* refactor(transcription): propagate request ctx through ModelTranscription*
Replaces context.Background() with the HTTP request ctx so client
disconnects start cancelling the gRPC call. No backend-side abort wiring
yet — that comes in a later commit. Pure plumbing.
Assisted-by: Claude:claude-haiku-4-5
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(cli): pass ctx to backend.ModelTranscription
Follow-up to e65d3e1f which threaded ctx through ModelTranscription
but missed the CLI caller. CLI commands have no request-scoped ctx,
so context.Background() is correct here.
Assisted-by: Claude:claude-haiku-4-5
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactor(audio): propagate request ctx into TTS, sound-gen, audio-transform
Same ctx-plumbing pattern applied to the rest of the audio path. CLI
callers use context.Background() since there is no request scope; HTTP
callers use c.Request().Context().
Assisted-by: Claude:claude-haiku-4-5
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactor(backend): propagate request ctx into biometric, detection, rerank, diarization paths
Replaces remaining context.Background() sites in core/backend with the
caller's ctx. After this commit, every core/backend/*.go entry point
threads the request ctx end-to-end to the gRPC client.
Assisted-by: Claude:claude-haiku-4-5
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactor(grpc): plumb ctx through AIModel.AudioTranscription{,Stream}
Adds context.Context as first parameter to the AIModel interface methods
that wrap whisper-style transcription. Server-side gRPC handler now
forwards the per-RPC ctx (server-streaming uses stream.Context()).
Whisper, Voxtral, vibevoice-cpp, and sherpa-onnx accept the parameter;
none uses it yet — the actual cancellation primitive lands in the next
commit so this is pure plumbing.
Assisted-by: Claude:claude-sonnet-4-6
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(whisper): add abort_callback hook in the C++ bridge
Installs a std::atomic<int> flag, wires it into
whisper_full_params.abort_callback, and exposes a set_abort(int) C
symbol so Go can flip the flag from a goroutine watching the request
context. transcribe() now distinguishes abort (return 2) from real
whisper_full failure (return 1).
Assisted-by: Claude:claude-haiku-4-5
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(whisper): register set_abort symbol in the purego loader
Adds the Go-side binding for the new C export so the next commit can
call CppSetAbort(1) from a watcher goroutine on ctx.Done().
Assisted-by: Claude:claude-haiku-4-5
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(whisper): honor ctx cancellation and return codes.Canceled
A watcher goroutine watches ctx.Done() during AudioTranscription and
calls CppSetAbort(1) on cancel. whisper_full sees abort_callback return
true at the next compute graph step, returns non-zero, and the bridge
returns 2 -> AudioTranscription maps that to codes.Canceled.
Adds an opt-in test (gated on WHISPER_MODEL_PATH / WHISPER_AUDIO_PATH)
that asserts cancellation latency under 5s and proves the abort flag
resets cleanly so the next transcription succeeds.
Assisted-by: Claude:claude-sonnet-4-6
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(whisper): join the cancel watcher goroutine before returning
Follow-up to 85edf9d2. The previous commit used `defer close(done)` and
called the watcher "joined synchronously" — but close() only signals,
it does not block until the goroutine exits. That left a window where
a late CppSetAbort(1) from a cancelled call could land on the next
call, after its C-side g_abort reset but before whisper_full() began
polling the abort callback, corrupting the second transcription.
Switch to a sync.WaitGroup join so wg.Wait() blocks until the watcher
has actually returned from its select.
Assisted-by: Claude:claude-sonnet-4-6
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(whisper): short-circuit pre-cancelled ctx in AudioTranscription
If ctx is already Done() at entry, return codes.Canceled immediately
instead of running the full transcription. The C-side g_abort reset
happens at the start of transcribe() and would otherwise overwrite a
watcher-set abort flag from an already-cancelled ctx, producing a
spurious successful transcription on a request the client has already
abandoned.
Assisted-by: Claude:claude-haiku-4-5
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(tests/distributed): update testLLM mock for new AudioTranscription signature
Phase B (93c48e19) added context.Context to AIModel.AudioTranscription
but missed the testLLM mock in tests/e2e/distributed. CI golangci-lint
caught it: *testLLM did not implement grpc.AIModel because the method
signature lacked the ctx parameter, which broke the distributed test
suite compilation and cascaded through every backend-build job that
runs `go build ./...`.
Assisted-by: Claude:claude-opus-4-7
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* test(whisper): port cancellation test to Ginkgo/Gomega
Project policy (.agents/coding-style.md, enforced by golangci-lint
forbidigo) is that all Go tests must use Ginkgo v2 + Gomega — no
stdlib testing patterns (t.Skip, t.Fatalf, etc.). Convert the
cancellation test to a Describe/It block with Skip(...) for env
gating and Expect/HaveOccurred for assertions.
Same coverage: cancel mid-flight returns codes.Canceled within 5s and
a follow-up transcription succeeds, proving the C-side g_abort flag
resets cleanly.
Assisted-by: Claude:claude-opus-4-7
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>
* feat(voice-recognition): add /v1/voice/{verify,analyze,embed} + speaker-recognition backend
Audio analog to face recognition. Adds three gRPC RPCs
(VoiceVerify / VoiceAnalyze / VoiceEmbed), their Go service and HTTP
layers, a new FLAG_SPEAKER_RECOGNITION capability flag, and a Python
backend scaffold under backend/python/speaker-recognition/ wrapping
SpeechBrain ECAPA-TDNN with a parallel OnnxDirectEngine for
WeSpeaker / 3D-Speaker ONNX exports.
The kokoros Rust backend gets matching unimplemented trait stubs —
tonic's async_trait has no defaults, so adding an RPC without Rust
stubs breaks the build (same regression fixed by eb01c772 for face).
Swagger, /api/instructions, and the auth RouteFeatureRegistry /
APIFeatures list are updated so the endpoints surface everywhere a
client or admin UI looks.
Assisted-by: Claude:claude-opus-4-7
* feat(voice-recognition): add 1:N identify + register/forget endpoints
Mirrors the face-recognition register/identify/forget surface. New
package core/services/voicerecognition/ carries a Registry interface
and a local-store-backed implementation (same in-memory vector-store
plumbing facerecognition uses, separate instance so the embedding
spaces stay isolated).
Handlers under /v1/voice/{register,identify,forget} reuse
backend.VoiceEmbed to compute the probe vector, then delegate the
nearest-neighbour search to the registry. Default cosine-distance
threshold is tuned for ECAPA-TDNN on VoxCeleb (0.25, EER ~1.9%).
As with the face registry, the current backing is in-memory only — a
pgvector implementation is a future constructor-level swap.
Assisted-by: Claude:claude-opus-4-7
* feat(voice-recognition): gallery, docs, CI and e2e coverage
- backend/index.yaml: speaker-recognition backend entry + CPU and
CUDA-12 image variants (plus matching development variants).
- gallery/index.yaml: speechbrain-ecapa-tdnn (default) and
wespeaker-resnet34 model entries. The WeSpeaker SHA-256 is a
deliberate placeholder — the HF URI must be curl'd and its hash
filled in before the entry installs.
- docs/content/features/voice-recognition.md: API reference + quickstart,
mirrors the face-recognition docs.
- React UI: CAP_SPEAKER_RECOGNITION flag export (consumers follow face's
precedent — no dedicated tab yet).
- tests/e2e-backends: voice_embed / voice_verify / voice_analyze specs.
Helper resolveFaceFixture is reused as-is — the only thing face/voice
share is "download a file into workDir", so no need for a new helper.
- Makefile: docker-build-speaker-recognition + test-extra-backend-
speaker-recognition-{ecapa,all} targets. Audio fixtures default to
VCTK p225/p226 samples from HuggingFace.
- CI: test-extra.yml grows a tests-speaker-recognition-grpc job
mirroring insightface. backend.yml matrix gains CPU + CUDA-12 image
build entries — scripts/changed-backends.js auto-picks these up.
Assisted-by: Claude:claude-opus-4-7
* feat(voice-recognition): wire a working /v1/voice/analyze head
Adds AnalysisHead: a lazy-loading age / gender / emotion inference
wrapper that plugs into both SpeechBrainEngine and OnnxDirectEngine.
Defaults to two open-licence HuggingFace checkpoints:
- audeering/wav2vec2-large-robust-24-ft-age-gender (Apache 2.0) —
age regression + 3-way gender (female / male / child).
- superb/wav2vec2-base-superb-er (Apache 2.0) — 4-way emotion.
Both are optional and degrade gracefully when transformers or the
model can't be loaded — the engine raises NotImplementedError so the
gRPC layer returns 501 instead of a generic 500.
Emotion classes pass through from the model (neutral/happy/angry/sad
on the default checkpoint); the e2e test now accepts any non-empty
dominant gender so custom age_gender_model overrides don't fail it.
Adds transformers to the backend's CPU and CUDA-12 requirements.
Assisted-by: Claude:claude-opus-4-7
* fix(voice-recognition): pin real WeSpeaker ResNet34 ONNX SHA-256
Replaces the placeholder hash in gallery/index.yaml with the actual
SHA-256 (7bb2f06e…) of the upstream
Wespeaker/wespeaker-voxceleb-resnet34-LM ONNX at ~25MB. `local-ai
models install wespeaker-resnet34` now succeeds.
Assisted-by: Claude:claude-opus-4-7
* fix(voice-recognition): soundfile loader + honest analyze default
Two issues surfaced on first end-to-end smoke with the actual backend
image:
1. torchaudio.load in torchaudio 2.8+ requires the torchcodec package
for audio decoding. Switch SpeechBrainEngine._load_waveform to the
already-present soundfile (listed in requirements.txt) plus a numpy
linear resample to 16kHz. Drops a heavy ffmpeg-linked dep and the
codepath we never exercise (torchaudio's ffmpeg backend).
2. The AnalysisHead was defaulting to audeering/wav2vec2-large-robust-
24-ft-age-gender, but AutoModelForAudioClassification silently
mangles that checkpoint — it reports the age head weights as
UNEXPECTED and re-initialises the classifier head with random
values, so the "gender" output is noise and there is no age output
at all. Make age/gender opt-in instead (empty default; users wire
a cleanly-loadable Wav2Vec2ForSequenceClassification checkpoint via
age_gender_model: option). Emotion keeps its working Superb default.
Also broaden _infer_age_gender's tensor-shape handling and catch
runtime exceptions so a dodgy age/gender head never takes down the
whole analyze call.
Docs and README updated to match the new policy.
Verified with the branch-scoped gallery on localhost:
- voice/embed → 192-d ECAPA-TDNN vector
- voice/verify → same-clip dist≈6e-08 verified=true; cross-speaker
dist 0.76–0.99 verified=false (as expected)
- voice/register/identify/forget → round-trip works, 404 on unknown id
- voice/analyze → emotion populated, age/gender omitted (opt-in)
Assisted-by: Claude:claude-opus-4-7
* fix(voice-recognition): real CI audio fixtures + fixture-agnostic verify spec
Two issues surfaced after CI actually ran the speaker-recognition e2e
target (I'd curl-tested against a running server but hadn't run the
make target locally):
1. The default BACKEND_TEST_VOICE_AUDIO_* URLs pointed at
huggingface.co/datasets/CSTR-Edinburgh/vctk paths that return 404
(the dataset is gated). Swap them for the speechbrain test samples
served from github.com/speechbrain/speechbrain/raw/develop/ —
public, no auth, correct 16kHz mono format.
2. The VoiceVerify spec required d(file1,file2) < 0.4, assuming
file1/file2 were same-speaker. The speechbrain samples are three
different speakers (example1/2/5), and there is no easy un-gated
source of true same-speaker audio pairs (VoxCeleb/VCTK/LibriSpeech
are all license- or size-gated for CI use). Replace the ceiling
check with a relative-ordering assertion: d(pair) > d(same-clip)
for both file2 and file3 — that's enough to prove the embeddings
encode speaker info, and it works with any three non-identical
clips. Actual speaker ordering d(1,2) vs d(1,3) is logged but not
asserted.
Local run: 4/4 voice specs pass (Health, LoadModel, VoiceEmbed,
VoiceVerify) on the built backend image. 12 non-voice specs skipped
as expected.
Assisted-by: Claude:claude-opus-4-7
* fix(ci): checkout with submodules in the reusable backend_build workflow
The kokoros Rust backend build fails with
failed to read .../sources/Kokoros/kokoros/Cargo.toml: No such file
because the reusable backend_build.yml workflow's actions/checkout
step was missing `submodules: true`. Dockerfile.rust does `COPY .
/LocalAI`, and without the submodule files the subsequent `cargo
build` can't find the vendored Kokoros crate.
The bug pre-dates this PR — scripts/changed-backends.js only triggers
the kokoros image job when something under backend/rust/kokoros or
the shared proto changes, so master had been coasting past it. The
voice-recognition proto addition re-broke it.
Other checkouts in backend.yml (llama-cpp-darwin) and test-extra.yml
(insightface, kokoros, speaker-recognition) already pass
`submodules: true`; this brings the shared backend image builder in
line.
Assisted-by: Claude:claude-opus-4-7
LocalAI [bot]