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 20:07:11 +00:00
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+++
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disableToc = false
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title = "Audio Transform"
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weight = 17
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url = "/features/audio-transform/"
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+++
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The audio-transform endpoints take **audio in** and emit **audio out**, optionally
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conditioned on a second reference audio signal. The category is generic by
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design — concrete operations include joint **acoustic echo cancellation +
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noise suppression + dereverberation** (LocalVQE), voice conversion (reference
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= target speaker), pitch shifting, audio super-resolution, and so on.
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The first shipping backend is [LocalVQE](https://github.com/localai-org/LocalVQE),
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a 1.3 M-parameter GGML-based model that performs joint AEC + noise suppression
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+ dereverberation on 16 kHz mono speech, ~9.6× realtime on a desktop CPU. It
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is a derivative of the Microsoft DeepVQE paper.
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## The mental model
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Every audio-transform request carries:
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- **`audio`** — the primary input file (required).
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- **`reference`** — an auxiliary signal whose meaning is backend-specific (optional).
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- For echo cancellation: the loopback / far-end signal played through the speakers.
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- For voice conversion: the target speaker's reference clip.
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- For pitch / style transfer: a tonal or style reference.
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- When omitted, the backend treats it as silence and degrades gracefully (LocalVQE,
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for example, does denoise + dereverb only when ref is empty).
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- **`params`** — a generic `key=value` map forwarded to the backend.
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- LocalVQE keys: `noise_gate=true|false`, `noise_gate_threshold_dbfs=<float>`.
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This shape mirrors WebRTC's `ProcessStream(near)` / `ProcessReverseStream(far)`
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APM API, NVIDIA Maxine's `NvAFX_Run` paired-stream signature, and the ICASSP
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AEC challenge 2-channel WAV convention.
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## Batch endpoint
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`POST /audio/transformations` (alias `POST /audio/transform`) — multipart
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form-data, returns audio bytes.
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| Field | Type | Required | Notes |
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|---|---|---|---|
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| `model` | string | yes | Audio-transform model id (e.g. `localvqe`) |
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| `audio` | file | yes | Primary input audio |
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| `reference` | file | no | Optional auxiliary signal |
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| `response_format` | string | no | `wav` (default), `mp3`, `ogg`, `flac` |
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| `sample_rate` | int | no | Desired output sample rate |
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| `params[<key>]` | string | no | Repeated; forwarded to backend |
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Example (LocalVQE: cancel echo, suppress noise, gate residual):
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```bash
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curl -X POST http://localhost:8080/audio/transformations \
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-F model=localvqe \
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-F audio=@mic.wav \
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-F reference=@loopback.wav \
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-F 'params[noise_gate]=true' \
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-F 'params[noise_gate_threshold_dbfs]=-50' \
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-o enhanced.wav
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```
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When `reference` is omitted, LocalVQE zero-fills the reference channel and
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the operation reduces to noise suppression + dereverberation.
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## Streaming endpoint
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`GET /audio/transformations/stream` — bidirectional WebSocket. The first
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client message is a JSON envelope; subsequent client messages are binary
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PCM frames; server emits binary PCM frames at the same cadence.
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### Wire format
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**Client → server** (text frame, first):
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```json
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{
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"type": "session.update",
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"model": "localvqe",
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"sample_format": "S16_LE",
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"sample_rate": 16000,
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"frame_samples": 256,
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"params": { "noise_gate": "true" }
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}
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```
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`sample_format` is `S16_LE` (16-bit signed little-endian) or `F32_LE` (32-bit
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float little-endian, [-1, 1]). `frame_samples` defaults to the backend's
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preferred hop length (256 = 16 ms for LocalVQE).
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**Client → server** (binary frames, subsequent): interleaved stereo PCM,
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channel 0 = audio (mic), channel 1 = reference. Frame size:
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`frame_samples × 2 channels × sample_size`. For `S16_LE` at 256 samples that
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is 1024 bytes per frame; for `F32_LE` it is 2048 bytes. If the reference is
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silent (no auxiliary signal), send zeros on channel 1.
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**Server → client** (binary frames): mono PCM in the same format,
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`frame_samples × sample_size` bytes (512 bytes for `S16_LE`, 1024 for `F32_LE`).
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**Mid-stream control** (text frame): another `session.update` resets the
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streaming state when its `reset` field is true; a `session.close` text frame
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ends the session cleanly.
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### Latency
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LocalVQE has 16 ms algorithmic latency (one hop). At runtime, ~1.66 ms of CPU
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time per frame on a modern desktop, leaving the rest of the budget for
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network and downstream playback.
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## Backend-specific tuning (LocalVQE)
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| `params[<key>]` | Type | Default | Effect |
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|---|---|---|---|
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| `noise_gate` | bool | `false` | Enable post-OLA RMS-based residual-echo gate |
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| `noise_gate_threshold_dbfs` | float | `-45.0` | Gate threshold in dBFS; frames below are zeroed |
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The gate is most useful in far-end-only / silent-near-end stretches where the
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model's residual would otherwise sound like buffering or amplified noise floor.
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A reasonable starting point is `-50` dBFS.
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## Configuring a model
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```yaml
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name: localvqe
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backend: localvqe
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parameters:
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model: localvqe-v1.1-1.3M-f32.gguf
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# Backend-specific defaults can be set in Options[]; per-request
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# params[*] form fields override.
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#
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# `backend` and `device` route through the upstream localvqe options
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# builder so you can force a non-default GGML backend (e.g. `Vulkan`) or
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# pin to a specific GPU index. Leave both unset to keep the CPU default.
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options:
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- noise_gate=true
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- noise_gate_threshold_dbfs=-50
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# - backend=Vulkan
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# - device=0
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```
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## See also
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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 08:36:59 +00:00
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- [Text to Audio (TTS)]({{< relref "text-to-audio.md" >}})
|
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 20:07:11 +00:00
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- [Audio to Text]({{< relref "audio-to-text.md" >}})
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- [LocalVQE upstream](https://github.com/localai-org/LocalVQE)
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- [DeepVQE paper (Indenbom et al., Interspeech 2023)](https://arxiv.org/abs/2306.03177)
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