@streaming-cdn/rtc-web 1.3.11 → 1.3.13

package/README.md

@@ -1,4 +1,4 @@
-# RTC Web SDK 1.3.11
+# RTC Web SDK 1.3.13
 
 The customer package contains compiled ESM/UMD JavaScript and TypeScript declarations. It intentionally excludes implementation source and source maps. Example source remains available under `examples/`.
 
@@ -147,6 +147,63 @@ side receives a terminal call state. Manual integrations must call
 `await media.leave(); media.destroy()` for `ended`, `cancelled`, `rejected`,
 `busy`, `failed`, and `missed`. Calling the same terminal action twice is safe.
 
+## Transports
+
+`transport: "mesh"` (the default) opens one connection per pair, which suits
+direct calls and small rooms. `transport: "sfu"` sends media once to the
+platform SFU and receives each remote participant from it, which is what a
+large room needs — media no longer grows with the participant count.
+`transport: "auto"` picks the SFU for `meeting` mode and the mesh for `voice`
+and `video` calls. The credential, the events and the rest of the API are
+identical, so an application switches transports without changing its UI or
+call flow. SDP for the SFU travels over `/v1/rtc/sfu/*` authenticated by the
+same signaling token; no separate credential exists.
+
+The SFU uplink publishes three simulcast layers (720p/360p/180p-class), so the
+SFU can hand every subscriber the layer their downlink carries. `setQuality`
+caps which layers are encoded; there is no per-connection tier on a single
+uplink, so `peerquality` is not emitted on this transport. To deliberately
+receive a lower layer for one participant — a thumbnail tile has no use for
+720p — call `setRemoteQuality(participantId, "low" | "medium" | "high")` on
+the room client (`getNativeClient()`); the publisher keeps sending every
+layer. The React Native adapter has the same transport option and the same
+`setRemoteQuality` on the client.
+
+## Active speaker
+
+The client samples WebRTC audio levels once a second on both transports and
+emits `activespeaker` (`{ participantId }`, or `null` when the room is
+silent). The choice has hysteresis — the highlight only moves when somebody is
+clearly louder than the current speaker — so two people at similar volume do
+not flap the indicator. Use it to light up the speaking tile:
+
+```ts
+media.addEventListener("activespeaker", ({ detail }) => {
+  highlightTile(detail.detail.participantId);
+});
+```
+
+## Picture-in-picture
+
+`createPictureInPictureController(media.getNativeClient())` floats a
+participant's video in the browser's always-on-top window while the user works
+elsewhere. `enter(participantId?)` must run inside a click handler — browsers
+require a user gesture — and picks the first remote participant by default,
+falling back to the local preview. `onLeave` fires when the user closes the
+floating window; `supported` is false where the browser has no PiP.
+
+```ts
+import { createPictureInPictureController } from "@streaming-cdn/rtc-web";
+
+const pip = createPictureInPictureController(media.getNativeClient(), {
+  onLeave: () => showInlineVideoAgain()
+});
+pipButton.onclick = () => { void pip.enter(); };
+```
+
+Android and iOS system PiP require native application support and are on the
+SDK roadmap.
+
 ## Quality and codecs
 
 Use `setQuality("low" | "medium" | "high" | "audio" | "auto")` at runtime.

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@streaming-cdn/rtc-web",
-  "version": "1.3.11",
+  "version": "1.3.13",
   "type": "module",
   "main": "dist/rtc-web.umd.js",
   "module": "dist/rtc-web.es.js",