@tencent-rtc/mcp 1.4.7 → 1.6.0-beta.1

package/resource/{doc → knowledge}/rtcengine/web/plugin/23-advanced-support-detection.md

@@ -212,7 +212,7 @@ Detection principle: Play an mp3 media file through an audio tag
 
 #### 6. Network Detection
 
-Reference: [Network Quality Detection before enterRoom](./tutorial-24-advanced-network-quality.html)
+Reference: [Network Quality Detection before enterRoom](tutorial-24-advanced-network-quality.md)
 
 ## TRTC Capability Detection Page
 

package/resource/{doc → knowledge}/rtcengine/web/plugin/24-advanced-network-quality.md

@@ -1,7 +1,7 @@
 
 Before entering the room or during the call, you can check the user's network quality to determine the current network quality. If the user's network quality is too poor, it is recommended that the user change the network environment to ensure normal call quality.
 
-This article mainly introduces how to implement network quality detection before the call or during the call based on the [NETWORK_QUALITY](./module-EVENT.html#.NETWORK_QUALITY) event.
+This article mainly introduces how to implement network quality detection before the call or during the call based on the [NETWORK_QUALITY](module-EVENT.md#.NETWORK_QUALITY) event.
 
 ## Detect network quality during the call
 
@@ -20,14 +20,14 @@ trtc.on(TRTC.EVENT.NETWORK_QUALITY, event => {
 
 ### Implementation process
 
-1. Call [TRTC.create](./TRTC.html#.createTRTC) to create two TRTC objects, referred to as uplinkTRTC and downlinkTRTC.
+1. Call [TRTC.create](TRTC.md#.createTRTC) to create two TRTC objects, referred to as uplinkTRTC and downlinkTRTC.
 2. Both TRTC objects enter the same room.
-3. Use uplinkTRTC to startLocalVideo, and listen to the [NETWORK_QUALITY](./module-EVENT.html#.NETWORK_QUALITY) event to detect the uplink network quality.
-4. Use downlinkTRTC to startRemoteVideo, and listen to the [NETWORK_QUALITY](./module-EVENT.html#.NETWORK_QUALITY) event to detect the downlink network quality.
+3. Use uplinkTRTC to startLocalVideo, and listen to the [NETWORK_QUALITY](module-EVENT.md#.NETWORK_QUALITY) event to detect the uplink network quality.
+4. Use downlinkTRTC to startRemoteVideo, and listen to the [NETWORK_QUALITY](module-EVENT.md#.NETWORK_QUALITY) event to detect the downlink network quality.
 5. The entire process lasts for about 15 seconds, and finally takes the average network quality to roughly determine the uplink and downlink network conditions.
 
 > !
-> - The detection process will generate a small amount of [basic service fees](https://trtc.io/document/59806?platform=android&product=rtcengine&menulabel=coresdk). If the push stream resolution is not specified, it defaults to pushing the stream with a resolution of 640*480.
+> - The detection process will generate a small amount of [audio and video duration fees](https://trtc.io/document/42734). If the push stream resolution is not specified, it defaults to pushing the stream with a resolution of 640*480.
 
 ### API call sequence
 
@@ -100,7 +100,7 @@ setTimeout(() => {
     );
   }
 
-  const validDownlinkNetworkQualitiesList = testResult.uplinkNetworkQualities.filter(value => value >= 1 && value <= 5);
+  const validDownlinkNetworkQualitiesList = testResult.downlinkNetworkQualities.filter(value => value >= 1 && value <= 5);
   if (validDownlinkNetworkQualitiesList.length > 0) {
     // Calculate the average downlink network quality
     testResult.average.downlinkNetworkQuality = Math.ceil(
@@ -130,5 +130,5 @@ After the above steps, you can get the average uplink network quality and the av
 
 > Suggestion: When the network quality is greater than 3, it is recommended to guide the user to check the network and try to change the network environment, otherwise it is difficult to ensure normal audio and video communication.
 > You can also reduce bandwidth consumption through the following strategies:
-> - If the uplink network quality is greater than 3, you can reduce the bitrate through the [TRTC.updateLocalVideo()](./TRTC.html#updateLocalVideo) interface or close the video through the [TRTC.stopLocalVideo()](./TRTC.html#stopLocalVideo) method to reduce uplink bandwidth consumption.
-> - If the downlink network quality is greater than 3, you can reduce the downlink bandwidth consumption by subscribing to a small stream (refer to: [Enable Small Stream Transmission](./tutorial-27-advanced-small-stream.html)) or only subscribing to audio.
+> - If the uplink network quality is greater than 3, you can reduce the bitrate through the [TRTC.updateLocalVideo()](TRTC.md#updateLocalVideo) interface or close the video through the [TRTC.stopLocalVideo()](TRTC.md#stopLocalVideo) method to reduce uplink bandwidth consumption.
+> - If the downlink network quality is greater than 3, you can reduce the downlink bandwidth consumption by subscribing to a small stream (refer to: [Enable Small Stream Transmission](tutorial-27-advanced-small-stream.md)) or only subscribing to audio.

package/resource/{doc → knowledge}/rtcengine/web/plugin/25-advanced-device-change.md

@@ -7,7 +7,7 @@ This article mainly introduces how to detect the behavior of users inserting, un
 > The SDK will detect the insertion and removal behavior of the camera and microphone, and automatically recover the collection at the appropriate time. The specific logic is as follows:
 > 1. If the currently streaming camera/microphone is unplugged, the SDK will try to use the remaining available media devices to restore the collection.
 > 2. If the camera/microphone is unplugged and there are no available media devices, the SDK will detect the device insertion behavior. If a new available media device is inserted, the SDK will try to use the newly inserted media device to restore the collection and push the stream.
-> 3. If the recovery fails, an error [TRTC.ERROR_CODE.DEVICE_ERROR](./module-ERROR_CODE.html#.DEVICE_ERROR) will be thrown, and the extraCode will be 5308 and 5309.
+> 3. If the recovery fails, an error [TRTC.ERROR_CODE.DEVICE_ERROR](module-ERROR_CODE.md#.DEVICE_ERROR) will be thrown, and the extraCode will be 5308 and 5309.
 
 ## Implementation Process
 

package/resource/{doc → knowledge}/rtcengine/web/plugin/26-advanced-publish-cdn-stream.md

@@ -42,7 +42,7 @@ For each mode, the following lifecycle applies:
 - `trtc.updatePlugin('CDNStreaming', options)` - Update parameters (0 or N times)
 - `trtc.stopPlugin('CDNStreaming', options)` - Stop the plugin
 
-For detailed `options` parameters, see [CDNStreamingOptions](./global.html#CDNStreamingOptions).
+For detailed `options` parameters, see [CDNStreamingOptions](global.md#CDNStreamingOptions).
 
 ## Relay Single Stream to CDN
 
@@ -155,7 +155,7 @@ This plugin sends a command to Tencent Cloud's transcoding server to mix multipl
 
 > Coordinate system: the origin (0, 0) is the **top-left corner** of the output canvas, in pixels. locationX/locationY is the top-left position of each stream; width/height is the layout region size. Coordinates/sizes are tied to encoding.videoWidth/videoHeight - if you change resolution, scale layout parameters accordingly to avoid misalignment.
 >
-> For full parameter details, see [CDNStreamingOptions](./global.html#CDNStreamingOptions).
+> For full parameter details, see [CDNStreamingOptions](global.md#CDNStreamingOptions).
 
 ### Start Mixing
 
@@ -319,7 +319,7 @@ Mix streams and push the mixed stream back to a specified TRTC room as a robot u
 > - A stream-pulling robot with userId `mcu_robot_${roomId}_${userId}` will automatically enter the current room. Avoid userId conflicts.
 > - This feature will incur transcoding fees for stream mixing.
 > - When `audioMixUserList` is empty, all users are mixed by default. Pass a non-existent userId if you don't need audio mixing.
-> - For full parameter details, see [CDNStreamingOptions](./global.html#CDNStreamingOptions).
+> - For full parameter details, see [CDNStreamingOptions](global.md#CDNStreamingOptions).
 
 ### Start Push Back
 
@@ -418,7 +418,7 @@ try {
 
 ## API Description
 
-For `startPlugin('CDNStreaming', options)` and `updatePlugin('CDNStreaming', options)`, parameter types can be found in [CDNStreamingOptions](./global.html#CDNStreamingOptions).
+For `startPlugin('CDNStreaming', options)` and `updatePlugin('CDNStreaming', options)`, parameter types can be found in [CDNStreamingOptions](global.md#CDNStreamingOptions).
 
 For `stopPlugin('CDNStreaming', options)`, the parameter only needs to fill in the `options.target.publishMode` field.
 
@@ -426,9 +426,9 @@ For `stopPlugin('CDNStreaming', options)`, the parameter only needs to fill in t
 
 1. To record mixed audio streams, please refer to [Cloud Recording](https://trtc.io/document/45169).
 
-2. Cloud-based stream mixing and transcoding requires decoding and re-encoding the audio and video streams input to the MCU cluster, which will incur additional service costs. Therefore, TRTC will charge users who use the MCU cluster for cloud-based stream mixing and transcoding an additional value-added fee. Cloud-based stream mixing and transcoding fees are based on the **resolution of the transcoded output** and **transcoding duration**. The higher the resolution of the transcoded output and the longer the transcoding time, the higher the cost. For details, please refer to [Cloud-Based Stream Mixing and Transcoding Billing Description](https://trtc.io/document/47858?product=rtcengine&menulabel=core%20sdk&platform=android#25e78f25-3514-4608-a85f-f6db86685684).
+2. Cloud-based stream mixing and transcoding requires decoding and re-encoding the audio and video streams input to the MCU cluster, which will incur additional service costs. Therefore, TRTC will charge users who use the MCU cluster for cloud-based stream mixing and transcoding an additional value-added fee. Cloud-based stream mixing and transcoding fees are based on the **resolution of the transcoded output** and **transcoding duration**. The higher the resolution of the transcoded output and the longer the transcoding time, the higher the cost. For details, please refer to [Cloud-Based Stream Mixing and Transcoding Billing Description](https://trtc.io/document/47858).
 
-3. For audio and video duration billing, please refer to [Billing Description](https://trtc.io/zh/document/47858?product=rtcengine&menulabel=core%20sdk&platform=android#25e78f25-3514-4608-a85f-f6db86685684).
+3. For audio and video duration billing, please refer to [Billing Description](https://trtc.io/zh/document/47858).
 
 4. After leaving the room, billing will not continue, but when re-entering the room, it will automatically continue to start. If you want to directly terminate CDN relay and mixing, please execute stopPlugin for the corresponding publishMode.
 

package/resource/{doc → knowledge}/rtcengine/web/plugin/27-advanced-small-stream.md

@@ -15,7 +15,7 @@ Applicable scenarios: The small stream is suitable for multi-person audio and vi
 
 #### Enable small streams on the publishing end
 
-When calling [startLocalVideo](./TRTC.html#startLocalVideo), set the small parameter to enable small stream encoding. In an environment that does not support enabling small streams, the small parameter will not take effect and will not report an error.
+When calling [startLocalVideo](TRTC.md#startLocalVideo), set the small parameter to enable small stream encoding. In an environment that does not support enabling small streams, the small parameter will not take effect and will not report an error.
 
 ```js
 await trtc.startLocalVideo({ option: { small: '120p' }});
@@ -35,7 +35,7 @@ await trtc.updateLocalVideo({ option: { small: false }});
 
 #### Subscribe the big stream or the small stream on the receiving end
 
-When calling [startRemoteVideo](./TRTC.html#startRemoteVideo), set the small parameter to subscribe the small stream. Otherwise, the big stream is subscribed by default. When subscribing the small stream, there is no need to check whether the remote user has published the small stream in advance. If the remote end does not publish the small stream, the SDK will automatically subscribe the big stream.
+When calling [startRemoteVideo](TRTC.md#startRemoteVideo), set the small parameter to subscribe the small stream. Otherwise, the big stream is subscribed by default. When subscribing the small stream, there is no need to check whether the remote user has published the small stream in advance. If the remote end does not publish the small stream, the SDK will automatically subscribe the big stream.
 
 ```js
 trtc.on(TRTC.EVENT.REMOTE_VIDEO_AVAILABLE, ({ userId, streamType }) => {
@@ -82,7 +82,7 @@ await trtc.updateRemoteVideo({
 
 In general, in a multi-person video scenario, due to limited page space, the streaming end will not display all videos on the page. Therefore, for those remote videos that do not need to be displayed on the page, they do not need to be streamed. Only when the view is visible should the video stream be pulled. This is to save traffic and reduce performance overhead.
 
-Implementing this feature is not easy. Starting with version 5.4.0, the SDK has built-in support for this feature. You only need to set the `receiveWhenViewVisible` and `viewRoot` parameters when calling [startRemoteVideo](./TRTC.html#startRemoteVideo) to enable this feature, which is very convenient and fast. The SDK will automatically check the visibility state of the `view` passed in when startRemoteVideo. Video streaming will start when the `view` is visible and stop when `view` is not visible.
+Implementing this feature is not easy. Starting with version 5.4.0, the SDK has built-in support for this feature. You only need to set the `receiveWhenViewVisible` and `viewRoot` parameters when calling [startRemoteVideo](TRTC.md#startRemoteVideo) to enable this feature, which is very convenient and fast. The SDK will automatically check the visibility state of the `view` passed in when startRemoteVideo. Video streaming will start when the `view` is visible and stop when `view` is not visible.
 
 ### Sample code
 

package/resource/{doc → knowledge}/rtcengine/web/plugin/29-advanced-water-mark.md

@@ -2,6 +2,24 @@
 
 This article introduces how to use the watermark plugin to add image watermarks on camera streams.
 
+## Run Demo
+
+<div style="margin: 20px 0;">
+<details class="trtc-demo-details" style="border-radius: 10px; border: 1px solid #e5e6eb; overflow: hidden; box-shadow: 0 4px 14px rgba(0,82,217,0.08); font-family: -apple-system, BlinkMacSystemFont, 'Segoe UI', Roboto, sans-serif;">
+  <summary style="padding: 14px 22px; background: linear-gradient(135deg, #006eff 0%, #0052d9 100%); color: #fff; font-size: 15px; font-weight: 600; cursor: pointer; list-style: none; user-select: none; display: flex; align-items: center; gap: 10px; transition: filter 0.2s;">
+    <span class="trtc-demo-arrow" style="font-size: 14px; opacity: 0.9;">▶</span>
+    <span class="trtc-demo-summary-text">Click to Run Demo</span>
+  </summary>
+  <div style="padding: 16px; background: #fff;">
+<script>
+(function(){var t=null;window.addEventListener('message',function(e){if(e.data&&e.data.type==='trtc-demo-resize'){var f=document.querySelector('.trtc-demo-iframe');if(f){f.style.height=e.data.height+'px';clearTimeout(t);t=setTimeout(function(){f.style.height=e.data.height+'px'},200)}}})})();
+document.querySelectorAll('.trtc-demo-details').forEach(function(d){d.addEventListener('toggle',function(){var s=d.querySelector('.trtc-demo-summary-text'),a=d.querySelector('.trtc-demo-arrow');if(d.open){s&&(s.textContent='Click to Close Demo');a&&(a.textContent='▼')}else{s&&(s.textContent='Click to Run Demo');a&&(a.textContent='▶')}})});
+</script>
+    <iframe allow="microphone; camera; display-capture;" class="trtc-demo-iframe" style="width:100%;border:none;min-height:700px" title="Enable Watermark Plugin" src="https://web.sdk.qcloud.com/trtc/webrtc/v5/demo/samples/advance-features/enable-watermark/index.html?lang=en" frameborder="no" loading="lazy" allowtransparency="true" allowfullscreen="true"></iframe>
+  </div>
+</details>
+</div>
+
 ## Prerequisites
 
 - TRTC Web SDK version >= 5.3.0.

package/resource/{doc → knowledge}/rtcengine/web/plugin/31-advanced-plugins-overview.md

@@ -13,7 +13,7 @@ TRTC Web SDK v5 provides a plugin system that supports on-demand loading of vari
 
 | Plugin Name | Description | Version Requirement | Package Requirement | Static Assets | Documentation |
 |---------|---------|---------|---------|---------|---------|
-| **Beauty** | Professional beauty filters with 6 parameters (smoothing, whitening, face slimming, etc.) + 10 2D stickers | ≥ 5.5.0 | Professional Edition | - | {@tutorial 28-advanced-beauty} |
+| **Beauty** | Professional beauty filters with 6 parameters (smoothing, whitening, face slimming, etc.) + 10 2D stickers | ≥ 5.5.0 | Professional Edition | - | [Enable Advanced Beauty](https://trtc.io/document/67329) |
 <!-- BasicBeauty is for Chinese market only, not exposed in English docs -->
 | **VirtualBackground** | Virtual background with blur, image background, and face centering | ≥ 5.2.0 | Premium Edition | Required | {@tutorial 36-advanced-virtual-background} |
 | **Watermark** | Video watermark with image overlay support | ≥ 5.3.2 | None | - | {@tutorial 29-advanced-water-mark} |

package/resource/{doc → knowledge}/rtcengine/web/plugin/34-advanced-proxy.md

@@ -22,7 +22,7 @@ This article mainly introduces the best practices for dealing with firewall rest
 
 When you call APIs such as startLocalVideo, startLocalAudio, startRemoteVideo, etc., the SDK will establish a media connection channel internally for transmitting audio and video data. When encountering firewall restrictions, the SDK may fail to establish a connection, and the SDK will throw a firewall-restricted error and continue to retry.
 
-You can refer to the following code example to listen for this error and guide users to change networks or check network firewalls and whitelist the domains and ports used by TRTC Web SDK. Reference: [TRTC Web SDK Domain and Port Whitelist](https://trtc.io/document/59667?platform=web&product=rtcengine&menulabel=coresdk).
+You can refer to the following code example to listen for this error and guide users to change networks or check network firewalls and whitelist the domains and ports used by TRTC Web SDK. Reference: [TRTC Web SDK Domain and Port Whitelist](https://trtc.io/document/59667).
 
 ```js
 trtc.on(TRTC.EVENT.ERROR, error => {
@@ -237,5 +237,5 @@ sudo . /turn.sh 10.0.0.4
 
 2. Firewall configuration:
 
-- For the Nginx server, the domain name whitelist needs to be configured in the company's intranet firewall to allow the Nginx server to access TRTC's related domain names. Refer to: [TRTC Web SDK domain and port whitelist](https://trtc.io/document/59667?platform=web&product=rtcengine&menulabel=coresdk)
+- For the Nginx server, the domain name whitelist needs to be configured in the company's intranet firewall to allow the Nginx server to access TRTC's related domain names. Refer to: [TRTC Web SDK domain and port whitelist](https://trtc.io/document/59667)
 - For the Turn Server, allow the Turn Server to access the external network.

package/resource/{doc → knowledge}/rtcengine/web/plugin/35-advanced-ai-denoiser.md

@@ -2,7 +2,7 @@
 
 The AI Denoiser plugin reduces noise during calls and minimizes the impact of environmental noise on communication. It effectively suppresses keyboard sounds, collisions, and other noises, making it particularly suitable for scenarios sensitive to transient noise, such as conference settings.
 
-This article introduces how to apply the AI Denoiser to the local audio stream in TRTC applications. [Click here to experience it online](https://trtc.io/demo/homepage/#/detail?scene=trtc&active=ai) with the denoiser demo.
+This article introduces how to apply the AI Denoiser to the local audio stream in TRTC applications. [Click here to experience it online](https://trtc.io/demo/homepage/#/detail) with the denoiser demo.
 
 ## Prerequisites