randos v0.11.0

randos v0.11.0

Choose every language you are happy to speak. Both the website and terminal
client now accept multiple speaking choices, alongside your hearing choices.
For each random match, randos selects one compatible speaking language per
person and labels the call with those choices. Your full selection remains
available for the next conversation. Existing single-language settings migrate
automatically, and older single-language clients remain supported.

The website has a calmer, centered connecting screen with your language pair,
live audio meters, and matching Mute and Cancel controls. It fits narrow phone
screens and supports light and dark themes.

The terminal client makes voice masking easier to inspect and try:

• The main header shows when masking is selected, including its style and tone.
  During a call it shows the frozen profile actually used for that call.
• The masking panel begins with the reminder that people may still recognize
  you. Off hides the unused tone option, and the preview meter stays inside
  the open panel.
• Use −/+ to adjust local Preview volume from 1 to 9. This session-only
  control makes quiet input easier to hear; it does not change the audio sent
  in calls. Press p to start or stop listening, and Esc to close.

The native helper also stops capture and reports failure when a WebRTC session
has permanently closed or its audio writer is unavailable, rather than keeping
an apparently connected but stalled call running. This is a lifecycle fix,
not automatic microphone leveling or a change to echo cancellation.

Opt-in local numeric diagnostics are available for investigating native call
input problems. They contain no recordings or participant identifiers and
are disabled by default; see the development guide.

Both clients include machine-selected translation updates. Labels without a
candidate that passes the established checks retain the English fallback.

Installation

Update with brew upgrade wasichu/randos/randos, or unpack the macOS arm64 or
Linux x86_64 archive and run bin/randos. Archives include ERTS, the native audio
helper, required licenses, and SHA-256 checksum files. Linux requires glibc 2.39
or newer, ALSA, OpenSSL 3, and the system C++ standard library.

Self-hosters should deploy the updated server before distributing this TUI:
multiple speaking languages extend the participant protocol. Existing
single-language clients can continue using the updated server.

Voice masking remains local and off by default. There are no accounts,
recordings, server-side audio, tracking, or new production voice models.