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3 Commits

Author SHA1 Message Date
Richard Palethorpe
718223f33b feat(localvqe/audio): v1.3 release and add spectrograms to audio transform UI (#10113) 
* chore(localvqe): update backend to v1.3, add v1.2/v1.3 gallery models

Bump the LocalVQE backend pin 72bfb4c6 -> b0f0378a, which adds the v1.2
(1.3 M) and v1.3 (4.8 M) GGUF SHA-256s to the upstream released-models
allowlist (and the arch_version=3 loader) so both load without
LOCALVQE_ALLOW_UNHASHED.

Add gallery entries for localvqe-v1.2-1.3m and localvqe-v1.3-4.8m
(SHA-256 verified against the downloaded weights) and update the
audio-transform docs to make v1.3 the current default while noting the
compact v1.1/v1.2 alternatives.

Assisted-by: Claude:claude-opus-4-8 Claude-Code
Signed-off-by: Richard Palethorpe <io@richiejp.com>

* chore(flake): add ffmpeg-headless to the dev shell

pkg/utils/ffmpeg_test.go shells out to the `ffmpeg` CLI, and the
pre-commit gate runs those tests via `make test-coverage`. Without
ffmpeg in the dev shell the gate fails with "executable file not found
in $PATH". The headless build provides the CLI without GUI/X deps.

Assisted-by: Claude:claude-opus-4-8 [Claude Code]
Signed-off-by: Richard Palethorpe <io@richiejp.com>

* fix(localvqe): parse WAV by walking RIFF sub-chunks

Walk the RIFF chunk list instead of assuming the canonical 44-byte
header layout. Real inputs (browser-recorded clips, ffmpeg output with
an 18/40-byte extensible `fmt ` chunk or trailing LIST/INFO metadata)
would otherwise splice header/metadata bytes into the PCM stream as an
audible impulse. Honour the `data` chunk size and validate that both
`fmt ` and `data` chunks are present.

Assisted-by: Claude:claude-opus-4-8 [Claude Code]
Signed-off-by: Richard Palethorpe <io@richiejp.com>

* fix(security-headers): allow blob: in connect-src for waveform fetch

The waveform renderer XHRs/fetches a freshly-created blob: object URL
(e.g. an uploaded or enhanced clip before it has a server URL). XHR/fetch
of blob: is governed by connect-src, not media-src, so it was blocked by
the CSP. Add blob: to connect-src.

Assisted-by: Claude:claude-opus-4-8 [Claude Code]
Signed-off-by: Richard Palethorpe <io@richiejp.com>

* feat(react-ui): add input/output spectrogram view to AudioTransform

The transform page only showed time-domain amplitude waveforms, so you
could see how loud a clip was but not which frequencies the model
touched. Add a time x frequency spectrogram heatmap and render the input
and output spectrums side by side, so it's visible which bands the
enhancement attenuates (bright input bands that go dark in the output).

Computed client-side via a Hann-windowed STFT over both clips (a small
dependency-free radix-2 FFT), defaulting to the LocalVQE 512/256 frame
geometry. This shows the net input->output spectral change; the model's
internal gain mask is not exposed by the backend.

- src/utils/fft.js            radix-2 FFT
- src/hooks/useSpectrogram.js decode + STFT -> normalised dB magnitude grid
- src/components/audio/Spectrogram.jsx  canvas heatmap (magma colormap)
- AudioTransform.jsx          dual-spectrogram panel + CSS
- e2e spec + UI coverage baseline bump (38.29 -> 39.0; measured ~39.4-40.2)

Assisted-by: Claude:claude-opus-4-8 [Claude Code]
Signed-off-by: Richard Palethorpe <io@richiejp.com>

* test(react-ui): make UI coverage deterministic, tighten the gate

UI e2e line coverage swung ~1pp run-to-run (39.1% <-> 40.2%), which forced
a loose 0.8pp tolerance on the monotonic gate — a band wide enough to let
a real ~300-line regression through silently. The swing was a bug, not
inherent jitter: the 'Create Agent navigates' spec ended on the URL
assertion, so AgentCreate.jsx's ~400 lines were collected only when its
render happened to beat the coverage teardown.

Wait for the page to actually render (assert its heading) so those lines
are covered every run. With the race gone, repeated runs land within
~0.013pp of each other, so:

- tighten UI_COVERAGE_TOLERANCE 0.8 -> 0.1 (noise floor, not a drift band)
- set the baseline to the real, reliably-achieved value (39.0 -> 39.86)

Localised by running the V8-coverage suite repeatedly and diffing per-file
line coverage; AgentCreate.jsx was the sole ~1pp flipper.

Assisted-by: Claude:claude-opus-4-8 [Claude Code]
Signed-off-by: Richard Palethorpe <io@richiejp.com>

---------

Signed-off-by: Richard Palethorpe <io@richiejp.com>
 2026-05-31 23:56:46 +02:00
LocalAI [bot]
a8d7d37a3c fix: unbreak master CI (docs, kokoros, vibevoice-cpp ABI) (#9682) 
* fix(docs): correct broken Hugo relrefs

The Hugo build has been failing on master since the relevant pages
landed:

- text-generation.md:720 referenced `/docs/features/distributed-mode`,
  but Hugo `relref` paths are relative to the content root, not the
  rendered URL. Drop the `/docs/` prefix so the lookup matches the
  existing `features/...` form used elsewhere in the file.
- audio-transform.md:144 referenced `tts.md`; the actual page is
  `text-to-audio.md`.

Assisted-by: Claude:claude-opus-4-7[1m]
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* fix(kokoros): stub Diarize and AudioTransform Backend trait methods

The recent backend.proto additions (Diarize, AudioTransform,
AudioTransformStream) extended the gRPC Backend trait, breaking
kokoros-grpc compilation with E0046 because the Rust implementation
hadn't picked up the new methods. Add Unimplemented stubs matching the
existing pattern for non-applicable RPCs in this TTS-only backend.

Assisted-by: Claude:claude-opus-4-7[1m]
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* fix(vibevoice-cpp): track upstream ABI + wire 1.5B voice cloning

Two recent commits in mudler/vibevoice.cpp reshaped the vv_capi_tts
signature without a corresponding bump on the LocalAI side:

  3bd759c "1.5b: unify into a single tts entry point" inserted a
          ref_audio_path parameter between voice_path and dst_wav_path.
  ad856bd "1.5b: multi-speaker dialog support" promoted that to a
          (const char* const* ref_audio_paths, int n_ref_audio_paths)
          pair for per-speaker conditioning.

Because purego resolves symbols by name and not by signature, the
build kept linking; at runtime the misaligned arguments turned the
TTS->ASR closed-loop test into a SIGSEGV inside cgo. Track HEAD
explicitly and bring the bridge in line with it:

  * Update the CppTTS purego binding to the 9-arg form. purego
    marshals []*byte as a **char by handing the C side the underlying
    array address; nil/empty maps to NULL, which matches the C
    contract for "no reference audio" on the realtime-0.5B path.
  * Add a `ref_audio` gallery option (comma-separated, repeatable)
    that the 1.5B path consumes for runtime voice cloning. Multiple
    entries are interpreted as one WAV per speaker (Speaker 0..n-1).
  * TTSRequest.Voice now routes by extension/shape: `.wav` or a
    comma-separated list goes to ref_audio_paths; anything else stays
    on voice_path (realtime-0.5B's pre-baked voice gguf).
  * Pin VIBEVOICE_CPP_VERSION to ad856bd and wire the Makefile into
    the existing bump_deps matrix so future upstream rolls land as
    reviewable PRs instead of a silent CI break.

Assisted-by: Claude:claude-opus-4-7[1m]
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* refactor(vibevoice-cpp): use ModelOptions.AudioPath for 1.5B ref audio

Use the existing audio_path field from ModelOptions (already plumbed
through config_file's `audio_path:` YAML and consumed by other audio
backends like kokoros) instead of inventing a custom `ref_audio:`
Options[] string. Multi-speaker setups stay on a single comma-
separated value.

No behavior change beyond the gallery key name; per-call routing via
TTSRequest.Voice is unchanged.

Assisted-by: Claude:claude-opus-4-7[1m]
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

---------

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
Co-authored-by: Ettore Di Giacinto <mudler@localai.io>
 2026-05-06 10:36:59 +02:00
Richard Palethorpe
bb033b16a9 feat: add LocalVQE backend and audio transformations UI (#9640) 
feat(audio-transform): add LocalVQE backend, bidi gRPC RPC, Studio UI

Introduce a generic "audio transform" capability for any audio-in / audio-out
operation (echo cancellation, noise suppression, dereverberation, voice
conversion, etc.) and ship LocalVQE as the first backend implementation.

Backend protocol:
- Two new gRPC RPCs in backend.proto: unary AudioTransform for batch and
  bidirectional AudioTransformStream for low-latency frame-by-frame use.
  This is the first bidi stream in the proto; per-frame unary at LocalVQE's
  16 ms hop would be RTT-bound. Wire it through pkg/grpc/{client,server,
  embed,interface,base} with paired-channel ergonomics.

LocalVQE backend (backend/go/localvqe/):
- Go-Purego wrapper around upstream liblocalvqe.so. CMake builds the upstream
  shared lib + its libggml-cpu-*.so runtime variants directly — no MODULE
  wrapper needed because LocalVQE handles CPU feature selection internally
  via GGML_BACKEND_DL.
- Sets GGML_NTHREADS from opts.Threads (or runtime.NumCPU()-1) — without it
  LocalVQE runs single-threaded at ~1× realtime instead of the documented
  ~9.6×.
- Reference-length policy: zero-pad short refs, truncate long ones (the
  trailing portion can't have leaked into a mic that wasn't recording).
- Ginkgo test suite (9 always-on specs + 2 model-gated).

HTTP layer:
- POST /audio/transformations (alias /audio/transform): multipart batch
  endpoint, accepts audio + optional reference + params[*]=v form fields.
  Persists inputs alongside the output in GeneratedContentDir/audio so the
  React UI history can replay past (audio, reference, output) triples.
- GET /audio/transformations/stream: WebSocket bidi, 16 ms PCM frames
  (interleaved stereo mic+ref in, mono out). JSON session.update envelope
  for config; constants hoisted in core/schema/audio_transform.go.
- ffmpeg-based input normalisation to 16 kHz mono s16 WAV via the existing
  utils.AudioToWav (with passthrough fast-path), so the user can upload any
  format / rate without seeing the model's strict 16 kHz constraint.
- BackendTraceAudioTransform integration so /api/backend-traces and the
  Traces UI light up with audio_snippet base64 and timing.
- Routes registered under routes/localai.go (LocalAI extension; OpenAI has
  no /audio/transformations endpoint), traced via TraceMiddleware.

Auth + capability + importer:
- FLAG_AUDIO_TRANSFORM (model_config.go), FeatureAudioTransform (default-on,
  in APIFeatures), three RouteFeatureRegistry rows.
- localvqe added to knownPrefOnlyBackends with modality "audio-transform".
- Gallery entry localvqe-v1-1.3m (sha256-pinned, hosted on
  huggingface.co/LocalAI-io/LocalVQE).

React UI:
- New /app/transform page surfaced via a dedicated "Enhance" sidebar
  section (sibling of Tools / Biometrics) — the page is enhancement, not
  generation, so it lives outside Studio. Two AudioInput components
  (Upload + Record tabs, drag-drop, mic capture).
- Echo-test button: records mic while playing the loaded reference through
  the speakers — the mic naturally picks up speaker bleed, giving a real
  (mic, ref) pair for AEC testing without leaving the UI.
- Reusable WaveformPlayer (canvas peaks + click-to-seek + audio controls)
  and useAudioPeaks hook (shared module-scoped AudioContext to avoid
  hitting browser context limits with three players on one page); migrated
  TTS, Sound, Traces audio blocks to use it.
- Past runs saved in localStorage via useMediaHistory('audio-transform') —
  the history entry stores all three URLs so clicking re-renders the full
  triple, not just the output.

Build + e2e:
- 11 matrix entries removed from .github/workflows/backend.yml (CUDA, ROCm,
  SYCL, Metal, L4T): upstream supports only CPU + Vulkan, so we ship those
  two and let GPU-class hardware route through Vulkan in the gallery
  capabilities map.
- tests-localvqe-grpc-transform job in test-extra.yml (gated on
  detect-changes.outputs.localvqe).
- New audio_transform capability + 4 specs in tests/e2e-backends.
- Playwright spec suite in core/http/react-ui/e2e/audio-transform.spec.js
  (8 specs covering tabs, file upload, multipart shape, history, errors).

Docs:
- New docs/content/features/audio-transform.md covering the (audio,
  reference) mental model, batch + WebSocket wire formats, LocalVQE param
  keys, and a YAML config example. Cross-links from text-to-audio and
  audio-to-text feature pages.

Assisted-by: Claude:claude-opus-4-7 [Bash Read Edit Write Agent TaskCreate]

Signed-off-by: Richard Palethorpe <io@richiejp.com>
 2026-05-04 22:07:11 +02:00