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feat(ced): sound-event classification backend (CED audio tagger) (#10425)

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* feat(ced): sketch sound-classification backend (CED audio tagger)

Wires ced.cpp (CED, 527-class AudioSet sound-event tagger; baby cry,
footsteps, glass, alarms, dog bark) into LocalAI as a Go/purego backend.

SKETCH (backend skeleton real; core REST wiring + CI/gallery is a checklist
in DESIGN.md):
- backend/backend.proto: new SoundDetection rpc + SoundClass messages
  (run `make protogen-go` to regenerate pkg/grpc/proto).
- backend/go/ced: main.go (purego dlopen libced.so + ced_capi.h),
  goced.go (Ced gRPC backend: Load + SoundDetection), Makefile
  (clone-at-pin CED_VERSION, ggml static-PIC shared build), run.sh,
  package.sh, .gitignore.
- DESIGN.md: REST /v1/audio/classification wiring (handler/route/capability
  registration checklist), gallery/index + CI registration, and a scoping
  note for the realtime/websocket live-recognition path (sliding-window
  classify over the existing ws transport + voicegate; the ced C-API
  per-PCM entry point is already window-friendly).

Backend code does not compile until protogen-go regenerates the pb types
and a libced.so is built (Makefile clones+builds it).

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* feat(ced): REST /v1/audio/classification endpoint + capability registration

Wires the ced sound-event classification backend (AudioSet audio tagger)
end to end through the REST surface, mirroring the transcription path.

- Handler: core/http/endpoints/openai/sound_classification.go parses the
  multipart audio upload, temp-files it, resolves the model config and
  calls the SoundDetection RPC; returns {model, detections[]} JSON.
- Backend wrapper: core/backend/sound_classification.go (ModelSoundDetection)
  loads the model and normalizes the proto response into schema types.
- Schema: core/schema/sound_classification.go (SoundClassificationResult).
- gRPC layer: SoundDetection wired through the LocalAI wrapper (interface,
  Backend client, Client, embed, server, base default) so the loader-typed
  client exposes the RPC; proto regenerated via make protogen-go.
- Route: POST /v1/audio/classification (+ /audio/classification alias) with
  the audio/multipart default-model middleware in routes/openai.go.
- Capability surfaces: swagger @Tags/@Router on the handler; FLAG_SOUND_
  CLASSIFICATION usecase flag + UsecaseSoundClassification + UsecaseInfoMap +
  GuessUsecases + ModalityGroups + GetAllModelConfigUsecases; meta usecase
  option; /api/instructions audio area updated; auth RouteFeatureRegistry +
  FeatureAudioClassification (APIFeatures, default ON) + FeatureMetas; UI
  usecaseFilters, capabilities.js CAP_SOUND_CLASSIFICATION, Models.jsx filter
  + i18n; docs page features/audio-classification.md + whats-new + crosslink.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* feat(ced): realtime sound-event detection over the websocket API

When a realtime pipeline configures a sound-classification model, each
VAD-committed utterance (the same window the transcription path produces)
is also run through the CED sound-event classifier and the scored AudioSet
tags are emitted as a new server event. No new backend rpc is needed: the
SoundDetection gRPC method already exists on this branch.

- config: add Pipeline.SoundDetection (yaml/json sound_detection,omitempty)
  beside Transcription/VAD.
- realtime: add Model.SoundDetection(ctx, audio, topK, threshold) to the
  ModelInterface; implement it on wrappedModel and transcriptOnlyModel by
  calling backend.ModelSoundDetection with the session's sound-classification
  model config (mirrors how Transcribe dispatches). Load the optional config
  in newModel / newTranscriptionOnlyModel; nil config keeps it additive.
- types: add ConversationItemSoundDetectionEvent (item_id, content_index,
  detections[]{label,score,index}) with type conversation.item.sound_detection,
  its ServerEventType constant and MarshalJSON, mirroring the transcription
  completed event.
- realtime: add emitSoundDetection (unary path: classify the committed window,
  build the event, t.SendEvent) and wire it at the utterance-commit hook right
  after emitTranscription; gated on session.SoundDetectionEnabled (resolved
  from Pipeline.SoundDetection at session setup, defaults top_k=5, threshold=0).
  Its error is logged via xlog but never aborts the turn.
- test: Ginkgo specs for emitSoundDetection (tags emitted, empty detections,
  classifier error) plus a SoundDetection method on the fakeModel double.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* fix(ced): implement SoundDetection in nodes backend test doubles

The SoundDetection method added to the grpc backend interface left two
test doubles (fakeBackendClient, fakeGRPCBackend) incomplete, so
core/services/nodes failed to compile under `go vet`/`go test` (go build
missed it: the doubles live in _test.go). Add the method to both,
mirroring their existing Detect mock. Repairs CI for the nodes package.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* feat(ced): decouple realtime sound detection from VAD (sound-only sessions)

Sound-event detection must activate on sounds, not speech, so it no longer
runs through the voice VAD/transcription path. A sound-detection-only
pipeline (sound_detection set, no transcription/LLM) now:

- is accepted by prepareRealtimeConfig (sound_detection counts as a pipeline
  stage),
- builds a lightweight model via newSoundDetectionOnlyModel (no VAD/STT/LLM/TTS
  loaded), and
- defaults the session to turn_detection none (no VAD) with no transcription
  stage, so the client drives windowing via input_audio_buffer.commit
  (option A: client-side sliding window). The per-PCM C-API already supports
  arbitrary windows.

commitUtterance gains a sound-only branch: it emits the
conversation.item.sound_detection event (scored AudioSet tags) and stops -
no transcription, no LLM response. generateResponse is now guarded on a
transcription stage being present, so a sound-only turn never invokes the LLM.

Existing transcription/VAD sessions are unchanged (additive). Added a
commitUtterance sound-only Ginkgo spec asserting it emits the sound event and
neither transcribes nor generates a response. go vet + golangci-lint
(new-from-merge-base) clean; openai suite green.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* feat(ced): register sound-classification backend in gallery + CI

Mechanical backend-image registration for the ced sound-event classifier,
mirroring the parakeet-cpp Go/purego backend everywhere it is wired up.

- .github/backend-matrix.yml: add the ced build matrix, field-for-field copies
  of the parakeet-cpp entries (cpu amd64/arm64, cublas cuda 12/13 amd64,
  l4t cuda-13 arm64, l4t-jetpack cuda-12 arm64, sycl f32/f16, vulkan
  amd64/arm64, rocm hipblas, and the metal darwin entry), changing only
  backend and tag-suffix. dockerfile stays ./backend/Dockerfile.golang.
- backend/index.yaml: add the &ced meta anchor (capabilities map per platform)
  plus ced-development and the per-arch image entries, each uri/mirror
  tag-suffix matching the matrix exactly. The model gallery (GGUF) entry is
  intentionally deferred pending the HuggingFace publish (TODO note inline).
- scripts/changed-backends.js: add an explicit item.backend === "ced" branch in
  inferBackendPath mapping to backend/go/ced/, same mechanism and ordering as
  the parakeet-cpp branch (before the generic golang fallthrough).
- .github/workflows/bump_deps.yaml: register mudler/ced.cpp -> CED_VERSION in
  backend/go/ced/Makefile so the daily bot bumps the pin.
- swagger/{docs.go,swagger.json,swagger.yaml}: regenerated via make swagger so
  the existing /v1/audio/classification annotations land in the generated spec.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* feat(ced): server-side windowing for realtime sound detection (option B)

Adds an optional server-driven sliding-window classifier so a sound-only
realtime client only has to stream audio (no input_audio_buffer.commit):

- Pipeline.sound_detection_window_ms / sound_detection_hop_ms config knobs.
  When both > 0 on a sound-only session, the server classifies the last
  window of streamed audio every hop and emits a conversation.item.sound_
  detection event; the input buffer is trimmed to one window so a long
  stream stays bounded. When unset, the session stays client-driven
  (option A). Runs independent of VAD (sound events are not speech).
- handleSoundWindow (ticker) + classifySoundWindow (one tick, extracted so
  it is unit-testable) + writeWindowWAV, which declares the true
  InputSampleRate (NewWAVHeaderWithRate) so the classifier resamples
  correctly. Goroutine is started after toggleVAD and torn down with the
  session (close + wg.Wait).
- Register pipeline.sound_detection (+window_ms/hop_ms) in the config meta
  registry; the earlier realtime commit added pipeline.sound_detection
  without a registry entry, failing TestAllFieldsHaveRegistryEntries. This
  fixes that and covers the two new knobs.

Tests: classifySoundWindow emits an event + trims the buffer to one window,
no-ops on too-little audio; writeWindowWAV declares the given sample rate.
go build/vet + golangci-lint (new-from-merge-base) clean; config + openai
suites green.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* feat(ced): add ced-base GGUF model gallery entries (f16 + q8_0)

The ced-base weights are now published at mudler/ced-base-gguf (Apache-2.0,
converted from mispeech/ced-base). Adds gallery/ced.yaml (backend: ced +
known_usecases: sound_classification) and two gallery/index.yaml entries
(ced-base-f16 default, ced-base-q8 smallest) with sha256-pinned files, and
removes the now-resolved TODO from backend/index.yaml.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* feat(ced): add tiny/mini/small GGUF model gallery entries

Publishes the rest of the CED family (same architecture, metadata-driven port
verified end-to-end on ced-tiny) to mudler/ced-{tiny,mini,small}-gguf and adds
their f16 + q8_0 gallery entries:

  ced-tiny  (5.5M, edge/Pi-class)  f16 11MB / q8_0 6MB
  ced-mini  (9.6M)                 f16 19MB / q8_0 11MB
  ced-small (22M)                  f16 42MB / q8_0 23MB

All sha256-pinned. ced-base remains the accuracy default.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* chore(ced): point gallery entries at the consolidated mudler/ced-gguf repo

All CED quantizations (tiny/mini/small/base, f16/q8_0) now live in a single
HuggingFace repo, mudler/ced-gguf, instead of per-model repos. Repoint the 8
gallery model entries' urls + file uris accordingly. sha256 and filenames are
unchanged.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* chore(ced): bump CED_VERSION to the short-clip fix

Pin the ced backend to ced.cpp 99c6ed3, which fixes a crash on any clip
shorter than target_length (~10.11s): time_pos_embed was added at its full
63-frame grid instead of being sliced to the clip's actual time grid, tripping
ggml_can_repeat in ggml_add. Surfaced by the live realtime e2e (sub-10s
windows) and gated with a short-clip parity test upstream.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* docs(ced): list ced.cpp as a LocalAI-team engine + backend-guide directive

- README.md: add ced.cpp to the "native C/C++/GGML engines developed and
  maintained by the LocalAI project" table.
- docs/content/features/backends.md: add a Sound Classification backend
  category (sound-event classification / audio tagging) listing ced.cpp.
- .agents/adding-backends.md: add a "Documenting the backend" section and two
  verification-checklist items requiring new backends to be documented in the
  backends.md category list, and in-house native engines to be added to the
  README maintained-engines table. This directive was missing.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* chore(ced): repin CED_VERSION to the v0.1.0 release commit

ced.cpp history was squashed into a single release commit (tagged v0.1.0), so
the previous pin (99c6ed3) no longer exists upstream. Pin to c04ac14, the
v0.1.0 release commit, so the backend builds against a commit that exists.

Signed-off-by: Ettore Di Giacinto <mudler@localai.io>

* fix(ced): silence gosec G304/G103 + govet unsafeptr on audited paths

- sound_classification.go: os.Create(dst) where dst = temp dir + path.Base of
  the upload (no traversal). #nosec G304, matching the depth-anything-cpp handler.
- goced.go: reading a NUL-terminated C string from a libced-owned buffer.
  #nosec G103 (gosec) + //nolint:govet (golangci-lint's unsafeptr check), since
  the uintptr is a C-owned malloc'd buffer, not Go-GC memory.

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>
This commit is contained in:
LocalAI [bot]
2026-06-22 01:00:28 +02:00
committed by GitHub
parent ce8a3e9266
commit 600dafd20b
52 changed files with 2161 additions and 152 deletions
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docs/content/features/audio-classification.md Normal file
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@@ -0,0 +1,55 @@
+++
disableToc = false
title = "Sound Classification"
weight = 18
url = "/features/audio-classification/"
+++
Sound-event classification (audio tagging) answers the question **"what am I hearing?"** - given an audio clip, it returns a list of scored [AudioSet](https://research.google.com/audioset/) labels (e.g. *Baby cry, infant cry*, *Glass breaking*, *Dog bark*, *Alarm*).
LocalAI exposes this through the `/v1/audio/classification` endpoint, modelled after `/v1/audio/transcriptions`. The reference backend is **[ced.cpp](https://github.com/mudler/ced.cpp)** (CED, a 527-class AudioSet tagger), a small ViT over a log-mel spectrogram ported to ggml with full PyTorch parity. Apache-2.0 weights are redistributable as GGUF.
Because classification is exposed as a regular OpenAI-style endpoint, any HTTP client works - there is no Python dependency on the consumer side.
## Endpoint
```
POST /v1/audio/classification
Content-Type: multipart/form-data
```
| Field | Type | Description |
|-------|------|-------------|
| `file` | file (required) | audio file in any format `ffmpeg` accepts |
| `model` | string (required) | name of the sound-classification-capable model (e.g. `ced-base`) |
| `top_k` | int | number of top tags to return (0 = backend default) |
| `threshold` | float | drop tags scoring below this value |
### Response
```json
{
"model": "ced-base",
"detections": [
{"index": 23, "label": "Baby cry, infant cry", "score": 0.87},
{"index": 22, "label": "Crying, sobbing", "score": 0.41}
]
}
```
Detections are returned in score-descending order. Scores are per-class probabilities (multi-label, independent), so they do not sum to 1.
## Example
```bash
curl http://localhost:8080/v1/audio/classification \
-H "Content-Type: multipart/form-data" \
-F file="@/path/to/clip.wav" \
-F model="ced-base" \
-F top_k=10
```
## See also
- [Audio to Text]({{% relref "audio-to-text" %}}) - speech transcription
- [Speaker Diarization]({{% relref "audio-diarization" %}}) - who spoke when

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docs/content/features/audio-diarization.md
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@@ -152,3 +152,7 @@ curl http://localhost:8080/v1/audio/diarization \
- **Speaker identity across files**: speaker IDs (`SPEAKER_00`, `SPEAKER_01`, …) are local to each request. To track the same person across multiple recordings, combine `/v1/audio/diarization` with `/v1/voice/embed` (speaker embedding) and maintain your own embedding store.
- **Hints vs. forces**: `num_speakers` overrides clustering when set; `min_speakers` / `max_speakers` are advisory and only honored by backends that expose a range hint. vibevoice.cpp ignores them — its model picks the count itself.
- **Sample rate**: input is automatically converted to 16 kHz mono via ffmpeg before the backend sees it; sherpa-onnx pyannote-3.0 requires 16 kHz.
## See also
- [Sound Classification]({{% relref "audio-classification" %}}) - tag non-speech sound events (alarms, glass breaking, baby cry) in a clip.

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docs/content/features/backends.md
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@@ -128,6 +128,7 @@ LocalAI supports various types of backends:
- **Speech-to-Text Backends**: For transcription (e.g., whisper.cpp, parakeet.cpp, faster-whisper, NeMo)
- **Text-to-Speech Backends**: For speech synthesis (e.g., piper, Kokoro, VibeVoice, Qwen3-TTS)
- **Sound Generation Backends**: For music and audio generation (e.g., ACE-Step)
- **Sound Classification Backends**: For sound-event classification / audio tagging - identifying everyday sounds like baby cry, glass breaking, alarms (e.g., ced.cpp)
- **Image & Video Generation Backends**: For diffusion models (e.g., stable-diffusion.cpp, diffusers)
- **Vision & Detection Backends**: For object detection, segmentation, depth, and face/voice recognition (e.g., rf-detr.cpp, locate-anything.cpp, sam3.cpp, insightface)
- **Audio Processing Backends**: For voice activity detection and audio enhancement (e.g., Silero VAD, LocalVQE)

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docs/content/whats-new.md
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@@ -15,6 +15,7 @@ You can see the release notes [here](https://github.com/mudler/LocalAI/releases)
- **April 2026**: [Audio Transform](/features/audio-transform/) — generic audio-in / audio-out endpoint with optional reference signal. First implementation: [LocalVQE](https://github.com/localai-org/LocalVQE) C++ backend (joint AEC + noise suppression + dereverberation, DeepVQE-style). Both batch (`POST /audio/transformations`) and bidirectional WebSocket streaming (`/audio/transformations/stream`). Studio "Transform" tab with synchronized waveform players for input / reference / output.
- **April 2026**: [Face recognition backend](/features/face-recognition/) — `insightface`-powered 1:1 verification, 1:N identification, face embedding, face detection, and demographic analysis. Ships both a non-commercial `buffalo_l` model and an Apache 2.0 OpenCV Zoo alternative.
- **May 2026**: [Speaker diarization](/features/audio-diarization/) — new `/v1/audio/diarization` endpoint returning "who spoke when" segments. Backed by `sherpa-onnx` (pyannote-3.0 + speaker embeddings + clustering) for pure diarization, and `vibevoice-cpp` for diarization bundled with long-form ASR. Supports `json` / `verbose_json` / `rttm` response formats.
- **June 2026**: [Sound classification](/features/audio-classification/) — new `/v1/audio/classification` endpoint for audio tagging / sound-event classification, returning scored [AudioSet](https://research.google.com/audioset/) labels (baby cry, glass breaking, alarms, ...). Backed by [ced.cpp](https://github.com/mudler/ced.cpp), a 527-class AudioSet tagger ported to ggml.
- **June 2026**: [PII analyze / redact API](/features/middleware/#analyze--redact-api) — the PII detection pipeline (NER + restricted-regex pattern tiers) is now a standalone service: `POST /api/pii/analyze` returns detected entity spans and `POST /api/pii/redact` returns the sanitised text (or `400 pii_blocked`), without routing a chat request through the middleware. Events gain an `origin` (`middleware` / `proxy` / `pii_analyze` / `pii_redact`) so `/api/pii/events` can be filtered by source.
- **June 2026**: Concurrent scoring and PII NER on llama.cpp — the `Score` (router classifier) and `TokenClassify` (PII NER) primitives now ride llama.cpp's server task queue instead of locking the context, so they run concurrently with chat/completion/embedding traffic and with each other. The `known_usecases` restriction that forced dedicated scorer/NER model configs on llama-cpp is lifted, repeated scoring calls reuse the prompt KV cache across candidates, and scoring inputs are no longer capped by the physical batch size.