try to fix vc delay
This commit is contained in:
parent
6a14104143
commit
ffd952b1f0
2 changed files with 424 additions and 136 deletions
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@ -1,7 +1,6 @@
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let currentVoiceWs = null;
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let currentVoiceCallId = null;
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let currentVoiceCallType = null;
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let voiceWsHeartbeat = null;
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let voiceActiveUsers = [];
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let voiceAesKeyBytes = null; // grupowy klucz rozmowy (root) - dystrybuowany hybrydą PQ
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let voiceUiContainer = null;
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@ -107,8 +106,8 @@ function getVoicePeerSenderKey(clientIdStr) {
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let micStream = null, audioContext = null, isAudioWorkletLoaded = false;
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let micSourceNode = null, senderNode = null, receiverNode = null, audioEncoder = null;
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let remoteAudio = {}, audioDecoders = {}, audioDecryptChains = {};
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let micSourceNode = null, senderNode = null, receiverNode = null;
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let remoteAudio = {}, audioDecryptChains = {};
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let knownVoiceUsers = new Set();
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let isMicMuted = false;
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let remoteVideos = {}, videoDecryptChains = {};
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@ -126,13 +125,14 @@ let isVoiceSpeakerMode = false;
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const SPEAKING_HANGOVER_MS = 400;
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// ===== Voice connection resilience =====
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// Every client broadcasts a tiny encrypted ping so peers can tell "quiet
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// user" apart from "crashed/half-open socket". Peers silent for too long are
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// dropped locally; a dead server socket triggers an automatic rejoin.
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// VOICE_SERVER_TIMEOUT_MS liczy tylko ŻYWYCH peerów (deadPeers są wykluczone):
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// ghost z rosteru (martwy TCP po stronie serwera, zamiata go idle-sweeper do
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// 15 s) nie może prowokować pętli fałszywych reconnectów = migające "Connecting".
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//Zawory kolejki socketu: TCP dowiezie zakolejkowane bajty tylko spóźnione,
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//więc progi są hierarchiczne - wideo ustępuje przy 12 KB, audio dopiero przy
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//14 KB (~270 ms głosu). Przy korku najpierw znika obraz, głos leci dalej.
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const VOICE_WS_VIDEO_QUEUE_LIMIT_BYTES = 12 * 1024;
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const VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES = 14 * 1024;
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//Resilience: peer bez pakietów >7 s = lokalnie martwy (media wycięte), socket
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//bez ruchu od ŻYWYCH peerów >15 s = wymuszony reconnect. Pings co 2 s.
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const VOICE_PEER_TIMEOUT_MS = 7000;
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const VOICE_PING_INTERVAL_MS = 2000;
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const VOICE_SERVER_TIMEOUT_MS = 15000;
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@ -141,8 +141,6 @@ const VOICE_ROSTER_POLL_MS = 5000;
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let peerLastSeen = {};
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let deadPeers = new Set();
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let lastVoiceServerPacketAt = 0;
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//Czy w bieżącym okresie socketu byli jacyś inni uczestnicy - baza dla
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//resetu baseline'u ciszy w watchdogu (patrz komentarz przy użyciu)
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let voiceHadRemotePeers = false;
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let voicePingTimer = null;
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let voiceRosterRetryTimer = null;
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@ -172,11 +170,21 @@ function cleanupPeerMedia(userId) {
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function sendVoicePing() {
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if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
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const iv = nextVoiceIv();
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crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, new Uint8Array([Date.now() & 255])).then(cipher => {
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//Typ 5: prośba o echo. Payload = 8B timestamp (ms, WŁASNY zegar). Odbiorca
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//odsyła te same bajty jako typ 6, a RTT liczę dopiero na jego powrocie.
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const tsData = BigInt(Date.now());
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const tsBuf = new ArrayBuffer(8);
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new DataView(tsBuf).setBigUint64(0, tsData);
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crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, new Uint8Array(tsBuf)).then(cipher => {
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if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
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const payload = new Uint8Array(14 + cipher.byteLength);
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payload[0] = 5; payload[1] = 0; payload.set(iv, 2); payload.set(new Uint8Array(cipher), 14);
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currentVoiceWs.send(payload.buffer);
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voiceOutstandingPings.set(tsData, Date.now());
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const nowMs = Date.now();
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for (const [k, sentAt] of voiceOutstandingPings) {
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if (nowMs - sentAt > 15000) voiceOutstandingPings.delete(k);
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}
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}).catch(() => {});
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}
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@ -222,9 +230,15 @@ const VOICE_FPS_OPTIONS = [15, 24, 30, 60];
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let voiceSettings = loadVoiceSettings();
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function loadVoiceSettings() {
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const defaults = { micId: "", camId: "", speakerId: "", camRes: "1280x720", camFps: 24, screenRes: "1920x1080", screenFps: 30, sensitivity: VOICE_SENSITIVITY_DEFAULT };
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//Globalny default 480p@24; stare zapisane defaulty (720p/1080p/30fps)
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//migrują na nowe, świadome wybory użytkownika zostają.
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const defaults = { micId: "", camId: "", speakerId: "", camRes: "640x480", camFps: 24, screenRes: "640x480", screenFps: 24, sensitivity: VOICE_SENSITIVITY_DEFAULT };
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try {
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return Object.assign(defaults, JSON.parse(localStorage.getItem(VOICE_SETTINGS_KEY)) || {});
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let s = Object.assign({}, JSON.parse(localStorage.getItem(VOICE_SETTINGS_KEY)) || {});
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if (s.camRes === "1280x720") s.camRes = defaults.camRes;
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if (s.screenRes === "1920x1080") s.screenRes = defaults.screenRes;
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if (s.screenFps === 30 || s.screenFps === "30") s.screenFps = defaults.screenFps;
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return Object.assign(defaults, s);
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} catch (e) {
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return defaults;
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}
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@ -279,6 +293,35 @@ async function applyVoiceSpeakerSink() {
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let voiceConnectedAt = null;
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let voiceTimerInterval = null;
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let voicePingMs = null;
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//Kontrola congestionu: bufferedAmount nie widzi backlogu kernela TCP, prawdziwy
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//obraz daje RTT z pingów. Baseline = MINIMUM z okna próbek (odporne na skoki
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//i na peera sztucznie opóźniającego własne echa - stałe kłamstwo podnosi minimum).
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let voiceRttFastMs = 0;
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let voiceRttSamples = [];
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let voiceRttBaseMs = 0;
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let voiceAudioHoldUntil = 0; //histereza zaworu audio
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//Echo-ping (typ 5 = prośba, typ 6 = echo): RTT wyłącznie na WŁASNYM zegarze -
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//wysyłam timestamp, peer odbija te same bajty, odejmuję czas nadania. Zegar
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//peera nigdy nie jest używany (może być przekręcony lub spreparowany).
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const voiceOutstandingPings = new Map(); // BigInt(ts) -> czas nadania
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let voiceLastEchoAt = 0; // rate-limit odpowiedzi echem
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let voiceLastRttSampleAt = 0; // kiedy ostatnio spłynął prawdziwy pomiar
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function voiceCongestionDelay() {
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if (!voiceRttFastMs || !voiceRttBaseMs) return 0;
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return Math.min(5000, Math.max(0, voiceRttFastMs - voiceRttBaseMs));
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}
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function voiceRecordRttSample(rtt) {
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voiceLastRttSampleAt = Date.now();
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voicePingMs = voicePingMs === null ? rtt : Math.round(voicePingMs * 0.7 + rtt * 0.3);
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voiceRttFastMs = voiceRttFastMs ? Math.round(voiceRttFastMs * 0.5 + rtt * 0.5) : rtt;
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voiceRttSamples.push(rtt);
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if (voiceRttSamples.length > 12) voiceRttSamples.shift();
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voiceRttBaseMs = Math.min.apply(null, voiceRttSamples);
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updateVoicePing();
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}
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function formatVoiceDuration(ms) {
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let totalSec = Math.floor(ms / 1000);
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@ -742,13 +785,8 @@ class AudioReceiver extends AudioWorkletProcessor {
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else {
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if (available > 6144) user.readPtr = (user.writePtr - 2048 + 65536) % 65536;
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else if (available > 2049) {
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//Adaptacyjny odzysk latencji: nadmiar nad target odprowadzany
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//proporcjonalnie (1 + excess/256 próbek na kwant), max ~17
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//= pełny powrót z 128 ms do ~43 ms w ułamku sekundy. Stale
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//"1 próbka/kwant" (~8 ms opóźnienia na sekundę) pozwalało,
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//by po każdym burstu sieciowym latency wisiał pod twardym
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//limitem przez kilkanaście sekund - odczuwalne jako "dziwnie
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//duże opóźnienie" na sieci komórkowej.
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//Adaptacyjny drain: nadmiar nad target odprowadzany proporcjonalnie
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//(1 + excess/256 /kwant) - powrót z ~128 ms do ~43 ms w ułamku sekundy
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const excess = available - 2048;
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user.readPtr = (user.readPtr + 1 + (excess >> 8)) % 65536;
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}
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@ -909,11 +947,9 @@ function bytesToB64u(a) {
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return btoa(bin).replace(/\+/g, "-").replace(/\//g, "_").replace(/=+$/, "");
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}
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//Pobiera skład pokoju z voice-servera (źródło prawdy o rosterze) i synchronizuje
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//UI. Wywoływana: na otwarciu socketu (plus szybkie retrye do czasu pojawienia
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//sie własnego kafelka), przy pakiecie od nieznanego clientId ORAZ cyklicznie co
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//VOICE_ROSTER_POLL_MS - bez steady-pollu zniknięcie ghosta z rosteru (sweep po
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//stronie serwera / leave bez call_update) nigdy nie docierało do naszych kafelków.
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//Roster pokoju z voice-servera (źródło prawdy) -> synchronizacja kafelków.
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//Steady-poll co VOICE_ROSTER_POLL_MS - bez tego sweep ghosta/leave bez
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//call_update nigdy nie docierał do UI.
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async function fetchVoiceRoster() {
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if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !currentVoiceCallId) return;
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try {
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@ -922,8 +958,11 @@ async function fetchVoiceRoster() {
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const t0 = performance.now();
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const res = await fetch(`${httpUrl}/room/users?room=${encodeURIComponent(currentVoiceCallId)}`);
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const usersRaw = await res.json();
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voicePingMs = Math.round(performance.now() - t0);
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updateVoicePing();
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//HTTP tylko jako szacunek gdy brak prawdziwych próbek RTT (solo / stary peer)
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if (voicePingMs === null || Date.now() - voiceLastRttSampleAt > 25000) {
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voicePingMs = Math.round(performance.now() - t0);
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updateVoicePing();
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}
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const rawUsers = usersRaw.map(u => String(u));
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//Serwer jest źródłem prawdy o składzie - nie filtrujemy po
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//deadPeers (audyt S15: ukryty peer nadal dostaje audio)
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@ -1062,12 +1101,18 @@ async function startVoiceSession(serverUrl, ticket) {
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}
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let vadHangover = 0;
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const myIdStr = String(window.id);
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senderNode.port.onmessage = async (e) => {
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if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
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// Backpressure valve: when the socket is congested, drop mic
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// frames instead of queueing them - queued audio would arrive
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// minutes late and drown the connection for every stream.
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if (currentVoiceWs.bufferedAmount > 256 * 1024) {
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//Zawór bajtowy + zawór RTT (kolejka kernela jest niewidoczna dla
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//bufferedAmount). Drop >450 ms nadwyżki z histerezą 700 ms - lepiej
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//urwać ułamek sekundy niż mówić z sekundami opóźnienia.
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if (currentVoiceWs.bufferedAmount > VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES) {
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vadHangover = 0;
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return;
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}
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if (voiceCongestionDelay() > 450) voiceAudioHoldUntil = Date.now() + 700;
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if (Date.now() < voiceAudioHoldUntil) {
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vadHangover = 0;
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return;
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}
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@ -1082,7 +1127,7 @@ async function startVoiceSession(serverUrl, ticket) {
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if (db >= sensitivity) vadHangover = 75;
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let isSpeaking = vadHangover > 0 && !isMicMuted;
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setTileSpeaking(String(window.id), isSpeaking);
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setTileSpeaking(myIdStr, isSpeaking);
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if (vadHangover > 0 && !isMicMuted) {
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vadHangover--;
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@ -1113,7 +1158,44 @@ async function startVoiceSession(serverUrl, ticket) {
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if (deadPeers.has(clientId)) deadPeers.delete(clientId);
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const type = view.getUint8(8);
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if (type === 5) return; // liveness ping - presence already recorded
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if (type === 5 || type === 6) {
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//Liveness już zapisany wyżej. Dalej tylko przy zainstalowanym kluczu.
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if (voiceAesKeyBytes) {
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const senderKey = getVoicePeerSenderKey(clientId);
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if (senderKey) {
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//Offsety ODBIORCZE: serwer dokleja 8B nadawcy, więc iv=[10..22),
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//ct=[22..). Format wysyłki (iv=2, ct=14) jest INNY - pomylenie
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//psuje każdą deszyfrację cicho (GCM auth fail -> .catch)
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const piv = new Uint8Array(arrayBuffer, 10, 12);
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const pct = new Uint8Array(arrayBuffer, 22);
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crypto.subtle.decrypt({name: "AES-GCM", iv: piv}, senderKey, pct).then(plain => {
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if (plain.byteLength !== 8) return; //stary klient: ping bez timestampu
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const tsData = new DataView(plain).getBigUint64(0);
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if (type === 5) {
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//Odbij te same bajty własnym podkluczem i świeżym IV; rate-limit
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const nowMs = Date.now();
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if (nowMs - voiceLastEchoAt < 150) return;
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voiceLastEchoAt = nowMs;
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if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
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const riv = nextVoiceIv();
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crypto.subtle.encrypt({name: "AES-GCM", iv: riv}, voiceMySenderCryptoKey, plain).then(rcipher => {
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if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
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const rpayload = new Uint8Array(14 + rcipher.byteLength);
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rpayload[0] = 6; rpayload[1] = 0; rpayload.set(riv, 2); rpayload.set(new Uint8Array(rcipher), 14);
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currentVoiceWs.send(rpayload.buffer);
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}).catch(() => {});
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} else {
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//Typ 6: powrót własnego timestampu - RTT na WŁASNYM zegarze.
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if (!voiceOutstandingPings.has(tsData)) return;
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voiceOutstandingPings.delete(tsData);
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const rtt = Date.now() - Number(tsData);
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if (rtt >= 1 && rtt < 15000) voiceRecordRttSample(rtt);
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}
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}).catch(() => {});
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}
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}
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return;
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}
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if (!voiceActiveUsers.includes(clientId) && clientId !== String(window.id)) {
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const baseUrl = currentVoiceWs.url.split('/ws')[0];
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@ -1365,8 +1447,6 @@ async function handleCallUpdate(data) {
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for (let u in remoteAudio) {
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let pureU = u.replace("_screen", "");
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if (!voiceActiveUsers.includes(pureU) && pureU !== myIdStr) {
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if (audioDecoders[u]) try { audioDecoders[u].close(); } catch(e){}
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delete audioDecoders[u];
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delete remoteAudio[u];
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const rn = u.endsWith("_screen") ? screenReceiverNode : receiverNode;
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if(rn) rn.port.postMessage({ action: 'remove', clientId: u });
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@ -2523,7 +2603,6 @@ function toggleVoiceMic() {
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// keeping the UI, mic stream and AudioContext alive for an automatic rejoin.
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function teardownVoiceTransport() {
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if (voicePingTimer) { clearInterval(voicePingTimer); voicePingTimer = null; }
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if (voiceWsHeartbeat) { clearInterval(voiceWsHeartbeat); voiceWsHeartbeat = null; }
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if (camControl) { try { camControl.stop(); } catch (e) {} camControl = null; }
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if (screenControl) { try { screenControl.stop(); } catch (e) {} screenControl = null; }
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if (currentVoiceWs) {
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@ -2550,6 +2629,12 @@ function teardownVoiceTransport() {
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setVoiceRootKey(null);
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voiceSendEpoch = null;
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voiceSendCounter = 0n;
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voiceRttFastMs = 0;
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voiceRttSamples = [];
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voiceRttBaseMs = 0;
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voiceAudioHoldUntil = 0;
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voiceOutstandingPings.clear();
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voiceLastRttSampleAt = 0;
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lastVoiceServerPacketAt = Date.now();
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}
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@ -2566,6 +2651,10 @@ function scheduleVoiceReconnect() {
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voiceReconnectAttempts++;
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setVoiceStatus("connecting");
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teardownVoiceTransport();
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//Zmiana sieci = stary RTT dotyczy martwej ścieżki; bez resetu wskaźnik
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//wisiałby zamrożony, a fallback HTTP nie miał prawa go nadpisać
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voicePingMs = null;
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updateVoicePing();
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setTimeout(async () => {
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if (!currentVoiceCallId || !currentVoiceCallType) return;
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try {
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@ -2617,9 +2706,7 @@ function endVoiceSession() {
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micStream = null;
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}
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if (micSourceNode) { micSourceNode.disconnect(); micSourceNode = null; }
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if (audioEncoder) { try { audioEncoder.close(); } catch(e){} audioEncoder = null; }
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if (voiceWsHeartbeat) { clearInterval(voiceWsHeartbeat); voiceWsHeartbeat = null; }
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if (voicePingTimer) { clearInterval(voicePingTimer); voicePingTimer = null; }
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if (voiceRosterRetryTimer) { clearInterval(voiceRosterRetryTimer); voiceRosterRetryTimer = null; }
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if (voiceRosterPollTimer) { clearInterval(voiceRosterPollTimer); voiceRosterPollTimer = null; }
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@ -2629,14 +2716,17 @@ function endVoiceSession() {
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voiceReconnectAttempts = 0;
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voiceActiveUsers = [];
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voiceHadRemotePeers = false;
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for (let id in audioDecoders) {
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try { audioDecoders[id].close(); } catch(e){}
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}
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voiceRttFastMs = 0;
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voiceRttSamples = [];
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voiceRttBaseMs = 0;
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voiceAudioHoldUntil = 0;
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voiceOutstandingPings.clear();
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voiceLastEchoAt = 0;
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voiceLastRttSampleAt = 0;
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for (let id in audioDecryptChains) {
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if(audioDecryptChains[id]) audioDecryptChains[id].clear();
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}
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audioDecoders = {};
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audioDecryptChains = {};
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remoteAudio = {};
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@ -2708,8 +2798,11 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
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screenSenderNode = new AudioWorkletNode(audioContext, 'audio-sender');
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screenSenderNode.port.onmessage = async (e) => {
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if (currentVoiceWs?.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
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// Same backpressure valve as the mic sender.
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if (currentVoiceWs.bufferedAmount > 256 * 1024) return;
|
||||
// Ten sam zawór co mikrofon (limit bajtów + korek RTT) - audio
|
||||
// ma pierwszeństwo przed wideo na wspólnym sockecie.
|
||||
if (currentVoiceWs.bufferedAmount > VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES) return;
|
||||
if (voiceCongestionDelay() > 450) voiceAudioHoldUntil = Date.now() + 700;
|
||||
if (Date.now() < voiceAudioHoldUntil) return;
|
||||
const iv = nextVoiceIv();
|
||||
const cipher = await crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, e.data.audioData);
|
||||
const payload = new Uint8Array(14 + cipher.byteLength);
|
||||
|
|
@ -2740,6 +2833,23 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
|
|||
};
|
||||
|
||||
let currentConfig = getCurrentConfig(), isRunning = true, lastKeyframeAt = 0;
|
||||
//AIMD + slow-start: start na 40% celu (pełną prędkością waliłby w nieznany
|
||||
//uplink, a bufory TCP w kernelu to MB-y = sekundy kolejki zanim cokolwiek
|
||||
//zmierzymy). Sygnał korka: skipy (zawór 12 KB) lub nadwyżka RTT.
|
||||
let brScale = 0.4, encTries = 0, encSkips = 0, brWindowAt = performance.now(), lastBrReconfAt = 0;
|
||||
//Korek trwa mimo podłogi bitrate'u → FPS spada do 1/2, 1/3
|
||||
let fpsDiv = 1, lowStreak = 0, cleanStreak = 0, tickCount = 0;
|
||||
let activeBitrate = Math.round(currentConfig.bitrate * brScale);
|
||||
const applyEncoderBitrate = (force = false) => {
|
||||
const target = Math.max(120000, Math.round(currentConfig.bitrate * brScale));
|
||||
if (!force && Math.abs(target - activeBitrate) / activeBitrate < 0.08) return;
|
||||
if (!force && performance.now() - lastBrReconfAt < 800) return;
|
||||
activeBitrate = target;
|
||||
lastBrReconfAt = performance.now();
|
||||
try {
|
||||
encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: activeBitrate, latencyMode: 'realtime'});
|
||||
} catch (e) {}
|
||||
};
|
||||
const encoder = new VideoEncoder({
|
||||
output: async (chunk) => {
|
||||
if (currentVoiceWs?.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
|
||||
|
|
@ -2751,7 +2861,7 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
|
|||
currentVoiceWs.send(payload.buffer);
|
||||
}, error: (e) => { voiceLogError('videoencoder', e); stopFunction(); }
|
||||
});
|
||||
encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: currentConfig.bitrate, latencyMode: 'realtime'});
|
||||
encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: activeBitrate, latencyMode: 'realtime'});
|
||||
|
||||
// Transient capture-mode wobbles must not thrash the encoder -
|
||||
// commit only after ~300 ms stable. Camera size/fps changes arrive
|
||||
|
|
@ -2769,31 +2879,65 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
|
|||
pendingCfg = newConfig;
|
||||
pendingAt = performance.now();
|
||||
} else if (performance.now() - pendingAt > 300) {
|
||||
//Zmiana geometrii źródła - uczony scale łącza zostaje,
|
||||
//resetujemy tylko redukcję FPS
|
||||
currentConfig = newConfig; forceKeyframe = true; lastKeyframeAt = 0; pendingCfg = null;
|
||||
encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: currentConfig.bitrate, latencyMode: 'realtime'});
|
||||
activeBitrate = Math.round(currentConfig.bitrate * brScale);
|
||||
fpsDiv = 1; lowStreak = 0; cleanStreak = 0;
|
||||
applyEncoderBitrate(true);
|
||||
}
|
||||
} else if (pendingCfg) {
|
||||
pendingCfg = null; // wiggle reverted on its own - ignore
|
||||
}
|
||||
|
||||
if (encoder.state === 'configured' && encoder.encodeQueueSize <= 2 && currentVoiceWs.bufferedAmount < 1024 * 256 && videoEl.readyState >= 2) {
|
||||
let frame = null;
|
||||
try {
|
||||
const nowMs = performance.now();
|
||||
frame = new VideoFrame(videoEl, {timestamp: Math.trunc(nowMs * 1000)});
|
||||
//Klatka kluczowa co ~1 s według ZEGARA, nie licznika zakodowanych
|
||||
//klatek: zawór congestion pomija klatki, więc licznik rozjeżdżał
|
||||
//się z realnym tempem i keyframes stawały się rzadkie. Widz
|
||||
//dołączający w trakcie udostępniania musi zaczekać na keyframe -
|
||||
//z licznikiem potrafiło to trwać bardzo długo (efekt "kamery
|
||||
//nie widać"), z zegarem to maksymalnie ~1 s.
|
||||
const wantKeyframe = forceKeyframe || (nowMs - lastKeyframeAt >= 1000);
|
||||
encoder.encode(frame, {keyFrame: wantKeyframe});
|
||||
if (wantKeyframe) lastKeyframeAt = nowMs;
|
||||
} catch (err) { } finally { if (frame) frame.close(); }
|
||||
}
|
||||
setTimeout(encodeLoop, 1000 / currentConfig.fps);
|
||||
};
|
||||
//Zawór kolejki socketu (12 KB) ustępuje DALEJ przed progiem audio
|
||||
//(14 KB) - przy korku najpierw znika obraz, głos płynie dalej.
|
||||
tickCount++;
|
||||
const policySkip = fpsDiv > 1 && (tickCount % fpsDiv !== 0);
|
||||
let skipped = false;
|
||||
if (policySkip) {
|
||||
//celowe tempo (fpsDiv) - nie liczy się do skipRatio
|
||||
} else if (encoder.state === 'configured' && encoder.encodeQueueSize <= 2 &&
|
||||
currentVoiceWs.readyState === WebSocket.OPEN &&
|
||||
currentVoiceWs.bufferedAmount < VOICE_WS_VIDEO_QUEUE_LIMIT_BYTES && videoEl.readyState >= 2) {
|
||||
let frame = null;
|
||||
try {
|
||||
const nowMs = performance.now();
|
||||
frame = new VideoFrame(videoEl, {timestamp: Math.trunc(nowMs * 1000)});
|
||||
//Keyframe wg zegara co ~1 s (nie licznika klatek - zawór je pomija)
|
||||
const wantKeyframe = forceKeyframe || (nowMs - lastKeyframeAt >= 1000);
|
||||
encoder.encode(frame, {keyFrame: wantKeyframe});
|
||||
if (wantKeyframe) lastKeyframeAt = nowMs;
|
||||
} catch (err) { } finally { if (frame) frame.close(); }
|
||||
} else if (encoder.state === 'configured') {
|
||||
skipped = true;
|
||||
}
|
||||
|
||||
encTries++;
|
||||
if (skipped) encSkips++;
|
||||
const winNow = performance.now();
|
||||
if (winNow - brWindowAt >= 800) {
|
||||
const skipRatio = encTries > 0 ? encSkips / encTries : 0;
|
||||
const congestionMs = voiceCongestionDelay();
|
||||
if (congestionMs > 450 || skipRatio > 0.3 || congestionMs > 250) {
|
||||
brScale = Math.max(0.15, brScale * (congestionMs > 450 ? 0.6 : 0.75));
|
||||
applyEncoderBitrate();
|
||||
if (brScale <= 0.17) lowStreak++; else lowStreak = 0;
|
||||
if (lowStreak >= 4 && fpsDiv < 3) { fpsDiv++; lowStreak = 0; }
|
||||
cleanStreak = 0;
|
||||
} else if (skipRatio === 0 && congestionMs < 80) {
|
||||
cleanStreak++;
|
||||
lowStreak = 0;
|
||||
if (cleanStreak >= 5) {
|
||||
if (fpsDiv > 1) { fpsDiv--; }
|
||||
else if (brScale < 1) { brScale = Math.min(1, brScale * 1.2); applyEncoderBitrate(); }
|
||||
cleanStreak = 0;
|
||||
}
|
||||
}
|
||||
encTries = 0; encSkips = 0; brWindowAt = winNow;
|
||||
}
|
||||
setTimeout(encodeLoop, 1000 / currentConfig.fps);
|
||||
};
|
||||
setTimeout(encodeLoop, 500);
|
||||
|
||||
videoTrack.onended = () => { if(isRunning) stopFunction(); };
|
||||
|
|
|
|||
280
webroot/voice.js
280
webroot/voice.js
|
|
@ -1,7 +1,6 @@
|
|||
let currentVoiceWs = null;
|
||||
let currentVoiceCallId = null;
|
||||
let currentVoiceCallType = null;
|
||||
let voiceWsHeartbeat = null;
|
||||
let voiceActiveUsers = [];
|
||||
let voiceAesKeyBytes = null; // grupowy klucz rozmowy (root) - dystrybuowany hybrydą PQ
|
||||
let voiceUiContainer = null;
|
||||
|
|
@ -107,8 +106,8 @@ function getVoicePeerSenderKey(clientIdStr) {
|
|||
|
||||
|
||||
let micStream = null, audioContext = null, isAudioWorkletLoaded = false;
|
||||
let micSourceNode = null, senderNode = null, receiverNode = null, audioEncoder = null;
|
||||
let remoteAudio = {}, audioDecoders = {}, audioDecryptChains = {};
|
||||
let micSourceNode = null, senderNode = null, receiverNode = null;
|
||||
let remoteAudio = {}, audioDecryptChains = {};
|
||||
let knownVoiceUsers = new Set();
|
||||
let isMicMuted = false;
|
||||
let remoteVideos = {}, videoDecryptChains = {};
|
||||
|
|
@ -126,13 +125,14 @@ let isVoiceSpeakerMode = false;
|
|||
|
||||
const SPEAKING_HANGOVER_MS = 400;
|
||||
|
||||
// ===== Voice connection resilience =====
|
||||
// Every client broadcasts a tiny encrypted ping so peers can tell "quiet
|
||||
// user" apart from "crashed/half-open socket". Peers silent for too long are
|
||||
// dropped locally; a dead server socket triggers an automatic rejoin.
|
||||
// VOICE_SERVER_TIMEOUT_MS liczy tylko ŻYWYCH peerów (deadPeers są wykluczone):
|
||||
// ghost z rosteru (martwy TCP po stronie serwera, zamiata go idle-sweeper do
|
||||
// 15 s) nie może prowokować pętli fałszywych reconnectów = migające "Connecting".
|
||||
//Zawory kolejki socketu: TCP dowiezie zakolejkowane bajty tylko spóźnione,
|
||||
//więc progi są hierarchiczne - wideo ustępuje przy 12 KB, audio dopiero przy
|
||||
//14 KB (~270 ms głosu). Przy korku najpierw znika obraz, głos leci dalej.
|
||||
const VOICE_WS_VIDEO_QUEUE_LIMIT_BYTES = 12 * 1024;
|
||||
const VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES = 14 * 1024;
|
||||
|
||||
//Resilience: peer bez pakietów >7 s = lokalnie martwy (media wycięte), socket
|
||||
//bez ruchu od ŻYWYCH peerów >15 s = wymuszony reconnect. Pings co 2 s.
|
||||
const VOICE_PEER_TIMEOUT_MS = 7000;
|
||||
const VOICE_PING_INTERVAL_MS = 2000;
|
||||
const VOICE_SERVER_TIMEOUT_MS = 15000;
|
||||
|
|
@ -141,8 +141,6 @@ const VOICE_ROSTER_POLL_MS = 5000;
|
|||
let peerLastSeen = {};
|
||||
let deadPeers = new Set();
|
||||
let lastVoiceServerPacketAt = 0;
|
||||
//Czy w bieżącym okresie socketu byli jacyś inni uczestnicy - baza dla
|
||||
//resetu baseline'u ciszy w watchdogu (patrz komentarz przy użyciu)
|
||||
let voiceHadRemotePeers = false;
|
||||
let voicePingTimer = null;
|
||||
let voiceRosterRetryTimer = null;
|
||||
|
|
@ -172,11 +170,21 @@ function cleanupPeerMedia(userId) {
|
|||
function sendVoicePing() {
|
||||
if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
|
||||
const iv = nextVoiceIv();
|
||||
crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, new Uint8Array([Date.now() & 255])).then(cipher => {
|
||||
//Typ 5: prośba o echo. Payload = 8B timestamp (ms, WŁASNY zegar). Odbiorca
|
||||
//odsyła te same bajty jako typ 6, a RTT liczę dopiero na jego powrocie.
|
||||
const tsData = BigInt(Date.now());
|
||||
const tsBuf = new ArrayBuffer(8);
|
||||
new DataView(tsBuf).setBigUint64(0, tsData);
|
||||
crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, new Uint8Array(tsBuf)).then(cipher => {
|
||||
if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
|
||||
const payload = new Uint8Array(14 + cipher.byteLength);
|
||||
payload[0] = 5; payload[1] = 0; payload.set(iv, 2); payload.set(new Uint8Array(cipher), 14);
|
||||
currentVoiceWs.send(payload.buffer);
|
||||
voiceOutstandingPings.set(tsData, Date.now());
|
||||
const nowMs = Date.now();
|
||||
for (const [k, sentAt] of voiceOutstandingPings) {
|
||||
if (nowMs - sentAt > 15000) voiceOutstandingPings.delete(k);
|
||||
}
|
||||
}).catch(() => {});
|
||||
}
|
||||
|
||||
|
|
@ -222,9 +230,15 @@ const VOICE_FPS_OPTIONS = [15, 24, 30, 60];
|
|||
let voiceSettings = loadVoiceSettings();
|
||||
|
||||
function loadVoiceSettings() {
|
||||
const defaults = { micId: "", camId: "", speakerId: "", camRes: "1280x720", camFps: 24, screenRes: "1920x1080", screenFps: 30, sensitivity: VOICE_SENSITIVITY_DEFAULT };
|
||||
//Globalny default 480p@24; stare zapisane defaulty (720p/1080p/30fps)
|
||||
//migrują na nowe, świadome wybory użytkownika zostają.
|
||||
const defaults = { micId: "", camId: "", speakerId: "", camRes: "640x480", camFps: 24, screenRes: "640x480", screenFps: 24, sensitivity: VOICE_SENSITIVITY_DEFAULT };
|
||||
try {
|
||||
return Object.assign(defaults, JSON.parse(localStorage.getItem(VOICE_SETTINGS_KEY)) || {});
|
||||
let s = Object.assign({}, JSON.parse(localStorage.getItem(VOICE_SETTINGS_KEY)) || {});
|
||||
if (s.camRes === "1280x720") s.camRes = defaults.camRes;
|
||||
if (s.screenRes === "1920x1080") s.screenRes = defaults.screenRes;
|
||||
if (s.screenFps === 30 || s.screenFps === "30") s.screenFps = defaults.screenFps;
|
||||
return Object.assign(defaults, s);
|
||||
} catch (e) {
|
||||
return defaults;
|
||||
}
|
||||
|
|
@ -279,6 +293,35 @@ async function applyVoiceSpeakerSink() {
|
|||
let voiceConnectedAt = null;
|
||||
let voiceTimerInterval = null;
|
||||
let voicePingMs = null;
|
||||
//Kontrola congestionu: bufferedAmount nie widzi backlogu kernela TCP, prawdziwy
|
||||
//obraz daje RTT z pingów. Baseline = MINIMUM z okna próbek (odporne na skoki
|
||||
//i na peera sztucznie opóźniającego własne echa - stałe kłamstwo podnosi minimum).
|
||||
let voiceRttFastMs = 0;
|
||||
let voiceRttSamples = [];
|
||||
let voiceRttBaseMs = 0;
|
||||
let voiceAudioHoldUntil = 0; //histereza zaworu audio
|
||||
|
||||
//Echo-ping (typ 5 = prośba, typ 6 = echo): RTT wyłącznie na WŁASNYM zegarze -
|
||||
//wysyłam timestamp, peer odbija te same bajty, odejmuję czas nadania. Zegar
|
||||
//peera nigdy nie jest używany (może być przekręcony lub spreparowany).
|
||||
const voiceOutstandingPings = new Map(); // BigInt(ts) -> czas nadania
|
||||
let voiceLastEchoAt = 0; // rate-limit odpowiedzi echem
|
||||
let voiceLastRttSampleAt = 0; // kiedy ostatnio spłynął prawdziwy pomiar
|
||||
|
||||
function voiceCongestionDelay() {
|
||||
if (!voiceRttFastMs || !voiceRttBaseMs) return 0;
|
||||
return Math.min(5000, Math.max(0, voiceRttFastMs - voiceRttBaseMs));
|
||||
}
|
||||
|
||||
function voiceRecordRttSample(rtt) {
|
||||
voiceLastRttSampleAt = Date.now();
|
||||
voicePingMs = voicePingMs === null ? rtt : Math.round(voicePingMs * 0.7 + rtt * 0.3);
|
||||
voiceRttFastMs = voiceRttFastMs ? Math.round(voiceRttFastMs * 0.5 + rtt * 0.5) : rtt;
|
||||
voiceRttSamples.push(rtt);
|
||||
if (voiceRttSamples.length > 12) voiceRttSamples.shift();
|
||||
voiceRttBaseMs = Math.min.apply(null, voiceRttSamples);
|
||||
updateVoicePing();
|
||||
}
|
||||
|
||||
function formatVoiceDuration(ms) {
|
||||
let totalSec = Math.floor(ms / 1000);
|
||||
|
|
@ -742,13 +785,8 @@ class AudioReceiver extends AudioWorkletProcessor {
|
|||
else {
|
||||
if (available > 6144) user.readPtr = (user.writePtr - 2048 + 65536) % 65536;
|
||||
else if (available > 2049) {
|
||||
//Adaptacyjny odzysk latencji: nadmiar nad target odprowadzany
|
||||
//proporcjonalnie (1 + excess/256 próbek na kwant), max ~17
|
||||
//= pełny powrót z 128 ms do ~43 ms w ułamku sekundy. Stale
|
||||
//"1 próbka/kwant" (~8 ms opóźnienia na sekundę) pozwalało,
|
||||
//by po każdym burstu sieciowym latency wisiał pod twardym
|
||||
//limitem przez kilkanaście sekund - odczuwalne jako "dziwnie
|
||||
//duże opóźnienie" na sieci komórkowej.
|
||||
//Adaptacyjny drain: nadmiar nad target odprowadzany proporcjonalnie
|
||||
//(1 + excess/256 /kwant) - powrót z ~128 ms do ~43 ms w ułamku sekundy
|
||||
const excess = available - 2048;
|
||||
user.readPtr = (user.readPtr + 1 + (excess >> 8)) % 65536;
|
||||
}
|
||||
|
|
@ -909,11 +947,9 @@ function bytesToB64u(a) {
|
|||
return btoa(bin).replace(/\+/g, "-").replace(/\//g, "_").replace(/=+$/, "");
|
||||
}
|
||||
|
||||
//Pobiera skład pokoju z voice-servera (źródło prawdy o rosterze) i synchronizuje
|
||||
//UI. Wywoływana: na otwarciu socketu (plus szybkie retrye do czasu pojawienia
|
||||
//sie własnego kafelka), przy pakiecie od nieznanego clientId ORAZ cyklicznie co
|
||||
//VOICE_ROSTER_POLL_MS - bez steady-pollu zniknięcie ghosta z rosteru (sweep po
|
||||
//stronie serwera / leave bez call_update) nigdy nie docierało do naszych kafelków.
|
||||
//Roster pokoju z voice-servera (źródło prawdy) -> synchronizacja kafelków.
|
||||
//Steady-poll co VOICE_ROSTER_POLL_MS - bez tego sweep ghosta/leave bez
|
||||
//call_update nigdy nie docierał do UI.
|
||||
async function fetchVoiceRoster() {
|
||||
if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !currentVoiceCallId) return;
|
||||
try {
|
||||
|
|
@ -922,8 +958,11 @@ async function fetchVoiceRoster() {
|
|||
const t0 = performance.now();
|
||||
const res = await fetch(`${httpUrl}/room/users?room=${encodeURIComponent(currentVoiceCallId)}`);
|
||||
const usersRaw = await res.json();
|
||||
voicePingMs = Math.round(performance.now() - t0);
|
||||
updateVoicePing();
|
||||
//HTTP tylko jako szacunek gdy brak prawdziwych próbek RTT (solo / stary peer)
|
||||
if (voicePingMs === null || Date.now() - voiceLastRttSampleAt > 25000) {
|
||||
voicePingMs = Math.round(performance.now() - t0);
|
||||
updateVoicePing();
|
||||
}
|
||||
const rawUsers = usersRaw.map(u => String(u));
|
||||
//Serwer jest źródłem prawdy o składzie - nie filtrujemy po
|
||||
//deadPeers (audyt S15: ukryty peer nadal dostaje audio)
|
||||
|
|
@ -1062,12 +1101,18 @@ async function startVoiceSession(serverUrl, ticket) {
|
|||
}
|
||||
|
||||
let vadHangover = 0;
|
||||
const myIdStr = String(window.id);
|
||||
senderNode.port.onmessage = async (e) => {
|
||||
if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
|
||||
// Backpressure valve: when the socket is congested, drop mic
|
||||
// frames instead of queueing them - queued audio would arrive
|
||||
// minutes late and drown the connection for every stream.
|
||||
if (currentVoiceWs.bufferedAmount > 256 * 1024) {
|
||||
//Zawór bajtowy + zawór RTT (kolejka kernela jest niewidoczna dla
|
||||
//bufferedAmount). Drop >450 ms nadwyżki z histerezą 700 ms - lepiej
|
||||
//urwać ułamek sekundy niż mówić z sekundami opóźnienia.
|
||||
if (currentVoiceWs.bufferedAmount > VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES) {
|
||||
vadHangover = 0;
|
||||
return;
|
||||
}
|
||||
if (voiceCongestionDelay() > 450) voiceAudioHoldUntil = Date.now() + 700;
|
||||
if (Date.now() < voiceAudioHoldUntil) {
|
||||
vadHangover = 0;
|
||||
return;
|
||||
}
|
||||
|
|
@ -1082,7 +1127,7 @@ async function startVoiceSession(serverUrl, ticket) {
|
|||
|
||||
if (db >= sensitivity) vadHangover = 75;
|
||||
let isSpeaking = vadHangover > 0 && !isMicMuted;
|
||||
setTileSpeaking(String(window.id), isSpeaking);
|
||||
setTileSpeaking(myIdStr, isSpeaking);
|
||||
|
||||
if (vadHangover > 0 && !isMicMuted) {
|
||||
vadHangover--;
|
||||
|
|
@ -1113,7 +1158,44 @@ async function startVoiceSession(serverUrl, ticket) {
|
|||
if (deadPeers.has(clientId)) deadPeers.delete(clientId);
|
||||
|
||||
const type = view.getUint8(8);
|
||||
if (type === 5) return; // liveness ping - presence already recorded
|
||||
if (type === 5 || type === 6) {
|
||||
//Liveness już zapisany wyżej. Dalej tylko przy zainstalowanym kluczu.
|
||||
if (voiceAesKeyBytes) {
|
||||
const senderKey = getVoicePeerSenderKey(clientId);
|
||||
if (senderKey) {
|
||||
//Offsety ODBIORCZE: serwer dokleja 8B nadawcy, więc iv=[10..22),
|
||||
//ct=[22..). Format wysyłki (iv=2, ct=14) jest INNY - pomylenie
|
||||
//psuje każdą deszyfrację cicho (GCM auth fail -> .catch)
|
||||
const piv = new Uint8Array(arrayBuffer, 10, 12);
|
||||
const pct = new Uint8Array(arrayBuffer, 22);
|
||||
crypto.subtle.decrypt({name: "AES-GCM", iv: piv}, senderKey, pct).then(plain => {
|
||||
if (plain.byteLength !== 8) return; //stary klient: ping bez timestampu
|
||||
const tsData = new DataView(plain).getBigUint64(0);
|
||||
if (type === 5) {
|
||||
//Odbij te same bajty własnym podkluczem i świeżym IV; rate-limit
|
||||
const nowMs = Date.now();
|
||||
if (nowMs - voiceLastEchoAt < 150) return;
|
||||
voiceLastEchoAt = nowMs;
|
||||
if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
|
||||
const riv = nextVoiceIv();
|
||||
crypto.subtle.encrypt({name: "AES-GCM", iv: riv}, voiceMySenderCryptoKey, plain).then(rcipher => {
|
||||
if (!currentVoiceWs || currentVoiceWs.readyState !== WebSocket.OPEN) return;
|
||||
const rpayload = new Uint8Array(14 + rcipher.byteLength);
|
||||
rpayload[0] = 6; rpayload[1] = 0; rpayload.set(riv, 2); rpayload.set(new Uint8Array(rcipher), 14);
|
||||
currentVoiceWs.send(rpayload.buffer);
|
||||
}).catch(() => {});
|
||||
} else {
|
||||
//Typ 6: powrót własnego timestampu - RTT na WŁASNYM zegarze.
|
||||
if (!voiceOutstandingPings.has(tsData)) return;
|
||||
voiceOutstandingPings.delete(tsData);
|
||||
const rtt = Date.now() - Number(tsData);
|
||||
if (rtt >= 1 && rtt < 15000) voiceRecordRttSample(rtt);
|
||||
}
|
||||
}).catch(() => {});
|
||||
}
|
||||
}
|
||||
return;
|
||||
}
|
||||
|
||||
if (!voiceActiveUsers.includes(clientId) && clientId !== String(window.id)) {
|
||||
const baseUrl = currentVoiceWs.url.split('/ws')[0];
|
||||
|
|
@ -1365,8 +1447,6 @@ async function handleCallUpdate(data) {
|
|||
for (let u in remoteAudio) {
|
||||
let pureU = u.replace("_screen", "");
|
||||
if (!voiceActiveUsers.includes(pureU) && pureU !== myIdStr) {
|
||||
if (audioDecoders[u]) try { audioDecoders[u].close(); } catch(e){}
|
||||
delete audioDecoders[u];
|
||||
delete remoteAudio[u];
|
||||
const rn = u.endsWith("_screen") ? screenReceiverNode : receiverNode;
|
||||
if(rn) rn.port.postMessage({ action: 'remove', clientId: u });
|
||||
|
|
@ -2523,7 +2603,6 @@ function toggleVoiceMic() {
|
|||
// keeping the UI, mic stream and AudioContext alive for an automatic rejoin.
|
||||
function teardownVoiceTransport() {
|
||||
if (voicePingTimer) { clearInterval(voicePingTimer); voicePingTimer = null; }
|
||||
if (voiceWsHeartbeat) { clearInterval(voiceWsHeartbeat); voiceWsHeartbeat = null; }
|
||||
if (camControl) { try { camControl.stop(); } catch (e) {} camControl = null; }
|
||||
if (screenControl) { try { screenControl.stop(); } catch (e) {} screenControl = null; }
|
||||
if (currentVoiceWs) {
|
||||
|
|
@ -2550,6 +2629,12 @@ function teardownVoiceTransport() {
|
|||
setVoiceRootKey(null);
|
||||
voiceSendEpoch = null;
|
||||
voiceSendCounter = 0n;
|
||||
voiceRttFastMs = 0;
|
||||
voiceRttSamples = [];
|
||||
voiceRttBaseMs = 0;
|
||||
voiceAudioHoldUntil = 0;
|
||||
voiceOutstandingPings.clear();
|
||||
voiceLastRttSampleAt = 0;
|
||||
lastVoiceServerPacketAt = Date.now();
|
||||
}
|
||||
|
||||
|
|
@ -2566,6 +2651,10 @@ function scheduleVoiceReconnect() {
|
|||
voiceReconnectAttempts++;
|
||||
setVoiceStatus("connecting");
|
||||
teardownVoiceTransport();
|
||||
//Zmiana sieci = stary RTT dotyczy martwej ścieżki; bez resetu wskaźnik
|
||||
//wisiałby zamrożony, a fallback HTTP nie miał prawa go nadpisać
|
||||
voicePingMs = null;
|
||||
updateVoicePing();
|
||||
setTimeout(async () => {
|
||||
if (!currentVoiceCallId || !currentVoiceCallType) return;
|
||||
try {
|
||||
|
|
@ -2617,9 +2706,7 @@ function endVoiceSession() {
|
|||
micStream = null;
|
||||
}
|
||||
if (micSourceNode) { micSourceNode.disconnect(); micSourceNode = null; }
|
||||
if (audioEncoder) { try { audioEncoder.close(); } catch(e){} audioEncoder = null; }
|
||||
|
||||
if (voiceWsHeartbeat) { clearInterval(voiceWsHeartbeat); voiceWsHeartbeat = null; }
|
||||
|
||||
if (voicePingTimer) { clearInterval(voicePingTimer); voicePingTimer = null; }
|
||||
if (voiceRosterRetryTimer) { clearInterval(voiceRosterRetryTimer); voiceRosterRetryTimer = null; }
|
||||
if (voiceRosterPollTimer) { clearInterval(voiceRosterPollTimer); voiceRosterPollTimer = null; }
|
||||
|
|
@ -2629,14 +2716,17 @@ function endVoiceSession() {
|
|||
voiceReconnectAttempts = 0;
|
||||
voiceActiveUsers = [];
|
||||
voiceHadRemotePeers = false;
|
||||
|
||||
for (let id in audioDecoders) {
|
||||
try { audioDecoders[id].close(); } catch(e){}
|
||||
}
|
||||
voiceRttFastMs = 0;
|
||||
voiceRttSamples = [];
|
||||
voiceRttBaseMs = 0;
|
||||
voiceAudioHoldUntil = 0;
|
||||
voiceOutstandingPings.clear();
|
||||
voiceLastEchoAt = 0;
|
||||
voiceLastRttSampleAt = 0;
|
||||
|
||||
for (let id in audioDecryptChains) {
|
||||
if(audioDecryptChains[id]) audioDecryptChains[id].clear();
|
||||
}
|
||||
audioDecoders = {};
|
||||
audioDecryptChains = {};
|
||||
remoteAudio = {};
|
||||
|
||||
|
|
@ -2708,8 +2798,11 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
|
|||
screenSenderNode = new AudioWorkletNode(audioContext, 'audio-sender');
|
||||
screenSenderNode.port.onmessage = async (e) => {
|
||||
if (currentVoiceWs?.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
|
||||
// Same backpressure valve as the mic sender.
|
||||
if (currentVoiceWs.bufferedAmount > 256 * 1024) return;
|
||||
// Ten sam zawór co mikrofon (limit bajtów + korek RTT) - audio
|
||||
// ma pierwszeństwo przed wideo na wspólnym sockecie.
|
||||
if (currentVoiceWs.bufferedAmount > VOICE_WS_AUDIO_QUEUE_LIMIT_BYTES) return;
|
||||
if (voiceCongestionDelay() > 450) voiceAudioHoldUntil = Date.now() + 700;
|
||||
if (Date.now() < voiceAudioHoldUntil) return;
|
||||
const iv = nextVoiceIv();
|
||||
const cipher = await crypto.subtle.encrypt({name: "AES-GCM", iv}, voiceMySenderCryptoKey, e.data.audioData);
|
||||
const payload = new Uint8Array(14 + cipher.byteLength);
|
||||
|
|
@ -2740,6 +2833,23 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
|
|||
};
|
||||
|
||||
let currentConfig = getCurrentConfig(), isRunning = true, lastKeyframeAt = 0;
|
||||
//AIMD + slow-start: start na 40% celu (pełną prędkością waliłby w nieznany
|
||||
//uplink, a bufory TCP w kernelu to MB-y = sekundy kolejki zanim cokolwiek
|
||||
//zmierzymy). Sygnał korka: skipy (zawór 12 KB) lub nadwyżka RTT.
|
||||
let brScale = 0.4, encTries = 0, encSkips = 0, brWindowAt = performance.now(), lastBrReconfAt = 0;
|
||||
//Korek trwa mimo podłogi bitrate'u → FPS spada do 1/2, 1/3
|
||||
let fpsDiv = 1, lowStreak = 0, cleanStreak = 0, tickCount = 0;
|
||||
let activeBitrate = Math.round(currentConfig.bitrate * brScale);
|
||||
const applyEncoderBitrate = (force = false) => {
|
||||
const target = Math.max(120000, Math.round(currentConfig.bitrate * brScale));
|
||||
if (!force && Math.abs(target - activeBitrate) / activeBitrate < 0.08) return;
|
||||
if (!force && performance.now() - lastBrReconfAt < 800) return;
|
||||
activeBitrate = target;
|
||||
lastBrReconfAt = performance.now();
|
||||
try {
|
||||
encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: activeBitrate, latencyMode: 'realtime'});
|
||||
} catch (e) {}
|
||||
};
|
||||
const encoder = new VideoEncoder({
|
||||
output: async (chunk) => {
|
||||
if (currentVoiceWs?.readyState !== WebSocket.OPEN || !voiceMySenderCryptoKey) return;
|
||||
|
|
@ -2751,7 +2861,7 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
|
|||
currentVoiceWs.send(payload.buffer);
|
||||
}, error: (e) => { voiceLogError('videoencoder', e); stopFunction(); }
|
||||
});
|
||||
encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: currentConfig.bitrate, latencyMode: 'realtime'});
|
||||
encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: activeBitrate, latencyMode: 'realtime'});
|
||||
|
||||
// Transient capture-mode wobbles must not thrash the encoder -
|
||||
// commit only after ~300 ms stable. Camera size/fps changes arrive
|
||||
|
|
@ -2769,31 +2879,65 @@ async function startHardwareVideo(btnEl, streamPromise, typeLabel, typeId) {
|
|||
pendingCfg = newConfig;
|
||||
pendingAt = performance.now();
|
||||
} else if (performance.now() - pendingAt > 300) {
|
||||
//Zmiana geometrii źródła - uczony scale łącza zostaje,
|
||||
//resetujemy tylko redukcję FPS
|
||||
currentConfig = newConfig; forceKeyframe = true; lastKeyframeAt = 0; pendingCfg = null;
|
||||
encoder.configure({codec: 'av01.0.04M.08', width: currentConfig.width, height: currentConfig.height, framerate: currentConfig.fps, bitrate: currentConfig.bitrate, latencyMode: 'realtime'});
|
||||
activeBitrate = Math.round(currentConfig.bitrate * brScale);
|
||||
fpsDiv = 1; lowStreak = 0; cleanStreak = 0;
|
||||
applyEncoderBitrate(true);
|
||||
}
|
||||
} else if (pendingCfg) {
|
||||
pendingCfg = null; // wiggle reverted on its own - ignore
|
||||
}
|
||||
|
||||
if (encoder.state === 'configured' && encoder.encodeQueueSize <= 2 && currentVoiceWs.bufferedAmount < 1024 * 256 && videoEl.readyState >= 2) {
|
||||
let frame = null;
|
||||
try {
|
||||
const nowMs = performance.now();
|
||||
frame = new VideoFrame(videoEl, {timestamp: Math.trunc(nowMs * 1000)});
|
||||
//Klatka kluczowa co ~1 s według ZEGARA, nie licznika zakodowanych
|
||||
//klatek: zawór congestion pomija klatki, więc licznik rozjeżdżał
|
||||
//się z realnym tempem i keyframes stawały się rzadkie. Widz
|
||||
//dołączający w trakcie udostępniania musi zaczekać na keyframe -
|
||||
//z licznikiem potrafiło to trwać bardzo długo (efekt "kamery
|
||||
//nie widać"), z zegarem to maksymalnie ~1 s.
|
||||
const wantKeyframe = forceKeyframe || (nowMs - lastKeyframeAt >= 1000);
|
||||
encoder.encode(frame, {keyFrame: wantKeyframe});
|
||||
if (wantKeyframe) lastKeyframeAt = nowMs;
|
||||
} catch (err) { } finally { if (frame) frame.close(); }
|
||||
}
|
||||
setTimeout(encodeLoop, 1000 / currentConfig.fps);
|
||||
};
|
||||
//Zawór kolejki socketu (12 KB) ustępuje DALEJ przed progiem audio
|
||||
//(14 KB) - przy korku najpierw znika obraz, głos płynie dalej.
|
||||
tickCount++;
|
||||
const policySkip = fpsDiv > 1 && (tickCount % fpsDiv !== 0);
|
||||
let skipped = false;
|
||||
if (policySkip) {
|
||||
//celowe tempo (fpsDiv) - nie liczy się do skipRatio
|
||||
} else if (encoder.state === 'configured' && encoder.encodeQueueSize <= 2 &&
|
||||
currentVoiceWs.readyState === WebSocket.OPEN &&
|
||||
currentVoiceWs.bufferedAmount < VOICE_WS_VIDEO_QUEUE_LIMIT_BYTES && videoEl.readyState >= 2) {
|
||||
let frame = null;
|
||||
try {
|
||||
const nowMs = performance.now();
|
||||
frame = new VideoFrame(videoEl, {timestamp: Math.trunc(nowMs * 1000)});
|
||||
//Keyframe wg zegara co ~1 s (nie licznika klatek - zawór je pomija)
|
||||
const wantKeyframe = forceKeyframe || (nowMs - lastKeyframeAt >= 1000);
|
||||
encoder.encode(frame, {keyFrame: wantKeyframe});
|
||||
if (wantKeyframe) lastKeyframeAt = nowMs;
|
||||
} catch (err) { } finally { if (frame) frame.close(); }
|
||||
} else if (encoder.state === 'configured') {
|
||||
skipped = true;
|
||||
}
|
||||
|
||||
encTries++;
|
||||
if (skipped) encSkips++;
|
||||
const winNow = performance.now();
|
||||
if (winNow - brWindowAt >= 800) {
|
||||
const skipRatio = encTries > 0 ? encSkips / encTries : 0;
|
||||
const congestionMs = voiceCongestionDelay();
|
||||
if (congestionMs > 450 || skipRatio > 0.3 || congestionMs > 250) {
|
||||
brScale = Math.max(0.15, brScale * (congestionMs > 450 ? 0.6 : 0.75));
|
||||
applyEncoderBitrate();
|
||||
if (brScale <= 0.17) lowStreak++; else lowStreak = 0;
|
||||
if (lowStreak >= 4 && fpsDiv < 3) { fpsDiv++; lowStreak = 0; }
|
||||
cleanStreak = 0;
|
||||
} else if (skipRatio === 0 && congestionMs < 80) {
|
||||
cleanStreak++;
|
||||
lowStreak = 0;
|
||||
if (cleanStreak >= 5) {
|
||||
if (fpsDiv > 1) { fpsDiv--; }
|
||||
else if (brScale < 1) { brScale = Math.min(1, brScale * 1.2); applyEncoderBitrate(); }
|
||||
cleanStreak = 0;
|
||||
}
|
||||
}
|
||||
encTries = 0; encSkips = 0; brWindowAt = winNow;
|
||||
}
|
||||
setTimeout(encodeLoop, 1000 / currentConfig.fps);
|
||||
};
|
||||
setTimeout(encodeLoop, 500);
|
||||
|
||||
videoTrack.onended = () => { if(isRunning) stopFunction(); };
|
||||
|
|
|
|||
Loading…
Add table
Add a link
Reference in a new issue