mirror of
https://github.com/blakeblackshear/frigate.git
synced 2026-10-24 15:35:46 +03:00
Josh HawkinsandGitHub
64d6366ac4
* allow users to select live streaming technology * fix webrtc being downgraded to mse on load `useUserPersistence` seeds state with the default and loads asynchronously, so the first render always used `mse` instead of the saved choice, and `useWebRTCGloballyAvailable` reports `checking` until the probe settles and re-enters that state on every consumer mount, so a saved `webrtc` was rewritten to `mse` even after the probe had already passed. On Safari the MSE player then timed out and latched the jsmpeg fallback. A pending probe now counts as available, the player waits on `autoLive` until the stored preferences load, and `handleError` gates on the mode in use since the fallback flag no longer implies webrtc is untried. A rejected IndexedDB read also resolves `loaded` now, so a blocked store can't leave the player waiting forever. * add support for configurable ICE servers in WebRTC player * add mic error state, fix dialog overwriting saved choice and dashboard ignoring stream * tweaks
109 lines
3.2 KiB
TypeScript
109 lines
3.2 KiB
TypeScript
import { FrigateConfig } from "@/types/frigateConfig";
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import { WebRTCUnavailableReason } from "@/types/live";
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/**
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* Helpers for detecting what the current browser can do over WebRTC.
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* Used to gate the WebRTC live streaming option (e.g. H.265 is only
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* decodable over WebRTC on Chrome 136+ / Safari 18+ with HEVC hardware).
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*/
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const DEFAULT_ICE_SERVERS: RTCIceServer[] = [
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{ urls: "stun:stun.l.google.com:19302" },
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];
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type ConfiguredIceServers = FrigateConfig["go2rtc"]["webrtc"]["ice_servers"];
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/** ICE servers for browser peer connections, falling back to public STUN. */
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export function webRTCIceServers(
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configured: ConfiguredIceServers,
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): RTCIceServer[] {
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if (!configured?.length) {
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return DEFAULT_ICE_SERVERS;
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}
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return configured.map((server) => ({
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urls: server.urls,
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username: server.username,
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credential: server.credential,
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}));
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}
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export function browserSupportsWebRTC(): boolean {
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return typeof window !== "undefined" && "RTCPeerConnection" in window;
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}
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/** Codec aliases that refer to the same underlying codec. */
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const CODEC_ALIASES: Record<string, string> = {
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HEVC: "H265",
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AVC: "H264",
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};
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function normalizeCodec(codec: string): string {
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const upper = codec.toUpperCase();
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return CODEC_ALIASES[upper] ?? upper;
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}
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/** The set of video codec names the browser can receive over WebRTC. */
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export function browserWebRTCVideoCodecs(): string[] {
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if (
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typeof RTCRtpReceiver === "undefined" ||
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typeof RTCRtpReceiver.getCapabilities !== "function"
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) {
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return [];
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}
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const caps = RTCRtpReceiver.getCapabilities("video");
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if (!caps) return [];
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const codecs = new Set<string>();
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for (const codec of caps.codecs) {
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// mimeType is like "video/H264"; ignore infra codecs (rtx, red, ulpfec)
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const name = codec.mimeType.split("/")[1]?.toUpperCase();
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if (!name || ["RTX", "RED", "ULPFEC", "FLEXFEC-03"].includes(name)) {
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continue;
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}
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codecs.add(normalizeCodec(name));
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}
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return Array.from(codecs);
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}
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export function browserSupportsWebRTCVideoCodec(codec: string): boolean {
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return browserWebRTCVideoCodecs().includes(normalizeCodec(codec));
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}
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const DOCS_URL =
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"https://docs.frigate.video/configuration/live/#selecting-a-streaming-technology";
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// Availability is consumed by several components at once, so without this the
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// same fact prints once per consumer.
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const loggedMessages = new Set<string>();
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/**
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* Logs why WebRTC is unavailable, supplementing the short message shown inline
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* in the technology selector with the underlying detail.
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*
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* @param scope - Stream name for per-stream reasons, omitted for global ones
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* @param reason - The reason surfaced in the UI
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* @param description - The detail the UI message leaves out
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*/
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export function logWebRTCUnavailable(
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scope: string | undefined,
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reason: WebRTCUnavailableReason,
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description: string,
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): void {
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const message = `${scope ? `${scope} - ` : ""}WebRTC unavailable '${reason}': ${description} See the documentation: ${DOCS_URL}`;
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if (loggedMessages.has(message)) {
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return;
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}
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loggedMessages.add(message);
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// eslint-disable-next-line no-console
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console.warn(message);
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}
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/** Clears the emitted messages (e.g. when go2rtc config changes). */
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export function resetWebRTCUnavailableLog(): void {
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loggedMessages.clear();
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}
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