@voltkit-rtc/client 0.3.0 → 0.5.0

package/CHANGELOG.md

@@ -1,5 +1,115 @@
 # Changelog
 
+## 0.5.0
+
+- All video tiers included by default: 1080p, 720p and 480p. Per-project restrictions still apply.
+- Authoritative 30 fps ladder: 6/3/1.2 Mbps top layers, 1080p lower layers 0.4/2 Mbps, 720p lower layers 0.2/0.8 Mbps. Phone/tablet defaults remain two layers.
+- `PublishOptions.frameRate` and `setVideoFrameRate(30 | 60)`. At 60 fps only the top layer rises to 9/4.5/1.8 Mbps; smaller layers stay at 30. Low capture fps or sustained device/upload limitation falls back to 30 with `frame-rate-fallback` on `error`.
+- Camera start bitrate is 2000 kbps.
+- The 60 fps monitor allows encoder/bitrate warm-up before counting sustained limitation, so the intentional 30 fps lower layers during startup do not immediately trigger fallback.
+- `publishScreen` / `unpublishScreen`, `ScreenPublishOptions`, `screenShareStarted` / `screenShareEnded`, separate `screen-audio`, native-size detail/motion ladders and reconnect restoration. Screen, camera, mic and music controls remain independent. Browser stop-sharing and interrupted capture clean up both screen producers.
+- `setReceiveQuality(producerId, 'auto' | '1080p' | '720p' | '480p')`, `ReceiveQuality` and producer `videoLayers` metadata. Presets follow actual dimensions; Auto clears both layer ceilings; reconnect restores the choice. Older publishers without metadata report `quality-unavailable` for explicit presets.
+- Same-size `replaceVideoTrack` stays smooth. A change in dimensions republishes camera video so advertised layer dimensions remain accurate.
+- `raiseHand(metadata?)`, `lowerHand(identity?)`, hand-event metadata/reason and roster `handMetadata` support role requests and moderated responses.
+- `participantMuted` normalizes moderation locks to `{ identity, kind, paused }`; `ParticipantInfo.videoBlocked` seeds camera locks on reconnect. Demotion while disconnected releases retained capture, and rejected unmute calls keep the device paused.
+- A screen-share republish that fails halfway cleans up that share while the camera continues.
+- No package publishing is part of this source update.
+
+## 0.4.0
+
+Sharper video, above all from phones. Nothing to change to upgrade: every 0.3.0
+call works as before, with new defaults.
+
+- **Higher bitrates.** Each layer gets about 4 Mbps at 1080p, 2.5 Mbps at
+  720p, 1.5 Mbps at 540p, 1 Mbps at 480p, 700 kbps at 360p and 400 kbps at
+  240p (were 2.5 Mbps, 1.8 Mbps, 1.2 Mbps, 750, 600 and 250 kbps). At 720p
+  that brought VP8's quantiser down from about 64 to 55: finer detail, most
+  of all in motion. A new stream's bandwidth estimate starts at 1.5 Mbps (was
+  1 Mbps). The service's per-publisher caps rose to match (4 Mbps up to 720p,
+  7 Mbps with 1080p), and viewers of 720p now start on the top layer instead
+  of taking about 15 s to climb to it.
+- **Phones keep the full picture.** New `PublishOptions.layers` (1, 2 or 3).
+  Phones and tablets now send two layers (half and full size) by default,
+  computers three as before. A busy phone shrinks its picture and then sends
+  fewer layers, dropping the top one: from three, viewers of a 720p phone got
+  180×320; from two, the full picture survives.
+- **H.264 on phones.** New `PublishOptions.codec`: `'auto'` (the default)
+  sends H.264 on phones and tablets that encode it in hardware and VP8
+  everywhere else; `'vp8'` and `'h264'` force one. If H.264 cannot be used,
+  VP8 is sent and the publish goes on; a forced `'h264'` that falls back also
+  fires `error`. A viewer whose browser cannot decode H.264 (rare: a few
+  browser builds leave it out) cannot watch a phone sending it: force `'vp8'`
+  if your audience uses such browsers.
+- New `PublishOptions.degradationPreference`: what the encoder gives up first
+  on a busy device. Unset leaves the browser's default.
+- **`getVideoStats()`**: what the camera is sending, layer by layer, and why
+  it is less than asked (`qualityLimitation`: `cpu` is the device,
+  `bandwidth` the upload). **`getReceiveStats(producerId)`**: what this client
+  receives of someone's video.
+- **`videoLimited` event**: the camera video is being held back by the device
+  (`cpu`), the upload (`bandwidth`) or something else (`other`), or that has
+  cleared (`none`). It fires once a reason has held for about 4 s, and is
+  checked only while you listen. Tell the publisher, so they know it is their
+  device or connection.
+- **`replaceVideoTrack(track)`**: send another video track, such as a canvas
+  with effects or the raw camera again. Viewers see no pause when the new
+  track takes the same layers. Your own stream is never modified, and a
+  track replaced while the camera was paused comes back to you switched on.
+- **`setPreferredLayers(producerId, spatialLayer, temporalLayer?)`**: receive
+  a smaller copy of someone's video for a small tile, as the Flutter and
+  native SDKs already could. Layer 0 is the smallest copy sent; at 720p the
+  640×360 copy needs about 0.76 Mbps instead of 2.6 for the full picture. The
+  choice is kept across reconnects and forgotten when the track ends.
+- `setVideoQuality`, `replaceVideoTrack` and publishing now change the camera
+  producer one at a time, so quick changes cannot leave two of them, and an
+  `unpublish()` in the middle of one leaves none.
+- Upload needed rises with the bitrates: about 3.4 Mbps for 720p from a
+  computer (3.2 from a phone), 6 Mbps for 1080p (5.5 from a phone) and
+  1.4 Mbps for 480p.
+- New exports: the `LayerCount`, `CodecPreference`, `VideoSendStats`,
+  `VideoLayerStats`, `VideoReceiveStats`, `VideoLimitedEvent` and
+  `VideoLimitReason` types. `videoEncodings` takes the most layers to send as
+  an optional fourth argument.
+- **Fix: reconnecting gives up when it should.** The attempt count restarted
+  whenever the socket opened, so a join refused with a code worth retrying
+  (`no-capacity`, `no-node`, `internal`) was retried about twice a second,
+  forever, and `closed` with `gave-up` never came: in a full cluster every
+  client kept at it. The count now restarts only once a join succeeds, the
+  gaps keep growing (half a second, doubling to 15 s), and after eight failed
+  attempts in a row the client stops with `closed` (`gave-up`). A connection
+  that drops again while a reconnect is restoring the session is retried
+  too; it was left disconnected, with no `closed`. `ChatClient` counted the
+  same way and is fixed the same way.
+- **Fix: a camera that is off stays off, on both ends.** A track handed to
+  `publish()` switched off, a camera switched off before a reconnect, or a
+  quality change or track swap to other layers could leave the camera paused
+  in the client but live on the server, so viewers got black video under a
+  camera shown as on (and silence, for a microphone or music switched off
+  before a reconnect). The server is now paused too; a camera producer that
+  is replaced starts in the camera's state, and while the camera is off no
+  frame of the new track is sent; a track swapped in during a reconnect, and
+  devices a reconnect has to ask for again, keep a camera that was off, off.
+  `camEnabled`, `micEnabled` and `localMediaChanged` report as before.
+- **Fix:** `unpublish()` while `publish()` or `publishMusic()` is still
+  making the microphone or music producer no longer leaves it published; the
+  camera was already covered. What was being made is closed and that call
+  rejects. A reconnect's republish under way when `unpublish()` runs
+  publishes nothing either.
+- **Fix: `unpublish()` forgets the devices.** It kept the capture constraints
+  given to `publish()`, so after `unpublish()` and then `publishMusic()`, a
+  reconnect asked for the camera and microphone again and published them.
+  Music published after `unpublish()` now reconnects alone.
+- **Fix:** `publish()` and `unpublish()` act in the order they are called. A
+  `publish()` or `publishMusic()` called while an `unpublish()` was still
+  waiting on the server had its new stream stopped and its state cleared when
+  that `unpublish()` finished. `unpublish()` now makes all its changes at once
+  and only then waits for the server.
+- **Fix:** a first `connect()` that fails at the join closes its socket and
+  starts no reconnect loop; only a session that has joined is reconnected.
+  Before, the server's join deadline closed that socket about 15 s later, and
+  the client began reconnecting a session the app had already been told had
+  failed. `ChatClient` too.
+
 ## 0.3.0
 
 - **Video quality.** `publish(stream, constraints, { quality })` takes

package/README.md

@@ -48,7 +48,7 @@ const joined = await client.connect();
 
 if (joined.role === 'host' || joined.role === 'cohost') {
   const media = await navigator.mediaDevices.getUserMedia({ audio: true, video: { width: 1280, height: 720 } });
-  await client.publish(media);                            // 720p, sent as 3 simulcast layers
+  await client.publish(media);                            // 720p: 3 simulcast layers from a computer, 2 from a phone
 }
 ```
 
@@ -69,6 +69,40 @@ client.publishing; client.micEnabled; client.camEnabled; client.local;   // curr
 A `viewer` cannot publish. That is enforced at the media server, not in this
 package — a tampered client gets an error, not a stream.
 
+`publish(media, constraints, options)` takes these options. The defaults suit
+phones and computers alike; they are kept for a reconnect's republish.
+
+| Option | Default | What it does |
+| --- | --- | --- |
+| `quality` | the highest the project allows | `'1080p'`, `'720p'` or `'480p'`; see [Video quality](#video-quality) |
+| `frameRate` | `30` | `30` or `60`; only the top layer runs at 60 fps; capture at 60 too |
+| `layers` | `2` on phones and tablets, `3` elsewhere | the most simulcast layers to send. When a phone is too busy, the browser shrinks the picture and may then send fewer layers, dropping the top one: asked for two, it keeps the full picture; asked for three, viewers were left with a quarter of it |
+| `codec` | `'auto'` | `'auto'` sends H.264 on phones and tablets that encode it in hardware, VP8 elsewhere. `'vp8'` or `'h264'` forces one; a forced H.264 the browser cannot send is VP8, and fires `error` |
+| `degradationPreference` | the browser's | what the encoder gives up first on a busy device: `'maintain-resolution'` (frame rate goes first), `'maintain-framerate'` or `'balanced'` |
+
+```ts
+await client.publish(media, undefined, { quality: '720p', layers: 2, codec: 'auto', degradationPreference: 'maintain-resolution' });
+```
+
+Chrome also reads the trade-off from the track itself:
+`track.contentHint = 'detail'` keeps the resolution, `'motion'` the frame rate.
+A viewer whose browser cannot decode the codec cannot watch, so force `'vp8'`
+if your audience uses browsers without H.264 (rare).
+
+To send another video track while live, such as a canvas with effects drawn on
+it or the raw camera again:
+
+```ts
+await client.replaceVideoTrack(canvas.captureStream(30).getVideoTracks()[0]);
+```
+
+When the new track has the same dimensions and layers as the old one, viewers see no pause; otherwise the camera's producer is replaced
+and they see a brief one. A paused camera stays paused. Your own stream is not
+modified: the client keeps one of its own (`client.local`, announced by
+`localStreamChanged`) with the new track, and a reconnect republishes that. The
+track you replaced is yours again: it is not stopped, and if pausing the camera
+had switched it off, it is switched back on.
+
 ### Video quality
 
 Choose 1080p, 720p or 480p when publishing, or change it while live:
@@ -78,30 +112,163 @@ const media = await navigator.mediaDevices.getUserMedia({ audio: true, video: {
 await client.publish(media, undefined, { quality: '1080p' });
 await client.setVideoQuality('480p');      // resolves to the quality actually used
 
-client.videoPolicy;   // { maxQuality: '720p' | '1080p', maxBitrate } for this project
+client.videoPolicy;   // { maxQuality: '480p' | '720p' | '1080p', maxFrameRate, maxBitrate } for this project
 client.videoQuality;  // what is being sent now
 ```
 
-| Quality | Layers sent (simulcast) | Top layer |
-|---|---|---|
-| 1080p | 480×270 · 960×540 · 1920×1080 | 2.5 Mbps |
-| 720p | 320×180 · 640×360 · 1280×720 | 1.8 Mbps |
-| 480p | 427×240 · 854×480 | 750 kbps |
-
-- 720p and 480p are available to every project. 1080p is available during the
-  free trial, and after that to projects that have it enabled (request it from
-  your project's Settings in the VoltKit dashboard). Asking for more than the project
-  allows sends the highest it does allow.
-- The default is the highest the project allows. The picture is scaled down to
-  the quality, never up: capture at least that size for the full picture.
-- The layers follow the picture's real size (a portrait phone, a 4:3 webcam),
-  so the browser sends every layer requested.
-- The server caps each publisher's incoming bitrate for the project's tier
-  (3.2 Mbps up to 720p, 5 Mbps with 1080p), whatever client is used.
-
-The full guide, with data use and troubleshooting:
+| Quality | Layers from a computer at 30 fps | From a phone or tablet | Video upload |
+|---|---|---|---|
+| 1080p | 480×270 @ 0.4 Mbps · 960×540 @ 2 Mbps · 1920×1080 @ **6 Mbps** | the top two | about 8.4 Mbps (phone 8) |
+| 720p | 320×180 @ 0.2 Mbps · 640×360 @ 0.8 Mbps · 1280×720 @ **3 Mbps** | the top two | about 4 Mbps (phone 3.8) |
+| 480p | 427×240 @ 0.4 Mbps · 854×480 @ **1.2 Mbps** | the same two | about 1.6 Mbps |
+
+All qualities are included on every plan. The default policy allows 1080p and
+60 fps, with a 25 Mbps incoming safety cap per publisher. A project can have
+an explicit administrative restriction; `videoPolicy` describes it.
+The picture is scaled down to the chosen quality, never up. Portrait phones
+keep their shape, for example 360×640 and 720×1280. Adaptive reception is the
+default: viewers receive the best layer their connection can carry. Viewers
+start with a 4 Mbps estimate, enough for 720p; 1080p ramps up as bandwidth allows.
+
+30 fps is the default. For 60 fps, capture at 60 and opt in:
+
+```ts
+await client.publish(media, undefined, { quality: '1080p', frameRate: 60 });
+const effectiveFps = await client.setVideoFrameRate(30); // camera only, other tracks continue
+```
+
+Only the top layer uses 60 fps: 1080p is **9 Mbps**, 720p **4.5 Mbps**, and
+480p **1.8 Mbps**; smaller layers stay at 30. This means roughly 1.5× the top
+layer's data and twice the encoding work. Phones can heat up and use more
+battery, and many front cameras capture only 30. If capture is below 50 fps,
+a project restricts it, or sustained device/upload limitation prevents 60,
+the SDK uses 30 and emits `error` with `code: 'frame-rate-fallback'` as a warning.
+`setVideoFrameRate(60)` retries the preference while preserving a paused camera.
+
+The full guide, with data use:
 [voltkit.in/docs/video-quality](https://voltkit.in/docs/video-quality).
 
+### When the picture is soft
+
+A soft or stuttering picture usually comes from the publisher's device (too
+busy to encode) or their upload (too slow for the top layer), not from the
+service. The client can tell you which.
+
+```ts
+const stats = await client.getVideoStats();   // null when no camera is published
+// { codec: 'H264', quality: '720p', width: 720, height: 1280, fps: 30, bitrate: 2400000,
+//   qualityLimitation: 'none', availableOutgoingBitrate: 3900000, layers: [ ... ] }
+```
+
+The top-level fields describe the biggest layer going out; `layers` lists each
+one (`active`, size, frame rate, bitrate). `qualityLimitation` is the browser's
+reason for sending less than asked: `cpu` is the device, `bandwidth` the
+upload, `none` neither (then look at the camera and the light). Bitrates are
+averaged since your previous call, so call it every second or two; the first
+call reports 0.
+
+`videoLimited` tells you when that changes, so you can tell the publisher:
+
+```ts
+client.on('videoLimited', ({ reason }) => {
+  if (reason === 'cpu') showNotice('Your device is busy, so the stream is softer. Turn off effects or switch to 480p.');
+  else if (reason === 'bandwidth') showNotice('Your upload is slow, so viewers get a lighter picture.');
+  else hideNotice();                                   // 'none': cleared; 'other': rare
+});
+```
+
+It fires once a new reason has held for about 4 seconds, so brief spikes stay
+quiet, and never fires `none` first. The client checks every 2 seconds, and
+while a camera is published and something listens, or while 60 fps needs fallback monitoring. `unpublish()` fires
+`none` if a limitation was showing.
+
+On a viewer, `getReceiveStats(producerId)` (the `producerId` from the `track`
+event) says what arrives: size, frame rate, bitrate, `packetsLost`, `jitter`
+and, where the browser counts them, `freezeCount`. A viewer receiving a
+smaller picture than the publisher's biggest layer is on a lower one, picked
+for their own connection; if every viewer is, look at the publisher's
+`getVideoStats()`.
+
+The [troubleshooting guide](https://voltkit.in/docs/troubleshooting) lists the
+usual causes and what to do about each.
+
+### Small tiles
+
+Each viewer gets the biggest layer their connection carries, which a thumbnail
+or a row of co-host tiles does not need. Ask for a smaller one with the
+`producerId` from the `track` event:
+
+```ts
+client.on('track', ({ kind, producerId, track }) => {
+  if (kind !== 'video') return;
+  showVideo(producerId, track);
+  if (isSmallTile(producerId)) client.setPreferredLayers(producerId, 0).catch(console.warn);
+});
+
+await client.setPreferredLayers(producerId, 2);   // the tile grew: the biggest layer again
+```
+
+Layer 0 is the smallest copy the publisher sends. At 720p a computer sends
+three (320×180, 640×360 and 1280×720) and a phone two (640×360 and 1280×720);
+a layer above the biggest means the biggest. The 640×360 copy needs about
+0.9 Mbps including audio, the full picture about 3.1. An optional third argument, the
+temporal layer, lowers the frame rate as well; without it the full frame rate
+is kept. It is the most this viewer is sent: a connection that cannot carry it
+still gets less. The choice is kept across reconnects and forgotten when the
+track ends, and `setPreferredLayers` rejects for a video this client is not
+receiving.
+
+### Receive quality
+
+```ts
+await client.setReceiveQuality(producerId, '480p');
+await client.setReceiveQuality(producerId, 'auto'); // recommended; clears spatial AND temporal ceilings
+```
+
+Offer **Auto (recommended) / 1080p / 720p / 480p**. The `ReceiveQuality` helper
+maps the request to the producer's actual `videoLayers` dimensions, including
+portrait phones, two-layer cameras and screen tracks. It chooses the largest
+available copy within the ceiling, or the smallest available if none fits.
+It never creates a missing resolution: a 1080p screen with 540p and 1080p copies
+uses 540p for the 480p choice. Connection adaptation still applies below the
+ceiling. The preference is remembered across reconnects and ends with that
+producer. Without layer metadata, explicit choices reject `quality-unavailable`;
+Auto and the numeric `setPreferredLayers` API still work.
+
+### Screen sharing
+
+Hosts and cohosts can share a screen beside their camera in SDK 0.5.0:
+
+```ts
+client.on('screenShareStarted', ({ stream, mode }) => showScreenPreview(stream, mode));
+client.on('screenShareEnded', ({ reason }) => hideScreenPreview(reason));
+await client.publishScreen({ mode: 'detail', audio: true }); // from a button click
+await client.unpublishScreen(); // camera, mic and music continue
+```
+
+`ScreenPublishOptions` accepts `mode: 'detail' | 'motion'` (default `detail`),
+`audio?: boolean` and `stream?: MediaStream`. Without `stream`, the SDK opens
+`getDisplayMedia` and returns its `MediaStream`; capture audio defaults to
+false. An existing stream's audio is used unless `audio: false` is supplied.
+The SDK stops the supplied screen tracks when sharing ends.
+
+Detail sends up to native 1920×1080 at 15 fps and 2.5 Mbps plus a half-size
+0.6 Mbps copy. Motion uses 30 fps, 6 Mbps plus 0.6 Mbps. Capture is never
+upscaled. Optional tab/system audio is separate, `source: 'screen-audio'`,
+128 kbps stereo. Viewers play it like `music`; screen video arrives with
+`source: 'screen'`. A browser may grant video without granting audio.
+
+The browser's **Stop sharing** automatically unpublishes and emits
+`screenShareEnded({ reason: 'track-ended' })`. Calling `unpublishScreen` emits
+`reason: 'stopped'`. A reconnect republishes a still-live screen without another
+picker. `unpublish()` and `close()` stop all local media. Unsupported capture
+(most mobile browsers) rejects `screen-share-unsupported`; overlapping shares
+reject `screen-share-active`; missing live video rejects `screen-share-no-video`.
+Browser cancellation retains its `NotAllowedError`. Capture requires HTTPS,
+a user gesture and permission; in iframes, also allow `display-capture`.
+
+[Setup, modes, native platforms and troubleshooting](https://voltkit.in/docs/screen-sharing).
+
 ### Music
 
 A host can play music to the room from inside the page. It does not go through
@@ -146,6 +313,8 @@ musicAudio.muted = true;      // "mute the music" for this viewer only
 ```ts
 await client.raiseHand();
 await client.lowerHand();
+await client.raiseHand({ requestRole: 'cohost' }); // metadata also supports 'moderator'
+await client.lowerHand(identity); // host/moderator: decline that raised hand
 await client.muteParticipant(identity);        // moderating roles: asks them to mute
 await client.forceMute(identity, 'video');     // moderating roles: camera off, locked until released
 await client.forceMute(identity, 'video', false); // release
@@ -159,19 +328,22 @@ await client.getRoomState();
 
 | Event | Fires when |
 | --- | --- |
-| `track` / `trackEnded` | someone's audio or video is ready to render, or went away; `source` is `mic`, `music`, `camera` or `screen` |
+| `track` / `trackEnded` | someone's audio or video is ready to render, or went away; `source` is `mic`, `music`, `camera`, `screen` or `screen-audio` |
 | `trackMuted` | someone paused or resumed a track — muted their mic, switched their music off |
 | `participantJoined` / `participantLeft` | roster changes |
+| `participantMuted` | moderation lock changed: `{ identity, kind, paused }`; reconnect roster `muted` / `videoBlocked` seed the locks |
 | `roleChanged` | someone was promoted or demoted |
-| `handRaised` | a hand went up or down |
+| `handRaised` | a hand went up or down, with optional `metadata` and `reason`; reconnect rosters include `handMetadata` |
+| `screenShareStarted` / `screenShareEnded` | local screen capture began (`stream`, `mode`) or stopped (`reason`) |
 | `forceMuted` | a moderator switched **your** mic or camera off |
 | `streamInterrupted` | media stopped but the connection is fine — the publisher is reconnecting |
 | `localMediaChanged` / `localStreamChanged` | your own mic, camera or stream changed |
 | `connected` / `disconnected` | the socket came up or dropped; reconnection is automatic |
 | `kicked` / `roomClosed` | you were removed, or the host ended the room |
-| `error` | something failed that the SDK could not recover from by itself |
+| `error` | a failure or recoverable warning such as `frame-rate-fallback`; inspect its code |
 | `usageLimit` | your project ran out of minutes or hit a usage limit; the session ends at `endsAt` unless it is topped up |
-| `closed` | terminal — `replaced`, `kicked`, `room-closed` or `gave-up`; your own `close()` does not fire it |
+| `videoLimited` | your camera video is held back by this device (`cpu`) or the upload (`bandwidth`), or that cleared (`none`) |
+| `closed` | terminal — `replaced`, `kicked`, `room-closed` or `gave-up`, or `left` when your own `close()` ends the session |
 
 `on()` returns a function that removes the listener.
 

package/dist/cjs/chat.d.ts

@@ -104,6 +104,8 @@ export declare class ChatClient {
     private closedByUs;
     private reconnecting;
     private reconnectAttempt;
+    /** Set by the first successful join. Only a session that has joined is reconnected. */
+    private joinedOnce;
     constructor(wsUrl: string, token: string);
     on<K extends keyof ChatClientEvents>(event: K, fn: ChatClientEvents[K]): () => void;
     private emit;

package/dist/cjs/chat.d.ts.map

@@ -1 +1 @@
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+{"version":3,"file":"chat.d.ts","sourceRoot":"","sources":["../../src/chat.ts"],"names":[],"mappings":"AAAA;;;;;;;;GAQG;AACH,OAAO,KAAK,EAAE,IAAI,EAAE,MAAM,YAAY,CAAC;AAEvC,MAAM,WAAW,WAAW;IAC1B,4EAA4E;IAC5E,EAAE,EAAE,MAAM,CAAC;IACX,OAAO,EAAE,MAAM,CAAC;IAChB,IAAI,EAAE,MAAM,CAAC;IACb,oEAAoE;IACpE,EAAE,CAAC,EAAE,MAAM,CAAC;IACZ,+EAA+E;IAC/E,IAAI,EAAE,MAAM,CAAC;IACb;;;OAGG;IACH,IAAI,CAAC,EAAE,MAAM,CAAC;IACd,QAAQ,CAAC,EAAE,OAAO,CAAC;IACnB,8EAA8E;IAC9E,WAAW,CAAC,EAAE,MAAM,CAAC;IACrB,EAAE,EAAE,MAAM,CAAC;CACZ;AAED,MAAM,WAAW,kBAAkB;IACjC,IAAI,CAAC,EAAE,MAAM,CAAC;IACd,gDAAgD;IAChD,EAAE,CAAC,EAAE,MAAM,CAAC;IACZ,QAAQ,CAAC,EAAE,OAAO,CAAC;IACnB,WAAW,CAAC,EAAE,MAAM,CAAC;IACrB,iFAAiF;IACjF,SAAS,CAAC,EAAE,OAAO,CAAC;CACrB;AAED,MAAM,WAAW,SAAS;IACxB,OAAO,EAAE,MAAM,CAAC;IAChB,sEAAsE;IACtE,OAAO,EAAE,MAAM,CAAC;IAChB,mDAAmD;IACnD,KAAK,EAAE,OAAO,CAAC;IACf,mCAAmC;IACnC,eAAe,CAAC,EAAE,MAAM,EAAE,CAAC;CAC5B;AAED,MAAM,WAAW,cAAc;IAC7B,OAAO,EAAE,MAAM,CAAC;IAChB,QAAQ,EAAE,MAAM,CAAC;IACjB,IAAI,EAAE,IAAI,CAAC;IACX,OAAO,EAAE,MAAM,CAAC;IAChB,KAAK,EAAE,OAAO,CAAC;IACf,8CAA8C;IAC9C,QAAQ,EAAE,WAAW,EAAE,CAAC;CACzB;AAED,MAAM,WAAW,UAAU;IACzB,OAAO,EAAE,CAAC,CAAC,EAAE,WAAW,KAAK,IAAI,CAAC;IAClC,cAAc,EAAE,CAAC,CAAC,EAAE;QAAE,EAAE,EAAE,MAAM,CAAC;QAAC,EAAE,EAAE,MAAM,CAAA;KAAE,KAAK,IAAI,CAAC;IACxD,SAAS,EAAE,CAAC,CAAC,EAAE;QAAE,QAAQ,EAAE,MAAM,CAAC;QAAC,KAAK,EAAE,OAAO,CAAC;QAAC,EAAE,EAAE,MAAM,CAAA;KAAE,KAAK,IAAI,CAAC;IACzE,MAAM,EAAE,CAAC,CAAC,EAAE;QAAE,QAAQ,EAAE,MAAM,CAAC;QAAC,MAAM,EAAE,OAAO,CAAA;KAAE,KAAK,IAAI,CAAC;CAC5D;AAED,MAAM,WAAW,gBAAiB,SAAQ,UAAU;IAClD,SAAS,EAAE,MAAM,IAAI,CAAC;IACtB,YAAY,EAAE,CAAC,MAAM,EAAE,MAAM,KAAK,IAAI,CAAC;IACvC,MAAM,EAAE,CAAC,MAAM,EAAE,cAAc,KAAK,IAAI,CAAC;IACzC,KAAK,EAAE,CAAC,GAAG,EAAE,KAAK,KAAK,IAAI,CAAC;IAC5B,+EAA+E;IAC/E,MAAM,EAAE,CAAC,MAAM,EAAE,SAAS,GAAG,MAAM,GAAG,OAAO,KAAK,IAAI,CAAC;IACvD,4EAA4E;IAC5E,UAAU,EAAE,CAAC,CAAC,EAAE;QAAE,MAAM,EAAE,MAAM,CAAC;QAAC,YAAY,EAAE,MAAM,CAAC;QAAC,MAAM,EAAE,MAAM,CAAA;KAAE,KAAK,IAAI,CAAC;CACnF;AAED,2EAA2E;AAC3E,qBAAa,YAAa,SAAQ,KAAK;IACzB,QAAQ,CAAC,IAAI,EAAE,MAAM;gBAAZ,IAAI,EAAE,MAAM,EAAE,OAAO,EAAE,MAAM;CAInD;AAyBD;;;GAGG;AACH,qBAAa,UAAU;IAaT,OAAO,CAAC,QAAQ,CAAC,KAAK;IAZlC,OAAO,CAAC,EAAE,CAA0B;IACpC,OAAO,CAAC,QAAQ,CAAC,OAAO,CAA8B;IACtD,OAAO,CAAC,QAAQ,CAAC,SAAS,CAAmC;IAC7D,OAAO,CAAC,MAAM,CAAK;IACnB,OAAO,CAAC,KAAK,CAAS;IACtB,OAAO,CAAC,UAAU,CAA+B;IACjD,OAAO,CAAC,UAAU,CAAS;IAC3B,OAAO,CAAC,YAAY,CAAS;IAC7B,OAAO,CAAC,gBAAgB,CAAK;IAC7B,uFAAuF;IACvF,OAAO,CAAC,UAAU,CAAS;gBAEE,KAAK,EAAE,MAAM,EAAE,KAAK,EAAE,MAAM;IAIzD,EAAE,CAAC,CAAC,SAAS,MAAM,gBAAgB,EAAE,KAAK,EAAE,CAAC,EAAE,EAAE,EAAE,gBAAgB,CAAC,CAAC,CAAC,GAAG,MAAM,IAAI;IAUnF,OAAO,CAAC,IAAI;IAUZ,gFAAgF;IAChF,IAAI,IAAI,IAAI,cAAc,GAAG,IAAI,CAEhC;IAEK,OAAO,IAAI,OAAO,CAAC,cAAc,CAAC;IAsBxC,OAAO,CAAC,UAAU;IAqClB,OAAO,CAAC,YAAY;IAYpB,wEAAwE;YAC1D,iBAAiB;IAgC/B,KAAK,IAAI,IAAI;IASb,OAAO,CAAC,GAAG;IAgBX,OAAO,CAAC,cAAc;IAQtB,OAAO,CAAC,SAAS;IA+CjB,WAAW,CAAC,IAAI,EAAE,MAAM,EAAE,IAAI,GAAE,kBAAuB,GAAG,OAAO,CAAC;QAAE,EAAE,EAAE,MAAM,CAAC;QAAC,EAAE,EAAE,MAAM,CAAA;KAAE,CAAC;IAI7F,6EAA6E;IACvE,WAAW,CAAC,IAAI,GAAE;QAAE,KAAK,CAAC,EAAE,MAAM,CAAC;QAAC,MAAM,CAAC,EAAE,MAAM,CAAA;KAAO,GAAG,OAAO,CAAC,WAAW,EAAE,CAAC;IAKzF,4DAA4D;IAC5D,aAAa,CAAC,SAAS,EAAE,MAAM,GAAG,OAAO,CAAC,OAAO,CAAC;IAIlD,sEAAsE;IACtE,QAAQ,CAAC,QAAQ,EAAE,MAAM,EAAE,KAAK,UAAO,GAAG,OAAO,CAAC,OAAO,CAAC;IAI1D,8EAA8E;IAC9E,SAAS,CAAC,MAAM,EAAE,OAAO,GAAG,OAAO,CAAC,OAAO,CAAC;IAI5C,YAAY,IAAI,OAAO,CAAC,SAAS,CAAC;CAGnC"}

package/dist/cjs/chat.js

@@ -39,6 +39,8 @@ class ChatClient {
     closedByUs = false;
     reconnecting = false;
     reconnectAttempt = 0;
+    /** Set by the first successful join. Only a session that has joined is reconnected. */
+    joinedOnce = false;
     constructor(wsUrl, token) {
         this.wsUrl = wsUrl;
         this.token = token;
@@ -67,8 +69,24 @@ class ChatClient {
         return this.joinResult;
     }
     async connect() {
-        await this.openSocket();
-        const result = (await this.rpc('join', { token: this.token }));
+        let result;
+        try {
+            await this.openSocket();
+            result = (await this.rpc('join', { token: this.token }));
+        }
+        catch (err) {
+            // Never joined: the app has the error. Left open, the socket would be
+            // closed by the server's join deadline, and that would start
+            // reconnecting a session that never began.
+            if (!this.joinedOnce)
+                this.detachSocket();
+            throw err;
+        }
+        // Joined: the next drop gets every reconnect attempt again. Not when the
+        // socket opens, or a join refused with a code worth retrying never uses
+        // them up and is retried about twice a second, forever.
+        this.reconnectAttempt = 0;
+        this.joinedOnce = true;
         this.joinResult = result;
         this.emit('joined', result);
         return result;
@@ -82,7 +100,6 @@ class ChatClient {
             this.ws = ws;
             const onError = () => reject(new Error('websocket failed to open'));
             ws.addEventListener('open', () => {
-                this.reconnectAttempt = 0;
                 this.emit('connected');
                 resolve();
             }, { once: true });
@@ -105,7 +122,9 @@ class ChatClient {
                     this.emit('closed', 'ended');
                     return;
                 }
-                if (!this.closedByUs)
+                // Before a first join the app is still waiting on connect(), which
+                // reports the failure; there is no session to restore.
+                if (!this.closedByUs && this.joinedOnce)
                     void this.scheduleReconnect();
             });
         });

package/dist/cjs/chat.js.map

@@ -1 +1 @@
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package/dist/cjs/device.d.ts

@@ -0,0 +1,85 @@
+/**
+ * What kind of device is publishing, and which video codec it should send.
+ *
+ * Phones and tablets get different defaults from computers: each simulcast
+ * layer is a separate encode, and a phone encoding VP8 does it in software.
+ * Under that load Chrome shrinks the picture, and with three layers asked for
+ * it then sends only two and drops the TOP one: viewers got 180×320 from a
+ * 720p phone. Two layers in hardware H.264 keep the full picture.
+ *
+ * Everything here reads the browser through small injectable parameters, so
+ * tests can stand in for any device.
+ */
+import type { types as ms } from 'mediasoup-client';
+type RtpCodecCapability = ms.RtpCodecCapability;
+/** What the SDK reads about the browser to choose its defaults. */
+export interface DeviceEnv {
+    userAgent: string;
+    /** `navigator.userAgentData.mobile`, where the browser has it (Chromium). */
+    uaMobile?: boolean;
+    maxTouchPoints: number;
+}
+/** The browser this runs in; an empty desktop-like one outside a browser. */
+export declare function currentEnv(): DeviceEnv;
+/**
+ * A phone or a tablet. `userAgentData.mobile` is false on Android tablets, so
+ * the user agent is checked as well.
+ */
+export declare function isPhoneOrTablet(env?: DeviceEnv): boolean;
+/** An iPhone, iPad or iPod: every one of them encodes H.264 in hardware. */
+export declare function isAppleMobile(env?: DeviceEnv): boolean;
+/** Two layers on phones and tablets, three elsewhere. */
+export declare function defaultLayerCount(env?: DeviceEnv): 2 | 3;
+/** `navigator.mediaCapabilities.encodingInfo`, or a stand-in for tests. */
+export type EncodingInfoProbe = (configuration: MediaEncodingConfiguration) => Promise<MediaCapabilitiesEncodingInfo>;
+/** The browser's own probe, when it has one. */
+export declare function currentProbe(): EncodingInfoProbe | undefined;
+/** The H.264 the media server takes: constrained baseline, packetization mode 1. */
+export declare const H264_CONTENT_TYPE = "video/H264;profile-level-id=42e01f;packetization-mode=1";
+/**
+ * Whether this browser encodes H.264 at this size and bitrate in hardware:
+ * true or false when it says so, null when it cannot say (no
+ * `mediaCapabilities`, no `webrtc` support in it, an error, or no answer in
+ * time). Publishing never waits on it for more than a moment.
+ */
+export declare function h264HardwareEncode(video: {
+    width: number;
+    height: number;
+    bitrate: number;
+}, probe?: EncodingInfoProbe | undefined, timeoutMs?: number): Promise<boolean | null>;
+/** What `PublishOptions.codec` may say. */
+export type CodecPreference = 'auto' | 'vp8' | 'h264';
+/** 'vp8' or 'h264' in any case; anything else, including nothing, is 'auto'. */
+export declare function normalizeCodec(codec: unknown): CodecPreference;
+export interface CodecChoiceInput {
+    requested: CodecPreference;
+    phoneOrTablet: boolean;
+    appleMobile: boolean;
+    /** This browser can send H.264 to this server (it is in the device's send capabilities). */
+    h264Available: boolean;
+    /** What the hardware probe said; null when it could not say. Only asked for 'auto' on phones and tablets. */
+    h264Hardware: boolean | null;
+}
+export interface CodecChoice {
+    codec: 'vp8' | 'h264';
+    /** Set when the codec asked for cannot be used: report it, and publish anyway. */
+    warning?: string;
+}
+/**
+ * The codec to publish with.
+ *
+ * 'auto' is H.264 on a phone or tablet that encodes it in hardware, and VP8
+ * everywhere else: computers encode VP8 well, and a few desktop browsers
+ * cannot decode H.264. When the probe cannot say, an Apple device is taken to
+ * have hardware H.264 (they all do) and anything else is not.
+ */
+export declare function chooseCodec(input: CodecChoiceInput): CodecChoice;
+/**
+ * The device's own capability entry for a codec, to hand to `produce()`.
+ * Only ever pass `produce()` an entry from the device's send capabilities:
+ * mediasoup-client adds the transceiver before it checks the codec, so one it
+ * cannot match leaves a transceiver behind that breaks every later produce.
+ */
+export declare function findCodec(codecs: readonly RtpCodecCapability[] | undefined, codec: 'vp8' | 'h264'): RtpCodecCapability | undefined;
+export {};
+//# sourceMappingURL=device.d.ts.map

package/dist/cjs/device.d.ts.map

@@ -0,0 +1 @@
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