Commit Graph
7 Commits
Author SHA1 Message Date
soubarnak ea7be5f111 feat(identity): rename user-visible identity to voculinux with one-time legacy import
The fork takes its own name everywhere a user can see it: binaries
(voculinux, voculinux-gui), desktop entry, autostart, tray, D-Bus
namespace (com.voculinux.Voculinux, no alias for the old name),
config/data/cache dirs, single-instance lock, IBus engine identity,
tray icons, resource prefixes, VOCULINUX_* environment names, and the
update-checker user agent. The internal Python module stays vocalinux
and the distribution metadata keeps the upstream URLs as provenance.

On first run a previous Vocalinux install's config.json, custom
dictionary, and portal restore token are copied once into the new
locations (new config always wins, no merge, no write-back); models
re-resolve through the pinned-checksum download path. An offline
naming-sweep test proves no old-name strings remain on user-visible
surfaces outside the allowlist.

Refs #18
2026-11-06 01:29:39 +05:30
e66bd218d9 fix(audio): prevent dictation hangs on stalled microphones (#974)
* fix(audio): bound capture reads and support cancellation

* fix(recognition): bound and serialize capture shutdown

* fix(recognition): cancel retries and report stalled microphones

* fix(audio): prefer pulse when the ALSA default is unavailable

* fix(recognition): guard late callbacks and finalize capture tests

* docs(audio): explain input selection and stall recovery

* fix(speech): enforce auto-pause across buffered reload timeouts

* fix(speech): recover auto-resume after reload timeout

* test(speech): fix auto-pause follow-up test regressions

* fix(speech): retain auto-resume after reconfigure timeout

* fix(tray): preserve auto-pause recovery across suspend

* test(capture-shutdown): give the bare tray the suspend path's shortcut managers

_on_system_suspend now releases the main and per-language hotkey
listeners before sleep (#1025); the fixture builds TrayIndicator without
__init__, so it needs the two attributes present.

* fix(tray): release hotkey listeners on suspend even while auto-paused

Merging #1025's suspend cleanup with this branch's auto-pause handoff
left an early return that skipped shortcut_manager.stop() whenever a
configured game was active, keeping EVIOCGRABs held through sleep. The
paused branch now only owns the model-lifecycle work; listener release
and stopping active recognition run for every suspend.

* fix(tray): hold the apply lock only around the download's verify and save

Wrapping the whole recommended-model download in _apply_settings_lock
blocked every GTK-thread settings apply (e.g. Test Dictation) for the
length of the transfer. The engine already serializes reconfigures, so
the lock only needs to cover verifying the download still owns the
receipt and saving it; an apply that lands mid-download now proceeds
instead of freezing the app.

---------

Co-authored-by: Jatin K Malik <jatinkrmalik@gmail.com>
2026-10-26 21:25:50 -07:00
4d78246174 fix(recognition): queue long dictation segments before the buffer cap (#976)
* fix(recognition): queue long dictation segments before the buffer cap

A held push-to-talk key (or a toggle session with no pause) keeps a single
segment growing until the 5000-chunk safety cap discards its oldest 25%.
The cap is a chunk count, not a duration, so the effective limit depends on
the capture format: at 16 kHz a chunk is 64 ms (5 min 20 s), but a device
that negotiates 44.1 kHz feeds 23 ms chunks and the same cap is 1 min 56 s.
A 3-minute hold therefore lost ~87 s of speech before transcription, with
only a log warning to show for it.

Flush the current segment onto the existing transcription queue once it
reaches _max_segment_seconds (90 s), or 75% of the chunk budget when chunks
are shorter, so the destructive trim can never fire on buffered speech.
Recording continues while queued segments are transcribed in order; reload
sessions keep the existing hold-until-reload behaviour.

Adds tests for the mid-hold flush, the buffer headroom, and the reload path.

* fix(recognition): hold push-to-talk text until the key is released

Text injection while PTT is held can release or modify the key.

* fix(recognition): hold PTT text without blocking the recognition worker

c7b9bd6 held transcribed text by waiting on the release gate inside
_process_audio_buffer, on the single recognition worker. Later segments
then sat in the 32-slot queue for the rest of the hold instead of being
transcribed: memory grows with every queued segment, and once 32 are
waiting (~48 min of 90 s segments) _enqueue_audio_segment starts dropping
the oldest speech.

Buffer the processed text instead. Every segment is transcribed as it
arrives; while the key is down its result is appended to a held list, and
the held results are delivered in capture order once the gate opens. The
final buffer carries the session gate too, so it cannot overtake text
that was transcribed before release. Held text is flushed when the worker
exits, at the end of stop_recognition, and after the reload drain. An
empty mid-hold result no longer flushes the held text early.

Tests assert that every queued segment is transcribed before release and
delivered in capture order afterwards.

* fix(recognition): transcribe held push-to-talk segments ahead of release

Keep transcription running during long push-to-talk holds while retaining each result until key release. Text injection during a hold can release or modify the PTT key.

* test(recognition): cover cross-session held text isolation

The per-session release gate must release only its own held text: a slow
worker from an earlier session finishing after a new push-to-talk session
starts must not deliver the new session's text while its key is down.

---------

Co-authored-by: Muhannad Abo Hajia <muhannadabohajia@gmail.com>
Co-authored-by: Jatin K Malik <jatinkrmalik@gmail.com>
2026-10-21 00:37:43 -07:00
5e531cbac0 refactor(audio): extract capture source from recognition manager (#883)
* refactor(audio): extract capture source from recognition manager

Move audio capture out of the silence-segmented dictation loop into
vocalinux.audio.capture: a reusable PortAudioCaptureSource that owns
device resolution, format negotiation, downmixing, resampling, and
stream lifecycle (open/read/reopen/close). The dictation loop now
consumes 16 kHz mono chunks and keeps only segmentation policy (VAD,
silence counters, buffer limits, reconnection backoff).

This unblocks alternate capture sources (PipeWire system-audio capture
for #751, tdrz long-form output for #756) which need the same capture
mechanics without dictation's silence segmentation.

Behavior is identical: the same helpers run in the same order, moved
verbatim. They are re-exported from recognition_manager for existing
imports, and a CaptureSource protocol is declared in common_types.py.
Tests repoint their patches of the moved helpers to the new module.

Closes #760

* fix(audio): annotate capture API and keep a superseded session's source

* fix(audio): reset retry counters when a new capture session starts

* fix(audio): keep last-error timestamp across capture sessions

The new-session reset wiped _last_audio_error_time, so an error within five seconds of the previous session's error always took the reconnect path instead of failing fast. Only the attempt count is per-session; the timestamp is also the rapid-failure throttle.

---------

Co-authored-by: Jatin K Malik <jatinkrmalik@gmail.com>
2026-09-29 23:04:43 -07:00
a89966a744 feat(recognition): record while an idle model reloads (#851)
* feat(recognition): buffer audio during model reload

* fix(recognition): recover buffered reload failures

* test(recognition): cover buffered reload edge cases

* fix(recognition): narrow buffered reload errors

* fix(settings): wait for IDLE before restoring Test Dictation callbacks

Buffered reload's stop_recognition() returns while PROCESSING. Test
Dictation must keep its text callbacks until recognition reaches IDLE
so the test utterance cannot inject into the focused app.

* style: apply Black formatting to PR 851 files

* fix(settings): cancel buffered reload before Test Dictation timeout restore

On the ~3min idle-wait timeout, do not restore live injector callbacks while
a buffered reload worker may still deliver text. Cancel via
_cancel_reload_recording off the GTK thread, then restore; if cancel is
unavailable or fails, leave test callbacks installed and show an error.
Add return/param annotations and regression coverage.

* fix(settings): keep Test Dictation waiting if buffered cancel fails

After the idle-wait timeout, if cancel is unavailable or raises, keep
polling with test callbacks and _test_active instead of marking the test
done. Ending early left the test callback as the only text destination,
so later normal dictation could land in the hidden test buffer. Restore
live injectors only once recognition reaches IDLE.

* fix(settings): reset Test Dictation timeout-cancel flag per session

Clear _test_timeout_cancel_attempted when a test starts, when entering
the buffered idle-wait, and when restoring UI so a later Test Dictation
can still cancel after timeout. Narrow the cancel-worker except clause
to AttributeError/OSError/RuntimeError/TypeError so unrelated defects
are not treated as ordinary cancel failure.

* fix(recognition): signal capture completion when buffered capture dies

A dead microphone during a buffered model reload left should_record=false
with no stop_recognition handoff, so the reload worker blocked forever on
_capture_finished.wait(). Signal the event from the audio thread's failure
teardown paths instead.

Refs VocaHQ/vocalinux#851

---------

Co-authored-by: Jatin K Malik <jatinkrmalik@gmail.com>
2026-09-29 22:02:56 -07:00
Jatin K MalikandGitHub 092dde105e fix(audio): open 3-8ch HDA mics at native layout (#813) (#829)
* fix(audio): open 3-8ch HDA mics at native layout (#813)

HDA analog capture often reports 4 channels. Opening below that native
count can abort PortAudio after read with free(): corrupted unsorted
chunks (same class of bug as #666/#673).

Try the reported channel count first when it is between 3 and 8, then
2ch and 1ch. Pulse devices that advertise 32/128 channels still open
stereo-first. Downmix N channels to mono so engines keep seeing 1ch.

Virtual-device filtering for pipewire/default (#629) is unchanged.

* fix(audio): downmix HDA capture from first channel pair

Averaging all N HDA channels halves speech when only the front pair is live.
Keep stereo mean-of-both, and for 3+ channels average channels 0 and 1.

* fix(audio): pick loudest channel for N-ch downmix

First-pair-only downmix silenced mics on ch2+. For 3+ channels, select
the loudest channel by per-buffer mean-square so speech stays at full
level whether it lands on the front or rear pair. Stereo still averages
both channels.

* fix(audio): sticky N-ch downmix channel for open stream

Pin the loudest-channel pick for the life of a capture stream so a
noise burst on another input cannot switch the mic mid-utterance.

* fix(audio): gate sticky downmix lock on speech energy

Don't pin the N-ch sticky channel from a silent/ambient first buffer;
only lock once the loudest channel clears a documented mean-square floor.

* style: isort sticky-lock import in device config tests
2026-09-14 00:21:17 -07:00
3f5a5effff feat(vad): add Silero VAD and drop silence-only buffers (#447)
* feat(vad): replace amplitude-based VAD with Silero neural network

Replace the simple amplitude threshold VAD with Silero VAD, a lightweight
neural network model that classifies audio as speech/non-speech. This
addresses the root cause of #302 (silence not stopping recording) and

- Add silero_vad.py wrapper using ONNX Runtime with bundled Silero v5 model
- Accumulate 512-sample chunks with 64-sample context window per official API
- Map existing vad_sensitivity (1-5) to Silero thresholds (0.8-0.3)
- Use max() across sub-chunks to avoid missing speech onsets
- Fall back to amplitude VAD when onnxruntime is unavailable
- Add onnxruntime as optional dependency (pip install vocalinux[vad])

Closes #344

* address review: add tests, LICENSE, logging, validation fixes

- Add unit tests for SileroVAD (process, reset, fallback, sensitivity mapping)
- Add MIT LICENSE for bundled Silero VAD model in data/
- Log which VAD backend is active on init
- Validate chunk length in SileroVAD.process()
- Skip is_speech check until a full Silero chunk is processed
- Add thread safety note to SileroVAD docstring
- Document resampling rate effect on VAD resolution
- Comment explaining why load_silero_vad() catches broad Exception

* address review: bundle license, document [vad] extra, surface backend in UI

- Pack data/LICENSE into the wheel (pyproject package-data) so attribution
  travels with the model
- README: short note on enabling neural VAD via vocalinux[vad]
- Settings: VAD Sensitivity row subtitle shows the active backend; if
  onnxruntime is missing, an info-row beneath the group spells out the
  install command and the restart requirement
- silero_vad.is_silero_available() probes onnxruntime + model presence
  w/o instantiating an InferenceSession
- CI installs .[dev,vad] so Silero paths get exercised
- Integration test for _record_audio Silero branch (mocked pyaudio +
  scripted SileroVAD), covering silence flush, speech reset, and
  sensitivity threshold mapping at sens=1 vs sens=5
- Tests no longer skip when other modules globally mock numpy: pop the
  top-level numpy ref so a fresh import gives the real package, and
  swap sys.modules["numpy"] inside the _record_audio call so the lazy
  import inside the function picks up real numpy too

* fix(vad): drop silence-only recording buffers

* test(vad): align stop buffer expectations

* test(vad): preserve real numpy module

* fix(install): install vad support opportunistically

* test(vad): cover recognition manager vad edges

---------

Co-authored-by: Sbenazar <127324689+Sbenazar@users.noreply.github.com>
2026-06-02 12:43:51 -07:00