInteger, TRTCVideoResolutionMode) {
+
+ /// Landscape resolution, such as TRTCVideoResolution_640_360 + TRTCVideoResolutionModeLandscape = 640x360.
+ TRTCVideoResolutionModeLandscape = 0,
+
+ /// Portrait resolution, such as TRTCVideoResolution_640_360 + TRTCVideoResolutionModePortrait = 360x640.
+ TRTCVideoResolutionModePortrait = 1,
+
+};
+
+/**
+ * 1.3 Video stream type
+ *
+ * TRTC provides three different video streams, including:
+ * - HD big image: it is generally used to transfer video data from the camera.
+ * - Smooth small image: it has the same content as the big image, but with lower resolution and bitrate and thus lower definition.
+ * - Substream image: it is generally used for screen sharing. Only one user in the room is allowed to publish the substream video image at any time, while other users must wait for this user to close the substream before they can publish their own
+ * substream.
+ * @note The SDK does not support enabling the smooth small image alone, which must be enabled together with the big image. It will automatically set the resolution and bitrate of the small image.
+ */
+typedef NS_ENUM(NSInteger, TRTCVideoStreamType) {
+
+ /// HD big image: it is generally used to transfer video data from the camera.
+ TRTCVideoStreamTypeBig = 0,
+
+ /// Smooth small image: it has the same content as the big image, but with lower resolution and bitrate and thus lower definition.
+ TRTCVideoStreamTypeSmall = 1,
+
+ /// Substream image: it is generally used for screen sharing. Only one user in the room is allowed to publish the substream video image at any time, while other users must wait for this user to close the substream before they can publish their
+ /// own substream.
+ TRTCVideoStreamTypeSub = 2,
+
+};
+
+/**
+ * 1.4 Video image fill mode
+ *
+ * If the aspect ratio of the video display area is not equal to that of the video image, you need to specify the fill mode:
+ */
+typedef NS_ENUM(NSInteger, TRTCVideoFillMode) {
+
+ /// Fill mode: the video image will be centered and scaled to fill the entire display area, where parts that exceed the area will be cropped. The displayed image may be incomplete in this mode.
+ TRTCVideoFillMode_Fill = 0,
+
+ /// Fit mode: the video image will be scaled based on its long side to fit the display area, where the short side will be filled with black bars. The displayed image is complete in this mode, but there may be black bars.
+ TRTCVideoFillMode_Fit = 1,
+
+};
+
+/**
+ * 1.5 Video image rotation direction
+ *
+ * TRTC provides rotation angle setting APIs for local and remote images. The following rotation angles are all clockwise.
+ */
+typedef NS_ENUM(NSInteger, TRTCVideoRotation) {
+
+ /// No rotation
+ TRTCVideoRotation_0 = 0,
+
+ /// Clockwise rotation by 90 degrees
+ TRTCVideoRotation_90 = 1,
+
+ /// Clockwise rotation by 180 degrees
+ TRTCVideoRotation_180 = 2,
+
+ /// Clockwise rotation by 270 degrees
+ TRTCVideoRotation_270 = 3,
+
+};
+
+/**
+ * 1.6 Beauty (skin smoothing) filter algorithm
+ *
+ * TRTC has multiple built-in skin smoothing algorithms. You can select the one most suitable for your product.
+ */
+typedef NS_ENUM(NSInteger, TRTCBeautyStyle) {
+
+ /// Smooth style, which uses a more radical algorithm for more obvious effect and is suitable for show live streaming.
+ TRTCBeautyStyleSmooth = 0,
+
+ /// Natural style, which retains more facial details for more natural effect and is suitable for most live streaming use cases.
+ TRTCBeautyStyleNature = 1,
+
+ /// Pitu style, which is provided by YouTu Lab. Its skin smoothing effect is between the smooth style and the natural style, that is, it retains more skin details than the smooth style and has a higher skin smoothing degree than the natural
+ /// style.
+ TRTCBeautyStylePitu = 2,
+
+};
+
+/**
+ * 1.7 Video pixel format
+ *
+ * TRTC provides custom video capturing and rendering features.
+ * - For the custom capturing feature, you can use the following enumerated values to describe the pixel format of the video you capture.
+ * - For the custom rendering feature, you can specify the pixel format of the video you expect the SDK to call back.
+ */
+typedef NS_ENUM(NSInteger, TRTCVideoPixelFormat) {
+
+ /// Undefined format
+ TRTCVideoPixelFormat_Unknown = 0,
+
+ /// YUV420P (I420) format
+ TRTCVideoPixelFormat_I420 = 1,
+
+ /// OpenGL 2D texture format
+ TRTCVideoPixelFormat_Texture_2D = 7,
+
+ /// BGRA32 format
+ TRTCVideoPixelFormat_32BGRA = 6,
+
+ /// YUV420SP (NV12) format
+ TRTCVideoPixelFormat_NV12 = 5,
+
+};
+
+/**
+ * 1.8 Video data transfer method
+ *
+ * For custom capturing and rendering features, you need to use the following enumerated values to specify the method of transferring video data:
+ * - Method 1. This method uses memory buffer to transfer video data. It is efficient on iOS but inefficient on Android. It is the only method supported on Windows currently.
+ * - Method 2. This method uses texture to transfer video data. It is efficient on both iOS and Android but is not supported on Windows. To use this method, you should have a general familiarity with OpenGL programming.
+ */
+typedef NS_ENUM(NSInteger, TRTCVideoBufferType) {
+
+ /// Undefined transfer method
+ TRTCVideoBufferType_Unknown = 0,
+
+ /// Use memory buffer to transfer video data. iOS: `PixelBuffer`; Android: `Direct Buffer` for JNI layer; Windows: memory data block.
+ TRTCVideoBufferType_PixelBuffer = 1,
+
+ /// Use memory buffer to transfer video data. iOS: more compact memory block in `NSData` type after additional processing; Android: `byte[]` for Java layer.
+ /// This transfer method has a lower efficiency than other methods.
+ TRTCVideoBufferType_NSData = 2,
+
+ /// Use texture to transfer video data
+ TRTCVideoBufferType_Texture = 3,
+
+};
+
+/**
+ * 1.9 Video mirror type
+ *
+ * Video mirroring refers to the left-to-right flipping of the video image, especially for the local camera preview image. After mirroring is enabled, it can bring anchors a familiar "look into the mirror" experience.
+ */
+typedef NS_ENUM(NSUInteger, TRTCVideoMirrorType) {
+
+ /// Auto mode: mirror the front camera's image but not the rear camera's image (for mobile devices only).
+ TRTCVideoMirrorTypeAuto = 0,
+
+ /// Mirror the images of both the front and rear cameras.
+ TRTCVideoMirrorTypeEnable = 1,
+
+ /// Disable mirroring for both the front and rear cameras.
+ TRTCVideoMirrorTypeDisable = 2,
+
+};
+
+/// Old version of TRTCVideoMirrorType, reserved for compatibility with older interface.
+typedef NS_ENUM(NSUInteger, TRTCLocalVideoMirrorType) {
+ TRTCLocalVideoMirrorType_Auto = TRTCVideoMirrorTypeAuto,
+ TRTCLocalVideoMirrorType_Enable = TRTCVideoMirrorTypeEnable,
+ TRTCLocalVideoMirrorType_Disable = TRTCVideoMirrorTypeDisable,
+} __attribute__((deprecated("use TRTCVideoMirrorType instead")));
+
+/**
+ * 1.10 Data source of local video screenshot
+ *
+ * The SDK can take screenshots from the following two data sources and save them as local files:
+ * - Video stream: the SDK screencaptures the native video content from the video stream. The screenshots are not controlled by the display of the rendering control.
+ * - Rendering layer: the SDK screencaptures the displayed video content from the rendering control, which can achieve the effect of WYSIWYG, but if the display area is too small, the screenshots will also be very small.
+ */
+typedef NS_ENUM(NSUInteger, TRTCSnapshotSourceType) {
+
+ /// The SDK screencaptures the native video content from the video stream. The screenshots are not controlled by the display of the rendering control.
+ TRTCSnapshotSourceTypeStream = 0,
+
+ /// The SDK screencaptures the displayed video content from the rendering control, which can achieve the effect of WYSIWYG, but if the display area is too small, the screenshots will also be very small.
+ TRTCSnapshotSourceTypeView = 1,
+
+};
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Definitions of network enumerated values
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * 2.1 Use cases
+ *
+ * TRTC features targeted optimizations for common audio/video application scenarios to meet the differentiated requirements in various verticals. The main scenarios can be divided into the following two categories:
+ * - Live streaming scenario (LIVE): including `LIVE` (audio + video) and `VoiceChatRoom` (pure audio).
+ * In the live streaming scenario, users are divided into two roles: "anchor" and "audience". A single room can sustain up to 100,000 concurrent online users. This is suitable for live streaming to a large audience.
+ * - Real-Time scenario (RTC): including `VideoCall` (audio + video) and `AudioCall` (pure audio).
+ * In the real-time scenario, there is no role difference between users, but a single room can sustain only up to 300 concurrent online users. This is suitable for small-scale real-time communication.
+ */
+typedef NS_ENUM(NSInteger, TRTCAppScene) {
+
+ /// In the video call scenario, 720p and 1080p HD image quality is supported. A single room can sustain up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
+ /// Use cases: [one-to-one video call], [video conferencing with up to 300 participants], [online medical diagnosis], [small class], [video interview], etc.
+ TRTCAppSceneVideoCall = 0,
+
+ /// In the interactive video live streaming scenario, mic can be turned on/off smoothly without waiting for switchover, and the anchor latency is as low as less than 300 ms. Live streaming to hundreds of thousands of concurrent users in the
+ /// audience role is supported with the playback latency down to 1,000 ms.
+ /// Use cases: [low-latency interactive live streaming], [big class], [anchor competition], [video dating room], [online interactive classroom], [remote training], [large-scale conferencing], etc.
+ ///@note In this scenario, you must use the `role` field in `TRTCParams` to specify the role of the current user.
+ TRTCAppSceneLIVE = 1,
+
+ /// Audio call scenario, where the `SPEECH` sound quality is used by default. A single room can sustain up to 300 concurrent online users, and up to 50 of them can speak simultaneously.
+ /// Use cases: [one-to-one audio call], [audio conferencing with up to 300 participants], [audio chat], [online Werewolf], etc.
+ TRTCAppSceneAudioCall = 2,
+
+ /// In the interactive audio live streaming scenario, mic can be turned on/off smoothly without waiting for switchover, and the anchor latency is as low as less than 300 ms. Live streaming to hundreds of thousands of concurrent users in the
+ /// audience role is supported with the playback latency down to 1,000 ms.
+ /// Use cases: [audio club], [online karaoke room], [music live room], [FM radio], etc.
+ ///@note In this scenario, you must use the `role` field in `TRTCParams` to specify the role of the current user.
+ TRTCAppSceneVoiceChatRoom = 3,
+
+};
+
+/**
+ * 2.2 Role
+ *
+ * Role is applicable only to live streaming scenarios (`TRTCAppSceneLIVE` and `TRTCAppSceneVoiceChatRoom`). Users are divided into two roles:
+ * - Anchor, who can publish their audio/video streams. There is a limit on the number of anchors. Up to 50 anchors are allowed to publish streams at the same time in one room.
+ * - Audience, who can only listen to or watch audio/video streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through {@link switchRole}. One room can sustain up to 100,000 concurrent
+ * online users in the audience role.
+ */
+typedef NS_ENUM(NSInteger, TRTCRoleType) {
+
+ /// An anchor can publish their audio/video streams. There is a limit on the number of anchors. Up to 50 anchors are allowed to publish streams at the same time in one room.
+ TRTCRoleAnchor = 20,
+
+ /// Audience can only listen to or watch audio/video streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through {@link switchRole}. One room can sustain up to 100,000 concurrent
+ /// online users in the audience role.
+ TRTCRoleAudience = 21,
+
+};
+
+/**
+ * 2.3 QoS control mode (disused)
+ */
+typedef NS_ENUM(NSInteger, TRTCQosControlMode) {
+
+ /// Client-based control, which is for internal debugging of SDK and shall not be used by users.
+ TRTCQosControlModeClient = 0,
+
+ /// On-cloud control, which is the default and recommended mode.
+ TRTCQosControlModeServer = 1,
+
+};
+
+/**
+ * 2.4 Image quality preference
+ *
+ * TRTC has two control modes in weak network environments: "ensuring clarity" and "ensuring smoothness". Both modes will give priority to the transfer of audio data.
+ */
+typedef NS_ENUM(NSInteger, TRTCVideoQosPreference) {
+
+ /// Ensuring smoothness: in this mode, when the current network is unable to transfer a clear and smooth video image, the smoothness of the image will be given priority, but there will be blurs.
+ TRTCVideoQosPreferenceSmooth = 1,
+
+ /// Ensuring clarity (default value): in this mode, when the current network is unable to transfer a clear and smooth video image, the clarity of the image will be given priority, but there will be lags.
+ TRTCVideoQosPreferenceClear = 2,
+
+};
+
+/**
+ * 2.5 Network quality
+ *
+ * TRTC evaluates the current network quality once every two seconds. The evaluation results are divided into six levels: `Excellent` indicates the best, and `Down` indicates the worst.
+ */
+typedef NS_ENUM(NSInteger, TRTCQuality) {
+
+ /// Undefined
+ TRTCQuality_Unknown = 0,
+
+ /// The current network is excellent
+ TRTCQuality_Excellent = 1,
+
+ /// The current network is good
+ TRTCQuality_Good = 2,
+
+ /// The current network is fair
+ TRTCQuality_Poor = 3,
+
+ /// The current network is bad
+ TRTCQuality_Bad = 4,
+
+ /// The current network is very bad
+ TRTCQuality_Vbad = 5,
+
+ /// The current network cannot meet the minimum requirements of TRTC
+ TRTCQuality_Down = 6,
+
+};
+
+/**
+ * 2.6 Audio/Video playback status
+ *
+ * This enumerated type is used in the video status changed API {@link onRemoteVideoStatusUpdated} to specify the current video status.
+ */
+typedef NS_ENUM(NSUInteger, TRTCAVStatusType) {
+
+ /// Stopped
+ TRTCAVStatusStopped = 0,
+
+ /// Playing
+ TRTCAVStatusPlaying = 1,
+
+ /// Loading
+ TRTCAVStatusLoading = 2,
+
+};
+
+/**
+ * 2.7 Reasons for playback status changes
+ *
+ * This enumerated type is used in the video status changed API {@link onRemoteVideoStatusUpdated} to specify the reason for the current video status change.
+ */
+typedef NS_ENUM(NSUInteger, TRTCAVStatusChangeReason) {
+
+ /// Default value
+ TRTCAVStatusChangeReasonInternal = 0,
+
+ /// The stream enters the "Loading" state due to network congestion
+ TRTCAVStatusChangeReasonBufferingBegin = 1,
+
+ /// The stream enters the "Playing" state after network recovery
+ TRTCAVStatusChangeReasonBufferingEnd = 2,
+
+ /// As a start-related API was directly called locally, the stream enters the "Playing" state
+ TRTCAVStatusChangeReasonLocalStarted = 3,
+
+ /// As a stop-related API was directly called locally, the stream enters the "Stopped" state
+ TRTCAVStatusChangeReasonLocalStopped = 4,
+
+ /// As the remote user started (or resumed) publishing the video stream, the stream enters the "Loading" or "Playing" state
+ TRTCAVStatusChangeReasonRemoteStarted = 5,
+
+ /// As the remote user stopped (or paused) publishing the video stream, the stream enters the "Stopped" state
+ TRTCAVStatusChangeReasonRemoteStopped = 6,
+
+};
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Definitions of audio enumerated values
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * 3.1 Audio sample rate
+ *
+ * The audio sample rate is used to measure the audio fidelity. A higher sample rate indicates higher fidelity. If there is music in the use case, `TRTCAudioSampleRate48000` is recommended.
+ */
+typedef NS_ENUM(NSInteger, TRTCAudioSampleRate) {
+
+ /// 16 kHz sample rate
+ TRTCAudioSampleRate16000 = 16000,
+
+ /// 32 kHz sample rate
+ TRTCAudioSampleRate32000 = 32000,
+
+ /// 44.1 kHz sample rate
+ TRTCAudioSampleRate44100 = 44100,
+
+ /// 48 kHz sample rate
+ TRTCAudioSampleRate48000 = 48000,
+
+};
+
+/**
+ * 3.2 Sound quality
+ *
+ * TRTC provides three well-tuned modes to meet the differentiated requirements for sound quality in various verticals:
+ * - Speech mode (Speech): it is suitable for application scenarios that focus on human communication. In this mode, the audio transfer is more resistant, and TRTC uses various voice processing technologies to ensure the optimal smoothness even in
+ * weak network environments.
+ * - Music mode (Music): it is suitable for scenarios with demanding requirements for music. In this mode, the amount of transferred audio data is very large, and TRTC uses various technologies to ensure that the high-fidelity details of music
+ * signals can be restored in each frequency band.
+ * - Default mode (Default): it is between `Speech` and `Music`. In this mode, the reproduction of music is better than that in `Speech` mode, and the amount of transferred data is much lower than that in `Music` mode; therefore, this mode has good
+ * adaptability to various scenarios.
+ */
+typedef NS_ENUM(NSInteger, TRTCAudioQuality) {
+
+ /// Speech mode: sample rate: 16 kHz; mono channel; bitrate: 16 Kbps. This mode has the best resistance among all modes and is suitable for audio call scenarios, such as online meeting and audio call.
+ TRTCAudioQualitySpeech = 1,
+
+ /// Default mode: sample rate: 48 kHz; mono channel; bitrate: 50 Kbps. This mode is between the speech mode and the music mode as the default mode in the SDK and is recommended.
+ TRTCAudioQualityDefault = 2,
+
+ /// Music mode: sample rate: 48 kHz; full-band stereo; bitrate: 128 Kbps. This mode is suitable for scenarios where Hi-Fi music transfer is required, such as online karaoke and music live streaming.
+ TRTCAudioQualityMusic = 3,
+
+};
+
+/**
+ * 3.3 Audio route (i.e., audio playback mode)
+ *
+ * "Audio route" determines whether the sound is played back from the speaker or receiver of a mobile device; therefore, this API is applicable only to mobile devices such as phones.
+ * Generally, a phone has two speakers: one is the receiver at the top, and the other is the stereo speaker at the bottom.
+ * - If the audio route is set to the receiver, the volume is relatively low, and the sound can be heard clearly only when the phone is put near the ear. This mode has a high level of privacy and is suitable for answering calls.
+ * - If the audio route is set to the speaker, the volume is relatively high, so there is no need to put the phone near the ear. Therefore, this mode can implement the "hands-free" feature.
+ */
+typedef NS_ENUM(NSInteger, TRTCAudioRoute) {
+
+ /// Speakerphone: the speaker at the bottom is used for playback (hands-free). With relatively high volume, it is used to play music out loud.
+ TRTCAudioModeSpeakerphone = 0,
+
+ /// Earpiece: the receiver at the top is used for playback. With relatively low volume, it is suitable for call scenarios that require privacy.
+ TRTCAudioModeEarpiece = 1,
+
+};
+
+/**
+ * 3.4 Audio reverb mode
+ *
+ * This enumerated value is used to set the audio reverb mode in the live streaming scenario and is often used in show live streaming.
+ */
+typedef NS_ENUM(NSInteger, TRTCReverbType) {
+
+ /// Disable reverb
+ TRTCReverbType_0 = 0,
+
+ /// KTV
+ TRTCReverbType_1 = 1,
+
+ /// Small room
+ TRTCReverbType_2 = 2,
+
+ /// Hall
+ TRTCReverbType_3 = 3,
+
+ /// Deep
+ TRTCReverbType_4 = 4,
+
+ /// Resonant
+ TRTCReverbType_5 = 5,
+
+ /// Metallic
+ TRTCReverbType_6 = 6,
+
+ /// Husky
+ TRTCReverbType_7 = 7,
+
+};
+
+/**
+ * 3.5 Voice changing type
+ *
+ * This enumerated value is used to set the voice changing mode in the live streaming scenario and is often used in show live streaming.
+ */
+typedef NS_ENUM(NSInteger, TRTCVoiceChangerType) {
+
+ /// Disable voice changing
+ TRTCVoiceChangerType_0 = 0,
+
+ /// Child
+ TRTCVoiceChangerType_1 = 1,
+
+ /// Girl
+ TRTCVoiceChangerType_2 = 2,
+
+ /// Middle-Aged man
+ TRTCVoiceChangerType_3 = 3,
+
+ /// Heavy metal
+ TRTCVoiceChangerType_4 = 4,
+
+ /// Nasal
+ TRTCVoiceChangerType_5 = 5,
+
+ /// Punk
+ TRTCVoiceChangerType_6 = 6,
+
+ /// Trapped beast
+ TRTCVoiceChangerType_7 = 7,
+
+ /// Otaku
+ TRTCVoiceChangerType_8 = 8,
+
+ /// Electronic
+ TRTCVoiceChangerType_9 = 9,
+
+ /// Robot
+ TRTCVoiceChangerType_10 = 10,
+
+ /// Ethereal
+ TRTCVoiceChangerType_11 = 11,
+
+};
+
+/**
+ * 3.6 System volume type (only for mobile devices)
+ *
+ * Smartphones usually have two types of system volume: call volume and media volume.
+ * - Call volume is designed for call scenarios. It comes with acoustic echo cancellation (AEC) and supports audio capturing by Bluetooth earphones, but its sound quality is average.
+ * If you cannot turn the volume down to 0 (i.e., mute the phone) using the volume buttons, then your phone is using call volume.
+ * - Media volume is designed for media scenarios such as music playback. AEC does not work when media volume is used, and Bluetooth earphones cannot be used for audio capturing. However, media volume delivers better music listening experience.
+ * If you are able to mute your phone using the volume buttons, then your phone is using media volume.
+ * The SDK offers three system volume control modes: auto, call volume, and media volume.
+ */
+typedef NS_ENUM(NSInteger, TRTCSystemVolumeType) {
+
+ /// Auto:
+ /// In the auto mode, call volume is used for anchors, and media volume for audience. This mode is suitable for live streaming scenarios.
+ /// If the scenario you select during `enterRoom` is `TRTCAppSceneLIVE` or `TRTCAppSceneVoiceChatRoom`, the SDK will automatically use this mode.
+ TRTCSystemVolumeTypeAuto = 0,
+
+ /// Media volume:
+ /// In this mode, media volume is used in all scenarios. It is rarely used, mainly suitable for music scenarios with demanding requirements on audio quality.
+ /// Use this mode if most of your users use peripheral devices such as audio cards. Otherwise, it is not recommended.
+ TRTCSystemVolumeTypeMedia = 1,
+
+ /// Call volume:
+ /// In this mode, the audio module does not change its work mode when users switch between anchors and audience, enabling seamless mic on/off. This mode is suitable for scenarios where users need to switch frequently between anchors and audience.
+ /// If the scenario you select during `enterRoom` is `TRTCAppSceneVideoCall` or `TRTCAppSceneAudioCall`, the SDK will automatically use this mode.
+ TRTCSystemVolumeTypeVOIP = 2,
+
+};
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Definitions of other enumerated values
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * 4.1 Log level
+ *
+ * Different log levels indicate different levels of details and number of logs. We recommend you set the log level to `TRTCLogLevelInfo` generally.
+ */
+typedef NS_ENUM(NSInteger, TRTCLogLevel) {
+
+ /// Output logs at all levels
+ TRTCLogLevelVerbose = 0,
+
+ /// Output logs at the DEBUG, INFO, WARNING, ERROR, and FATAL levels
+ TRTCLogLevelDebug = 1,
+
+ /// Output logs at the INFO, WARNING, ERROR, and FATAL levels
+ TRTCLogLevelInfo = 2,
+
+ /// Output logs at the WARNING, ERROR, and FATAL levels
+ TRTCLogLevelWarn = 3,
+
+ /// Output logs at the ERROR and FATAL levels
+ TRTCLogLevelError = 4,
+
+ /// Output logs at the FATAL level
+ TRTCLogLevelFatal = 5,
+
+ /// Do not output any SDK logs
+ TRTCLogLevelNone = 6,
+
+};
+
+/**
+ * 4.2 G-sensor switch (for mobile devices only)
+ */
+typedef NS_ENUM(NSInteger, TRTCGSensorMode) {
+
+ /// Do not adapt to G-sensor orientation
+ /// This mode is the default value for desktop platforms. In this mode, the video image published by the current user is not affected by the change of the G-sensor orientation.
+ TRTCGSensorMode_Disable = 0,
+
+ /// Adapt to G-sensor orientation
+ /// This mode is the default value on mobile platforms. In this mode, the video image published by the current user is adjusted according to the G-sensor orientation, while the orientation of the local preview image remains unchanged.
+ /// One of the adaptation modes currently supported by the SDK is as follows: when the phone or tablet is upside down, in order to ensure that the screen orientation seen by the remote user is normal, the SDK will automatically rotate the
+ /// published video image by 180 degrees.
+ /// If the UI layer of your application has enabled G-sensor adaption, we recommend you use the `UIFixLayout` mode.
+ TRTCGSensorMode_UIAutoLayout = 1,
+
+ /// Adapt to G-sensor orientation
+ /// In this mode, the video image published by the current user is adjusted according to the G-sensor orientation, and the local preview image will also be rotated accordingly.
+ /// One of the features currently supported is as follows: when the phone or tablet is upside down, in order to ensure that the screen orientation seen by the remote user is normal, the SDK will automatically rotate the published video image by
+ /// 180 degrees.
+ /// If the UI layer of your application doesn't support G-sensor adaption, but you want the video image in the SDK to adapt to the G-sensor orientation, we recommend you use the `UIFixLayout` mode.
+ TRTCGSensorMode_UIFixLayout = 2,
+
+};
+
+/**
+ * 4.3 Screen sharing target type (for desktops only)
+ */
+typedef NS_ENUM(NSInteger, TRTCScreenCaptureSourceType) {
+
+ /// Undefined
+ TRTCScreenCaptureSourceTypeUnknown = -1,
+
+ /// The screen sharing target is the window of an application
+ TRTCScreenCaptureSourceTypeWindow = 0,
+
+ /// The screen sharing target is the entire screen
+ TRTCScreenCaptureSourceTypeScreen = 1,
+
+};
+
+/**
+ * 4.4 Layout mode of On-Cloud MixTranscoding
+ *
+ * TRTC's On-Cloud MixTranscoding service can mix multiple audio/video streams in the room into one stream. Therefore, you need to specify the layout scheme of the video images. The following layout modes are provided:
+ */
+typedef NS_ENUM(NSInteger, TRTCTranscodingConfigMode) {
+
+ /// Undefined
+ TRTCTranscodingConfigMode_Unknown = 0,
+
+ /// Manual layout mode
+ /// In this mode, you need to specify the precise position of each video image. This mode has the highest degree of freedom, but its ease of use is the worst:
+ ///- You need to enter all the parameters in `TRTCTranscodingConfig`, including the position coordinates of each video image (TRTCMixUser).
+ ///- You need to listen on the `onUserVideoAvailable()` and `onUserAudioAvailable()` event callbacks in `TRTCCloudDelegate` and constantly adjust the `mixUsers` parameter according to the audio/video status of each user with mic on in the current
+ /// room.
+ TRTCTranscodingConfigMode_Manual = 1,
+
+ /// Pure audio mode
+ /// This mode is suitable for pure audio scenarios such as audio call (AudioCall) and audio chat room (VoiceChatRoom).
+ ///- You only need to set it once through the `setMixTranscodingConfig()` API after room entry, and then the SDK will automatically mix the audio of all mic-on users in the room into the current user's live stream.
+ ///- You don't need to set the `mixUsers` parameter in `TRTCTranscodingConfig`; instead, you only need to set the `audioSampleRate`, `audioBitrate` and `audioChannels` parameters.
+ TRTCTranscodingConfigMode_Template_PureAudio = 2,
+
+ /// Preset layout mode
+ /// This is the most popular layout mode, because it allows you to set the position of each video image in advance through placeholders, and then the SDK automatically adjusts it dynamically according to the number of video images in the room.
+ /// In this mode, you still need to set the `mixUsers` parameter, but you can set `userId` as a "placeholder". Placeholder values include:
+ /// - "$PLACE_HOLDER_REMOTE$": image of remote user. Multiple images can be set.
+ /// - "$PLACE_HOLDER_LOCAL_MAIN$": local camera image. Only one image can be set.
+ /// - "$PLACE_HOLDER_LOCAL_SUB$": local screen sharing image. Only one image can be set.
+ /// In this mode, you don't need to listen on the `onUserVideoAvailable()` and `onUserAudioAvailable()` callbacks in `TRTCCloudDelegate` to make real-time adjustments.
+ /// Instead, you only need to call `setMixTranscodingConfig()` once after successful room entry. Then, the SDK will automatically populate the placeholders you set with real `userId` values.
+ TRTCTranscodingConfigMode_Template_PresetLayout = 3,
+
+ /// Screen sharing mode
+ /// This mode is suitable for screen sharing-based use cases such as online education and supported only by the SDKs for Windows and macOS.
+ /// In this mode, the SDK will first build a canvas according to the target resolution you set (through the `videoWidth` and `videoHeight` parameters).
+ ///- Before the teacher enables screen sharing, the SDK will scale up the teacher's camera image and draw it onto the canvas.
+ ///- After the teacher enables screen sharing, the SDK will draw the video image shared on the screen onto the same canvas.
+ /// The purpose of this layout mode is to ensure consistency in the output resolution of the mixtranscoding module and avoid problems with blurred screen during course replay and webpage playback (web players don't support adjustable resolution).
+ /// Meanwhile, the audio of mic-on students will be mixed into the teacher's audio/video stream by default.
+ ///< br>
+ /// Video content is primarily the shared screen in teaching mode, and it is a waste of bandwidth to transfer camera image and screen image at the same time.
+ /// Therefore, the recommended practice is to directly draw the camera image onto the current screen through the `setLocalVideoRenderCallback` API.
+ /// In this mode, you don't need to set the `mixUsers` parameter in `TRTCTranscodingConfig`, and the SDK will not mix students' images so as not to interfere with the screen sharing effect.
+ ///< br>
+ /// You can set width x height in `TRTCTranscodingConfig` to 0 px x 0 px, and the SDK will automatically calculate a suitable resolution based on the aspect ratio of the user's current screen.
+ ///- If the teacher's current screen width is less than or equal to 1920 px, the SDK will use the actual resolution of the teacher's current screen.
+ ///- If the teacher's current screen width is greater than 1920 px, the SDK will select one of the three resolutions of 1920x1080 (16:9), 1920x1200 (16:10), and 1920x1440 (4:3) according to the current screen aspect ratio.
+ TRTCTranscodingConfigMode_Template_ScreenSharing = 4,
+
+};
+
+/**
+ * 4.5 Media recording type
+ *
+ * This enumerated type is used in the local media recording API {@link startLocalRecording} to specify whether to record audio/video files or pure audio files.
+ */
+typedef NS_ENUM(NSUInteger, TRTCRecordType) {
+
+ /// Record audio only
+ TRTCRecordTypeAudio = 0,
+
+ /// Record video only
+ TRTCRecordTypeVideo = 1,
+
+ /// Record both audio and video
+ TRTCRecordTypeBoth = 2,
+
+};
+
+/**
+ * 4.6 Stream mix input type
+ */
+typedef NS_ENUM(NSUInteger, TRTCMixInputType) {
+
+ /// Unspecified. The SDK will determine the stream mix input type according to the value of the `pureAudio` parameter
+ TRTCMixInputTypeUndefined = 0,
+
+ /// Mix both audio and video
+ TRTCMixInputTypeAudioVideo = 1,
+
+ /// Mix video only
+ TRTCMixInputTypePureVideo = 2,
+
+ /// Mix audio only
+ TRTCMixInputTypePureAudio = 3,
+
+};
+
+/**
+ * 4.7 Device type (for desktop platforms only)
+ *
+ * This enumerated value is used to define three types of audio/video devices, namely, camera, mic, and speaker, so that the same device management API can control the three different types of devices.
+ * Starting from v8.0, TRTC redefines `TXMediaDeviceType` in `TXDeviceManager` to replace `TRTCMediaDeviceType` on legacy versions.
+ * Only the definition of `TRTCMediaDeviceType` is retained here for compatibility with customer code on legacy versions.
+ */
+#if TARGET_OS_MAC && !TARGET_OS_IPHONE
+typedef NS_ENUM(NSInteger, TRTCMediaDeviceType) {
+ TRTCMediaDeviceTypeUnknown = -1, ///< undefined device type
+ TRTCMediaDeviceTypeAudioInput = 0, ///< microphone
+ TRTCMediaDeviceTypeAudioOutput = 1, ///< speaker
+ TRTCMediaDeviceTypeVideoCamera = 2, ///< camera
+ TRTCMediaDeviceTypeVideoWindow = 3, ///< windows(for screen share)
+ TRTCMediaDeviceTypeVideoScreen = 4, ///< screen (for screen share)
+} __attribute__((deprecated("use TXDeviceManager#TXMediaDeviceType instead")));
+
+typedef TXMediaDeviceInfo TRTCMediaDeviceInfo __attribute__((deprecated("use TXDeviceManager#TXMediaDeviceInfo instead")));
+#endif
+
+/**
+ * 4.11 Audio recording content type
+ *
+ * This enumerated type is used in the audio recording API {@link startAudioRecording} to specify the content of the recorded audio.
+ */
+typedef NS_ENUM(NSUInteger, TRTCAudioRecordingContent) {
+
+ /// Record both local and remote audio
+ TRTCAudioRecordingContentAll = 0,
+
+ /// Record local audio only
+ TRTCAudioRecordingContentLocal = 1,
+
+ /// Record remote audio only
+ TRTCAudioRecordingContentRemote = 2,
+
+};
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Definitions of core TRTC classes
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+/**
+ * 5.1 Room entry parameters
+ *
+ * As the room entry parameters in the TRTC SDK, these parameters must be correctly set so that the user can successfully enter the audio/video room specified by `roomId` or `strRoomId`.
+ * For historical reasons, TRTC supports two types of room IDs: `roomId` and `strRoomId`.
+ * Note: do not mix `roomId` and `strRoomId`, because they are not interchangeable. For example, the number `123` and the string `123` are two completely different rooms in TRTC.
+ */
+@interface TRTCParams : NSObject
+
+///**Field description:** application ID, which is required. Tencent Cloud generates bills based on `sdkAppId`.
+///**Recommended value:** the ID can be obtained on the account information page in the [TRTC console](https://console.cloud.tencent.com/rav/) after the corresponding application is created.
+@property(nonatomic, assign) UInt32 sdkAppId;
+
+///**Field description:** user ID, which is required. It is the `userId` of the local user in UTF-8 encoding and acts as the username.
+///**Recommended value:** if the ID of a user in your account system is "mike", `userId` can be set to "mike".
+@property(nonatomic, copy, nonnull) NSString *userId;
+
+///**Field description:** user signature, which is required. It is the authentication signature corresponding to the current `userId` and acts as the login password for Tencent Cloud services.
+///**Recommended value:** for the calculation method, please see [UserSig](https://cloud.tencent.com/document/product/647/17275).
+@property(nonatomic, copy, nonnull) NSString *userSig;
+
+///**Field description:** numeric room ID. Users (userId) in the same room can see one another and make audio/video calls.
+///**Recommended value:** value range: 1–4294967294.
+///**Note:** `roomId` and `strRoomId` are mutually exclusive. If you decide to use `strRoomId`, then `roomId` should be entered as 0. If both are entered, `roomId` will be used.
+///**Note:** do not mix `roomId` and `strRoomId`, because they are not interchangeable. For example, the number `123` and the string `123` are two completely different rooms in TRTC.
+@property(nonatomic, assign) UInt32 roomId;
+
+///**Field description:** string-type room ID. Users (userId) in the same room can see one another and make audio/video calls.
+///**Note:** `roomId` and `strRoomId` are mutually exclusive. If you decide to use `strRoomId`, then `roomId` should be entered as 0. If both are entered, `roomId` will be used.
+///**Note:** do not mix `roomId` and `strRoomId`, because they are not interchangeable. For example, the number `123` and the string `123` are two completely different rooms in TRTC.
+///**Recommended value:** the length limit is 64 bytes. The following 89 characters are supported:
+/// - Uppercase and lowercase letters (a–z and A–Z)
+/// - Digits (0–9)
+/// - Space, "!", "#", "$", "%", "&", "(", ")", "+", "-", ":", ";", "<", "=", ".", ">", "?", "@", "[", "]", "^", "_", "{", "}", "|", "~", and ",".
+@property(nonatomic, copy, nonnull) NSString *strRoomId;
+
+///**Field description:** role in the live streaming scenario, which is applicable only to the live streaming scenario ({@link TRTCAppSceneLIVE} or {@link TRTCAppSceneVoiceChatRoom}) but doesn't take effect in the call scenario.
+///**Recommended value:** default value: anchor ({@link TRTCRoleAnchor}).
+@property(nonatomic, assign) TRTCRoleType role;
+
+///**Field description:** specified `streamId` in Tencent Cloud CSS, which is optional. After setting this field, you can play back the user's audio/video stream on Tencent Cloud CSS CDN through a standard pull scheme (FLV or HLS).
+///**Recommended value:** this parameter can contain up to 64 bytes and can be left empty. We recommend you use `sdkappid_roomid_userid_main` as the `streamid`, which is easier to identify and will not cause conflicts in your multiple applications.
+///**Note:** to use Tencent Cloud CSS CDN, you need to enable the auto-relayed live streaming feature on the "Function Configuration" page in the [console](https://console.cloud.tencent.com/trtc/) first.
+/// For more information, please see [CDN Relayed Live Streaming](https://cloud.tencent.com/document/product/647/16826).
+@property(nonatomic, copy, nullable) NSString *streamId;
+
+///**Field description:** on-cloud recording field, which is optional and used to specify whether to record the user's audio/video stream in the cloud.
+/// For more information, please see [On-Cloud Recording and Playback](https://cloud.tencent.com/document/product/647/16823).
+///**Recommended value:** it can contain up to 64 bytes. Letters (a–z and A–Z), digits (0–9), underscores, and hyphens are allowed.
+///< p>
+/// Scheme 1. Manual recording
+/// 1. Enable on-cloud recording in "Application Management" > "On-cloud Recording Configuration" in the [console](https://console.cloud.tencent.com/trtc).
+/// 2. Set "Recording Mode" to "Manual Recording".
+/// 3. After manual recording is set, in a TRTC room, only users with the `userDefineRecordId` parameter set will have video recording files in the cloud, while users without this parameter set will not.
+/// 4. The recording file will be named in the format of "userDefineRecordId_start time_end time" in the cloud.
+///< p>
+/// Scheme 2. Auto-recording
+/// 1. You need to enable on-cloud recording in "Application Management" > "On-cloud Recording Configuration" in the [console](https://console.cloud.tencent.com/trtc).
+/// 2. Set "Recording Mode" to "Auto-recording".
+/// 3. After auto-recording is set, any user who upstreams audio/video in a TRTC room will have a video recording file in the cloud.
+/// 4. The file will be named in the format of "userDefineRecordId_start time_end time". If `userDefineRecordId` is not specified, the file will be named in the format of "streamId_start time_end time".
+///< br>
+@property(nonatomic, copy, nullable) NSString *userDefineRecordId;
+
+///**Field description:** permission credential used for permission control, which is optional. If you want only users with the specified `userId` values to enter a room, you need to use `privateMapKey` to restrict the permission.
+///**Recommended value:** we recommend you use this parameter only if you have high security requirements. For more information, please see [Enabling Advanced Permission Control](https://cloud.tencent.com/document/product/647/32240).
+@property(nonatomic, copy, nullable) NSString *privateMapKey;
+
+///**Field description:** business data, which is optional. This field is needed only by some advanced features.
+///**Recommended value:** do not set this field on your own.
+@property(nonatomic, copy, nullable) NSString *bussInfo;
+@end
+
+/**
+ * 5.2 Video encoding parameters
+ *
+ * These settings determine the quality of image viewed by remote users as well as the image quality of recorded video files in the cloud.
+ */
+@interface TRTCVideoEncParam : NSObject
+
+///**Field description:** video resolution
+///**Recommended value**
+/// - For mobile video call, we recommend you select a resolution of 360x640 or below and select `Portrait` (portrait resolution) for `resMode`.
+/// - For mobile live streaming, we recommend you select a resolution of 540x960 and select `Portrait` (portrait resolution) for `resMode`.
+/// - For desktop platforms (Windows and macOS), we recommend you select a resolution of 640x360 or above and select `Landscape` (landscape resolution) for `resMode`.
+///**Note:** to use a portrait resolution, please specify `resMode` as `Portrait`; for example, when used together with `Portrait`, 640x360 represents 360x640.
+@property(nonatomic, assign) TRTCVideoResolution videoResolution;
+
+///**Field description:** resolution mode (landscape/portrait)
+///**Recommended value:** for mobile platforms (iOS and Android), `Portrait` is recommended; for desktop platforms (Windows and macOS), `Landscape` is recommended.
+///**Note:** to use a portrait resolution, please specify `resMode` as `Portrait`; for example, when used together with `Portrait`, 640x360 represents 360x640.
+@property(nonatomic, assign) TRTCVideoResolutionMode resMode;
+
+////// **Field description:** video capturing frame rate
+////// **Recommended value:** 15 or 20 fps. If the frame rate is lower than 5 fps, there will be obvious lagging; if lower than 10 fps but higher than 5 fps, there will be slight lagging; if higher than 20 fps, the bandwidth will be wasted (the frame
+/// rate of movies is generally 24 fps).
+////// **Note:** the front cameras on certain Android phones do not support a capturing frame rate higher than 15 fps. For some Android phones that focus on beautification features, the capturing frame rate of the front cameras may be lower than 10
+/// fps.
+@property(nonatomic, assign) int videoFps;
+
+///**Field description:** target video bitrate. The SDK encodes streams at the target video bitrate and will actively reduce the bitrate only in weak network environments.
+///**Recommended value:** please see the optimal bitrate for each specification in `TRTCVideoResolution`. You can also slightly increase the optimal bitrate.
+/// For example, `TRTCVideoResolution_1280_720` corresponds to the target bitrate of 1,200 Kbps. You can also set the bitrate to 1,500 Kbps for higher definition.
+///**Note:** you can set the `videoBitrate` and `minVideoBitrate` parameters at the same time to restrict the SDK's adjustment range of the video bitrate:
+/// - If you want to "ensure clarity while allowing lag in weak network environments", you can set `minVideoBitrate` to 60% of `videoBitrate`.
+/// - If you want to "ensure smoothness while allowing blur in weak network environments", you can set `minVideoBitrate` to a low value, for example, 100 Kbps.
+/// - If you set `videoBitrate` and `minVideoBitrate` to the same value, it is equivalent to disabling the adaptive adjustment capability of the SDK for the video bitrate.
+@property(nonatomic, assign) int videoBitrate;
+
+///**Field description:** minimum video bitrate. The SDK will reduce the bitrate to as low as the value specified by `minVideoBitrate` to ensure the smoothness only if the network conditions are poor.
+///**Note:** default value: 0, indicating that a reasonable value of the lowest bitrate will be automatically calculated by the SDK according to the resolution you specify.
+///**Recommended value:** you can set the `videoBitrate` and `minVideoBitrate` parameters at the same time to restrict the SDK's adjustment range of the video bitrate:
+/// - If you want to "ensure clarity while allowing lag in weak network environments", you can set `minVideoBitrate` to 60% of `videoBitrate`.
+/// - If you want to "ensure smoothness while allowing blur in weak network environments", you can set `minVideoBitrate` to a low value, for example, 100 Kbps.
+/// - If you set `videoBitrate` and `minVideoBitrate` to the same value, it is equivalent to disabling the adaptive adjustment capability of the SDK for the video bitrate.
+@property(nonatomic, assign) int minVideoBitrate;
+
+///**Field description:** whether to allow dynamic resolution adjustment. Once enabled, this field will affect on-cloud recording.
+///**Recommended value:** this feature is suitable for scenarios that don't require on-cloud recording. After it is enabled, the SDK will intelligently select a suitable resolution according to the current network conditions to avoid the inefficient
+/// encoding mode of "large resolution + small bitrate".
+///**Note:** default value: NO. If you need on-cloud recording, please do not enable this feature, because if the video resolution changes, the MP4 file recorded in the cloud cannot be played back normally by common players.
+@property(nonatomic, assign) BOOL enableAdjustRes;
+@end
+
+/**
+ * 5.3 Network QoS control parameter set
+ *
+ * Network QoS control parameter. The settings determine the QoS control policy of the SDK in weak network conditions (e.g., whether to "ensure clarity" or "ensure smoothness").
+ */
+@interface TRTCNetworkQosParam : NSObject
+
+///**Field description:** whether to ensure smoothness or clarity
+///**Recommended value:** ensuring clarity
+///**Note:** this parameter mainly affects the audio/video performance of TRTC in weak network environments:
+/// - Ensuring smoothness: in this mode, when the current network is unable to transfer a clear and smooth video image, the smoothness of the image will be given priority, but there will be blurs.
+/// - Ensuring clarity (default value): in this mode, when the current network is unable to transfer a clear and smooth video image, the clarity of the image will be given priority, but there will be lags.
+@property(nonatomic, assign) TRTCVideoQosPreference preference;
+
+///**Field description:** QoS control mode (disused)
+///**Recommended value:** on-cloud control
+///**Note:** please set the on-cloud control mode (TRTCQosControlModeServer).
+@property(nonatomic, assign) TRTCQosControlMode controlMode;
+@end
+
+/**
+ * 5.4 Rendering parameters of video image
+ *
+ * You can use these parameters to control the video image rotation angle, fill mode, and mirror mode.
+ */
+@interface TRTCRenderParams : NSObject
+
+///**Field description:** clockwise image rotation angle
+///**Recommended value:** rotation angles of 90, 180, and 270 degrees are supported. Default value: {@link TRTCVideoRotation_0}
+@property(nonatomic) TRTCVideoRotation rotation;
+
+///**Field description:** image fill mode
+///**Recommended value:** fill (the image may be stretched or cropped) or fit (there may be black bars in unmatched areas). Default value: {@link TRTCVideoFillMode_Fill}
+@property(nonatomic) TRTCVideoFillMode fillMode;
+
+///**Field description:** image mirror mode
+///**Recommended value:** default value: {@link TRTCVideoMirrorType_Auto}
+@property(nonatomic) TRTCVideoMirrorType mirrorType;
+
+@end
+
+/**
+ * 5.5 Network quality
+ *
+ * This indicates the quality of the network. You can use it to display the network quality of each user on the UI.
+ */
+@interface TRTCQualityInfo : NSObject
+
+/// User ID
+@property(nonatomic, copy, nullable) NSString *userId;
+
+/// Network quality
+@property(nonatomic, assign) TRTCQuality quality;
+@end
+
+/**
+ * 5.6 Volume
+ *
+ * This indicates the audio volume value. You can use it to display the volume of each user in the UI.
+ */
+@interface TRTCVolumeInfo : NSObject
+
+///`userId` of the speaker. An empty value indicates the local user.
+@property(nonatomic, copy, nullable) NSString *userId;
+
+/// Volume of the speaker. Value range: 0–100.
+@property(assign, nonatomic) NSUInteger volume;
+
+@end
+
+/**
+ * 5.7 Network speed testing parameters
+ *
+ * You can test the network speed through the {@link startSpeedTest:} interface before the user enters the room (this API cannot be called during a call).
+ */
+@interface TRTCSpeedTestParams : NSObject
+
+/// Application identification, please refer to the relevant instructions in {@link TRTCParams}.
+@property(nonatomic) uint32_t sdkAppId;
+
+/// User identification, please refer to the relevant instructions in {@link TRTCParams}.
+@property(nonatomic, copy, nonnull) NSString *userId;
+
+/// User signature, please refer to the relevant instructions in {@link TRTCParams}.
+@property(nonatomic, copy, nonnull) NSString *userSig;
+
+/// Expected upstream bandwidth (kbps, value range: 10 to 5000, no uplink bandwidth test when it is 0).
+@property(nonatomic) NSInteger expectedUpBandwidth;
+
+/// Expected downstream bandwidth (kbps, value range: 10 to 5000, no downlink bandwidth test when it is 0).
+@property(nonatomic) NSInteger expectedDownBandwidth;
+@end
+
+/**
+ * 5.8 Network speed test result
+ *
+ * The {@link startSpeedTest:} API can be used to test the network speed before a user enters a room (this API cannot be called during a call).
+ */
+@interface TRTCSpeedTestResult : NSObject
+
+/// Whether the network speed test is successful.
+@property(nonatomic) BOOL success;
+
+/// Error message for network speed test.
+@property(nonatomic, copy, nonnull) NSString *errMsg;
+
+/// Server IP address.
+@property(nonatomic, copy, nonnull) NSString *ip;
+
+/// Network quality, which is tested and calculated based on the internal evaluation algorithm. For more information, please see {@link TRTCQuality}
+@property(nonatomic) TRTCQuality quality;
+
+/// Upstream packet loss rate between 0 and 1.0. For example, 0.3 indicates that 3 data packets may be lost in every 10 packets sent to the server.
+@property(nonatomic) float upLostRate;
+
+/// Downstream packet loss rate between 0 and 1.0. For example, 0.2 indicates that 2 data packets may be lost in every 10 packets received from the server.
+@property(nonatomic) float downLostRate;
+
+/// Delay in milliseconds, which is the round-trip time between the current device and TRTC server. The smaller the value, the better. The normal value range is 10–100 ms.
+@property(nonatomic) uint32_t rtt;
+
+/// Upstream bandwidth (in kbps, -1: invalid value).
+@property(nonatomic) NSInteger availableUpBandwidth;
+
+/// Downstream bandwidth (in kbps, -1: invalid value).
+@property(nonatomic) NSInteger availableDownBandwidth;
+@end
+
+/**
+ * 5.10 Video frame information
+ *
+ * `TRTCVideoFrame` is used to describe the raw data of a frame of the video image, which is the image data before frame encoding or after frame decoding.
+ */
+@interface TRTCVideoFrame : NSObject
+
+///**Field description:** video pixel format
+@property(nonatomic, assign) TRTCVideoPixelFormat pixelFormat;
+
+///**Field description:** video data structure type
+@property(nonatomic, assign) TRTCVideoBufferType bufferType;
+
+///**Field description:** video data when `bufferType` is {@link TRTCVideoBufferType_PixelBuffer}, which carries the `PixelBuffer` unique to iOS.
+@property(nonatomic, assign, nullable) CVPixelBufferRef pixelBuffer;
+
+///**Field description:** video data when `bufferType` is {@link TRTCVideoBufferType_NSData}, which carries the memory data blocks in `NSData` type.
+@property(nonatomic, retain, nullable) NSData *data;
+
+///**Field description:** video texture ID, i.e., video data when `bufferType` is {@link TRTCVideoBufferType_Texture}, which carries the texture data used for OpenGL rendering.
+@property(nonatomic, assign) GLuint textureId;
+
+///**Field description:** video width
+///**Recommended value:** please enter the width of the video data passed in.
+@property(nonatomic, assign) uint32_t width;
+
+///**Field description:** video height
+///**Recommended value:** please enter the height of the video data passed in.
+@property(nonatomic, assign) uint32_t height;
+
+///**Field description:** video frame timestamp in milliseconds
+///**Recommended value:** this parameter can be set to 0 for custom video capturing. In this case, the SDK will automatically set the `timestamp` field. However, please "evenly" set the calling interval of `sendCustomVideoData`.
+@property(nonatomic, assign) uint64_t timestamp;
+
+///**Field description:** clockwise rotation angle of video pixels
+@property(nonatomic, assign) TRTCVideoRotation rotation;
+
+@end
+
+/**
+ * 5.11 Audio frame data
+ */
+@interface TRTCAudioFrame : NSObject
+
+///**Field description:** audio data
+@property(nonatomic, retain, nonnull) NSData *data;
+
+///**Field description:** sample rate
+@property(nonatomic, assign) TRTCAudioSampleRate sampleRate;
+
+///**Field description:** number of sound channels
+@property(nonatomic, assign) int channels;
+
+///**Field description:** timestamp in ms
+@property(nonatomic, assign) uint64_t timestamp;
+
+///**Field description:** extra data in audio frame, message sent by remote users through `onLocalProcessedAudioFrame` that add to audio frame will be callback through this field.
+@property(nonatomic, retain, nullable) NSData *extraData;
+@end
+
+/**
+ * 5.12 Description information of each video image in On-Cloud MixTranscoding
+ *
+ * `TRTCMixUser` is used to specify the location, size, layer, and stream type of each video image in On-Cloud MixTranscoding.
+ */
+@interface TRTCMixUser : NSObject
+
+///**Field description:** user ID
+@property(nonatomic, copy, nonnull) NSString *userId;
+
+///**Field description:** ID of the room where this audio/video stream is located (an empty value indicates the local room ID)
+@property(nonatomic, copy, nullable) NSString *roomID;
+
+///**Field description:** specify the coordinate area of this video image in px
+@property(nonatomic, assign) CGRect rect;
+
+///**Field description:** specify the level of this video image (value range: 1–15; the value must be unique)
+@property(nonatomic, assign) int zOrder;
+
+///**Field description:** specify whether this video image is the primary stream image ({@link TRTCVideoStreamTypeBig}) or substream image ({@link TRTCVideoStreamTypeSub}).
+@property(nonatomic) TRTCVideoStreamType streamType;
+
+///**Field description:** specify whether this stream mixes audio only
+///**Recommended value:** default value: NO
+///**Note:** this field has been disused. We recommend you use the new field `inputType` introduced in v8.5.
+@property(nonatomic, assign) BOOL pureAudio;
+
+///**Field description:** specify the mixed content of this stream (audio only, video only, or audio and video). This field is an upgrade to the `pureAudio` field.
+///**Recommended value:** default value: TRTCMixInputTypeUndefined, which represents the value of `pureAudio`.
+/// - If you are using TRTC for the first time and have not set the `pureAudio` field before, you can set this field according to your actual needs. We recommend you not set `pureAudio` again.
+/// - If you have already used the `pureAudio` field on a legacy version and want to keep its setting, you can set `inputType` to `TRTCMixInputTypeUndefined`.
+@property(nonatomic, assign) TRTCMixInputType inputType;
+
+@end
+
+/**
+ * 5.13 Layout and transcoding parameters of On-Cloud MixTranscoding
+ *
+ * These parameters are used to specify the layout position information of each video image and the encoding parameters of mixtranscoding during On-Cloud MixTranscoding.
+ */
+@interface TRTCTranscodingConfig : NSObject
+
+///**Field description:** layout mode
+///**Recommended value:** please choose a value according to your business needs. The preset mode has better applicability.
+@property(nonatomic, assign) TRTCTranscodingConfigMode mode;
+
+///**Field description:** `appId` of Tencent Cloud CSS
+///**Recommended value:** please click **Application Management** > **Application Information** in the [TRTC console](https://console.cloud.tencent.com/trtc) and get the `appId` in **Relayed Live Streaming Info**.
+@property(nonatomic) int appId;
+
+///**Field description:** `bizId` of Tencent Cloud CSS
+///**Recommended value:** please click **Application Management** > **Application Information** in the [TRTC console](https://console.cloud.tencent.com/trtc) and get the `bizId` in **Relayed Live Streaming Info**.
+@property(nonatomic) int bizId;
+
+///**Field description:** specify the target resolution (width) of On-Cloud MixTranscoding
+///**Recommended value:** 360 px. If you only mix audio streams, please set both `width` and `height` to 0; otherwise, there will be a black background in the live stream after mixtranscoding.
+@property(nonatomic, assign) int videoWidth;
+
+///**Field description:** specify the target resolution (height) of On-Cloud MixTranscoding
+///**Recommended value:** 640 px. If you only mix audio streams, please set both `width` and `height` to 0; otherwise, there will be a black background in the live stream after mixtranscoding.
+@property(nonatomic, assign) int videoHeight;
+
+///**Field description:** specify the target video bitrate (Kbps) of On-Cloud MixTranscoding
+///**Recommended value:** if you enter 0, TRTC will estimate a reasonable bitrate value based on `videoWidth` and `videoHeight`. You can also refer to the recommended bitrate value in the video resolution enumeration definition (in the comment
+/// section).
+@property(nonatomic, assign) int videoBitrate;
+
+///**Field description:** specify the target video frame rate (fps) of On-Cloud MixTranscoding
+///**Recommended value:** default value: 15 fps. Value range: (0,30].
+@property(nonatomic, assign) int videoFramerate;
+
+///**Field description:** specify the target video keyframe interval (GOP) of On-Cloud MixTranscoding
+///**Recommended value:** default value: 2 (in seconds). Value range: [1,8].
+@property(nonatomic, assign) int videoGOP;
+
+///**Field description:** specify the background color of the mixed video image.
+///**Recommended value:** default value: 0x000000, which means black and is in the format of hex number; for example: "0x61B9F1" represents the RGB color (97,158,241).
+@property(nonatomic, assign) int backgroundColor;
+
+///**Field description:** specify the background image of the mixed video image.
+///**Recommended value:** default value: null, indicating not to set the background image.
+///**Note:** you need to upload the background image by clicking **Add image** in "Application Management" > "Function Configuration" > "Material Management" in the [console](https://console.cloud.tencent.com/trtc) in advance.
+/// After the upload is successful, you can get the corresponding "image ID". Then, you need to convert it into a string and set it as `backgroundImage`.
+/// For example, if the "image ID" is 63, you can set `backgroundImage = @"63"`;
+@property(nonatomic, copy, nullable) NSString *backgroundImage;
+
+///**Field description:** specify the target audio sample rate of On-Cloud MixTranscoding
+///**Recommended value:** default value: 48000 Hz. Valid values: 12000 Hz, 16000 Hz, 22050 Hz, 24000 Hz, 32000 Hz, 44100 Hz, 48000 Hz.
+@property(nonatomic, assign) int audioSampleRate;
+
+///**Field description:** specify the target audio bitrate of On-Cloud MixTranscoding
+///**Recommended value:** default value: 64 Kbps. Value range: [32,192].
+@property(nonatomic, assign) int audioBitrate;
+
+///**Field description:** specify the number of sound channels of On-Cloud MixTranscoding
+///**Recommended value:** default value: 1, which means mono channel. Valid values: 1: mono channel; 2: dual channel.
+@property(nonatomic, assign) int audioChannels;
+
+///**Field description:** specify the audio encoding type of On-Cloud MixTranscoding
+///**Recommended value:** default value: 0, which means LC-AAC. Valid values: 0: LC-AAC; 1: HE-AAC; 2: HE-AACv2.
+///**Note:** HE-AAC and HE-AACv2 only support [48000, 44100, 32000, 24000, 16000] sample rate.
+///**Note:** HE-AACv2 only support dual channel.
+///**Note:** HE-AAC and HE-AACv2 take effects iff the output streamId is specified.
+@property(nonatomic, assign) int audioCodec;
+
+///**Field description:** specify the position, size, layer, and stream type of each video image in On-Cloud MixTranscoding
+///**Recommended value:** this field is an array in `TRTCMixUser` type, where each element represents the information of a video image.
+@property(nonatomic, copy, nonnull) NSArray *mixUsers;
+
+///**Field description:** ID of the live stream output to CDN
+///**Recommended value:** default value: null, that is, the audio/video streams in the room will be mixed into the audio/video stream of the caller of this API.
+/// - If you don't set this parameter, the SDK will execute the default logic, that is, it will mix the multiple audio/video streams in the room into the audio/video stream of the caller of this API, i.e., A + B => A.
+/// - If you set this parameter, the SDK will mix the audio/video streams in the room into the live stream you specify, i.e., A + B => C (C is the `streamId` you specify).
+@property(nonatomic, copy, nullable) NSString *streamId;
+
+@end
+
+/**
+ * 5.14 Push parameters required to be set when publishing audio/video streams to non-Tencent Cloud CDN
+ *
+ * TRTC's backend service supports publishing audio/video streams to third-party live CDN service providers through the standard RTMP protocol.
+ * If you use the Tencent Cloud CSS CDN service, you don't need to care about this parameter; instead, just use the {@link startPublish} API.
+ */
+@interface TRTCPublishCDNParam : NSObject
+
+///**Field description:** `appId` of Tencent Cloud CSS
+///**Recommended value:** please click **Application Management** > **Application Information** in the [TRTC console](https://console.cloud.tencent.com/trtc) and get the `appId` in **Relayed Live Streaming Info**.
+@property(nonatomic) int appId;
+
+///**Field description:** `bizId` of Tencent Cloud CSS
+///**Recommended value:** please click **Application Management** > **Application Information** in the [TRTC console](https://console.cloud.tencent.com/trtc) and get the `bizId` in **Relayed Live Streaming Info**.
+@property(nonatomic) int bizId;
+
+///**Field description:** specify the push address (in RTMP format) of this audio/video stream at the third-party live streaming service provider
+///**Recommended value:** the push URL rules vary greatly by service provider. Please enter a valid push URL according to the requirements of the target service provider. TRTC's backend server will push audio/video streams in the standard format to
+/// the third-party service provider according to the URL you enter.
+///**Note:** the push URL must be in RTMP format and meet the specifications of your target live streaming service provider; otherwise, the target service provider will reject the push requests from TRTC's backend service.
+@property(nonatomic, strong, nonnull) NSString *url;
+
+///**Field description:** specify the push address (in RTMP format) of this audio/video stream at the third-party live streaming service provider
+///**Recommended value:** default value: null,that is, the audio/video streams in the room will be pushed to the target service provider of the caller of this API.
+@property(nonatomic, strong, nonnull) NSString *streamId;
+
+@end
+
+/**
+ * 5.15 Local audio file recording parameters
+ *
+ * This parameter is used to specify the recording parameters in the audio recording API {@link startAudioRecording}.
+ */
+@interface TRTCAudioRecordingParams : NSObject
+
+///**Field description:** storage path of the audio recording file, which is required.
+///**Note:** this path must be accurate to the file name and extension. The extension determines the format of the audio recording file. Currently, supported formats include PCM, WAV, and AAC.
+/// For example, if you specify the path as `mypath/record/audio.aac`, it means that you want the SDK to generate an audio recording file in AAC format.
+/// Please specify a valid path with read/write permissions; otherwise, the audio recording file cannot be generated.
+@property(nonatomic, strong, nonnull) NSString *filePath;
+
+///**Field description:** Audio recording content type.
+///**Note:** Record all local and remote audio by default.
+@property(nonatomic, assign) TRTCAudioRecordingContent recordingContent;
+
+@end
+
+/**
+ * 5.16 Local media file recording parameters
+ *
+ * This parameter is used to specify the recording parameters in the local media file recording API {@link startLocalRecording}.
+ * The `startLocalRecording` API is an enhanced version of the `startAudioRecording` API. The former can record video files, while the latter can only record audio files.
+ */
+@interface TRTCLocalRecordingParams : NSObject
+
+///**Field description:** address of the recording file, which is required. Please ensure that the path is valid with read/write permissions; otherwise, the recording file cannot be generated.
+///**Note:** this path must be accurate to the file name and extension. The extension determines the format of the recording file. Currently, only the MP4 format is supported.
+/// For example, if you specify the path as `mypath/record/test.mp4`, it means that you want the SDK to generate a local video file in MP4 format.
+/// Please specify a valid path with read/write permissions; otherwise, the recording file cannot be generated.
+@property(nonatomic, copy, nonnull) NSString *filePath;
+
+///**Field description:** media recording type, which is `TRTCRecordTypeBoth` by default, indicating to record both audio and video.
+@property(nonatomic, assign) TRTCRecordType recordType;
+
+///**Field description:** `interval` is the update frequency of the recording information in milliseconds. Value range: 1000–10000. Default value: -1, indicating not to call back
+@property(nonatomic, assign) int interval;
+
+@end
+
+/**
+ * 5.17 Sound effect parameter (disused)
+ *
+ * "Sound effects" in TRTC refer to some short audio files (usually only a few seconds), such as "applause" and "laughter".
+ * This parameter is used to specify the path and number of playback times of a sound effect file (short audio file) in the sound effect playback API {@link TRTCCloud#playAudioEffect} on legacy versions.
+ * After v7.3, the sound effect API has been replaced by a new {@link TXAudioEffectManager#startPlayMusic} API.
+ * When you specify the {@link TXAudioMusicParam} parameter of `startPlayMusic`, if `isShortFile` is set to `YES`, the file is a "sound effect" file.
+ */
+@interface TRTCAudioEffectParam : NSObject
+
++ (_Nonnull instancetype)new __attribute__((unavailable("Use -initWith:(int)effectId path:(NSString * )path instead")));
+- (_Nonnull instancetype)init __attribute__((unavailable("Use -initWith:(int)effectId path:(NSString *)path instead")));
+
+///**Field description:** sound effect ID
+///**Note:** the SDK supports playing multiple sound effects. IDs are used to distinguish different sound effects and control their start, end, volume, etc.
+@property(nonatomic, assign) int effectId;
+
+///**Field description:** sound effect file path. Supported file formats include AAC, MP3, and M4A.
+@property(nonatomic, copy, nonnull) NSString *path;
+
+///**Field description:** number of times the sound effect is looped
+///**Valid values:** 0 or any positive integer. 0 (default) indicates that the sound effect is played once, 1 twice, and so on.
+@property(nonatomic, assign) int loopCount;
+
+///**Field description:** whether the sound effect is upstreamed
+///**Recommended value:** YES: when the sound effect is played back locally, it will be upstreamed to the cloud and can be heard by remote users. NO: the sound effect will not be upstreamed to the cloud and can only be heard locally. Default value:
+/// NO
+@property(nonatomic, assign) BOOL publish;
+
+///**Field description:** sound effect volume
+///**Recommended value:** value range: 0–100. Default value: 100
+@property(nonatomic, assign) int volume;
+
+- (_Nonnull instancetype)initWith:(int)effectId path:(NSString *_Nonnull)path;
+@end
+
+/**
+ * 5.18 Room switch parameter
+ *
+ * This parameter is used for the room switch API {@link switchRoom}, which can quickly switch a user from one room to another.
+ */
+@interface TRTCSwitchRoomConfig : NSObject
+
+///**Field description:** numeric room ID, which is optional. Users in the same room can see one another and make audio/video calls.
+///**Recommended value:** value range: 1–4294967294.
+///**Note:** either `roomId` or `strRoomId` must be entered. If both are entered, `roomId` will be used.
+@property(nonatomic, assign) UInt32 roomId;
+
+///**Field description:** string-type room ID, which is optional. Users in the same room can see one another and make audio/video calls.
+///**Note:** either `roomId` or `strRoomId` must be entered. If both are entered, `roomId` will be used.
+@property(nonatomic, copy, nullable) NSString *strRoomId;
+
+///**Field description:** user signature, which is optional. It is the authentication signature corresponding to the current `userId` and acts as the login password.
+/// If you don't specify the newly calculated `userSig` during room switch, the SDK will continue to use the `userSig` you specified during room entry (enterRoom).
+/// This requires you to ensure that the old `userSig` is still within the validity period allowed by the signature at the moment of room switch; otherwise, room switch will fail.
+///**Recommended value:** for the calculation method, please see [UserSig](https://cloud.tencent.com/document/product/647/17275).
+@property(nonatomic, copy, nullable) NSString *userSig;
+
+///**Field description:** permission credential used for permission control, which is optional. If you want only users with the specified `userId` values to enter a room, you need to use `privateMapKey` to restrict the permission.
+///**Recommended value:** we recommend you use this parameter only if you have high security requirements. For more information, please see [Enabling Advanced Permission Control](https://cloud.tencent.com/document/product/647/32240).
+@property(nonatomic, copy, nullable) NSString *privateMapKey;
+
+@end
+
+/**
+ * 5.19 Format parameter of custom audio callback
+ *
+ * This parameter is used to set the relevant format (including sample rate and number of channels) of the audio data called back by the SDK in the APIs related to custom audio callback.
+ */
+@interface TRTCAudioFrameDelegateFormat : NSObject
+
+///**Field description:** sample rate
+///**Recommended value:** default value: 48000 Hz. Valid values: 16000, 32000, 44100, 48000.
+@property(nonatomic, assign) TRTCAudioSampleRate sampleRate;
+
+///**Field description:** number of sound channels
+///**Recommended value:** default value: 1, which means mono channel. Valid values: 1: mono channel; 2: dual channel.
+@property(nonatomic, assign) int channels;
+
+///**Field description:** number of sample points
+///**Recommended value:** the value must be an integer multiple of sampleRate/100.
+@property(nonatomic, assign) int samplesPerCall;
+
+@end
+
+/**
+ * 5.21 Screen sharing target information (for desktop systems only)
+ *
+ * When users share the screen, they can choose to share the entire desktop or only the window of a certain program.
+ * `TRTCScreenCaptureSourceInfo` is used to describe the information of the target to be shared, including ID, name, and thumbnail. The fields in this structure are read-only.
+ */
+#if TARGET_OS_MAC && !TARGET_OS_IPHONE
+@interface TRTCScreenCaptureSourceInfo : NSObject
+
+///**Field description:** capturing source type (i.e., whether to share the entire screen or a certain window)
+@property(assign, nonatomic) TRTCScreenCaptureSourceType type;
+
+///**Field description:** capturing source ID. For a window, this field indicates a window ID; for a screen, this field indicates a display ID.
+@property(copy, nonatomic, nullable) NSString *sourceId;
+
+///**Field description:** capturing source name (encoded in UTF-8)
+@property(copy, nonatomic, nullable) NSString *sourceName;
+
+///**Field description:** extended window information
+@property(nonatomic, strong, nullable) NSDictionary *extInfo;
+
+///**Field description:** thumbnail of the shared window
+@property(nonatomic, readonly, nullable) NSImage *thumbnail;
+
+///**Field description:** icon of the shared window
+@property(nonatomic, readonly, nullable) NSImage *icon;
+
+@end
+#endif
+
+/**
+ * 5.24 parameter of the parallel strategy of remote audio streams
+ *
+ * This parameter is used to set the parallel strategy of remote audio streams.
+ */
+@interface TRTCAudioParallelParams : NSObject
+
+///**Field description:** Max number of remote audio streams. Default value: 0
+/// if maxCount > 0 and the number of people in the room is more than `maxCount`，SDK will select `maxCount` of remote audio streams in real time, which can reduce performance cost greatly.
+/// if maxCount = 0，SDK won't limit the number of remote audio streams, which may cause performance cost when there are many speakers in one room.
+@property(assign, nonatomic) UInt32 maxCount;
+
+///**Field description:** Users included that must be able to play.
+///**Note:** A list of user IDs. These users must be able to play and do not participate in smart selection.
+/// The number of `incluseUsers` must be less than `maxCount`. Otherwise, the setting of the parallel strategy of remote audio streams is invalid.
+///`incluseUsers` is valid when `maxCount` > 0. When `incluseUsers` takes effect, the max number of remote audio streams is (`maxCount` - the number of valid users in `incluseUsers`).
+@property(nonatomic, strong, nullable) NSArray *includeUsers;
+
+@end
+
+/// @}
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TRTCCloudDelegate.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TRTCCloudDelegate.h
new file mode 100644
index 0000000..4014f3f
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TRTCCloudDelegate.h
@@ -0,0 +1,901 @@
+/**
+ * Module: TRTCCloudDelegate @ TXLiteAVSDK
+ * Function: event callback APIs for TRTC’s video call feature
+ */
+/// @defgroup TRTCCloudDelegate_ios TRTCCloudDelegate
+/// Tencent Cloud TRTC Event Notification Interface
+/// @{
+#import
+#import "TRTCCloudDef.h"
+#import "TXLiteAVCode.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class TRTCCloud;
+@class TRTCStatistics;
+
+@protocol TRTCCloudDelegate
+@optional
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Error and warning events
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Error and warning events
+/// @{
+
+/**
+ * 1.1 Error event callback
+ *
+ * Error event, which indicates that the SDK threw an irrecoverable error such as room entry failure or failure to start device
+ * For more information, see [Error Codes](https://intl.cloud.tencent.com/document/product/647/35124).
+ *
+ * @param errCode Error code
+ * @param errMsg Error message
+ * @param extInfo Extended field. Certain error codes may carry extra information for troubleshooting.
+ */
+- (void)onError:(TXLiteAVError)errCode errMsg:(nullable NSString *)errMsg extInfo:(nullable NSDictionary *)extInfo;
+
+/**
+ * 1.2 Warning event callback
+ *
+ * Warning event, which indicates that the SDK threw an error requiring attention, such as video lag or high CPU usage
+ * For more information, see [Error Codes](https://intl.cloud.tencent.com/document/product/647/35124).
+ *
+ * @param warningCode Warning code
+ * @param warningMsg Warning message
+ * @param extInfo Extended field. Certain warning codes may carry extra information for troubleshooting.
+ */
+- (void)onWarning:(TXLiteAVWarning)warningCode warningMsg:(nullable NSString *)warningMsg extInfo:(nullable NSDictionary *)extInfo;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Room event callback
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Room event callback
+/// @{
+
+/**
+ * 2.1 Whether room entry is successful
+ *
+ * After calling the `enterRoom()` API in `TRTCCloud` to enter a room, you will receive the `onEnterRoom(result)` callback from `TRTCCloudDelegate`.
+ * - If room entry succeeded, `result` will be a positive number (`result` > 0), indicating the time in milliseconds (ms) the room entry takes.
+ * - If room entry failed, `result` will be a negative number (result < 0), indicating the error code for the failure.
+ * For more information on the error codes for room entry failure, see [Error Codes](https://intl.cloud.tencent.com/document/product/647/35124).
+ *
+ * @note
+ * 1. In TRTC versions below 6.6, the `onEnterRoom(result)` callback is returned only if room entry succeeds, and the `onError()` callback is returned if room entry fails.
+ * 2. In TRTC 6.6 and above, the `onEnterRoom(result)` callback is returned regardless of whether room entry succeeds or fails, and the `onError()` callback is also returned if room entry fails.
+ *
+ * @param result If `result` is greater than 0, it indicates the time (in ms) the room entry takes; if `result` is less than 0, it represents the error code for room entry.
+ */
+- (void)onEnterRoom:(NSInteger)result;
+
+/**
+ * 2.2 Room exit
+ *
+ * Calling the `exitRoom()` API in `TRTCCloud` will trigger the execution of room exit-related logic, such as releasing resources of audio/video devices and codecs.
+ * After all resources occupied by the SDK are released, the SDK will return the `onExitRoom()` callback.
+ *
+ * If you need to call `enterRoom()` again or switch to another audio/video SDK, please wait until you receive the `onExitRoom()` callback.
+ * Otherwise, you may encounter problems such as the camera or mic being occupied.
+ *
+ * @param reason Reason for room exit. `0`: the user called `exitRoom` to exit the room; `1`: the user was removed from the room by the server; `2`: the room was dismissed.
+ */
+- (void)onExitRoom:(NSInteger)reason;
+
+/**
+ * 2.3 Role switching
+ *
+ * You can call the `switchRole()` API in `TRTCCloud` to switch between the anchor and audience roles. This is accompanied by a line switching process.
+ * After the switching, the SDK will return the `onSwitchRole()` event callback.
+ *
+ * @param errCode Error code. `ERR_NULL` indicates a successful switch. For more information, please see [Error Codes](https://intl.cloud.tencent.com/document/product/647/35124).
+ * @param errMsg Error message
+ */
+- (void)onSwitchRole:(TXLiteAVError)errCode errMsg:(nullable NSString *)errMsg;
+
+/**
+ * 2.4 Result of room switching
+ *
+ * You can call the `switchRoom()` API in `TRTCCloud` to switch from one room to another.
+ * After the switching, the SDK will return the `onSwitchRoom()` event callback.
+ *
+ * @param errCode Error code. `ERR_NULL` indicates a successful switch. For more information, please see [Error Codes](https://intl.cloud.tencent.com/document/product/647/35124).
+ * @param errMsg Error message
+ */
+- (void)onSwitchRoom:(TXLiteAVError)errCode errMsg:(nullable NSString *)errMsg;
+
+/**
+ * 2.5 Result of requesting cross-room call
+ *
+ * You can call the `connectOtherRoom()` API in `TRTCCloud` to establish a video call with the anchor of another room. This is the “anchor competition” feature.
+ * The caller will receive the `onConnectOtherRoom()` callback, which can be used to determine whether the cross-room call is successful.
+ * If it is successful, all users in either room will receive the `onUserVideoAvailable()` callback from the anchor of the other room.
+ *
+ * @param userId The user ID of the anchor (in another room) to be called
+ * @param errCode Error code. `ERR_NULL` indicates that cross-room connection is established successfully. For more information, please see [Error Codes](https://intl.cloud.tencent.com/document/product/647/35124).
+ * @param errMsg Error message
+ */
+- (void)onConnectOtherRoom:(NSString *)userId errCode:(TXLiteAVError)errCode errMsg:(nullable NSString *)errMsg;
+
+/**
+ * 2.6 Result of ending cross-room call
+ */
+- (void)onDisconnectOtherRoom:(TXLiteAVError)errCode errMsg:(nullable NSString *)errMsg;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// User event callback
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name User event callback
+/// @{
+
+/**
+ * 3.1 A user entered the room
+ *
+ * Due to performance concerns, this callback works differently in different scenarios (i.e., `AppScene`, which you can specify by setting the second parameter when calling `enterRoom`).
+ * - Live streaming scenarios (`TRTCAppSceneLIVE` or `TRTCAppSceneVoiceChatRoom`): in live streaming scenarios, a user is either in the role of an anchor or audience. The callback is returned only when an anchor enters the room.
+ * - Call scenarios (`TRTCAppSceneVideoCall` or `TRTCAppSceneAudioCall`): in call scenarios, the concept of roles does not apply (all users can be considered as anchors), and the callback is returned when any user enters the room.
+ *
+ * @note
+ * 1. The `onRemoteUserEnterRoom` callback indicates that a user entered the room, but it does not necessarily mean that the user enabled audio or video.
+ * 2. If you want to know whether a user enabled video, we recommend you use the `onUserVideoAvailable()` callback.
+ * @param userId User ID of the remote user
+ */
+- (void)onRemoteUserEnterRoom:(NSString *)userId;
+
+/**
+ * 3.2 A user exited the room
+ *
+ * As with `onRemoteUserEnterRoom`, this callback works differently in different scenarios (i.e., `AppScene`, which you can specify by setting the second parameter when calling `enterRoom`).
+ * - Live streaming scenarios (`TRTCAppSceneLIVE` or `TRTCAppSceneVoiceChatRoom`): the callback is triggered only when an anchor exits the room.
+ * - Call scenarios (`TRTCAppSceneVideoCall` or `TRTCAppSceneAudioCall`): in call scenarios, the concept of roles does not apply, and the callback is returned when any user exits the room.
+ *
+ * @param userId User ID of the remote user
+ * @param reason Reason for room exit. `0`: the user exited the room voluntarily; `1`: the user exited the room due to timeout; `2`: the user was removed from the room.
+ */
+- (void)onRemoteUserLeaveRoom:(NSString *)userId reason:(NSInteger)reason;
+
+/**
+ * 3.3 A remote user published/unpublished primary stream video
+ *
+ * The primary stream is usually used for camera images. If you receive the `onUserVideoAvailable(userId, YES)` callback, it indicates that the user has available primary stream video.
+ * You can then call {@link startRemoteView} to subscribe to the remote user’s video. If the subscription is successful, you will receive the `onFirstVideoFrame(userid)` callback, which indicates that the first video frame of the user is rendered.
+ *
+ * If you receive the `onUserVideoAvailable(userId, NO)` callback, it indicates that the video of the remote user is disabled, which may be because the user called {@link muteLocalVideo} or {@link stopLocalPreview}.
+ *
+ * @param userId User ID of the remote user
+ * @param available Whether the user published (or unpublished) primary stream video. `YES`: published; `NO`: unpublished
+ */
+- (void)onUserVideoAvailable:(NSString *)userId available:(BOOL)available;
+
+/**
+ * 3.4 A remote user published/unpublished substream video
+ *
+ * The substream is usually used for screen sharing images. If you receive the `onUserSubStreamAvailable(userId, YES)` callback, it indicates that the user has available substream video.
+ * You can then call {@link startRemoteSubStreamView} to subscribe to the remote user’s video. If the subscription is successful, you will receive the `onFirstVideoFrame(userid)` callback, which indicates that the first frame of the user is rendered.
+ *
+ * @note The API used to display substream images is {@link startRemoteSubStreamView}, not {@link startRemoteView}.
+ *
+ * @param userId User ID of the remote user
+ * @param available Whether the user published (or unpublished) substream video. `YES`: published; `NO`: unpublished
+ */
+- (void)onUserSubStreamAvailable:(NSString *)userId available:(BOOL)available;
+
+/**
+ * 3.5 A remote user published/unpublished audio
+ *
+ * If you receive the `onUserAudioAvailable(userId, YES)` callback, it indicates that the user published audio.
+ * - In auto-subscription mode, the SDK will play the user’s audio automatically.
+ * - In manual subscription mode, you can call {@link muteRemoteAudio}(userid, NO) to play the user’s audio.
+ *
+ * @note The auto-subscription mode is used by default. You can switch to the manual subscription mode by calling {@link setDefaultStreamRecvMode}, but it must be called before room entry for the switch to take effect.
+ *
+ * @param userId User ID of the remote user
+ * @param available Whether the user published (or unpublished) audio. `YES`: published; `NO`: unpublished
+ */
+- (void)onUserAudioAvailable:(NSString *)userId available:(BOOL)available;
+
+/**
+ * 3.6 The SDK started rendering the first video frame of the local or a remote user
+ *
+ * The SDK returns this event callback when it starts rendering your first video frame or that of a remote user. The `userId` in the callback can help you determine whether the frame is yours or a remote user’s.
+ * - If `userId` is empty, it indicates that the SDK has started rendering your first video frame. The precondition is that you have called {@link startLocalPreview} or {@link startScreenCapture}.
+ * - If `userId` is not empty, it indicates that the SDK has started rendering the first video frame of a remote user. The precondition is that you have called {@link startRemoteView} to subscribe to the user’s video.
+ *
+ * @note
+ * 1. The callback of the first local video frame being rendered is triggered only after you call {@link startLocalPreview} or {@link startScreenCapture}.
+ * 2. The callback of the first video frame of a remote user being rendered is triggered only after you call {@link startRemoteView} or {@link startRemoteSubStreamView}.
+ *
+ * @param userId The user ID of the local or a remote user. If it is empty, it indicates that the first local video frame is available; if it is not empty, it indicates that the first video frame of a remote user is available.
+ * @param streamType Video stream type. The primary stream (`Main`) is usually used for camera images, and the substream (`Sub`) for screen sharing images.
+ * @param width Video width
+ * @param height Video height
+ */
+- (void)onFirstVideoFrame:(NSString *)userId streamType:(TRTCVideoStreamType)streamType width:(int)width height:(int)height;
+
+/**
+ * 3.7 The SDK started playing the first audio frame of a remote user
+ *
+ * The SDK returns this callback when it plays the first audio frame of a remote user. The callback is not returned for the playing of the first audio frame of the local user.
+ *
+ * @param userId User ID of the remote user
+ */
+- (void)onFirstAudioFrame:(NSString *)userId;
+
+/**
+ * 3.8 The first local video frame was published
+ *
+ * After you enter a room and call {@link startLocalPreview} or {@link startScreenCapture} to enable local video capturing (whichever happens first),
+ * the SDK will start video encoding and publish the local video data via its network module to the cloud.
+ * It returns the `onSendFirstLocalVideoFrame` callback after publishing the first local video frame.
+ *
+ * @param streamType Video stream type. The primary stream (`Main`) is usually used for camera images, and the substream (`Sub`) for screen sharing images.
+ */
+- (void)onSendFirstLocalVideoFrame:(TRTCVideoStreamType)streamType;
+
+/**
+ * 3.9 The first local audio frame was published
+ *
+ * After you enter a room and call {@link startLocalAudio} to enable audio capturing (whichever happens first),
+ * the SDK will start audio encoding and publish the local audio data via its network module to the cloud.
+ * The SDK returns the `onSendFirstLocalAudioFrame` callback after sending the first local audio frame.
+ */
+- (void)onSendFirstLocalAudioFrame;
+
+/**
+ * 3.10 Change of remote video status
+ *
+ * You can use this callback to get the status (`Playing`, `Loading`, or `Stopped`) of the video of each remote user and display it on the UI.
+ * @param userId User ID
+ * @param streamType Video stream type. The primary stream (`Main`) is usually used for camera images, and the substream (`Sub`) for screen sharing images.
+ * @param status Video status, which may be `Playing`, `Loading`, or `Stopped`
+ * @param reason Reason for the change of status
+ * @param extraInfo Extra information
+ */
+- (void)onRemoteVideoStatusUpdated:(NSString *)userId streamType:(TRTCVideoStreamType)streamType streamStatus:(TRTCAVStatusType)status reason:(TRTCAVStatusChangeReason)reason extrainfo:(nullable NSDictionary *)info;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Callback of statistics on network and technical metrics
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Callback of statistics on network and technical metrics
+/// @{
+
+/**
+ * 4.1 Real-time network quality statistics
+ *
+ * This callback is returned every 2 seconds and notifies you of the upstream and downstream network quality detected by the SDK.
+ * The SDK uses a built-in proprietary algorithm to assess the current latency, bandwidth, and stability of the network and returns a result.
+ * If the result is `1` (excellent), it means that the current network conditions are excellent; if it is `6` (down), it means that the current network conditions are too bad to support TRTC calls.
+ *
+ * @note In the returned parameters `localQuality` and `remoteQuality`, if `userId` is empty, it indicates that the network quality statistics of the local user are returned. Otherwise, the network quality statistics of a remote user are returned.
+ *
+ * @param localQuality Upstream network quality
+ * @param remoteQuality Downstream network quality
+ */
+- (void)onNetworkQuality:(TRTCQualityInfo *)localQuality remoteQuality:(NSArray *)remoteQuality;
+
+/**
+ * 4.2 Real-time statistics on technical metrics
+ *
+ * This callback is returned every 2 seconds and notifies you of the statistics on technical metrics related to video, audio, and network. The metrics are listed in {@link TRTCStatistics}:
+ * - Video statistics: video resolution (`resolution`), frame rate (`FPS`), bitrate (`bitrate`), etc.
+ * - Audio statistics: audio sample rate (`samplerate`), number of audio channels (`channel`), bitrate (`bitrate`), etc.
+ * - Network statistics: the round trip time (`rtt`) between the SDK and the cloud (SDK -> Cloud -> SDK), package loss rate (`loss`), upstream traffic (`sentBytes`), downstream traffic (`receivedBytes`), etc.
+ *
+ * @note If you want to learn about only the current network quality and do not want to spend much time analyzing the statistics returned by this callback, we recommend you use {@link onNetworkQuality}.
+ * @param statistics Statistics, including local statistics and the statistics of remote users. For details, please see {@link TRTCStatistics}.
+ */
+- (void)onStatistics:(TRTCStatistics *)statistics;
+
+/**
+ * 4.3 Callback of network speed test
+ *
+ * The callback is triggered by {@link startSpeedTest:}.
+ *
+ * @param result Speed test data, including loss rates, rtt and bandwidth rates, please refer to {@link TRTCSpeedTestResult} for details.
+ */
+- (void)onSpeedTestResult:(TRTCSpeedTestResult *)result;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Callback of connection to the cloud
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Callback of connection to the cloud
+/// @{
+
+/**
+ * 5.1 The SDK was disconnected from the cloud
+ *
+ * The SDK returns this callback when it is disconnected from the cloud, which may be caused by network unavailability or change of network, for example, when the user walks into an elevator.
+ * After returning this callback, the SDK will attempt to reconnect to the cloud, and will return the {@link onTryToReconnect} callback. When it is reconnected, it will return the {@link onConnectionRecovery} callback.
+ * In other words, the SDK proceeds from one event to the next in the following order:
+ *
+ * [onConnectionLost] =====> [onTryToReconnect] =====> [onConnectionRecovery]
+ * /|\ |
+ * |------------------------------------------------------|
+ *
+ */
+- (void)onConnectionLost;
+
+/**
+ * 5.2 The SDK is reconnecting to the cloud
+ *
+ * When the SDK is disconnected from the cloud, it returns the {@link onConnectionLost} callback. It then attempts to reconnect and returns this callback ({@link onTryToReconnect}). After it is reconnected, it returns the {@link onConnectionRecovery}
+ * callback.
+ */
+- (void)onTryToReconnect;
+
+/**
+ * 5.3 The SDK is reconnected to the cloud
+ *
+ * When the SDK is disconnected from the cloud, it returns the {@link onConnectionLost} callback. It then attempts to reconnect and returns the {@link onTryToReconnect} callback. After it is reconnected, it returns this callback ({@link
+ * onConnectionRecovery}).
+ */
+- (void)onConnectionRecovery;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Callback of hardware events
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Callback of hardware events
+/// @{
+
+/**
+ * 6.1 The camera is ready
+ *
+ * After you call {@link startLocalPreivew}, the SDK will try to start the camera and return this callback if the camera is started.
+ * If it fails to start the camera, it’s probably because the application does not have access to the camera or the camera is being used.
+ * You can capture the {@link onError} callback to learn about the exception and let users know via UI messages.
+ */
+- (void)onCameraDidReady;
+
+/**
+ * 6.2 The mic is ready
+ *
+ * After you call {@link startLocalAudio}, the SDK will try to start the mic and return this callback if the mic is started.
+ * If it fails to start the mic, it’s probably because the application does not have access to the mic or the mic is being used.
+ * You can capture the {@link onError} callback to learn about the exception and let users know via UI messages.
+ */
+- (void)onMicDidReady;
+
+/**
+ * 6.3 The audio route changed (for mobile devices only)
+ *
+ * Audio route is the route (speaker or receiver) through which audio is played.
+ * - When audio is played through the receiver, the volume is relatively low, and the sound can be heard only when the phone is put near the ear. This mode has a high level of privacy and is suitable for answering calls.
+ * - When audio is played through the speaker, the volume is relatively high, and there is no need to put the phone near the ear. This mode enables the "hands-free" feature.
+ *
+ * @param route Audio route, i.e., the route (speaker or receiver) through which audio is played
+ * @param fromRoute The audio route used before the change
+ */
+#if TARGET_OS_IPHONE
+- (void)onAudioRouteChanged:(TRTCAudioRoute)route fromRoute:(TRTCAudioRoute)fromRoute;
+#endif
+
+/**
+ * 6.4 Volume
+ *
+ * The SDK can assess the volume of each channel and return this callback on a regular basis. You can display, for example, a waveform or volume bar on the UI based on the statistics returned.
+ * You need to first call {@link enableAudioVolumeEvaluation} to enable the feature and set the interval for the callback.
+ * Note that the SDK returns this callback at the specified interval regardless of whether someone is speaking in the room. When no one is speaking in the room, `userVolumes` is empty, and `totalVolume` is `0`.
+ *
+ * @note `userVolumes` is an array. If `userId` is empty, the elements in the array represent the volume of the local user’s audio. Otherwise, they represent the volume of a remote user’s audio.
+ *
+ * @param userVolumes An array that represents the volume of all users who are speaking in the room. Value range: 0-100
+ * @param totalVolume The total volume of all remote users. Value range: 0-100
+ */
+- (void)onUserVoiceVolume:(NSArray *)userVolumes totalVolume:(NSInteger)totalVolume;
+
+/**
+ * 6.5 The status of a local device changed (for desktop OS only)
+ *
+ * The SDK returns this callback when a local device (camera, mic, or speaker) is connected or disconnected.
+ *
+ * @param deviceId Device ID
+ * @param deviceType Device type
+ * @param state Device status. `0`: disconnected; `1`: connected
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)onDevice:(NSString *)deviceId type:(TRTCMediaDeviceType)deviceType stateChanged:(NSInteger)state;
+#endif
+
+/**
+ * 6.6 The capturing volume of the mic changed
+ *
+ * On desktop OS such as macOS and Windows, users can set the capturing volume of the mic in the audio control panel.
+ * The higher volume a user sets, the higher the volume of raw audio captured by the mic.
+ * On some keyboards and laptops, users can also mute the mic by pressing a key (whose icon is a crossed out mic).
+ *
+ * When users set the mic capturing volume via the UI or a keyboard shortcut, the SDK will return this callback.
+ *
+ * @note You need to call {@link enableAudioVolumeEvaluation} and set the callback interval (`interval` > 0) to enable the callback. To disable the callback, set `interval` to `0`.
+ *
+ * @param volume System audio capturing volume, which users can set in the audio control panel. Value range: 0-100
+ * @param muted Whether the mic is muted. `YES`: muted; `NO`: unmuted
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)onAudioDeviceCaptureVolumeChanged:(NSInteger)volume muted:(BOOL)muted;
+#endif
+
+/**
+ * 6.7 The playback volume changed
+ *
+ * On desktop OS such as macOS and Windows, users can set the system’s playback volume in the audio control panel.
+ * On some keyboards and laptops, users can also mute the speaker by pressing a key (whose icon is a crossed out speaker).
+ *
+ * When users set the system’s playback volume via the UI or a keyboard shortcut, the SDK will return this callback.
+ *
+ * @note You need to call {@link enableAudioVolumeEvaluation} and set the callback interval (`interval` > 0) to enable the callback. To disable the callback, set `interval` to `0`.
+ *
+ * @param volume The system playback volume, which users can set in the audio control panel. Value range: 0-100
+ * @param muted Whether the speaker is muted. `YES`: muted; `NO`: unmuted
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)onAudioDevicePlayoutVolumeChanged:(NSInteger)volume muted:(BOOL)muted;
+#endif
+
+/**
+ * 6.8 Whether system audio capturing is enabled successfully (for macOS only)
+ *
+ * On macOS, you can call {@link startSystemAudioLoopback} to install an audio driver and have the SDK capture the audio played back by the system.
+ * In use cases such as video teaching and music live streaming, the teacher can use this feature to let the SDK capture the sound of the video played by his or her computer, so that students in the room can hear the sound too.
+ * The SDK returns this callback after trying to enable system audio capturing. To determine whether it is actually enabled, pay attention to the error parameter in the callback.
+ *
+ * @param err If it is `ERR_NULL`, system audio capturing is enabled successfully. Otherwise, it is not.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (void)onSystemAudioLoopbackError:(TXLiteAVError)err;
+#endif
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Callback of the receipt of a custom message
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Callback of the receipt of a custom message
+/// @{
+
+/**
+ * 7.1 Receipt of custom message
+ *
+ * When a user in a room uses {@link sendCustomCmdMsg} to send a custom message, other users in the room can receive the message through the `onRecvCustomCmdMsg` callback.
+ *
+ * @param userId User ID
+ * @param cmdID Command ID
+ * @param seq Message serial number
+ * @param message Message data
+ */
+- (void)onRecvCustomCmdMsgUserId:(NSString *)userId cmdID:(NSInteger)cmdID seq:(UInt32)seq message:(NSData *)message;
+
+/**
+ * 7.2 Loss of custom message
+ *
+ * When you use {@link sendCustomCmdMsg} to send a custom UDP message, even if you enable reliable transfer (by setting `reliable` to `YES`), there is still a chance of message loss. Reliable transfer only helps maintain a low probability of message
+ * loss, which meets the reliability requirements in most cases. If the sender sets `reliable` to `YES`, the SDK will use this callback to notify the recipient of the number of custom messages lost during a specified time period (usually 5s) in the
+ * past.
+ *
+ * @note The recipient receives this callback only if the sender sets `reliable` to `YES`.
+ * @param userId User ID
+ * @param cmdID Command ID
+ * @param errCode Error code
+ * @param missed Number of lost messages
+ */
+- (void)onMissCustomCmdMsgUserId:(NSString *)userId cmdID:(NSInteger)cmdID errCode:(NSInteger)errCode missed:(NSInteger)missed;
+
+/**
+ * 7.3 Receipt of SEI message
+ *
+ * If a user in the room uses {@link sendSEIMsg} to send an SEI message via video frames, other users in the room can receive the message through the `onRecvSEIMsg` callback.
+ *
+ * @param userId User ID
+ * @param message Data
+ */
+- (void)onRecvSEIMsg:(NSString *)userId message:(NSData *)message;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// CDN event callback
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name CDN event callback
+/// @{
+
+/**
+ * 8.1 Started publishing to Tencent Cloud CSS CDN
+ *
+ * When you call {@link startPublishing} to publish streams to Tencent Cloud CSS CDN, the SDK will sync the command to the CVM immediately.
+ * The SDK will then receive the execution result from the CVM and return the result to you via this callback.
+ *
+ * @param err `0`: successful; other values: failed
+ * @param errMsg Error message
+ */
+- (void)onStartPublishing:(int)err errMsg:(NSString *)errMsg;
+
+/**
+ * 8.2 Stopped publishing to Tencent Cloud CSS CDN
+ *
+ * When you call {@link stopPublishing} to stop publishing streams to Tencent Cloud CSS CDN, the SDK will sync the command to the CVM immediately.
+ * The SDK will then receive the execution result from the CVM and return the result to you via this callback.
+ *
+ * @param err `0`: successful; other values: failed
+ * @param errMsg Error message
+ */
+- (void)onStopPublishing:(int)err errMsg:(NSString *)errMsg;
+
+/**
+ * 8.3 Started publishing to non-Tencent Cloud’s live streaming CDN
+ *
+ * When you call {@link startPublishCDNStream} to start publishing streams to a non-Tencent Cloud’s live streaming CDN, the SDK will sync the command to the CVM immediately.
+ * The SDK will then receive the execution result from the CVM and return the result to you via this callback.
+ *
+ * @note If you receive a callback that the command is executed successfully, it only means that your command was sent to Tencent Cloud’s backend server. If the CDN vendor does not accept your streams, the publishing will still fail.
+ * @param err `0`: successful; other values: failed
+ * @param errMsg Error message
+ */
+- (void)onStartPublishCDNStream:(int)err errMsg:(NSString *)errMsg;
+
+/**
+ * 8.4 Stopped publishing to non-Tencent Cloud’s live streaming CDN
+ *
+ * When you call {@link stopPublishCDNStream} to stop publishing to a non-Tencent Cloud’s live streaming CDN, the SDK will sync the command to the CVM immediately.
+ * The SDK will then receive the execution result from the CVM and return the result to you via this callback.
+ *
+ * @param err `0`: successful; other values: failed
+ * @param errMsg Error message
+ */
+- (void)onStopPublishCDNStream:(int)err errMsg:(NSString *)errMsg;
+
+/**
+ * 8.5 Set the layout and transcoding parameters for On-Cloud MixTranscoding
+ *
+ * When you call {@link setMixTranscodingConfig} to modify the layout and transcoding parameters for On-Cloud MixTranscoding, the SDK will sync the command to the CVM immediately.
+ * The SDK will then receive the execution result from the CVM and return the result to you via this callback.
+ *
+ * @param err `0`: successful; other values: failed
+ * @param errMsg Error message
+ */
+- (void)onSetMixTranscodingConfig:(int)err errMsg:(NSString *)errMsg;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Screen sharing event callback
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Screen sharing event callback
+/// @{
+
+/**
+ * 9.1 Screen sharing started
+ *
+ * The SDK returns this callback when you call {@link startScreenCapture} and other APIs to start screen sharing.
+ */
+- (void)onScreenCaptureStarted;
+
+/**
+ * 9.2 Screen sharing was paused
+ *
+ * The SDK returns this callback when you call {@link pauseScreenCapture} to pause screen sharing.
+ * @param reason Reason.
+ * - `0`: the user paused screen sharing.
+ * - `1`: screen sharing was paused because the shared window became invisible(Mac). screen sharing was paused because setting parameters(Windows).
+ * - `2`: screen sharing was paused because the shared window became minimum(only for Windows).
+ * - `3`: screen sharing was paused because the shared window became invisible(only for Windows).
+ */
+- (void)onScreenCapturePaused:(int)reason;
+
+/**
+ * 9.3 Screen sharing was resumed
+ *
+ * The SDK returns this callback when you call {@link resumeScreenCapture} to resume screen sharing.
+ * @param reason Reason.
+ * - `0`: the user resumed screen sharing.
+ * - `1`: screen sharing was resumed automatically after the shared window became visible again(Mac). screen sharing was resumed automatically after setting parameters(Windows).
+ * - `2`: screen sharing was resumed automatically after the shared window became minimize recovery(only for Windows).
+ * - `3`: screen sharing was resumed automatically after the shared window became visible again(only for Windows).
+ */
+- (void)onScreenCaptureResumed:(int)reason;
+
+/**
+ * 9.4 Screen sharing stopped
+ *
+ * The SDK returns this callback when you call {@link stopScreenCapture} to stop screen sharing.
+ * @param reason Reason. `0`: the user stopped screen sharing; `1`: screen sharing stopped because the shared window was closed.
+ */
+- (void)onScreenCaptureStoped:(int)reason;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Callback of local recording and screenshot events
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Callback of local recording and screenshot events
+/// @{
+
+/**
+ * 10.1 Local recording started
+ *
+ * When you call {@link startLocalRecording} to start local recording, the SDK returns this callback to notify you whether recording is started successfully.
+ * @param errCode Error code. `0`: recording started successfully; `-1`: failed to start recording; `-2`: incorrect file extension
+ * @param storagePath Storage path of recording file
+ */
+- (void)onLocalRecordBegin:(NSInteger)errCode storagePath:(NSString *)storagePath;
+
+/**
+ * 10.2 Local media is being recorded
+ *
+ * The SDK returns this callback regularly after local recording is started successfully via the calling of {@link startLocalRecording}.
+ * You can capture this callback to stay up to date with the status of the recording task.
+ * You can set the callback interval when calling {@link startLocalRecording}.
+ *
+ * @param duration Cumulative duration of recording, in milliseconds
+ * @param storagePath Storage path of recording file
+ */
+- (void)onLocalRecording:(NSInteger)duration storagePath:(NSString *)storagePath;
+
+/**
+ * 10.3 Local recording stopped
+ *
+ * When you call {@link stopLocalRecording} to stop local recording, the SDK returns this callback to notify you of the recording result.
+ * @param errCode Error code. `0`: recording succeeded; `-1`: recording failed; `-2`: recording was ended due to change of resolution or switch between the landscape and portrait mode.
+ * @param storagePath Storage path of recording file
+ */
+- (void)onLocalRecordComplete:(NSInteger)errCode storagePath:(NSString *)storagePath;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Disused callbacks (please use the new ones)
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Disused callbacks (please use the new ones)
+/// @{
+
+/**
+ * An anchor entered the room (disused)
+ *
+ * @deprecated This callback is not recommended in the new version. Please use {@link onRemoteUserEnterRoom} instead.
+ */
+- (void)onUserEnter:(NSString *)userId __attribute__((deprecated("use onRemoteUserLeaveRoom instead")));
+
+/**
+ * An anchor left the room (disused)
+ *
+ * @deprecated This callback is not recommended in the new version. Please use {@link onRemoteUserLeaveRoom} instead.
+ */
+- (void)onUserExit:(NSString *)userId reason:(NSInteger)reason __attribute__((deprecated("use onRemoteUserLeaveRoom instead")));
+
+/**
+ * Audio effects ended (disused)
+ *
+ * @deprecated This callback is not recommended in the new version. Please use {@link ITXAudioEffectManager} instead.
+ * Audio effects and background music can be started using the same API ({@link startPlayMusic}) now instead of separate ones.
+ */
+- (void)onAudioEffectFinished:(int)effectId code:(int)code __attribute__((deprecated("use ITXAudioEffectManager.startPlayMusic instead")));
+
+/// @}
+@end // End of class TRTCCloudDelegate
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Callback of custom video processing
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Callback of custom video processing
+/// @{
+
+@protocol TRTCVideoRenderDelegate
+
+/**
+ * Custom video rendering
+ *
+ * If you have configured the callback of custom rendering for local or remote video, the SDK will return to you via this callback video frames that are otherwise sent to the rendering control, so that you can customize rendering.
+ * @param frame Video frames to be rendered
+ * @param userId `userId` of the video source. This parameter can be ignored if the callback is for local video (`setLocalVideoRenderDelegate`).
+ * @param streamType Stream type. The primary stream (`Main`) is usually used for camera images, and the substream (`Sub`) for screen sharing images.
+ */
+@optional
+- (void)onRenderVideoFrame:(TRTCVideoFrame *_Nonnull)frame userId:(NSString *__nullable)userId streamType:(TRTCVideoStreamType)streamType;
+
+@end // End of class TRTCVideoRenderDelegate
+
+@protocol TRTCVideoFrameDelegate
+
+/**
+ * Video processing by third-party beauty filters
+ *
+ * If you use a third-party beauty filter component, you need to configure this callback in `TRTCCloud` to have the SDK return to you video frames that are otherwise pre-processed by TRTC.
+ * You can then send the video frames to the third-party beauty filter component for processing. As the data returned can be read and modified, the result of processing can be synced to TRTC for subsequent encoding and publishing.
+ *
+ * @param srcFrame Used to carry images captured by TRTC via the camera
+ * @param dstFrame Used to receive video images processed by third-party beauty filters
+ * @note Currently, only the OpenGL texture scheme is supported(PC supports TRTCVideoBufferType_Buffer format Only)
+ *
+ * Case 1: the beauty filter component generates new textures
+ * If the beauty filter component you use generates a frame of new texture (for the processed image) during image processing, please set `dstFrame.textureId` to the ID of the new texture in the callback function.
+ *
+ * uint32_t onProcessVideoFrame(TRTCVideoFrame * _Nonnull)srcFrame dstFrame:(TRTCVideoFrame * _Nonnull)dstFrame{
+ * self.frameID += 1;
+ * dstFrame.pixelBuffer = [[FURenderer shareRenderer] renderPixelBuffer:srcFrame.pixelBuffer
+ * withFrameId:self.frameID
+ * items:self.renderItems
+ * itemCount:self.renderItems.count];
+ * return 0;
+ * }
+ *
+ *
+ *
+ * Case 2: you need to provide target textures to the beauty filter component
+ * If the third-party beauty filter component you use does not generate new textures and you need to manually set an input texture and an output texture for the component, you can consider the following scheme:
+ * ```ObjectiveC
+ * uint32_t onProcessVideoFrame(TRTCVideoFrame * _Nonnull)srcFrame dstFrame:(TRTCVideoFrame * _Nonnull)dstFrame{
+ * thirdparty_process(srcFrame.textureId, srcFrame.width, srcFrame.height, dstFrame.textureId);
+ * return 0;
+ * }
+ * ```
+ * ```java
+ * int onProcessVideoFrame(TRTCCloudDef.TRTCVideoFrame srcFrame, TRTCCloudDef.TRTCVideoFrame dstFrame) {
+ * thirdparty_process(srcFrame.texture.textureId, srcFrame.width, srcFrame.height, dstFrame.texture.textureId);
+ * return 0;
+ * }
+ * ```
+ */
+@optional
+- (uint32_t)onProcessVideoFrame:(TRTCVideoFrame *_Nonnull)srcFrame dstFrame:(TRTCVideoFrame *_Nonnull)dstFrame;
+
+/**
+ * The OpenGL context in the SDK was destroyed
+ */
+@optional
+- (void)onGLContextDestory;
+
+@end // End of class TRTCVideoFrameDelegate
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Callback of custom audio processing
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Callback of custom audio processing
+/// @{
+
+@protocol TRTCAudioFrameDelegate
+@optional
+
+/**
+ * Audio data captured by the local mic and pre-processed by the audio module
+ *
+ * After you configure the callback of custom audio processing, the SDK will return via this callback the data captured and pre-processed (ANS, AEC, and AGC) in PCM format.
+ * - The audio returned is in PCM format and has a fixed frame length (time) of 0.02s.
+ * - The formula to convert a frame length in seconds to one in bytes is **sample rate * frame length in seconds * number of sound channels * audio bit depth**.
+ * - Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be **48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920
+ * bytes**.
+ *
+ * @param frame Audio frames in PCM format
+ * @note
+ * 1. Please avoid time-consuming operations in this callback function. The SDK processes an audio frame every 20 ms, so if your operation takes more than 20 ms, it will cause audio exceptions.
+ * 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
+ * 3. The audio data is returned via this callback after ANS, AEC and AGC, but it **does not include** pre-processing effects like background music, audio effects, or reverb, and therefore has a short delay.
+ */
+- (void)onCapturedRawAudioFrame:(TRTCAudioFrame *)frame;
+
+/**
+ * Audio data captured by the local mic, pre-processed by the audio module, effect-processed and BGM-mixed
+ *
+ * After you configure the callback of custom audio processing, the SDK will return via this callback the data captured, pre-processed (ANS, AEC, and AGC), effect-processed and BGM-mixed in PCM format, before it is submitted to the network module for
+ * encoding.
+ * - The audio data returned via this callback is in PCM format and has a fixed frame length (time) of 0.02s.
+ * - The formula to convert a frame length in seconds to one in bytes is **sample rate * frame length in seconds * number of sound channels * audio bit depth**.
+ * - Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be **48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920
+ * bytes**.
+ *
+ * Instructions:
+ * You could write data to the `TRTCAudioFrame.extraData` filed, in order to achieve the purpose of transmitting signaling.
+ * Because the data block of the audio frame header cannot be too large, we recommend you limit the size of the signaling data to only a few bytes when using this API. If extra data more than 100 bytes, it won't be sent.
+ * Other users in the room can receive the message through the `TRTCAudioFrame.extraData` in `onRemoteUserAudioFrame` callback in {@link TRTCAudioFrameDelegate}.
+ *
+ * @param frame Audio frames in PCM format
+ * @note
+ * 1. Please avoid time-consuming operations in this callback function. The SDK processes an audio frame every 20 ms, so if your operation takes more than 20 ms, it will cause audio exceptions.
+ * 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
+ * 3. Audio data is returned via this callback after ANS, AEC, AGC, effect-processing and BGM-mixing, and therefore the delay is longer than that with {@link onCapturedRawAudioFrame}.
+ */
+- (void)onLocalProcessedAudioFrame:(TRTCAudioFrame *)frame;
+
+/**
+ * Audio data of each remote user before audio mixing
+ *
+ * After you configure the callback of custom audio processing, the SDK will return via this callback the raw audio data (PCM format) of each remote user before mixing.
+ * - The audio data returned via this callback is in PCM format and has a fixed frame length (time) of 0.02s.
+ * - The formula to convert a frame length in seconds to one in bytes is **sample rate * frame length in seconds * number of sound channels * audio bit depth**.
+ * - Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be **48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920
+ * bytes**.
+ *
+ * @param frame Audio frames in PCM format
+ * @param userId User ID
+ * @note The audio data returned via this callback can be read but not modified.
+ */
+- (void)onRemoteUserAudioFrame:(TRTCAudioFrame *)frame userId:(NSString *)userId;
+
+/**
+ * Data mixed from each channel before being submitted to the system for playback
+ *
+ * After you configure the callback of custom audio processing, the SDK will return to you via this callback the data (PCM format) mixed from each channel before it is submitted to the system for playback.
+ * - The audio data returned via this callback is in PCM format and has a fixed frame length (time) of 0.02s.
+ * - The formula to convert a frame length in seconds to one in bytes is **sample rate * frame length in seconds * number of sound channels * audio bit depth**.
+ * - Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be **48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920
+ * bytes**.
+ *
+ * @param frame Audio frames in PCM format
+ * @note
+ * 1. Please avoid time-consuming operations in this callback function. The SDK processes an audio frame every 20 ms, so if your operation takes more than 20 ms, it will cause audio exceptions.
+ * 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
+ * 3. The audio data returned via this callback is the audio data mixed from each channel before it is played. It does not include the in-ear monitoring data.
+ */
+- (void)onMixedPlayAudioFrame:(TRTCAudioFrame *)frame;
+
+/**
+ * Data mixed from all the captured and to-be-played audio in the SDK
+ *
+ * After you configure the callback of custom audio processing, the SDK will return via this callback the data (PCM format) mixed from all captured and to-be-played audio in the SDK, so that you can customize recording.
+ * - The audio data returned via this callback is in PCM format and has a fixed frame length (time) of 0.02s.
+ * - The formula to convert a frame length in seconds to one in bytes is **sample rate * frame length in seconds * number of sound channels * audio bit depth**.
+ * - Assume that the audio is recorded on a single channel with a sample rate of 48,000 Hz and audio bit depth of 16 bits, which are the default settings of TRTC. The frame length in bytes will be **48000 * 0.02s * 1 * 16 bits = 15360 bits = 1920
+ * bytes**.
+ *
+ * @param frame Audio frames in PCM format
+ * @note
+ * 1. This data returned via this callback is mixed from all audio in the SDK, including local audio after pre-processing (ANS, AEC, and AGC), special effects application, and music mixing, as well as all remote audio, but it does not include the
+ * in-ear monitoring data.
+ * 2. The audio data returned via this callback cannot be modified.
+ */
+- (void)onMixedAllAudioFrame:(TRTCAudioFrame *)frame;
+
+@end // End of class TRTCAudioFrameDelegate
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Other event callbacks
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Other event callbacks
+/// @{
+
+@protocol TRTCLogDelegate
+
+@optional
+
+/**
+ * Printing of local log
+ *
+ * If you want to capture the local log printing event, you can configure the log callback to have the SDK return to you via this callback all logs that are to be printed.
+ * @param log Log content
+ * @param level Log level. For more information, please see `TRTC_LOG_LEVEL`.
+ * @param module Reserved field, which is not defined at the moment and has a fixed value of `TXLiteAVSDK`.
+ */
+- (void)onLog:(nullable NSString *)log LogLevel:(TRTCLogLevel)level WhichModule:(nullable NSString *)module;
+
+@end // End of class TRTCLogDelegate
+/// @}
+NS_ASSUME_NONNULL_END
+
+/// @}
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TRTCStatistics.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TRTCStatistics.h
new file mode 100644
index 0000000..b3141d8
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TRTCStatistics.h
@@ -0,0 +1,177 @@
+/**
+ * Module: TRTC audio/video metrics (read-only)
+ * Function: the TRTC SDK reports to you the current real-time audio/video metrics (frame rate, bitrate, lag, etc.) once every two seconds
+ */
+/// @defgroup TRTCStatisic_ios TRTCStatisic
+/// Tencent Cloud TRTC : audio, video and network related statistical indicators
+/// @{
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Local audio/video metrics
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Local audio/video metrics
+/// @{
+
+@interface TRTCLocalStatistics : NSObject
+
+///**Field description:** local video width in px
+@property(nonatomic, assign) uint32_t width;
+
+///**Field description:** local video height in px
+@property(nonatomic, assign) uint32_t height;
+
+///**Field description:** local video frame rate in fps, i.e., how many video frames there are per second
+@property(nonatomic, assign) uint32_t frameRate;
+
+///**Field description:** remote video bitrate in Kbps, i.e., how much video data is generated per second
+@property(nonatomic, assign) uint32_t videoBitrate;
+
+///**Field description:** remote audio sample rate (Hz)
+@property(nonatomic, assign) uint32_t audioSampleRate;
+
+///**Field description:** local audio bitrate in Kbps, i.e., how much audio data is generated per second
+@property(nonatomic, assign) uint32_t audioBitrate;
+
+///**Field description:** video stream type (HD big image | smooth small image | substream image)
+@property(nonatomic, assign) TRTCVideoStreamType streamType;
+
+///**Field description:**Audio equipment collection status(
+/// 0：Normal；1：Long silence detected；2：Broken sound detected；3：Abnormal intermittent sound detected;)
+@property(nonatomic, assign) uint32_t audioCaptureState;
+@end
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Remote audio/video metrics
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Remote audio/video metrics
+/// @{
+
+@interface TRTCRemoteStatistics : NSObject
+
+///**Field description:** user ID
+@property(nonatomic, retain) NSString* userId;
+
+///**Field description:** total packet loss rate (%) of the audio stream
+///`audioPacketLoss ` represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience".
+/// The smaller the `audioPacketLoss `, the better. The packet loss rate of 0 indicates that all data of the audio stream has entirely reached the audience.
+/// If `downLoss` is `0` but `audioPacketLoss ` isn't, there is no packet loss on the linkage of "cloud -> audience" for the audiostream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud".
+@property(nonatomic, assign) uint32_t audioPacketLoss;
+
+///**Field description:** total packet loss rate (%) of the video stream
+///`videoPacketLoss ` represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience".
+/// The smaller the `videoPacketLoss `, the better. The packet loss rate of 0 indicates that all data of the video stream has entirely reached the audience.
+/// If `downLoss` is `0` but `videoPacketLoss ` isn't, there is no packet loss on the linkage of "cloud -> audience" for the video stream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud".
+@property(nonatomic, assign) uint32_t videoPacketLoss;
+
+///**Field description:** remote video width in px
+@property(nonatomic, assign) uint32_t width;
+
+///**Field description:** remote video height in px
+@property(nonatomic, assign) uint32_t height;
+
+///**Field description:** remote video frame rate (fps)
+@property(nonatomic, assign) uint32_t frameRate;
+
+///**Field description:** remote video bitrate (Kbps)
+@property(nonatomic, assign) uint32_t videoBitrate;
+
+///**Field description:** local audio sample rate (Hz)
+@property(nonatomic, assign) uint32_t audioSampleRate;
+
+///**Field description:** local audio bitrate (Kbps)
+@property(nonatomic, assign) uint32_t audioBitrate;
+
+///**Field description:** playback delay (ms)
+/// In order to avoid audio/video lags caused by network jitters and network packet disorders, TRTC maintains a playback buffer on the playback side to organize the received network data packets.
+/// The size of the buffer is adaptively adjusted according to the current network quality and converted to the length of time in milliseconds, i.e., `jitterBufferDelay`.
+@property(nonatomic, assign) uint32_t jitterBufferDelay;
+
+///**Field description:** end-to-end delay (ms)
+///`point2PointDelay` represents the delay of "anchor -> cloud -> audience". To be more precise, it represents the delay of the entire linkage of "collection -> encoding -> network transfer -> receiving -> buffering -> decoding -> playback".
+///`point2PointDelay` works only if both the local and remote SDKs are on version 8.5 or above. If the remote SDK is on a version below 8.5, this value will always be 0 and thus meaningless.
+@property(nonatomic, assign) uint32_t point2PointDelay;
+
+///**Field description:** cumulative audio playback lag duration (ms)
+@property(nonatomic, assign) uint32_t audioTotalBlockTime;
+
+///**Field description:** audio playback lag rate (%)
+/// Audio playback lag rate (audioBlockRate) = cumulative audio playback lag duration (audioTotalBlockTime)/total audio playback duration
+@property(nonatomic, assign) uint32_t audioBlockRate;
+
+///**Field description:** cumulative video playback lag duration (ms)
+@property(nonatomic, assign) uint32_t videoTotalBlockTime;
+
+///**Field description:** video playback lag rate (%)
+/// Video playback lag rate (videoBlockRate) = cumulative video playback lag duration (videoTotalBlockTime)/total video playback duration
+@property(nonatomic, assign) uint32_t videoBlockRate;
+
+///**Field description:** total packet loss rate (%) of the audio/video stream
+/// Deprecated, please use audioPacketLoss and videoPacketLoss instead.
+@property(nonatomic, assign) uint32_t finalLoss __attribute__((deprecated("Use audioPacketLoss and videoPacketLoss instead.")));
+
+///**Field description:** video stream type (HD big image | smooth small image | substream image)
+@property(nonatomic, assign) TRTCVideoStreamType streamType;
+@end
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Network and performance metrics
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Network and performance metrics
+/// @{
+
+@interface TRTCStatistics : NSObject
+
+///**Field description:** CPU utilization (%) of the current application
+@property(nonatomic, assign) uint32_t appCpu;
+
+///**Field description:** CPU utilization (%) of the current system
+@property(nonatomic, assign) uint32_t systemCpu;
+
+///**Field description:** upstream packet loss rate (%) from the SDK to cloud
+/// The smaller the value, the better. If `upLoss` is `0%`, the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost.
+/// If `upLoss` is `30%`, 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage.
+@property(nonatomic, assign) uint32_t upLoss;
+
+///**Field description:** downstream packet loss rate (%) from cloud to the SDK
+/// The smaller the value, the better. If `downLoss` is `0%`, the downstream network quality is very good, and the data packets received from the cloud are basically not lost.
+/// If `downLoss` is `30%`, 30% of the audio/video data packets sent to the SDK by the cloud are lost on the transfer linkage.
+@property(nonatomic, assign) uint32_t downLoss;
+
+///**Field description:** round-trip delay (ms) from the SDK to cloud
+/// This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK
+/// -> cloud -> SDK". The smaller the value, the better. If `rtt` is below 50 ms, it means a short audio/video call delay; if `rtt` is above 200 ms, it means a long audio/video call delay. It should be explained that `rtt` represents the total time
+/// spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between `upRtt` and `downRtt`.
+@property(nonatomic, assign) uint32_t rtt;
+
+///**Field description:** round-trip delay (ms) from the SDK to gateway
+/// This value represents the total time it takes to send a network packet from the SDK to the gateway and then send a network packet back from the gateway to the SDK, i.e., the total time it takes for a network packet to go through the linkage of
+/// "SDK -> gateway -> SDK". The smaller the value, the better. If `gatewayRtt` is below 50 ms, it means a short audio/video call delay; if `gatewayRtt` is above 200 ms, it means a long audio/video call delay. It should be explained that `gatewayRtt`
+/// is invalid for cellular network.
+@property(nonatomic, assign) uint32_t gatewayRtt;
+
+///**Field description:** total number of sent bytes (including signaling data and audio/video data)
+@property(nonatomic, assign) uint64_t sentBytes;
+
+///**Field description:** total number of received bytes (including signaling data and audio/video data)
+@property(nonatomic, assign) uint64_t receivedBytes;
+
+///**Field description:** local audio/video statistics
+/// As there may be three local audio/video streams (i.e., HD big image, smooth small image, and substream image), the local audio/video statistics are an array.
+@property(nonatomic, strong) NSArray* localStatistics;
+
+///**Field description:** remote audio/video statistics
+/// As there may be multiple concurrent remote users, and each of them may have multiple concurrent audio/video streams (i.e., HD big image, smooth small image, and substream image), the remote audio/video statistics are an array.
+@property(nonatomic, strong) NSArray* remoteStatistics;
+
+@end
+/// @}
+
+/// @}
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXAudioCustomProcessDelegate.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXAudioCustomProcessDelegate.h
new file mode 100644
index 0000000..9155c38
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXAudioCustomProcessDelegate.h
@@ -0,0 +1,38 @@
+//
+// TXAudioCustomProcessDelegate.h
+// TXLiteAVSDK
+//
+// Created by realingzhou on 2018/1/15.
+// Copyright © 2018年 Tencent. All rights reserved.
+//
+
+#ifndef TXAudioCustomProcessDelegate_h
+#define TXAudioCustomProcessDelegate_h
+#import
+
+@protocol TXAudioCustomProcessDelegate
+
+/**
+ * 原始声音的回调
+ * @param data pcm数据
+ * @param timeStamp 时间戳
+ * @param sampleRate 采样率
+ * @param channels 声道数
+ * @param withBgm 回调的数据是否包含bgm,当不开启回声消除时，回调的raw pcm会包含bgm
+ */
+@optional
+- (void)onRecordRawPcmData:(NSData *)data timeStamp:(unsigned long long)timeStamp sampleRate:(int)sampleRate channels:(int)channels withBgm:(BOOL)withBgm;
+
+/**
+ * 经过特效处理的声音回调
+ * @param data pcm数据
+ * @param timeStamp 时间戳
+ * @param sampleRate 采样率
+ * @param channels 声道数
+ */
+@optional
+- (void)onRecordPcmData:(NSData *)data timeStamp:(unsigned long long)timeStamp sampleRate:(int)sampleRate channels:(int)channels;
+
+@end
+
+#endif /* TXAudioCustomProcessDelegate_h */
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXAudioEffectManager.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXAudioEffectManager.h
new file mode 100644
index 0000000..13ce621
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXAudioEffectManager.h
@@ -0,0 +1,331 @@
+/**
+ * Module: management class for background music, short audio effects, and voice effects
+ * Description: sets background music, short audio effects, and voice effects
+ */
+/// @defgroup TXAudioEffectManager_ios TXAudioEffectManager
+/// Tencent Cloud Audio Effect Management Module
+/// @{
+#import
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Definitions of enumerated values related to audio effects
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Definitions of enumerated values related to audio effects
+/// @{
+
+/**
+ * 1.1 Reverb effects
+ *
+ * Reverb effects can be applied to human voice. Based on acoustic algorithms, they can mimic voice in different environments. The following effects are supported currently:
+ * 0: original; 1: karaoke; 2: room; 3: hall; 4: low and deep; 5: resonant; 6: metal; 7: husky; 8: ethereal; 9: studio; 10: melodious; 11: phonograph; 12: nature
+ */
+typedef NS_ENUM(NSInteger, TXVoiceReverbType) {
+ TXVoiceReverbType_0 = 0, ///< Disable
+ TXVoiceReverbType_1 = 1, ///< Karaoke
+ TXVoiceReverbType_2 = 2, ///< Room
+ TXVoiceReverbType_3 = 3, ///< Hall
+ TXVoiceReverbType_4 = 4, ///< Low and deep
+ TXVoiceReverbType_5 = 5, ///< Resonant
+ TXVoiceReverbType_6 = 6, ///< Metal
+ TXVoiceReverbType_7 = 7, ///< Husky
+ TXVoiceReverbType_8 = 8, ///< ethereal
+ TXVoiceReverbType_9 = 9, ///< studio
+ TXVoiceReverbType_10 = 10, ///< melodious
+ TXVoiceReverbType_11 = 11, ///< phonograph
+ TXVoiceReverbType_12 = 12, ///< nature
+};
+
+/**
+ * 1.2 Voice changing effects
+ *
+ * Voice changing effects can be applied to human voice. Based on acoustic algorithms, they change the tone of voice. The following effects are supported currently:
+ * 0: original; 1: child; 2: little girl; 3: middle-aged man; 4: metal; 5: nasal; 6: foreign accent; 7: trapped beast; 8: otaku; 9: electric; 10: robot; 11: ethereal
+ */
+typedef NS_ENUM(NSInteger, TXVoiceChangeType) {
+ TXVoiceChangeType_0 = 0, ///< Disable
+ TXVoiceChangeType_1 = 1, ///< Child
+ TXVoiceChangeType_2 = 2, ///< Little girl
+ TXVoiceChangeType_3 = 3, ///< Middle-aged man
+ TXVoiceChangeType_4 = 4, ///< Metal
+ TXVoiceChangeType_5 = 5, ///< Nasal
+ TXVoiceChangeType_6 = 6, ///< Foreign accent
+ TXVoiceChangeType_7 = 7, ///< Trapped beast
+ TXVoiceChangeType_8 = 8, ///< Otaku
+ TXVoiceChangeType_9 = 9, ///< Electric
+ TXVoiceChangeType_10 = 10, ///< Robot
+ TXVoiceChangeType_11 = 11, ///< Ethereal
+};
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Callback of playing background music
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Callback of playing background music
+/// @{
+
+// Playback progress block of background music
+
+/// Background music started.
+typedef void (^TXAudioMusicStartBlock)(NSInteger errCode);
+
+/// Playback progress of background music
+typedef void (^TXAudioMusicProgressBlock)(NSInteger progressMs, NSInteger durationMs);
+
+/// Background music ended
+typedef void (^TXAudioMusicCompleteBlock)(NSInteger errCode);
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Background music playback information
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Background music playback information
+/// @{
+
+/**
+ * Background music playback information
+ *
+ * The information, including playback ID, file path, and loop times, is passed in the {@link startPlayMusic} API.
+ * 1. If you play the same music track multiple times, please use the same ID instead of a separate ID for each playback.
+ * 2. If you want to play different music tracks at the same time, use different IDs for them.
+ * 3. If you use the same ID to play a music track different from the current one, the SDK will stop the current one before playing the new one.
+ */
+@interface TXAudioMusicParam : NSObject
+
+///**Field description:** music ID
+///**Note:** the SDK supports playing multiple music tracks. IDs are used to distinguish different music tracks and control their start, end, volume, etc.
+@property(nonatomic) int32_t ID;
+
+///**Field description:** absolute path of the music file or url.the mp3,aac,m4a,wav supported.
+@property(nonatomic, copy) NSString *path;
+
+///**Field description:** number of times the music track is looped
+///**Valid values:** 0 or any positive integer. 0 (default) indicates that the music is played once, 1 twice, and so on.
+@property(nonatomic) NSInteger loopCount;
+
+///**Field description:** whether to send the music to remote users
+///**Valid values:** `YES`: remote users can hear the music played locally; `NO` (default): only the local user can hear the music.
+@property(nonatomic) BOOL publish;
+
+///**Field description:** whether the music played is a short music track
+///**Valid values:** `YES`: short music track that needs to be looped; `NO` (default): normal-length music track
+@property(nonatomic) BOOL isShortFile;
+
+///**Field description:** the point in time in milliseconds for starting music playback
+@property(nonatomic) NSInteger startTimeMS;
+
+///**Field description:** the point in time in milliseconds for ending music playback. 0 indicates that playback continues till the end of the music track.
+@property(nonatomic) NSInteger endTimeMS;
+@end
+/// @}
+
+// Definition of audio effect management module
+@interface TXAudioEffectManager : NSObject
+
+/**
+ * You cannot create a `TXAudioEffectManager` object.
+ * You need to obtain the object using the `getAudioEffectManager` API of `TRTCCloud` or `TXLivePush`.
+ */
+- (instancetype)init NS_UNAVAILABLE;
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Voice effect APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Voice effect APIs
+/// @{
+
+/**
+ * 1.1 Enabling in-ear monitoring
+ *
+ * After enabling in-ear monitoring, anchors can hear in earphones their own voice captured by the mic. This is designed for singing scenarios.
+ *
+ * In-ear monitoring cannot be enabled for Bluetooth earphones. This is because Bluetooth earphones have high latency. Please ask anchors to use wired earphones via a UI reminder.
+ * Given that not all phones deliver excellent in-ear monitoring effects, we have blocked this feature on some phones.
+ *
+ * @note In-ear monitoring can be enabled only when earphones are used. Please remind anchors to use wired earphones.
+ * @param enable `YES:` enable; `NO`: disable
+ */
+- (void)enableVoiceEarMonitor:(BOOL)enable;
+
+/**
+ * 1.2 Setting in-ear monitoring volume
+ *
+ * This API is used to set the volume of in-ear monitoring.
+ *
+ * @param volume Volume. Value range: 0-100; default: 100
+ * @note If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
+ */
+- (void)setVoiceEarMonitorVolume:(NSInteger)volume;
+
+/**
+ * 1.3 Setting voice reverb effects
+ *
+ * This API is used to set reverb effects for human voice. For the effects supported, please see {@link TXVoiceReverbType}.
+ *
+ * @note Effects become invalid after room exit. If you want to use the same effect after you enter the room again, you need to set the effect again using this API.
+ */
+- (void)setVoiceReverbType:(TXVoiceReverbType)reverbType;
+
+/**
+ * 1.4 Setting voice changing effects
+ *
+ * This API is used to set voice changing effects. For the effects supported, please see {@link TXVoiceChangeType}.
+ *
+ * @note Effects become invalid after room exit. If you want to use the same effect after you enter the room again, you need to set the effect again using this API.
+ */
+- (void)setVoiceChangerType:(TXVoiceChangeType)changerType;
+
+/**
+ * 1.5 Setting speech volume
+ *
+ * This API is used to set the volume of speech. It is often used together with the music volume setting API {@link setAllMusicVolume} to balance between the volume of music and speech.
+ *
+ * @param volume Volume. Value range: 0-100; default: 100
+ * @note If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
+ */
+- (void)setVoiceVolume:(NSInteger)volume;
+
+/**
+ * 1.6 Setting speech pitch
+ *
+ * This API is used to set the pitch of speech.
+ *
+ * @param pitch Ptich，Value range: -1.0f~1.0f; default: 0.0f。
+ */
+- (void)setVoicePitch:(double)pitch;
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Background music APIs
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Background music APIs
+/// @{
+
+/**
+ * 2.1 Starting background music
+ *
+ * You must assign an ID to each music track so that you can start, stop, or set the volume of music tracks by ID.
+ *
+ * @note
+ * 1. If you play the same music track multiple times, please use the same ID instead of a separate ID for each playback.
+ * 2. If you want to play different music tracks at the same time, use different IDs for them.
+ * 3. If you use the same ID to play a music track different from the current one, the SDK will stop the current one before playing the new one.
+ *
+ * @param musicParam Music parameter
+ * @param startBlock Callback of starting music
+ * @param progressBlock Callback of playback progress
+ * @param completeBlock Callback of ending music
+ */
+- (void)startPlayMusic:(TXAudioMusicParam *)musicParam onStart:(TXAudioMusicStartBlock _Nullable)startBlock onProgress:(TXAudioMusicProgressBlock _Nullable)progressBlock onComplete:(TXAudioMusicCompleteBlock _Nullable)completeBlock;
+
+/**
+ * 2.2 Stopping background music
+ *
+ * @param id Music ID
+ */
+- (void)stopPlayMusic:(int32_t)id;
+
+/**
+ * 2.3 Pausing background music
+ *
+ * @param id Music ID
+ */
+- (void)pausePlayMusic:(int32_t)id;
+
+/**
+ * 2.4 Resuming background music
+ *
+ * @param id Music ID
+ */
+- (void)resumePlayMusic:(int32_t)id;
+
+/**
+ * 2.5 Setting the local and remote playback volume of background music
+ *
+ * This API is used to set the local and remote playback volume of background music.
+ * - Local volume: the volume of music heard by anchors
+ * - Remote volume: the volume of music heard by audience
+ *
+ * @param volume Volume. Value range: 0-100; default: 100
+ * @note If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
+ */
+- (void)setAllMusicVolume:(NSInteger)volume;
+
+/**
+ * 2.6 Setting the remote playback volume of a specific music track
+ *
+ * This API is used to control the remote playback volume (the volume heard by audience) of a specific music track.
+ *
+ * @param id Music ID
+ * @param volume Volume. Value range: 0-100; default: 100
+ * @note If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
+ */
+- (void)setMusicPublishVolume:(int32_t)id volume:(NSInteger)volume;
+
+/**
+ * 2.7 Setting the local playback volume of a specific music track
+ *
+ * This API is used to control the local playback volume (the volume heard by anchors) of a specific music track.
+ *
+ * @param id Music ID
+ * @param volume Volume. Value range: 0-100. default: 100
+ * @note If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
+ */
+- (void)setMusicPlayoutVolume:(int32_t)id volume:(NSInteger)volume;
+
+/**
+ * 2.8 Adjusting the pitch of background music
+ *
+ * @param id Music ID
+ * @param pitch Pitch. Value range: floating point numbers in the range of [-1, 1]; default: 0.0f
+ */
+- (void)setMusicPitch:(int32_t)id pitch:(double)pitch;
+
+/**
+ * 2.9 Changing the speed of background music
+ *
+ * @param id Music ID
+ * @param speedRate Music speed. Value range: floating point numbers in the range of [0.5, 2]; default: 1.0f
+ */
+- (void)setMusicSpeedRate:(int32_t)id speedRate:(double)speedRate;
+
+/**
+ * 2.10 Getting the playback progress (ms) of background music
+ *
+ * @param id Music ID
+ * @return The milliseconds that have passed since playback started. -1 indicates failure to get the the playback progress.
+ */
+- (NSInteger)getMusicCurrentPosInMS:(int32_t)id;
+
+/**
+ * 2.11 Getting the total length (ms) of background music
+ *
+ * @param path Path of the music file.
+ * @return The length of the specified music file is returned. -1 indicates failure to get the length.
+ */
+- (NSInteger)getMusicDurationInMS:(NSString *)path;
+
+/**
+ * 2.12 Setting the playback progress (ms) of background music
+ *
+ * @note Do not call this API frequently as the music file may be read and written to each time the API is called, which can be time-consuming.
+ * Wait till users finish dragging the progress bar before you call this API.
+ * The progress bar controller on the UI tends to update the progress at a high frequency as users drag the progress bar. This will result in poor user experience unless you limit the frequency.
+ *
+ * @param id Music ID
+ * @param pts Unit: millisecond
+ */
+- (void)seekMusicToPosInMS:(int32_t)id pts:(NSInteger)pts;
+
+/// @}
+@end // End of interface TXAudioEffectManager
+/// @}
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXAudioRawDataDelegate.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXAudioRawDataDelegate.h
new file mode 100644
index 0000000..9d834fe
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXAudioRawDataDelegate.h
@@ -0,0 +1,38 @@
+//
+// TXAudioRawDataDelegate.h
+// TXLiteAVSDK
+//
+// Created by realingzhou on 2018/2/24.
+// Copyright © 2018年 Tencent. All rights reserved.
+//
+
+#ifndef TXAudioRawDataDelegate_h
+#define TXAudioRawDataDelegate_h
+
+@protocol TXAudioRawDataDelegate
+
+/**
+ * 音频播放信息回调
+ *
+ * @param sampleRate 采样率
+ * @param channels 声道数
+ */
+@optional
+- (void)onAudioInfoChanged:(int)sampleRate channels:(int)channels;
+
+/**
+ * 音频播放数据回调，数据格式 ：PCM
+ *
+ * <！！！注意！！！> 该函数内不要做耗时操作<！！！注意！！！>
+ * 音频播放器会在播放数据的前一刻，调用此函数，同步回调将要播放的数据。因此在函数内部做耗时操作可能会影响播放
+ *
+ *
+ * @param data pcm数据
+ * @param timestamp 时间戳。注 ：会有连续相同的时间戳回调出来，超过2048字节，时间戳才会变化。
+ */
+@optional
+- (void)onPcmDataAvailable:(NSData *)data pts:(unsigned long long)timestamp;
+
+@end
+
+#endif /* TXAudioRawDataDelegate_h */
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXBeautyManager.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXBeautyManager.h
new file mode 100644
index 0000000..ff08ddb
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXBeautyManager.h
@@ -0,0 +1,328 @@
+/**
+ * Module: beauty filter and image processing parameter configurations
+ * Function: you can modify parameters such as beautification, filter, and green screen
+ */
+#import
+#import
+#if TARGET_OS_IPHONE
+#import
+typedef UIImage TXImage;
+#else
+#import
+typedef NSImage TXImage;
+#endif
+
+NS_ASSUME_NONNULL_BEGIN
+
+/// @defgroup TXBeautyManager_ios TXBeautyManager
+/// Beauty filter and animated effect parameter management
+/// @{
+
+/**
+ * Beauty (skin smoothing) filter algorithm
+ * TRTC has multiple built-in skin smoothing algorithms. You can select the one most suitable for your product needs.
+ */
+typedef NS_ENUM(NSInteger, TXBeautyStyle) {
+
+ /// Smooth style, which uses a more radical algorithm for more obvious effect and is suitable for show live streaming.
+ TXBeautyStyleSmooth = 0,
+
+ /// Natural style, which retains more facial details for more natural effect and is suitable for most live streaming use cases.
+ TXBeautyStyleNature = 1,
+
+ /// Pitu style, which is provided by YouTu Lab. Its skin smoothing effect is between the smooth style and the natural style, that is, it retains more skin details than the smooth style and has a higher skin smoothing degree than the natural
+ /// style.
+ TXBeautyStylePitu = 2
+};
+
+@interface TXBeautyManager : NSObject
+
+/**
+ * Sets the beauty (skin smoothing) filter algorithm.
+ *
+ * TRTC has multiple built-in skin smoothing algorithms. You can select the one most suitable for your product needs:
+ *
+ * @param beautyStyle Beauty filter style. `TXBeautyStyleSmooth`: smooth; `TXBeautyStyleNature`: natural; `TXBeautyStylePitu`: Pitu
+ */
+- (void)setBeautyStyle:(TXBeautyStyle)beautyStyle;
+
+/**
+ * Sets the strength of the beauty filter.
+ *
+ * @param beautyLevel Strength of the beauty filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ */
+- (void)setBeautyLevel:(float)beautyLevel;
+
+/**
+ * Sets the strength of the brightening filter.
+ *
+ * @param whitenessLevel Strength of the brightening filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ */
+- (void)setWhitenessLevel:(float)whitenessLevel;
+
+/**
+ * Enables clarity enhancement.
+ */
+- (void)enableSharpnessEnhancement:(BOOL)enable;
+
+/**
+ * Sets the strength of the rosy skin filter.
+ *
+ * @param ruddyLevel Strength of the rosy skin filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ */
+- (void)setRuddyLevel:(float)ruddyLevel;
+
+/**
+ * Sets color filter.
+ *
+ * The color filter is a color lookup table image containing color mapping relationships. You can find several predefined filter images in the official demo we provide.
+ * The SDK performs secondary processing on the original video image captured by the camera according to the mapping relationships in the lookup table to achieve the expected filter effect.
+ * @param image Color lookup table containing color mapping relationships. The image must be in PNG format.
+ */
+- (void)setFilter:(nullable TXImage *)image;
+
+/**
+ * Sets the strength of color filter.
+ *
+ * The larger this value, the more obvious the effect of the color filter, and the greater the color difference between the video image processed by the filter and the original video image.
+ * The default strength is 0.5, and if it is not sufficient, it can be adjusted to a value above 0.5. The maximum value is 1.
+ *
+ * @param strength Value range: 0–1. The greater the value, the more obvious the effect. Default value: 0.5
+ */
+- (void)setFilterStrength:(float)strength;
+
+/**
+ * Sets green screen video. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * The green screen feature enabled by this API is not capable of intelligent keying. It requires that there be a green screen behind the videoed person or object for further chroma keying.
+ *
+ * @param path Path of the video file in MP4 format. An empty value indicates to disable the effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+- (int)setGreenScreenFile:(nullable NSString *)path;
+
+/**
+ * Sets the strength of the eye enlarging filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param eyeScaleLevel Strength of the eye enlarging filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setEyeScaleLevel:(float)eyeScaleLevel;
+#endif
+
+/**
+ * Sets the strength of the face slimming filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param faceSlimLevel Strength of the face slimming filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setFaceSlimLevel:(float)faceSlimLevel;
+#endif
+
+/**
+ * Sets the strength of the chin slimming filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param faceVLevel Strength of the chin slimming filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setFaceVLevel:(float)faceVLevel;
+#endif
+
+/**
+ * Sets the strength of the chin lengthening/shortening filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param chinLevel Strength of the chin lengthening/shortening filter. Value range: -9–9. `0` indicates to disable the filter, a value smaller than 0 indicates that the chin is shortened, and a value greater than 0 indicates that the chin is
+ * lengthened.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setChinLevel:(float)chinLevel;
+#endif
+
+/**
+ * Sets the strength of the face shortening filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param faceShortLevel Strength of the face shortening filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setFaceShortLevel:(float)faceShortLevel;
+#endif
+
+/**
+ * Sets the strength of the face narrowing filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param level Strength of the face narrowing filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setFaceNarrowLevel:(float)faceNarrowLevel;
+#endif
+
+/**
+ * Sets the strength of the nose slimming filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param noseSlimLevel Strength of the nose slimming filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setNoseSlimLevel:(float)noseSlimLevel;
+#endif
+
+/**
+ * Sets the strength of the eye brightening filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param eyeLightenLevel Strength of the eye brightening filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setEyeLightenLevel:(float)eyeLightenLevel;
+#endif
+
+/**
+ * Sets the strength of the teeth whitening filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param toothWhitenLevel Strength of the teeth whitening filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setToothWhitenLevel:(float)toothWhitenLevel;
+#endif
+
+/**
+ * Sets the strength of the wrinkle removal filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param wrinkleRemoveLevel Strength of the wrinkle removal filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setWrinkleRemoveLevel:(float)wrinkleRemoveLevel;
+#endif
+
+/**
+ * Sets the strength of the eye bag removal filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param pounchRemoveLevel Strength of the eye bag removal filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setPounchRemoveLevel:(float)pounchRemoveLevel;
+#endif
+
+/**
+ * Sets the strength of the smile line removal filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param smileLinesRemoveLevel Strength of the smile line removal filter. Value range: 0–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setSmileLinesRemoveLevel:(float)smileLinesRemoveLevel;
+#endif
+
+/**
+ * Sets the strength of the hairline adjustment filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param foreheadLevel Strength of the hairline adjustment filter. Value range: -9–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setForeheadLevel:(float)foreheadLevel;
+#endif
+
+/**
+ * Sets the strength of the eye distance adjustment filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param eyeDistanceLevel Strength of the eye distance adjustment filter. Value range: -9–9. `0` indicates to disable the filter, a value smaller than 0 indicates to widen, and a value greater than 0 indicates to narrow.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setEyeDistanceLevel:(float)eyeDistanceLevel;
+#endif
+
+/**
+ * Sets the strength of the eye corner adjustment filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param eyeAngleLevel Strength of the eye corner adjustment filter. Value range: -9–9. `0` indicates to disable the filter, and `9` indicates the most obvious effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setEyeAngleLevel:(float)eyeAngleLevel;
+#endif
+
+/**
+ * Sets the strength of the mouth shape adjustment filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param mouthShapeLevel Strength of the mouth shape adjustment filter. Value range: -9–9. `0` indicates to disable the filter, a value smaller than 0 indicates to widen, and a value greater than 0 indicates to narrow.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setMouthShapeLevel:(float)mouthShapeLevel;
+#endif
+
+/**
+ * Sets the strength of the nose wing narrowing filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param noseWingLevel Strength of the nose wing adjustment filter. Value range: -9–9. `0` indicates to disable the filter, a value smaller than 0 indicates to widen, and a value greater than 0 indicates to narrow.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setNoseWingLevel:(float)noseWingLevel;
+#endif
+
+/**
+ * Sets the strength of the nose position adjustment filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param nosePositionLevel Strength of the nose position adjustment filter. Value range: -9–9. `0` indicates to disable the filter, a value smaller than 0 indicates to lift, and a value greater than 0 indicates to lower.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setNosePositionLevel:(float)nosePositionLevel;
+#endif
+
+/**
+ * Sets the strength of the lip thickness adjustment filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param lipsThicknessLevel Strength of the lip thickness adjustment filter. Value range: -9–9. `0` indicates to disable the filter, a value smaller than 0 indicates to thicken, and a value greater than 0 indicates to thin.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setLipsThicknessLevel:(float)lipsThicknessLevel;
+#endif
+
+/**
+ * Sets the strength of the face shape adjustment filter. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param faceBeautyLevel Strength of the face shape adjustment filter. Value range: 0–9. `0` indicates to disable the filter, and the greater the value, the more obvious the effect.
+ * @return 0: Success; -5: feature of license not supported.
+ */
+#if TARGET_OS_IPHONE
+- (int)setFaceBeautyLevel:(float)faceBeautyLevel;
+#endif
+
+/**
+ * Selects the AI animated effect pendant. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ *
+ * @param tmplName Animated effect pendant name
+ * @param tmplDir Directory of the animated effect material file
+ */
+#if TARGET_OS_IPHONE
+- (void)setMotionTmpl:(nullable NSString *)tmplName inDir:(nullable NSString *)tmplDir;
+#endif
+
+/**
+ * Sets whether to mute during animated effect playback. This API takes effect only in the [Enterprise Edition SDK](https://cloud.tencent.com/document/product/647/32689#Enterprise).
+ * Some animated effects have audio effects, which can be disabled through this API when they are played back.
+ *
+ * @param motionMute `YES`: mute; `NO`: unmute
+ */
+#if TARGET_OS_IPHONE
+- (void)setMotionMute:(BOOL)motionMute;
+#endif
+
+@end
+/// @}
+
+NS_ASSUME_NONNULL_END
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXDeviceManager.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXDeviceManager.h
new file mode 100644
index 0000000..30bf73c
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXDeviceManager.h
@@ -0,0 +1,354 @@
+/**
+ * Module: audio/video device management module
+ * Description: manages audio/video devices such as camera, mic, and speaker.
+ */
+/// @defgroup TXDeviceManager_ios TXDeviceManager
+/// Tencent Cloud Device Management Module
+/// @{
+#import
+#if TARGET_OS_IPHONE
+#import
+#elif TARGET_OS_MAC
+#import
+#endif
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Type definitions of audio/video devices
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Type definitions of audio/video devices
+/// @{
+
+/**
+ * System volume type (for mobile devices only)
+ *
+ * @deprecated This API is not recommended after v9.5.
+ *
+ * Smartphones usually have two types of system volume: call volume and media volume.
+ * - Call volume is designed for call scenarios. It comes with acoustic echo cancellation (AEC) and supports audio capturing by Bluetooth earphones, but its sound quality is average.
+ * If you cannot turn the volume down to 0 (i.e., mute the phone) using the volume buttons, then your phone is using call volume.
+ * - Media volume is designed for media scenarios such as music playback. AEC does not work when media volume is used, and Bluetooth earphones cannot be used for audio capturing. However, media volume delivers better music listening experience.
+ * If you are able to mute your phone using the volume buttons, then your phone is using media volume.
+ *
+ * The SDK offers three system volume control modes: auto, call volume, and media volume.
+ */
+#if TARGET_OS_IPHONE
+typedef NS_ENUM(NSInteger, TXSystemVolumeType) {
+
+ /// Auto
+ TXSystemVolumeTypeAuto = 0,
+
+ /// Media volume
+ TXSystemVolumeTypeMedia = 1,
+
+ /// Call volume
+ TXSystemVolumeTypeVOIP = 2,
+
+};
+#endif
+
+/**
+ * Audio route (the route via which audio is played)
+ *
+ * Audio route is the route (speaker or receiver) via which audio is played. It applies only to mobile devices such as mobile phones.
+ * A mobile phone has two speakers: one at the top (receiver) and the other the bottom.
+ * - If the audio route is set to the receiver, the volume is relatively low, and audio can be heard only when the phone is put near the ear. This mode has a high level of privacy and is suitable for answering calls.
+ * - If the audio route is set to the speaker, the volume is relatively high, and there is no need to put the phone near the ear. This mode enables the "hands-free" feature.
+ */
+#if TARGET_OS_IPHONE
+typedef NS_ENUM(NSInteger, TXAudioRoute) {
+
+ /// Speakerphone: the speaker at the bottom is used for playback (hands-free). With relatively high volume, it is used to play music out loud.
+ TXAudioRouteSpeakerphone = 0,
+
+ /// Earpiece: the receiver at the top is used for playback. With relatively low volume, it is suitable for call scenarios that require privacy.
+ TXAudioRouteEarpiece = 1,
+
+};
+#endif
+
+/**
+ * Device type (for desktop OS)
+ *
+ * This enumerated type defines three types of audio/video devices, namely camera, mic and speaker, so that you can use the same device management API to manage three types of devices.
+ */
+#if TARGET_OS_MAC && !TARGET_OS_IPHONE
+typedef NS_ENUM(NSInteger, TXMediaDeviceType) {
+ TXMediaDeviceTypeUnknown = -1, ///< undefined device type
+ TXMediaDeviceTypeAudioInput = 0, ///< microphone
+ TXMediaDeviceTypeAudioOutput = 1, ///< speaker or earpiece
+ TXMediaDeviceTypeVideoCamera = 2, ///< camera
+};
+#endif
+
+/**
+ * Device operation
+ *
+ * This enumerated value is used to notify the status change of the local device {@link onDeviceChanged}.
+ */
+#if TARGET_OS_MAC && !TARGET_OS_IPHONE
+typedef NS_ENUM(NSInteger, TXMediaDeviceState) {
+
+ /// The device has been plugged in
+ TXMediaDeviceStateAdd = 0,
+
+ /// The device has been removed
+ TXMediaDeviceStateRemove = 1,
+
+ /// The device has been enabled
+ TXMediaDeviceStateActive = 2,
+
+};
+#endif
+
+/**
+ * Audio/Video device information (for desktop OS)
+ *
+ * This structure describes key information (such as device ID and device name) of an audio/video device, so that users can choose on the UI the device to use.
+ */
+#if TARGET_OS_MAC && !TARGET_OS_IPHONE
+@interface TXMediaDeviceInfo : NSObject
+/// device type
+@property(assign, nonatomic) TXMediaDeviceType type;
+/// device id
+@property(copy, nonatomic, nullable) NSString *deviceId;
+/// device name
+@property(copy, nonatomic, nullable) NSString *deviceName;
+/// device properties
+@property(copy, nonatomic, nullable) NSString *deviceProperties;
+@end
+#endif
+/// @}
+
+/**
+ * The status of a local device changed (for desktop OS only)
+ *
+ * The SDK returns this callback when a local device (camera, mic, or speaker) is connected or disconnected.
+ *
+ * @param deviceId Device ID
+ * @param type Device type
+ * @param state Device status. `0`: connected; `1`: disconnected; `2`: started
+ */
+#if TARGET_OS_MAC && !TARGET_OS_IPHONE
+@protocol TXDeviceObserver
+
+- (void)onDeviceChanged:(NSString *)deviceId type:(TXMediaDeviceType)mediaType state:(TXMediaDeviceState)mediaState;
+
+@end
+#endif
+
+@interface TXDeviceManager : NSObject
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Device APIs for mobile OS (iOS and Android)
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Device APIs for mobile OS
+/// @{
+
+/**
+ * 1.1 Querying whether the front camera is being used
+ */
+#if TARGET_OS_IPHONE
+- (BOOL)isFrontCamera;
+
+/**
+ * 1.2 Switching to the front/rear camera (for mobile OS)
+ */
+- (NSInteger)switchCamera:(BOOL)frontCamera;
+
+/**
+ * 1.3 Querying whether the current camera supports zooming (for mobile OS)
+ */
+- (BOOL)isCameraZoomSupported;
+
+/**
+ * 1.3 Getting the maximum zoom ratio of the camera (for mobile OS)
+ */
+- (CGFloat)getCameraZoomMaxRatio;
+
+/**
+ * 1.4 Setting the camera zoom ratio (for mobile OS)
+ *
+ * @param zoomRatio Value range: 1-5. 1 indicates the widest angle of view (original), and 5 the narrowest angle of view (zoomed in).
+ */
+- (NSInteger)setCameraZoomRatio:(CGFloat)zoomRatio;
+
+/**
+ * 1.5 Querying whether automatic face detection is supported (for mobile OS)
+ */
+- (BOOL)isAutoFocusEnabled;
+
+/**
+ * 1.6 Enabling auto focus (for mobile OS)
+ *
+ * After auto focus is enabled, the camera will automatically detect and always focus on faces.
+ */
+- (NSInteger)enableCameraAutoFocus:(BOOL)enabled;
+
+/**
+ * 1.7 Adjusting the focus (for mobile OS)
+ *
+ * This API can be used to achieve the following:
+ * 1. A user can tap on the camera preview.
+ * 2. A rectangle will appear where the user taps, indicating the spot the camera will focus on.
+ * 3. The user passes the coordinates of the spot to the SDK using this API, and the SDK will instruct the camera to focus as required.
+ * @note Before using this API, you must first disable auto focus using {@link enableCameraAutoFocus}.
+ * @param position The spot to focus on. Pass in the coordinates of the spot you want to focus on.
+ * @return 0: operation successful; negative number: operation failed.
+ */
+- (NSInteger)setCameraFocusPosition:(CGPoint)position;
+
+/**
+ * 1.8 Querying whether flash is supported (for mobile OS)
+ */
+- (BOOL)isCameraTorchSupported;
+
+/**
+ * 1.8 Enabling/Disabling flash, i.e., the torch mode (for mobile OS)
+ */
+- (NSInteger)enableCameraTorch:(BOOL)enabled;
+
+/**
+ * 1.9 Setting the audio route (for mobile OS)
+ *
+ * A mobile phone has two audio playback devices: the receiver at the top and the speaker at the bottom.
+ * If the audio route is set to the receiver, the volume is relatively low, and audio can be heard only when the phone is put near the ear. This mode has a high level of privacy and is suitable for answering calls.
+ * If the audio route is set to the speaker, the volume is relatively high, and there is no need to put the phone near the ear. This mode enables the "hands-free" feature.
+ */
+- (NSInteger)setAudioRoute:(TXAudioRoute)route;
+#endif
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Device APIs for desktop OS (Windows & macOS)
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Device APIs for desktop OS
+/// @{
+
+/**
+ * 2.1 Getting the device list (for desktop OS)
+ *
+ * @param type Device type. Set it to the type of device you want to get. For details, please see the definition of `TXMediaDeviceType`.
+ * @note
+ * - To ensure that the SDK can manage the lifecycle of the `ITXDeviceCollection` object, after using this API, please call the `release` method to release the resources.
+ * - Do not use `delete` to release the Collection object returned as deleting the ITXDeviceCollection* pointer will cause crash.
+ * - The valid values of `type` are `TXMediaDeviceTypeMic`, `TXMediaDeviceTypeSpeaker`, and `TXMediaDeviceTypeCamera`.
+ * - This API can be used only on macOS and Windows.
+ */
+#if !TARGET_OS_IPHONE && TARGET_OS_MAC
+- (NSArray *_Nullable)getDevicesList:(TXMediaDeviceType)type;
+
+/**
+ * 2.2 Setting the device to use (for desktop OS)
+ *
+ * @param type Device type. For details, please see the definition of `TXMediaDeviceType`.
+ * @param deviceId Device ID. You can get the ID of a device using the {@link getDevicesList} API.
+ * @return 0: operation successful; negative number: operation failed.
+ */
+- (NSInteger)setCurrentDevice:(TXMediaDeviceType)type deviceId:(NSString *)deviceId;
+
+/**
+ * 2.3 Getting the device currently in use (for desktop OS)
+ */
+- (TXMediaDeviceInfo *_Nullable)getCurrentDevice:(TXMediaDeviceType)type;
+
+/**
+ * 2.4 Setting the volume of the current device (for desktop OS)
+ *
+ * This API is used to set the capturing volume of the mic or playback volume of the speaker, but not the volume of the camera.
+ * @param volume Volume. Value range: 0-100; default: 100
+ * @note If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
+ */
+- (NSInteger)setCurrentDeviceVolume:(NSInteger)volume deviceType:(TXMediaDeviceType)type;
+
+/**
+ * 2.5 Getting the volume of the current device (for desktop OS)
+ *
+ * This API is used to get the capturing volume of the mic or playback volume of the speaker, but not the volume of the camera.
+ */
+- (NSInteger)getCurrentDeviceVolume:(TXMediaDeviceType)type;
+
+/**
+ * 2.6 Muting the current device (for desktop OS)
+ *
+ * This API is used to mute the mic or speaker, but not the camera.
+ */
+- (NSInteger)setCurrentDeviceMute:(BOOL)mute deviceType:(TXMediaDeviceType)type;
+
+/**
+ * 2.7 Querying whether the current device is muted (for desktop OS)
+ *
+ * This API is used to query whether the mic or speaker is muted. Camera muting is not supported.
+ */
+- (BOOL)getCurrentDeviceMute:(TXMediaDeviceType)type;
+
+/**
+ * 2.8 Starting camera testing (for desktop OS)
+ *
+ * @note You can use the {@link setCurrentDevice} API to switch between cameras during testing.
+ */
+- (NSInteger)startCameraDeviceTest:(NSView *)view;
+
+/**
+ * 2.9 Ending camera testing (for desktop OS)
+ */
+- (NSInteger)stopCameraDeviceTest;
+
+/**
+ * 2.10 Starting mic testing (for desktop OS)
+ *
+ * This API is used to test whether the mic functions properly. The mic volume detected (value range: 0-100) is returned via a callback.
+ * @param interval Interval of volume callbacks
+ */
+- (NSInteger)startMicDeviceTest:(NSInteger)interval testEcho:(void (^)(NSInteger volume))testEcho;
+
+/**
+ * 2.11 Ending mic testing (for desktop OS)
+ */
+- (NSInteger)stopMicDeviceTest;
+
+/**
+ * 2.12 Starting speaker testing (for desktop OS)
+ *
+ * This API is used to test whether the audio playback device functions properly by playing a specified audio file. If users can hear audio during testing, the device functions properly.
+ * @param filePath Path of the audio file
+ */
+- (NSInteger)startSpeakerDeviceTest:(NSString *)audioFilePath onVolumeChanged:(void (^)(NSInteger volume, BOOL isLastFrame))volumeBlock;
+
+/**
+ * 2.13 Ending speaker testing (for desktop OS)
+ */
+- (NSInteger)stopSpeakerDeviceTest;
+
+/**
+ * 2.14 set onDeviceChanged callback (for Mac)
+ */
+- (void)setObserver:(nullable id)observer;
+#endif
+
+/// @}
+/////////////////////////////////////////////////////////////////////////////////
+//
+// Disused APIs (the corresponding new APIs are recommended)
+//
+/////////////////////////////////////////////////////////////////////////////////
+/// @name Disused APIs (the corresponding new APIs are recommended)
+/// @{
+
+/**
+ * Setting the system volume type (for mobile OS)
+ *
+ * @deprecated This API is not recommended after v9.5. Please use the `startLocalAudio(quality)` API in {@link TRTCCloud} instead, which param `quality` is used to decide audio quality.
+ */
+#if TARGET_OS_IPHONE
+- (NSInteger)setSystemVolumeType:(TXSystemVolumeType)type __attribute__((deprecated("use TRTCCloud#startLocalAudio:quality instead")));
+#endif
+
+/// @}
+@end
+/// @}
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVBuffer.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVBuffer.h
new file mode 100644
index 0000000..5ca84ea
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVBuffer.h
@@ -0,0 +1,49 @@
+
+
+#ifndef TXLiteAVBuffer_h
+#define TXLiteAVBuffer_h
+
+#include
+#include
+#include
+
+namespace liteav {
+
+/**
+ * Buffer 数据类型
+ */
+class TXLiteAVBuffer {
+
+public:
+ virtual ~TXLiteAVBuffer() {}
+
+ /**
+ * 获取 buffer 的内存地址
+ */
+ virtual uint8_t * data() = 0;
+
+ /**
+ * 获取 buffer 的内存地址
+ */
+ virtual const uint8_t * cdata() const = 0;
+
+ /**
+ * 获取 buffer 的内存size
+ */
+ virtual size_t size() const = 0;
+
+ /**
+ * 设置 buffe 的有效数据 size
+ * 如果此 size 超过当前 capacity，会造成重新分配内存，并复制数据
+ */
+ virtual void SetSize(size_t size) = 0;
+
+ /**
+ * 确保 buffer 分配的内存空间足够，不用多次分配拷贝内存。此方法会引起内存分配，data / cdata 方法获取的指针失效
+ * @param capacity buffer 预分配的内存size
+ */
+ virtual void EnsureCapacity(size_t capacity) = 0;
+};
+}
+
+#endif /* TXLiteAVBuffer_h */
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVCode.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVCode.h
new file mode 100644
index 0000000..d518a35
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVCode.h
@@ -0,0 +1,430 @@
+#ifndef __TXLITEAVCODE_H__
+#define __TXLITEAVCODE_H__
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// 错误码
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+typedef enum TXLiteAVError
+{
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 基础错误码
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_NULL = 0, ///< 无错误
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 进房（enterRoom）相关错误码
+ // NOTE: 通过回调函数 TRTCCloudDelegate##onEnterRoom() 和 TRTCCloudDelegate##OnError() 通知
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_ROOM_ENTER_FAIL = -3301, ///< 进入房间失败，请查看 onError 中的 -3301 对应的 msg 提示确认失败原因
+ ERR_ROOM_REQUEST_IP_TIMEOUT = -3307, ///< 请求 IP 和 sig 超时，请检查网络是否正常，或网络防火墙是否放行 UDP。可尝试访问下列 IP：162.14.22.165:8000 162.14.6.105:8000 和域名：default-query.trtc.tencent-cloud.com:8000
+ ERR_ROOM_REQUEST_ENTER_ROOM_TIMEOUT = -3308, ///< 请求进房超时，请检查是否断网或者是否开启vpn，您也可以切换4G进行测试确认
+ ERR_ENTER_ROOM_PARAM_NULL = -3316, ///< 进房参数为空，请检查： enterRoom:appScene: 接口调用是否传入有效的 param
+ ERR_SDK_APPID_INVALID = -3317, ///< 进房参数 sdkAppId 错误，请检查 TRTCParams.sdkAppId 是否为空
+ ERR_ROOM_ID_INVALID = -3318, ///< 进房参数 roomId 错误，请检查 TRTCParams.roomId 或 TRTCParams.strRoomId 是否为空，注意 roomId 和 strRoomId 不可混用
+ ERR_USER_ID_INVALID = -3319, ///< 进房参数 userId 不正确，请检查 TRTCParams.userId 是否为空
+ ERR_USER_SIG_INVALID = -3320, ///< 进房参数 userSig 不正确，请检查 TRTCParams.userSig 是否为空
+ ERR_ROOM_REQUEST_ENTER_ROOM_REFUSED = -3340, ///< 进房请求被拒绝，请检查是否连续调用 enterRoom 进入相同 Id 的房间
+ ERR_SERVER_INFO_PRIVILEGE_FLAG_ERROR = -100006, ///< 您开启了高级权限控制，但参数 TRTCParams.privateMapKey 校验失败，您可参考 https://cloud.tencent.com/document/product/647/32240 进行检查
+ ERR_SERVER_INFO_SERVICE_SUSPENDED = -100013, ///< 服务不可用。请检查：套餐包剩余分钟数是否大于0，腾讯云账号是否欠费。您可参考 https://cloud.tencent.com/document/product/647/50492 进行查看与配置
+ ERR_SERVER_INFO_ECDH_GET_TINYID = -100018, ///< UserSig 校验失败，请检查参数 TRTCParams.userSig 是否填写正确，或是否已经过期。您可参考 https://cloud.tencent.com/document/product/647/50686 进行校验
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 退房（exitRoom）相关错误码
+ // NOTE: 通过回调函数 TRTCCloudDelegate##OnError() 通知
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_ROOM_REQUEST_QUIT_ROOM_TIMEOUT = -3325, ///< 请求退房超时
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 设备（摄像头、麦克风、扬声器）相关错误码
+ // NOTE: 通过回调函数 TRTCCloudDelegate##OnError() 通知
+ // 区段：-6000 ~ -6999
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_CAMERA_START_FAIL = -1301, ///< 打开摄像头失败，例如在 Windows 或 Mac 设备，摄像头的配置程序（驱动程序）异常，禁用后重新启用设备，或者重启机器，或者更新配置程序
+ ERR_CAMERA_NOT_AUTHORIZED = -1314, ///< 摄像头设备未授权，通常在移动设备出现，可能是权限被用户拒绝了
+ ERR_CAMERA_SET_PARAM_FAIL = -1315, ///< 摄像头参数设置出错（参数不支持或其它）
+ ERR_CAMERA_OCCUPY = -1316, ///< 摄像头正在被占用中，可尝试打开其他摄像头
+ ERR_MIC_START_FAIL = -1302, ///< 打开麦克风失败，例如在 Windows 或 Mac 设备，麦克风的配置程序（驱动程序）异常，禁用后重新启用设备，或者重启机器，或者更新配置程序
+ ERR_MIC_NOT_AUTHORIZED = -1317, ///< 麦克风设备未授权，通常在移动设备出现，可能是权限被用户拒绝了
+ ERR_MIC_SET_PARAM_FAIL = -1318, ///< 麦克风设置参数失败
+ ERR_MIC_OCCUPY = -1319, ///< 麦克风正在被占用中，例如移动设备正在通话时，打开麦克风会失败
+ ERR_MIC_STOP_FAIL = -1320, ///< 停止麦克风失败
+ ERR_SPEAKER_START_FAIL = -1321, ///< 打开扬声器失败，例如在 Windows 或 Mac 设备，扬声器的配置程序（驱动程序）异常，禁用后重新启用设备，或者重启机器，或者更新配置程序
+ ERR_SPEAKER_SET_PARAM_FAIL = -1322, ///< 扬声器设置参数失败
+ ERR_SPEAKER_STOP_FAIL = -1323, ///< 停止扬声器失败
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 系统声音采集相关错误码
+ // NOTE: 通过回调函数 TRTCCloudDelegate##onSystemAudioLoopbackError() 通知
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_AUDIO_PLUGIN_START_FAIL = -1330, ///< 开启系统声音录制失败，例如音频驱动插件不可用
+ ERR_AUDIO_PLUGIN_INSTALL_NOT_AUTHORIZED = -1331, ///< 安装音频驱动插件未授权
+ ERR_AUDIO_PLUGIN_INSTALL_FAILED = -1332, ///< 安装音频驱动插件失败
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 屏幕分享相关错误码
+ // NOTE: 通过回调函数 TRTCCloudDelegate##OnError() 通知
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_SCREEN_CAPTURE_START_FAIL = -1308, ///< 开始录屏失败，如果在移动设备出现，可能是权限被用户拒绝了，如果在 Windows 或 Mac 系统的设备出现，请检查录屏接口的参数是否符合要求
+ ERR_SCREEN_CAPTURE_UNSURPORT = -1309, ///< 录屏失败，在 Android 平台，需要5.0以上的系统，在 iOS 平台，需要11.0以上的系统
+ ERR_SERVER_CENTER_NO_PRIVILEDGE_PUSH_SUB_VIDEO = -102015, ///< 没有权限上行辅路
+ ERR_SERVER_CENTER_ANOTHER_USER_PUSH_SUB_VIDEO = -102016, ///< 其他用户正在上行辅路
+ ERR_SCREEN_CAPTURE_STOPPED = -7001, ///< 录屏被系统中止
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 编解码相关错误码
+ // NOTE: 通过回调函数 TRTCCloudDelegate##OnError() 通知
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_VIDEO_ENCODE_FAIL = -1303, ///< 视频帧编码失败，例如 iOS 设备切换到其他应用时，硬编码器可能被系统释放，再切换回来时，硬编码器重启前，可能会抛出
+ ERR_UNSUPPORTED_RESOLUTION = -1305, ///< 不支持的视频分辨率
+ ERR_AUDIO_ENCODE_FAIL = -1304, ///< 音频帧编码失败，例如传入自定义音频数据，SDK 无法处理
+ ERR_UNSUPPORTED_SAMPLERATE = -1306, ///< 不支持的音频采样率
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 自定义采集相关错误码
+ // NOTE: 通过回调函数 TRTCCloudDelegate##OnError() 通知
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_PIXEL_FORMAT_UNSUPPORTED = -1327, ///< 设置的 pixel format 不支持
+ ERR_BUFFER_TYPE_UNSUPPORTED = -1328, ///< 设置的 buffer type 不支持
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // CDN 绑定和混流相关错误码
+ // NOTE: 通过回调函数 TRTCCloudDelegate##onStartPublishing() 和 TRTCCloudDelegate##onSetMixTranscodingConfig 通知。
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_PUBLISH_CDN_STREAM_REQUEST_TIME_OUT = -3321, ///< 旁路转推请求超时
+ ERR_CLOUD_MIX_TRANSCODING_REQUEST_TIME_OUT = -3322, ///< 云端混流请求超时
+ ERR_PUBLISH_CDN_STREAM_SERVER_FAILED = -3323, ///< 旁路转推回包异常
+ ERR_CLOUD_MIX_TRANSCODING_SERVER_FAILED = -3324, ///< 云端混流回包异常
+ ERR_ROOM_REQUEST_START_PUBLISHING_TIMEOUT = -3333, ///< 开始向腾讯云的直播 CDN 推流信令超时
+ ERR_ROOM_REQUEST_START_PUBLISHING_ERROR = -3334, ///< 开始向腾讯云的直播 CDN 推流信令异常
+ ERR_ROOM_REQUEST_STOP_PUBLISHING_TIMEOUT = -3335, ///< 停止向腾讯云的直播 CDN 推流信令超时
+ ERR_ROOM_REQUEST_STOP_PUBLISHING_ERROR = -3336, ///< 停止向腾讯云的直播 CDN 推流信令异常
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 跨房连麦（ConnectOtherRoom）相关错误码
+ // NOTE: 通过回调函数 TRTCCloudDelegate##onConnectOtherRoom() 通知。
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+ ERR_ROOM_REQUEST_CONN_ROOM_TIMEOUT = -3326, ///< 请求连麦超时
+ ERR_ROOM_REQUEST_DISCONN_ROOM_TIMEOUT = -3327, ///< 请求退出连麦超时
+ ERR_ROOM_REQUEST_CONN_ROOM_INVALID_PARAM = -3328, ///< 无效参数
+ ERR_CONNECT_OTHER_ROOM_AS_AUDIENCE = -3330, ///< 当前是观众角色，不能请求或断开跨房连麦，需要先 switchRole() 到主播
+ ERR_SERVER_CENTER_CONN_ROOM_NOT_SUPPORT = -102031, ///< 不支持跨房间连麦
+ ERR_SERVER_CENTER_CONN_ROOM_REACH_MAX_NUM = -102032, ///< 达到跨房间连麦上限
+ ERR_SERVER_CENTER_CONN_ROOM_REACH_MAX_RETRY_TIMES = -102033, ///< 跨房间连麦重试次数耗尽
+ ERR_SERVER_CENTER_CONN_ROOM_REQ_TIMEOUT = -102034, ///< 跨房间连麦请求超时
+ ERR_SERVER_CENTER_CONN_ROOM_REQ = -102035, ///< 跨房间连麦请求格式错误
+ ERR_SERVER_CENTER_CONN_ROOM_NO_SIG = -102036, ///< 跨房间连麦无签名
+ ERR_SERVER_CENTER_CONN_ROOM_DECRYPT_SIG = -102037, ///< 跨房间连麦签名解密失败
+ ERR_SERVER_CENTER_CONN_ROOM_NO_KEY = -102038, ///< 未找到跨房间连麦签名解密密钥
+ ERR_SERVER_CENTER_CONN_ROOM_PARSE_SIG = -102039, ///< 跨房间连麦签名解析错误
+ ERR_SERVER_CENTER_CONN_ROOM_INVALID_SIG_TIME = -102040, ///< 跨房间连麦签名时间戳错误
+ ERR_SERVER_CENTER_CONN_ROOM_SIG_GROUPID = -102041, ///< 跨房间连麦签名不匹配
+ ERR_SERVER_CENTER_CONN_ROOM_NOT_CONNED = -102042, ///< 本房间无连麦
+ ERR_SERVER_CENTER_CONN_ROOM_USER_NOT_CONNED = -102043, ///< 本用户未发起连麦
+ ERR_SERVER_CENTER_CONN_ROOM_FAILED = -102044, ///< 跨房间连麦失败
+ ERR_SERVER_CENTER_CONN_ROOM_CANCEL_FAILED = -102045, ///< 取消跨房间连麦失败
+ ERR_SERVER_CENTER_CONN_ROOM_CONNED_ROOM_NOT_EXIST = -102046, ///< 被连麦房间不存在
+ ERR_SERVER_CENTER_CONN_ROOM_CONNED_REACH_MAX_ROOM = -102047, ///< 被连麦房间达到连麦上限
+ ERR_SERVER_CENTER_CONN_ROOM_CONNED_USER_NOT_EXIST = -102048, ///< 被连麦用户不存在
+ ERR_SERVER_CENTER_CONN_ROOM_CONNED_USER_DELETED = -102049, ///< 被连麦用户已被删除
+ ERR_SERVER_CENTER_CONN_ROOM_CONNED_USER_FULL = -102050, ///< 被连麦用户达到资源上限
+ ERR_SERVER_CENTER_CONN_ROOM_INVALID_SEQ = -102051, ///< 连麦请求序号错乱
+
+ /////////////////////////////////////////////////////////////////////////////////
+ //
+ // 客户无需关心的内部错误码
+ //
+ /////////////////////////////////////////////////////////////////////////////////
+
+ // - Remove From Head
+ ERR_RTMP_PUSH_NET_DISCONNECT = -1307, ///< 直播，推流出现网络断开，且经过多次重试无法恢复
+ ERR_RTMP_PUSH_INVALID_ADDRESS = -1313, ///< 直播，推流地址非法，例如不是 RTMP 协议的地址
+ ERR_RTMP_PUSH_NET_ALLADDRESS_FAIL = -1324, ///< 直播，连接推流服务器失败（若支持智能选路，IP 全部失败）
+ ERR_RTMP_PUSH_NO_NETWORK = -1325, ///< 直播，网络不可用，请确认 Wi-Fi、移动数据或者有线网络是否正常
+ ERR_RTMP_PUSH_SERVER_REFUSE = -1326, ///< 直播，服务器拒绝连接请求，可能是该推流地址已经被占用，或者 TXSecret 校验失败，或者是过期了，或者是欠费了
+
+ ERR_PLAY_LIVE_STREAM_NET_DISCONNECT = -2301, ///< 直播，网络断连，且经多次重连抢救无效，可以放弃治疗，更多重试请自行重启播放
+ ERR_GET_RTMP_ACC_URL_FAIL = -2302, ///< 直播，获取加速拉流的地址失败
+ ERR_FILE_NOT_FOUND = -2303, ///< 播放的文件不存在
+ ERR_HEVC_DECODE_FAIL = -2304, ///< H265 解码失败
+ ERR_VOD_DECRYPT_FAIL = -2305, ///< 点播，音视频流解密失败
+ ERR_GET_VODFILE_MEDIAINFO_FAIL = -2306, ///< 点播，获取点播文件信息失败
+ ERR_PLAY_LIVE_STREAM_SWITCH_FAIL = -2307, ///< 直播，切流失败（切流可以播放不同画面大小的视频）
+ ERR_PLAY_LIVE_STREAM_SERVER_REFUSE = -2308, ///< 直播，服务器拒绝连接请求
+ ERR_RTMP_ACC_FETCH_STREAM_FAIL = -2309, ///< 直播，RTMPACC 低延时拉流失败，且经过多次重试无法恢复
+ ERR_HEVC_ENCODE_FAIL = -2310, ///< 265编码失败
+ ERR_HEVC_ENCODE_NOT_SUPPORT = -2311, ///< 265编码判断不支持
+ ERR_HEVC_SOFTDECODER_START_FAIL = -2312, ///< 265软解启动失败
+
+ ERR_ROOM_HEARTBEAT_FAIL = -3302, ///< 心跳失败，客户端定时向服务器发送数据包，告诉服务器自己活着，这个错误通常是发包超时
+ ERR_ROOM_REQUEST_IP_FAIL = -3303, ///< 拉取接口机服务器地址失败
+ ERR_ROOM_CONNECT_FAIL = -3304, ///< 连接接口机服务器失败
+ ERR_ROOM_REQUEST_AVSEAT_FAIL = -3305, ///< 请求视频位失败
+ ERR_ROOM_REQUEST_TOKEN_HTTPS_TIMEOUT = -3306, ///< 请求 token HTTPS 超时，请检查网络是否正常，或网络防火墙是否放行 HTTPS 访问 official.opensso.tencent-cloud.com:443
+ ERR_ROOM_REQUEST_VIDEO_FLAG_TIMEOUT = -3309, ///< 请求视频位超时
+ ERR_ROOM_REQUEST_VIDEO_DATA_ROOM_TIMEOUT = -3310, ///< 请求视频数据超时
+ ERR_ROOM_REQUEST_CHANGE_ABILITY_TIMEOUT = -3311, ///< 请求修改视频能力项超时
+ ERR_ROOM_REQUEST_STATUS_REPORT_TIMEOUT = -3312, ///< 请求状态上报超时
+ ERR_ROOM_REQUEST_CLOSE_VIDEO_TIMEOUT = -3313, ///< 请求关闭视频超时
+ ERR_ROOM_REQUEST_SET_RECEIVE_TIMEOUT = -3314, ///< 请求接收视频项超时
+ ERR_ROOM_REQUEST_TOKEN_INVALID_PARAMETER = -3315, ///< 请求 token 无效参数，请检查 TRTCParams.userSig 是否填写正确
+ ERR_ROOM_REQUEST_EXIT_ROOM_WHEN_ENTERING_ROOM = -3341, ///< 进房尚未成功时，收到了退房请求
+
+ ERR_ROOM_REQUEST_AES_TOKEN_RETURN_ERROR = -3329, ///< 请求 AES TOKEN 时，server 返回的内容是空的
+ ERR_ACCIP_LIST_EMPTY = -3331, ///< 请求接口机 IP 返回的列表为空的
+ ERR_ROOM_REQUEST_SEND_JSON_CMD_TIMEOUT = -3332, ///< 请求发送Json 信令超时
+
+ // Info 服务器（查询接口机 IP）, 服务器错误码，数值范围[-100000, -110000]
+ ERR_SERVER_INFO_UNPACKING_ERROR = -100000, ///< server 解包错误，可能请求数据被篡改
+ ERR_SERVER_INFO_TOKEN_ERROR = -100001, ///< TOKEN 错误
+ ERR_SERVER_INFO_ALLOCATE_ACCESS_FAILED = -100002, ///< 分配接口机错误
+ ERR_SERVER_INFO_GENERATE_SIGN_FAILED = -100003, ///< 生成签名错误
+ ERR_SERVER_INFO_TOKEN_TIMEOUT = -100004, ///< HTTPS token 超时
+ ERR_SERVER_INFO_INVALID_COMMAND = -100005, ///< 无效的命令字
+ ERR_SERVER_INFO_GENERATE_KEN_ERROR = -100007, ///< HTTPS 请求时，生成加密 key 错误
+ ERR_SERVER_INFO_GENERATE_TOKEN_ERROR = -100008, ///< HTTPS 请求时，生成 token 错误
+ ERR_SERVER_INFO_DATABASE = -100009, ///< 数据库查询失败（房间相关存储信息）
+ ERR_SERVER_INFO_BAD_ROOMID = -100010, ///< 房间号错误
+ ERR_SERVER_INFO_BAD_SCENE_OR_ROLE = -100011, ///< 场景或角色错误
+ ERR_SERVER_INFO_ROOMID_EXCHANGE_FAILED = -100012, ///< 房间号转换出错
+ ERR_SERVER_INFO_STRGROUP_HAS_INVALID_CHARS = -100014, ///< 房间号非法
+ ERR_SERVER_INFO_LACK_SDKAPPID = -100015, ///< 非法SDKAppid
+ ERR_SERVER_INFO_INVALID = -100016, ///< 无效请求, 分配接口机失败
+ ERR_SERVER_INFO_ECDH_GET_KEY = -100017, ///< 生成公钥失败
+
+ // Access 接口机
+ ERR_SERVER_ACC_TOKEN_TIMEOUT = -101000, ///< token 过期
+ ERR_SERVER_ACC_SIGN_ERROR = -101001, ///< 签名错误
+ ERR_SERVER_ACC_SIGN_TIMEOUT = -101002, ///< 签名超时
+ ERR_SERVER_ACC_ROOM_NOT_EXIST = -101003, ///< 房间不存在
+ ERR_SERVER_ACC_ROOMID = -101004, ///< 后台房间标识 roomId 错误
+ ERR_SERVER_ACC_LOCATIONID = -101005, ///< 后台用户位置标识 locationId 错误
+ ERR_SERVER_ACC_TOKEN_EORROR = -101006, ///< token里面的tinyid和进房信令tinyid不同 或是 进房信令没有token
+
+ // Center 服务器（信令和流控处理等任务）
+ ERR_SERVER_CENTER_SYSTEM_ERROR = -102000, ///< 后台错误
+
+ ERR_SERVER_CENTER_INVALID_ROOMID = -102001, ///< 无效的房间 Id
+ ERR_SERVER_CENTER_CREATE_ROOM_FAILED = -102002, ///< 创建房间失败
+ ERR_SERVER_CENTER_SIGN_ERROR = -102003, ///< 签名错误
+ ERR_SERVER_CENTER_SIGN_TIMEOUT = -102004, ///< 签名过期
+ ERR_SERVER_CENTER_ROOM_NOT_EXIST = -102005, ///< 房间不存在
+ ERR_SERVER_CENTER_ADD_USER_FAILED = -102006, ///< 房间添加用户失败
+ ERR_SERVER_CENTER_FIND_USER_FAILED = -102007, ///< 查找用户失败
+ ERR_SERVER_CENTER_SWITCH_TERMINATION_FREQUENTLY = -102008, ///< 频繁切换终端
+ ERR_SERVER_CENTER_LOCATION_NOT_EXIST = -102009, ///< locationid 错误
+ ERR_SERVER_CENTER_NO_PRIVILEDGE_CREATE_ROOM = -102010, ///< 没有权限创建房间
+ ERR_SERVER_CENTER_NO_PRIVILEDGE_ENTER_ROOM = -102011, ///< 没有权限进入房间
+ ERR_SERVER_CENTER_INVALID_PARAMETER_SUB_VIDEO = -102012, ///< 辅路抢视频位、申请辅路请求类型参数错误
+ ERR_SERVER_CENTER_NO_PRIVILEDGE_PUSH_VIDEO = -102013, ///< 没有权限上视频
+ ERR_SERVER_CENTER_ROUTE_TABLE_ERROR = -102014, ///< 没有空闲路由表
+ ERR_SERVER_CENTER_NOT_PUSH_SUB_VIDEO = -102017, ///< 当前用户没有上行辅路
+ ERR_SERVER_CENTER_USER_WAS_DELETED = -102018, ///< 用户被删除状态
+ ERR_SERVER_CENTER_NO_PRIVILEDGE_REQUEST_VIDEO = -102019, ///< 没有权限请求视频
+ ERR_SERVER_CENTER_INVALID_PARAMETER = -102023, ///< 进房参数 bussInfo 错误
+ ERR_SERVER_CENTER_I_FRAME_UNKNOW_TYPE = -102024, ///< 请求 I 帧未知 opType
+ ERR_SERVER_CENTER_I_FRAME_INVALID_PACKET = -102025, ///< 请求 I 帧包格式错误
+ ERR_SERVER_CENTER_I_FRAME_DEST_USER_NOT_EXIST = -102026, ///< 请求 I 帧目标用户不存在
+ ERR_SERVER_CENTER_I_FRAME_ROOM_TOO_BIG = -102027, ///< 请求 I 帧房间用户太多
+ ERR_SERVER_CENTER_I_FRAME_RPS_INVALID_PARAMETER = -102028, ///< 请求 I 帧参数错误
+ ERR_SERVER_CENTER_INVALID_ROOM_ID = -102029, ///< 房间号非法
+ ERR_SERVER_CENTER_ROOM_ID_TOO_LONG = -102030, ///< 房间号超过限制
+ ERR_SERVER_CENTER_ROOM_FULL = -102052, ///< 房间满员
+ ERR_SERVER_CENTER_DECODE_JSON_FAIL = -102053, ///< JSON 串解析失败
+ ERR_SERVER_CENTER_UNKNOWN_SUB_CMD = -102054, ///< 未定义命令字
+ ERR_SERVER_CENTER_INVALID_ROLE = -102055, ///< 未定义角色
+ ERR_SERVER_CENTER_REACH_PROXY_MAX = -102056, ///< 代理机超出限制
+ ERR_SERVER_CENTER_RECORDID_STORE = -102057, ///< 无法保存用户自定义 recordId
+ ERR_SERVER_CENTER_PB_SERIALIZE = -102058, ///< Protobuf 序列化错误
+
+ ERR_SERVER_SSO_SIG_EXPIRED = -70001, ///< sig 过期，请尝试重新生成。如果是刚生成，就过期，请检查有效期填写的是否过小，或者填的 0
+ ERR_SERVER_SSO_SIG_VERIFICATION_FAILED_1 = -70003, ///< sig 校验失败，请确认下 sig 内容是否被截断，如缓冲区长度不够导致的内容截断
+ ERR_SERVER_SSO_SIG_VERIFICATION_FAILED_2 = -70004, ///< sig 校验失败，请确认下 sig 内容是否被截断，如缓冲区长度不够导致的内容截断
+ ERR_SERVER_SSO_SIG_VERIFICATION_FAILED_3 = -70005, ///< sig 校验失败，可用工具自行验证生成的 sig 是否正确
+ ERR_SERVER_SSO_SIG_VERIFICATION_FAILED_4 = -70006, ///< sig 校验失败，可用工具自行验证生成的 sig 是否正确
+ ERR_SERVER_SSO_SIG_VERIFICATION_FAILED_5 = -70007, ///< sig 校验失败，可用工具自行验证生成的 sig 是否正确
+ ERR_SERVER_SSO_SIG_VERIFICATION_FAILED_6 = -70008, ///< sig 校验失败，可用工具自行验证生成的 sig 是否正确
+ ERR_SERVER_SSO_SIG_VERIFICATION_FAILED_7 = -70009, ///< 用业务公钥验证 sig 失败，请确认生成的 usersig 使用的私钥和 sdkAppId 是否对应
+ ERR_SERVER_SSO_SIG_VERIFICATION_FAILED_8 = -70010, ///< sig 校验失败，可用工具自行验证生成的 sig 是否正确
+ ERR_SERVER_SSO_SIG_VERIFICATION_ID_NOT_MATCH = -70013, ///< sig 中 identifier 与请求时的 identifier 不匹配，请检查登录时填写的 identifier 与 sig 中的是否一致
+ ERR_SERVER_SSO_APPID_NOT_MATCH = -70014, ///< sig 中 sdkAppId 与请求时的 sdkAppId 不匹配，请检查登录时填写的 sdkAppId 与 sig 中的是否一致
+ ERR_SERVER_SSO_VERIFICATION_EXPIRED = -70017, ///< 内部第三方票据验证超时，请重试，如多次重试不成功，请@TLS 帐号支持，QQ 3268519604
+ ERR_SERVER_SSO_VERIFICATION_FAILED = -70018, ///< 内部第三方票据验证超时，请重试，如多次重试不成功，请@TLS 帐号支持，QQ 3268519604
+
+ ERR_SERVER_SSO_APPID_NOT_FOUND = -70020, ///< sdkAppId 未找到，请确认是否已经在腾讯云上配置
+ ERR_SERVER_SSO_ACCOUNT_IN_BLACKLIST = -70051, ///< 帐号已被拉入黑名单，请联系 TLS 帐号支持 QQ 3268519604
+ ERR_SERVER_SSO_SIG_INVALID = -70052, ///< usersig 已经失效，请重新生成，再次尝试
+ ERR_SERVER_SSO_LIMITED_BY_SECURITY = -70114, ///< 安全原因被限制
+ ERR_SERVER_SSO_INVALID_LOGIN_STATUS = -70221, ///< 登录状态无效，请使用 usersig 重新鉴权
+ ERR_SERVER_SSO_APPID_ERROR = -70252, ///< sdkAppId 填写错误
+ ERR_SERVER_SSO_TICKET_VERIFICATION_FAILED = -70346, ///< 票据校验失败，请检查各项参数是否正确
+ ERR_SERVER_SSO_TICKET_EXPIRED = -70347, ///< 票据因过期原因校验失败
+ ERR_SERVER_SSO_ACCOUNT_EXCEED_PURCHASES = -70398, ///< 创建账号数量超过已购买预付费数量限制
+ ERR_SERVER_SSO_INTERNAL_ERROR = -70500, ///< 服务器内部错误，请重试
+
+ //秒级监控上报错误码
+ ERR_REQUEST_QUERY_CONFIG_TIMEOUT = -4001, ///< 请求通用配置超时
+ ERR_CUSTOM_STREAM_INVALID = -4002, ///< 自定义流id错误
+ ERR_USER_DEFINE_RECORD_ID_INVALID = -4003, ///< userDefineRecordId错误
+ ERR_MIX_PARAM_INVALID = -4004, ///< 混流参数校验失败
+ ERR_REQUEST_ACC_BY_HOST_IP = -4005, ///< 通过域名进行0x1请求
+ // - /Remove From Head
+} TXLiteAVError;
+
+/////////////////////////////////////////////////////////////////////////////////
+//
+// 警告码
+//
+//> 不需要特别关注，但您可以根据其中某些感兴趣的警告码，对当前用户进行相应的提示
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+typedef enum TXLiteAVWarning
+{
+ WARNING_HW_ENCODER_START_FAIL = 1103, ///< 硬编码启动出现问题，自动切换到软编码
+ WARNING_CURRENT_ENCODE_TYPE_CHANGED = 1104, ///< 当前编码格式, 通过key 为type获取，值为1时是265编码，值为0时是264编码
+ WARNING_VIDEO_ENCODER_SW_TO_HW = 1107, ///< 当前 CPU 使用率太高，无法满足软件编码需求，自动切换到硬件编码
+ WARNING_INSUFFICIENT_CAPTURE_FPS = 1108, ///< 摄像头采集帧率不足，部分自带美颜算法的 Android 手机上会出现
+ WARNING_SW_ENCODER_START_FAIL = 1109, ///< 软编码启动失败
+ WARNING_REDUCE_CAPTURE_RESOLUTION = 1110, ///< 摄像头采集分辨率被降低，以满足当前帧率和性能最优解。
+ WARNING_CAMERA_DEVICE_EMPTY = 1111, ///< 没有检测到可用的摄像头设备
+ WARNING_CAMERA_NOT_AUTHORIZED = 1112, ///< 用户未授权当前应用使用摄像头
+ WARNING_MICROPHONE_DEVICE_EMPTY = 1201, ///< 没有检测到可用的麦克风设备
+ WARNING_SPEAKER_DEVICE_EMPTY = 1202, ///< 没有检测到可用的扬声器设备
+ WARNING_MICROPHONE_NOT_AUTHORIZED = 1203, ///< 用户未授权当前应用使用麦克风
+ WARNING_MICROPHONE_DEVICE_ABNORMAL = 1204, ///< 音频采集设备不可用（例如被占用或者PC判定无效设备）
+ WARNING_SPEAKER_DEVICE_ABNORMAL = 1205, ///< 音频播放设备不可用（例如被占用或者PC判定无效设备）
+ WARNING_SCREEN_CAPTURE_NOT_AUTHORIZED = 1206, ///< 用户未授权当前应用使用屏幕录制
+ WARNING_VIDEO_FRAME_DECODE_FAIL = 2101, ///< 当前视频帧解码失败
+ WARNING_AUDIO_FRAME_DECODE_FAIL = 2102, ///< 当前音频帧解码失败
+ WARNING_VIDEO_PLAY_LAG = 2105, ///< 当前视频播放出现卡顿
+ WARNING_HW_DECODER_START_FAIL = 2106, ///< 硬解启动失败，采用软解码
+ WARNING_VIDEO_DECODER_HW_TO_SW = 2108, ///< 当前流硬解第一个 I 帧失败，SDK 自动切软解
+ WARNING_SW_DECODER_START_FAIL = 2109, ///< 软解码器启动失败
+ WARNING_VIDEO_RENDER_FAIL = 2110, ///< 视频渲染失败
+ WARNING_START_CAPTURE_IGNORED = 4000, ///< 已经在采集，启动采集被忽略
+ WARNING_AUDIO_RECORDING_WRITE_FAIL = 7001, ///< 音频录制写入文件失败
+ WARNING_ROOM_DISCONNECT = 5101, ///< 网络断开连接
+ WARNING_IGNORE_UPSTREAM_FOR_AUDIENCE = 6001, ///< 当前是观众角色，忽略上行音视频数据
+ WARNING_MICROPHONE_HOWLING_DETECTED = 7002, ///< 录制音频时监测到啸叫。请调节两台客户端之间的距离或降低播放音量，检测到啸叫后，5s后会再次进行重新检测
+
+ // - Remove From Head
+ WARNING_NET_BUSY = 1101, ///< 网络状况不佳：上行带宽太小，上传数据受阻
+ WARNING_RTMP_SERVER_RECONNECT = 1102, ///< 直播，网络断连, 已启动自动重连（自动重连连续失败超过三次会放弃）
+ WARNING_LIVE_STREAM_SERVER_RECONNECT = 2103, ///< 直播，网络断连, 已启动自动重连（自动重连连续失败超过三次会放弃）
+ WARNING_RECV_DATA_LAG = 2104, ///< 网络来包不稳：可能是下行带宽不足，或由于主播端出流不均匀
+ WARNING_RTMP_DNS_FAIL = 3001, ///< 直播，DNS 解析失败
+ WARNING_RTMP_SEVER_CONN_FAIL = 3002, ///< 直播，服务器连接失败
+ WARNING_RTMP_SHAKE_FAIL = 3003, ///< 直播，与 RTMP 服务器握手失败
+ WARNING_RTMP_SERVER_BREAK_CONNECT = 3004, ///< 直播，服务器主动断开
+ WARNING_RTMP_READ_WRITE_FAIL = 3005, ///< 直播，RTMP 读/写失败，将会断开连接
+ WARNING_RTMP_WRITE_FAIL = 3006, ///< 直播，RTMP 写失败（SDK 内部错误码，不会对外抛出）
+ WARNING_RTMP_READ_FAIL = 3007, ///< 直播，RTMP 读失败（SDK 内部错误码，不会对外抛出）
+ WARNING_RTMP_NO_DATA = 3008, ///< 直播，超过30s 没有数据发送，主动断开连接
+ WARNING_PLAY_LIVE_STREAM_INFO_CONNECT_FAIL = 3009, ///< 直播，connect 服务器调用失败（SDK 内部错误码，不会对外抛出）
+ WARNING_NO_STEAM_SOURCE_FAIL = 3010, ///< 直播，连接失败，该流地址无视频（SDK 内部错误码，不会对外抛出）
+ WARNING_ROOM_RECONNECT = 5102, ///< 网络断连，已启动自动重连
+ WARNING_ROOM_NET_BUSY = 5103, ///< 网络状况不佳：上行带宽太小，上传数据受阻
+ // - /Remove From Head
+} TXLiteAVWarning;
+
+// - Remove From Head
+/////////////////////////////////////////////////////////////////////////////////
+//
+// （三）事件列表
+//
+/////////////////////////////////////////////////////////////////////////////////
+
+typedef enum TXLiteAVEvent
+{
+ EVT_RTMP_PUSH_CONNECT_SUCC = 1001, ///< 直播，已经连接 RTMP 推流服务器
+ EVT_RTMP_PUSH_BEGIN = 1002, ///< 直播，已经与 RTMP 服务器握手完毕，开始推流
+ EVT_CAMERA_START_SUCC = 1003, ///< 打开摄像头成功
+ EVT_SCREEN_CAPTURE_SUCC = 1004, ///< 录屏启动成功
+ EVT_UP_CHANGE_RESOLUTION = 1005, ///< 上行动态调整分辨率
+ EVT_UP_CHANGE_BITRATE = 1006, ///< 码率动态调整
+ EVT_FIRST_FRAME_AVAILABLE = 1007, ///< 首帧画面采集完成
+ EVT_START_VIDEO_ENCODER = 1008, ///< 编码器启动成功
+ EVT_SNAPSHOT_COMPLETE = 1022, ///< 一帧截图完成
+ EVT_CAMERA_REMOVED = 1023, ///< 摄像头设备已被移出（Windows 和 Mac 版 SDK 使用）
+ EVT_CAMERA_AVAILABLE = 1024, ///< 摄像头设备重新可用（Windows 和 Mac 版 SDK 使用）
+ EVT_CAMERA_CLOSE = 1025, ///< 关闭摄像头完成（Windows 和 Mac 版 SDK 使用）
+ EVT_RTMP_PUSH_PUBLISH_START = 1026, ///< 直播，与 RTMP 服务器连接后，收到 NetStream.Publish.Start 消息，表明流发布成功（SDK 内部事件，不会对外抛出）
+ EVT_HW_ENCODER_START_SUCC = 1027, ///< 硬编码器启动成功
+ EVT_SW_ENCODER_START_SUCC = 1028, ///< 软编码器启动成功
+ EVT_LOCAL_RECORD_RESULT = 1029, ///< 本地录制结果
+ EVT_LOCAL_RECORD_PROGRESS = 1030, ///< 本地录制状态通知
+
+ EVT_PLAY_LIVE_STREAM_CONNECT_SUCC = 2001, ///< 直播，已经连接 RTMP 拉流服务器
+ EVT_PLAY_LIVE_STREAM_BEGIN = 2002, ///< 直播，已经与 RTMP 服务器握手完毕，开始拉流
+ EVT_RENDER_FIRST_I_FRAME = 2003, ///< 渲染首个视频数据包（IDR）
+ EVT_VIDEO_PLAY_BEGIN = 2004, ///< 视频播放开始
+ EVT_VIDEO_PLAY_PROGRESS = 2005, ///< 视频播放进度
+ EVT_VIDEO_PLAY_END = 2006, ///< 视频播放结束
+ EVT_VIDEO_PLAY_LOADING = 2007, ///< 视频播放 loading
+ EVT_START_VIDEO_DECODER = 2008, ///< 解码器启动
+ EVT_DOWN_CHANGE_RESOLUTION = 2009, ///< 下行视频分辨率改变
+ EVT_GET_VODFILE_MEDIAINFO_SUCC = 2010, ///< 点播，获取点播文件信息成功
+ EVT_VIDEO_CHANGE_ROTATION = 2011, ///< 视频旋转角度发生改变
+ EVT_PLAY_GET_MESSAGE = 2012, ///< 消息事件
+ EVT_VOD_PLAY_PREPARED = 2013, ///< 点播，视频加载完毕
+ EVT_VOD_PLAY_LOADING_END = 2014, ///< 点播，loading 结束
+ EVT_PLAY_LIVE_STREAM_SWITCH_SUCC = 2015, ///< 直播，切流成功（切流可以播放不同画面大小的视频）
+ EVT_VOD_PLAY_TCP_CONNECT_SUCC = 2016, ///< 点播，TCP 连接成功（SDK 内部事件，不会对外抛出）
+ EVT_VOD_PLAY_FIRST_VIDEO_PACKET = 2017, ///< 点播，收到首帧数据（SDK 内部事件，不会对外抛出）
+ EVT_VOD_PLAY_DNS_RESOLVED = 2018, ///< 点播，DNS 解析完成（SDK 内部事件，不会对外抛出）
+ EVT_VOD_PLAY_SEEK_COMPLETE = 2019, ///< 点播，视频播放 Seek 完成（SDK 内部事件，不会对外抛出）
+ EVT_VIDEO_DECODER_CACHE_TOO_MANY_FRAMES = 2020, ///< 视频解码器缓存帧数过多，超过40帧（SDK 内部事件，不会对外抛出）
+ EVT_HW_DECODER_START_SUCC = 2021, ///< 硬解码器启动成功（SDK 内部事件，不会对外抛出）
+ EVT_SW_DECODER_START_SUCC = 2022, ///< 软解码器启动成功（SDK 内部事件，不会对外抛出）
+ EVT_AUDIO_JITTER_STATE_FIRST_LOADING = 2023, ///< 音频首次加载（SDK 内部事件，不会对外抛出）
+ EVT_AUDIO_JITTER_STATE_LOADING = 2024, ///< 音频正在加载（SDK 内部事件，不会对外抛出）
+ EVT_AUDIO_JITTER_STATE_PLAYING = 2025, ///< 音频正在播放（SDK 内部事件，不会对外抛出）
+ EVT_AUDIO_JITTER_STATE_FIRST_PLAY = 2026, ///< 音频首次播放（SDK 内部事件，不会对外抛出）
+ EVT_MIC_START_SUCC = 2027, ///< 麦克风启动成功
+ EVT_PLAY_GET_METADATA = 2028, ///< 视频流MetaData事件
+ EVT_MIC_RELEASE_SUCC = 2029, ///< 释放麦克风占用
+ EVT_AUDIO_DEVICE_ROUTE_CHANGED = 2030, ///< 音频设备的route发生改变，即当前的输入输出设备发生改变，比如耳机被拔出
+ EVT_PLAY_GET_FLVSESSIONKEY = 2031, ///< TXLivePlayer 接收到http响应头中的 flvSessionKey 信息
+ EVT_AUDIO_SESSION_INTERRUPT = 2032, ///< Audio Session Interrupt事件
+
+
+ EVT_ROOM_ENTER = 1018, ///< 进入房间成功
+ EVT_ROOM_EXIT = 1019, ///< 退出房间
+ EVT_ROOM_USERLIST = 1020, ///< 下发房间成员列表（不包括自己）
+ EVT_ROOM_NEED_REENTER = 1021, ///< WiFi 切换到4G 会触发断线重连，此时需要重新进入房间（拉取最优的服务器地址）
+ EVT_ROOM_ENTER_FAILED = 1022, ///< 自己进入房间失败
+ EVT_ROOM_USER_ENTER = 1031, ///< 进房通知
+ EVT_ROOM_USER_EXIT = 1032, ///< 退房通知
+ EVT_ROOM_USER_VIDEO_STATE = 1033, ///< 视频状态位变化通知
+ EVT_ROOM_USER_AUDIO_STATE = 1034, ///< 音频状态位变化通知
+
+ EVT_ROOM_REQUEST_IP_SUCC = 8001, ///< 拉取接口机服务器地址成功
+ EVT_ROOM_CONNECT_SUCC = 8002, ///< 连接接口机服务器成功
+ EVT_ROOM_REQUEST_AVSEAT_SUCC = 8003, ///< 请求视频位成功
+} TXLiteAVEvent;
+// - /Remove From Head
+
+#endif /* __TXLITEAVCODE_H__ */
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVEncodedDataProcessingListener.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVEncodedDataProcessingListener.h
new file mode 100644
index 0000000..33e79d2
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVEncodedDataProcessingListener.h
@@ -0,0 +1,54 @@
+/*
+* Module: live 编码数据回调
+*
+* Function: 回调推流端编码完，和 播放端解码前的数据
+*
+*/
+
+
+#ifndef TXLiteAVEncodedDataProcessingListener_h
+#define TXLiteAVEncodedDataProcessingListener_h
+
+#include
+#include "TXLiteAVBuffer.h"
+
+namespace liteav {
+
+struct TXLiteAVEncodedData {
+ const char * userId; // didEncodeVideo 和 didEncodeAudio 回调时，此字段为null；
+ int streamType; // 视频流类型，参考 TRTCVideoStreamType，audio时，此字段为0
+ const liteav::TXLiteAVBuffer * originData; // 原始数据
+ liteav::TXLiteAVBuffer * processedData; // 写回处理后的数据
+};
+
+class ITXLiteAVEncodedDataProcessingListener {
+public:
+ virtual ~ITXLiteAVEncodedDataProcessingListener() {}
+
+ /**
+ * 回调编码完的视频数据。
+ * @note videoData.userId = nullptr
+ */
+ virtual bool didEncodeVideo(TXLiteAVEncodedData & videoData) { return false; }
+
+ /**
+ * 回调解码前的视频数据。
+ * @note videoData.userId 表示对应的user，当userId 为 nullptr时，表示此时先接收到数据了，对应的userId还未完成同步。获取到userId之后会回调正确的userId
+ */
+ virtual bool willDecodeVideo(TXLiteAVEncodedData & videoData) { return false; }
+
+ /**
+ * 回调编码完的音频数据。
+ * @note audioData.userId = nullptr
+ */
+ virtual bool didEncodeAudio(TXLiteAVEncodedData & audioData) { return false; }
+
+ /**
+ * 回调解码前的音频数据。
+ * @note audioData.userId 表示对应的user，当userId 为 nullptr时，表示此时先接收到数据了，对应的userId还未完成同步。获取到userId之后会回调正确的userId
+ */
+ virtual bool willDecodeAudio(TXLiteAVEncodedData & audioData) { return false; }
+};
+}
+
+#endif /* TXLiteAVEncodedDataProcessingListener_h */
diff --git a/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVSDK.h b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVSDK.h
new file mode 100644
index 0000000..a949948
--- /dev/null
+++ b/src/ios/TXLiteAVSDK_TRTC.framework/Headers/TXLiteAVSDK.h
@@ -0,0 +1,35 @@
+//
+// TXLiteAVSDK.h
+// TXLiteAVSDK
+//
+// Created by alderzhang on 2017/6/9.
+// Copyright © 2017年 Tencent. All rights reserved.
+//
+
+
+
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import
+#import