14 Commits

Author SHA1 Message Date
Miroslav Šedivý 3191dbb058 Remove legacy mode.
- Deleted the legacy WebSocket client implementation (`wstoclient.go`) and its test file (`wstoclient_test.go`).
- Removed legacy handling from the HTTP manager, including legacy proxy setup and routing.
- Cleaned up session management by removing legacy host checks and related functions.
- Eliminated the legacy WebRTC handler and its associated logic.
- Updated session interface to remove legacy methods.
- Removed legacy API documentation and configuration options from migration guide and help files.
2026-10-16 20:23:56 +02:00
a f4d632df2c feat: create capture subscription api to support more transports (#699)
* feat: introduce encoded media subscriptions

* chore: preserve sample field comments

* perf: reduce encoded sample dispatch copies

* docs: define encoded sample buffer ownership

* fix: avoid pipeline recreation errors

Reuse pipelines started by a new subscription during resize.

* fix: synchronize stream bitrate state

Protect bitrate updates, reads, and resets with one mutex.

* simplify

* group stream sink fields

* fix: remove unnecessary blank line in removeSubscription method

* remove defensive check

* remove noop

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Co-authored-by: Miroslav Šedivý <sedivy.miro@gmail.com>
2026-09-24 10:58:39 +02:00
KЯØNE 371f5c2c7b fix: prevent goroutine leak by adding exit signal for RTCP PLI ticker (#697)
Co-authored-by: alvarorichard <alvarorichard@tuta.io>
2026-09-10 13:33:49 +02:00
Miroslav Šedivý 827944349b fix: reduce audio jitter buffer latency (#673)
* fix: reduce audio jitter buffer latency

- Release listenersMu before dispatching samples to listeners
- Buffer Track.sample channel (size 2) with non-blocking WriteSample
- Buffer pipeline.sample channel (size 4) to unblock GStreamer thread
- Add leaky=downstream max-size-buffers=5 to audio queue to prevent burst accumulation
- Set sync=false on appsink to deliver buffers immediately without clock wait

These changes break the backpressure chain that caused the GStreamer
streaming thread to stall. When the Go scheduler has a hiccup, buffered
frames were delivered in a burst, causing the browser audio jitter buffer
to grow to ~10 seconds over time.

Fixes: #598

* fix: log message when dropping samples due to full track channel
2026-08-04 14:59:11 +02:00
Miroslav Šedivý 2c124c506e Upgrade go dependencies (#644)
* update pion WebRTC from v3 to v4.

* update mapstructure dependency to use go-viper version.

* update go-chi dependency to v5.
2026-04-25 15:20:43 +02:00
Miroslav Šedivý a545115156 fix regression #639, #638. 2026-04-10 10:58:34 +02:00
Miroslav Šedivý a4f1cfbe0d refactor: standardize constant names and improve goroutine handling across multiple files. 2026-04-08 15:16:47 +02:00
Hani bc19ab866f feat(client): add microphone passthrough button to controls toolbar (#620)
* feat(client): add microphone passthrough button to controls toolbar

Add mic toggle button to the bottom controls bar that enables users to
share their local microphone with the remote neko session via WebRTC.

The server already supports microphone capture (capture.microphone.enabled)
but the legacy client had no UI to trigger getUserMedia and send an audio
track to the peer connection.

Changes:
- base.ts: Add enableMicrophone/disableMicrophone methods that call
  getUserMedia and addTrack/removeTrack on the RTCPeerConnection.
  Mic is cleaned up automatically on disconnect.
- controls.vue: Add mic button (fa-microphone/fa-microphone-slash) between
  play/pause and volume controls with tooltip and error handling.
- en-us.ts: Add i18n strings for mic tooltips and error dialog.

* if the error is not io.EOF, log it. Otherwise, it's a normal closure of the track.

* tie microphone to active host and auto-disable on control loss

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Co-authored-by: h1n054ur <admin@haniumer.com>
Co-authored-by: Miroslav Šedivý <sedivy.miro@gmail.com>
2026-02-17 17:02:51 +01:00
Miroslav Šedivý 1c63b56b94 legacyhandler: key/button action log only in trace level. 2025-04-03 21:50:09 +02:00
Miroslav Šedivý 51bbad7650 rename package to github name. 2025-03-26 23:51:19 +01:00
Miroslav Šedivý f8e9c20ff1 enable WebRTC legacy conditionally. 2025-02-20 19:33:57 +01:00
Miroslav Šedivý 15e805c9b2 rename package. 2024-09-06 23:26:29 +02:00
Miroslav Šedivý 9ea9c1c208 add legacy webrtc handler. 2024-09-06 22:59:04 +02:00
Miroslav Šedivý cfb423b13d move server to server directory. 2024-09-06 22:56:24 +02:00