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