* test(whisper): wire e2e streaming transcription target
Adds test-extra-backend-whisper-transcription, mirroring the existing
llama-cpp / sherpa-onnx / vibevoice-cpp targets. The generic
AudioTranscriptionStream spec at tests/e2e-backends/backend_test.go:644
fails today because backend/go/whisper has no streaming impl - this
target is the failing TDD gate that the next phase makes pass.
Confirmed RED locally: 3 Passed (health, load, offline transcription),
1 Failed (streaming spec hits its 300s context deadline because the
base implementation returns 'unimplemented' but doesn't close the
result channel, leaving the gRPC stream open until the client times
out).
Assisted-by: Claude:claude-opus-4-7
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(whisper-cpp): expose new_segment_callback to the Go side
Adds set_new_segment_callback() and a C-side trampoline that whisper.cpp
invokes once per new text segment during whisper_full(). The trampoline
dispatches (idx_first, n_new, user_data) to a Go function pointer
registered via purego.NewCallback - text and timings are pulled by Go
through the existing get_segment_text/get_segment_t0/get_segment_t1
getters.
Wires the hook only when streaming is actually requested, to avoid a
per-segment function-pointer dispatch on the offline path.
Assisted-by: Claude:claude-opus-4-7
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* feat(whisper-cpp): implement AudioTranscriptionStream
Wires whisper.cpp's new_segment_callback through purego back to Go so
the streaming transcription RPC produces real, time-correlated deltas
while whisper_full() is still decoding. Each segment becomes one
TranscriptStreamResponse{Delta}; whisper_full's return is the
TranscriptStreamResponse{FinalResult} carrying the full segment list,
language, and duration.
Per-call state is tracked in a sync.Map keyed by an atomic counter; the
Go callback registered via purego.NewCallback is a singleton, dispatched
through user_data. SingleThread today means only one entry is ever live,
but the map shape matches the sherpa-onnx TTS callback pattern.
The streaming path's final.Text is the literal concat of every emitted
delta (a strings.Builder accumulated by onNewSegment) so the e2e
invariant `final.Text == concat(deltas)` holds exactly. The first delta
has no leading space; subsequent deltas are space-prefixed. The offline
AudioTranscription path is unchanged.
Closes the gap with sherpa-onnx, vibevoice-cpp, llama-cpp, and tinygrad,
which already implement AudioTranscriptionStream.
Verified GREEN locally: make test-extra-backend-whisper-transcription
passes 4/4 specs (3 Passed initially under RED, +1 streaming spec now).
Assisted-by: Claude:claude-opus-4-7
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* test(whisper-cpp): assert progressive multi-segment streaming
Drives AudioTranscriptionStream against a real long-audio fixture and
asserts len(deltas) >= 2. The generic e2e spec at
tests/e2e-backends/backend_test.go:644 only checks len(deltas) >= 1
which is satisfied by both real and faked streaming - this spec is the
guardrail that a future "fake" impl can't sneak past.
Skipped by default (env-gated, like the cancellation spec); set
WHISPER_LIBRARY, WHISPER_MODEL_PATH, and WHISPER_AUDIO_PATH to a 30+
second clip to run.
Verified locally with a 55s 5x-JFK concat against ggml-base.en.bin:
1 Passed in 7.3s, deltas >= 2, finalSegmentCount >= 2,
concat(deltas) == final.Text.
Assisted-by: Claude:claude-opus-4-7
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* ci(whisper-cpp): add transcription gRPC e2e job
Mirrors tests-sherpa-onnx-grpc-transcription /
tests-llama-cpp-grpc-transcription. Runs make
test-extra-backend-whisper-transcription whenever the whisper backend
or the run-all switch fires, so a pin-bump or refactor that breaks
streaming transcription gets caught before merge.
The whisper output on detect-changes is already emitted by
scripts/changed-backends.js (it iterates allBackendPaths); this PR
just exposes it as a workflow output and consumes it.
Assisted-by: Claude:claude-opus-4-7
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(whisper-cpp): silence errcheck on AudioTranscriptionStream defers
golangci-lint runs with new-from-merge-base=origin/master, so the
identical defer patterns in the existing offline AudioTranscription
path are grandfathered while the new ones in AudioTranscriptionStream
trip errcheck. Wrap both defers in `func() { _ = ... }()` to match what
errcheck wants without altering behavior. The errors from os.RemoveAll
and *os.File.Close are not actionable inside a defer here (we're
already returning), matching the offline path's contract.
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>
* 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>
* 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>
* feat(api): add /v1/audio/diarization endpoint with sherpa-onnx + vibevoice.cpp
Closes#1648.
OpenAI-style multipart endpoint that returns "who spoke when". Single
endpoint instead of the issue's three-endpoint sketch (refactor /vad,
/vad/embedding, /diarization) — the typical client wants one call, and
embeddings can land later as a sibling without breaking this surface.
Response shape borrows from Pyannote/Deepgram: segments carry a
normalised SPEAKER_NN id (zero-padded, stable across the response) plus
the raw backend label, optional per-segment text when the backend bundles
ASR, and a speakers summary in verbose_json. response_format also accepts
rttm so consumers can pipe straight into pyannote.metrics / dscore.
Backends:
* vibevoice-cpp — Diarize() reuses the existing vv_capi_asr pass.
vibevoice's ASR prompt asks the model to emit
[{Start,End,Speaker,Content}] natively, so diarization is a by-product
of the same pass; include_text=true preserves the transcript per
segment, otherwise we drop it.
* sherpa-onnx — wraps the upstream SherpaOnnxOfflineSpeakerDiarization
C API (pyannote segmentation + speaker-embedding extractor + fast
clustering). libsherpa-shim grew config builders, a SetClustering
wrapper for per-call num_clusters/threshold overrides, and a
segment_at accessor (purego can't read field arrays out of
SherpaOnnxOfflineSpeakerDiarizationSegment[] directly).
Plumbing: new Diarize gRPC RPC + DiarizeRequest / DiarizeSegment /
DiarizeResponse messages, threaded through interface.go, base, server,
client, embed. Default Base impl returns unimplemented.
Capability surfaces all updated: FLAG_DIARIZATION usecase,
FeatureAudioDiarization permission (default-on), RouteFeatureRegistry
entries for /v1/audio/diarization and /audio/diarization, audio
instruction-def description widened, CAP_DIARIZATION JS symbol,
swagger regenerated, /api/instructions discovery map updated.
Tests:
* core/backend: speaker-label normalisation (first-seen → SPEAKER_NN,
per-speaker totals, nil-safety, fallback to backend NumSpeakers when
no segments).
* core/http/endpoints/openai: RTTM rendering (file-id basename, negative
duration clamping, fallback id).
* tests/e2e: mock-backend grew a deterministic Diarize that emits
raw labels "5","2","5" so the e2e suite verifies SPEAKER_NN
remapping, verbose_json speakers summary + transcript pass-through
(gated by include_text), RTTM bytes content-type, and rejection of
unknown response_format. mock-diarize model config registered with
known_usecases=[FLAG_DIARIZATION] to bypass the backend-name guard.
Docs: new features/audio-diarization.md (request/response, RTTM example,
sherpa-onnx + vibevoice setup), cross-link from audio-to-text.md, entry
in whats-new.md.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
Assisted-by: Claude:claude-opus-4-7 [Claude Code]
* fix(diarization): correct sherpa-onnx symbol name + lint cleanup
CI failures on #9654:
* sherpa-onnx-grpc-{tts,transcription} and sherpa-onnx-realtime panicked
at backend startup with `undefined symbol: SherpaOnnxDestroyOfflineSpeakerDiarizationResult`.
Upstream's actual symbol is SherpaOnnxOfflineSpeakerDiarizationDestroyResult
(Destroy in the middle, not the prefix); the rest of the diarization
surface follows the same naming pattern. The mismatched name made
purego.RegisterLibFunc fail at dlopen time and crashed the gRPC server
before the BeforeAll could probe Health, taking down every sherpa-onnx
test job — not just the diarization-related ones.
* golangci-lint flagged 5 errcheck violations on new defer cleanups
(os.RemoveAll / Close / conn.Close); wrap each in a `defer func() { _ = X() }()`
closure (matches the pattern other LocalAI files use for new code, since
pre-existing bare defers are grandfathered in via new-from-merge-base).
* golangci-lint also flagged forbidigo violations: the new
diarization_test.go files used testing.T-style `t.Errorf` / `t.Fatalf`,
which are forbidden by the project's coding-style policy
(.agents/coding-style.md). Convert both files to Ginkgo/Gomega
Describe/It with Expect(...) — they get picked up by the existing
TestBackend / TestOpenAI suites, no new suite plumbing needed.
* modernize linter: tightened the diarization segment loop to
`for i := range int(numSegments)` (Go 1.22+ idiom).
Verified locally: golangci-lint with new-from-merge-base=origin/master
reports 0 issues across all touched packages, and the four mocked
diarization e2e specs in tests/e2e/mock_backend_test.go still pass.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
Assisted-by: Claude:claude-opus-4-7 [Claude Code]
* fix(vibevoice-cpp): convert non-WAV input via ffmpeg + raise ASR token budget
Confirmed end-to-end against a real LocalAI instance with vibevoice-asr-q4_k
loaded and the multi-speaker MP3 sample at vibevoice.cpp/samples/2p_argument.mp3:
both /v1/audio/transcriptions and /v1/audio/diarization now succeed and
return correctly attributed speaker turns for the full clip.
Two latent issues surfaced once the diarization endpoint actually exercised
the backend with a non-trivial input:
1. vv_capi_asr only accepts WAV via load_wav_24k_mono. The previous code
passed the uploaded path straight through, so anything that wasn't
already a 24 kHz mono s16le WAV failed at the C side with rc=-8 and
the very unhelpful "vv_capi_asr failed". prepareWavInput shells out
to ffmpeg ("-ar 24000 -ac 1 -acodec pcm_s16le") in a per-call temp
dir, matching the rate the model was trained on; both AudioTranscription
and Diarize now route through it. This is the same shape sherpa-onnx
uses (utils.AudioToWav), but vibevoice needs 24 kHz rather than 16 kHz
so we don't reuse that helper.
2. The C ABI's max_new_tokens defaults to 256 when 0 is passed. That's
fine for a five-second clip but not for anything past ~10 s — vibevoice
stops mid-JSON, the parse fails, and the caller sees a hard error.
Pass a much larger budget (16 384 ≈ ~9 minutes of speech at the
model's ~30 tok/s rate); generation stops at EOS so this is a cap
rather than a target.
3. As a defensive belt-and-braces, mirror AudioTranscription's existing
"fall back to a single segment if the model emits non-JSON text"
pattern in Diarize, so partial / unusual model output never produces
a 500. This kept the endpoint usable while diagnosing (1) and (2),
and is the right behaviour to keep.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
Assisted-by: Claude:claude-opus-4-7 [Claude Code]
* fix(vibevoice-cpp): pass valid WAVs through directly so ffmpeg is not required at runtime
Spotted by tests-e2e-backend (1.25.x): the previous fix forced every
incoming audio file through `ffmpeg -ar 24000 ...`, which meant the
backend container — which does not ship ffmpeg — failed even for the
existing happy path where the caller already uploads a WAV. The
container-side error was:
rpc error: code = Unknown desc = vibevoice-cpp: ffmpeg convert to
24k mono wav: exec: "ffmpeg": executable file not found in $PATH
Reading vibevoice.cpp's audio_io.cpp, `load_wav_24k_mono` uses drwav and
already accepts any PCM/IEEE-float WAV at any sample rate, downmixes
multi-channel input to mono, and resamples to 24 kHz internally. So the
only inputs that genuinely need an external converter are non-WAV
formats (MP3, OGG, FLAC, ...).
Detect WAVs by RIFF/WAVE magic at bytes 0..3 / 8..11 and pass them
straight through with a no-op cleanup; everything else still goes
through ffmpeg with the same 24 kHz mono s16le target. The result:
* Container builds without ffmpeg keep working for WAV uploads
(the e2e-backends fixture is jfk.wav at 16 kHz mono s16le).
* MP3 and other non-WAV inputs still get the new ffmpeg conversion
path so the diarization endpoint stays useful.
* If the caller uploads a non-WAV but ffmpeg isn't on PATH, the
surfaced error is still descriptive enough to act on.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
Assisted-by: Claude:claude-opus-4-7 [Claude Code]
* fix(ci): make gcc-14 install in Dockerfile.golang best-effort for jammy bases
The LocalVQE PR (bb033b16) made `gcc-14 g++-14` an unconditional apt
install in backend/Dockerfile.golang and pointed update-alternatives at
them. That works on the default `BASE_IMAGE=ubuntu:24.04` (noble has
gcc-14 in main), but every Go backend that builds on
`nvcr.io/nvidia/l4t-jetpack:r36.4.0` — jammy under the hood — now fails
at the apt step:
E: Unable to locate package gcc-14
This blocked unrelated jobs:
backend-jobs(*-nvidia-l4t-arm64-{stablediffusion-ggml, sam3-cpp, whisper,
acestep-cpp, qwen3-tts-cpp, vibevoice-cpp}). LocalVQE itself is only
matrix-built on ubuntu:24.04 (CPU + Vulkan), so it doesn't actually
need gcc-14 anywhere else.
Make the gcc-14 install conditional on the package being available in
the configured apt repos. On noble: identical behaviour to today (gcc-14
installed, update-alternatives points at it). On jammy: skip the
gcc-14 stanza entirely and let build-essential's default gcc take over,
which is what the other Go backends compile with anyway.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
Assisted-by: Claude:claude-opus-4-7 [Claude Code]
---------
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
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>
* feat(vibevoice-cpp): add purego TTS+ASR backend
Wire up Microsoft VibeVoice via the vibevoice.cpp C ABI as a new
purego-based Go backend that serves both Backend.TTS and
Backend.AudioTranscription from a single gRPC binary. Mirrors the
qwen3-tts-cpp / sherpa-onnx pattern so the variant matrix
(cpu/cuda12/cuda13/metal/rocm/sycl-f16/f32/vulkan/l4t) and the
e2e-backends gRPC harness reuse existing infrastructure.
- backend/go/vibevoice-cpp/ - Makefile, CMakeLists, purego shim, gRPC
Backend with model-dir auto-detection, closed-loop TTS->ASR smoke test
- backend/index.yaml - &vibevoicecpp meta + 18 image entries
- Makefile - .NOTPARALLEL, BACKEND_VIBEVOICE_CPP, docker-build wiring,
test-extra-backend-vibevoice-cpp-{tts,transcription} e2e wrappers
- .github/workflows/backend.yml - matrix entries for all variants
- .github/workflows/test-extra.yml - per-backend smoke + 2 gRPC e2e jobs
* feat(vibevoice-cpp): drop hardcoded glob detection, add gallery entries
Refactor backend Load() to follow the standard Options[] convention
used by sherpa-onnx and the rest of the multi-role backends:
ModelFile is the primary gguf, supplementary paths come through
opts.Options[] as key=value (or key:value for Make-target compat),
resolved against opts.ModelPath. type=asr/tts decides the role of
ModelFile when neither tts_model nor asr_model is set explicitly.
Add gallery/index.yaml entries:
- vibevoice-cpp - realtime 0.5B Q8_0 TTS + tokenizer + Carter voice
- vibevoice-cpp-asr - long-form ASR Q8_0 + tokenizer
Both pull from huggingface://mudler/vibevoice.cpp-models with sha256
verification. parameters.model + Options[] paths are siblings under
{models_dir} per the qwen3-tts-cpp convention.
Update Makefile e2e wrappers to pass BACKEND_TEST_OPTIONS comma+colon
style, and tighten the per-backend Go closed-loop test to use the
explicit Options API.
* fix(vibevoice-cpp): force whole-archive link so vv_capi_* exports survive
libvibevoice is a STATIC archive linked into the MODULE library.
Without --whole-archive (or -force_load on Apple, /WHOLEARCHIVE on
MSVC), the linker garbage-collects symbols not referenced from this
translation unit - which means dlopen+RegisterLibFunc panics with
'undefined symbol: vv_capi_load' at backend startup, since purego
looks them up by name and our cpp/govibevoicecpp.cpp doesn't call
them directly.
* test(vibevoice-cpp): rewrite suite with Ginkgo v2
Match the convention used by backend/go/sherpa-onnx/backend_test.go.
The suite now covers backend semantics that don't need purego (Locking,
empty-ModelFile rejection, TTS/ASR-without-loaded-model errors) on top
of the gRPC lifecycle specs (Health, Load, closed-loop TTS->ASR).
Model-dependent specs Skip() when VIBEVOICE_MODEL_DIR is unset, so
`go test ./backend/go/vibevoice-cpp/` is green on a clean checkout
and runs the heavyweight closed-loop spec when test.sh has staged
the bundle.
* fix(vibevoice-cpp): implement TTSStream + AudioTranscriptionStream
The gRPC server's stream handlers (pkg/grpc/server.go) spawn a
goroutine that ranges over a chan; the only thing closing that chan
is the backend's own *Stream method. With the default Base stub
returning 'unimplemented' and never touching the chan, the server
goroutine hangs forever and the client hits DeadlineExceeded - which
is exactly what the e2e harness saw in the test-extra-backend-vibevoice-cpp-tts
matrix run.
TTSStream synthesizes via vv_capi_tts to a tempfile, then emits a
streaming WAV header (chunk sizes 0xFFFFFFFF so HTTP clients can
start playback before the full PCM lands) followed by the PCM body
in 64 KB slices. The header + >=2 PCM frames satisfy the harness's
'expected >=2 chunks' assertion and give a real progressive stream.
AudioTranscriptionStream runs the offline transcription, emits each
segment as a delta, and closes with a final_result whose Text equals
the concatenated deltas (the harness asserts those match).
Two new Ginkgo specs guard the close-channel-on-error path so the
deadline-exceeded regression can't come back silently.
* fix(vibevoice-cpp): silence errcheck on cleanup paths
Lint flagged six unchecked Close()/Remove()/RemoveAll() calls along
purely-cleanup deferred paths. Wrap each in '_ = ...' (or a closure
for defers that take args) - matches what the rest of the LocalAI
backend/go/* tree already does for these callsites.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(vibevoice-cpp): closed-loop slot fill + modelRoot-relative path resolution
Two bugs the test-extra-backend-vibevoice-cpp-* CI matrix surfaced:
1. Closed-loop Load with ModelFile=tts.gguf + Options[asr_model=...] left
v.ttsModel empty, because the default-fill block only ran when BOTH
slots were empty. vv_capi_load then got tts="" + a voice and the
C side rejected it with rc=-3 'TTS model required to load a voice'.
Fix: ModelFile fills the *primary* role-slot (decided by 'type=' in
Options, defaulting to tts) independently of the secondary, so
ModelFile + asr_model resolves to both.
2. resolvePath stat'd CWD before falling back to relTo. With LocalAI
launched from a directory that happens to contain a same-named
file, supplementary Options[] paths could leak away from the
models dir. Drop the CWD probe entirely - relative paths now
*always* join onto opts.ModelPath (the gallery convention).
New Ginkgo coverage:
* 'ModelFile slot resolution' (4 specs) - asr_model+ModelFile, type=asr,
explicit tts_model override, key:value variant.
* 'resolvePath (relative-to-modelRoot)' (5 specs) - join, abs passthrough,
empty input, empty relTo, and the CWD-trap regression test.
* 'Load resolves relative Options paths against opts.ModelPath' - end-
to-end gallery layout round-trip.
Verified locally: 19/19 specs pass (with model bundle, including the
closed-loop TTS->ASR; without bundle, 17 pass + 2 model-dependent skip).
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* test(vibevoice-cpp): use gallery convention in closed-loop spec
The 'loads the realtime TTS model' / closed-loop specs were passing
already-prefixed paths into Options[]:
Options: ['tokenizer=' + filepath.Join(modelDir, 'tokenizer.gguf')]
Combined with no ModelPath set on the request, the backend's
modelRoot fell back to filepath.Dir(ModelFile) = modelDir, then
resolvePath joined the prefixed Options path on top of it -
producing 'vibevoice-models/vibevoice-models/tokenizer.gguf' when
the CI's VIBEVOICE_MODEL_DIR is the relative './vibevoice-models'.
The fix is to mirror the gallery contract LocalAI core actually
sends in production: ModelPath is the models root (absolute),
ModelFile is a name *under* it, every Options[] path is relative
to ModelPath. Uses filepath.Base() to get bare filenames.
Verified locally with both VIBEVOICE_MODEL_DIR=/tmp/vv-bundle (abs)
and VIBEVOICE_MODEL_DIR=vibevoice-models (the relative shape that
broke CI). Both: 19/19 specs pass, ~55-60s.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* ci(vibevoice-cpp): switch ASR to Q4_K + bump transcription timeout
The Q8_0 ASR gguf is ~14 GB - too big to fit alongside the runner
image, the docker build cache, and the test artifacts on a free
ubuntu-latest GHA runner; 'test-extra-backend-vibevoice-cpp-transcription'
was getting SIGTERM'd at 90 min before the model could finish loading.
Switch to Q4_K (~10 GB on disk, slightly faster CPU decode) for:
* the e2e harness Make target
* the gallery 'vibevoice-cpp-asr' entry (parameters + files block)
* the per-backend test.sh auto-download list
Bump tests-vibevoice-cpp-grpc-transcription's timeout-minutes from
90 to 150 - even with Q4_K, the 30 s JFK clip on a CPU runner needs
runway above the previous 90 min cap.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* ci(vibevoice-cpp): drop transcription gRPC e2e job - too heavy for free runners
The vibevoice ASR is a 7B-parameter model. Even on Q4_K (~10 GB on
disk) a single 30 s transcription saturates the per-test 30 min
timeout in the e2e-backends harness on a 4-core ubuntu-latest, and
the 10 GB download + Docker layer + working space leaves no headroom
on the runner's free disk. Two attempts in CI got SIGTERM'd at the
LoadModel boundary - the bottleneck isn't tunable from the workflow
side without a paid-tier runner.
The per-backend tests-vibevoice-cpp job already runs the same
AudioTranscription path via a closed-loop TTS->ASR Ginkgo spec - same
gRPC contract, same model, single process - so the standalone
tests-vibevoice-cpp-grpc-transcription job was redundant on top of
the disk/CPU pressure.
The Makefile target test-extra-backend-vibevoice-cpp-transcription
stays for local invocation on workstations that can afford it -
useful when developing the streaming codepaths.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* ci(vibevoice-cpp): restore transcription gRPC e2e on bigger-runner
Switch tests-vibevoice-cpp-grpc-transcription from ubuntu-latest to
the self-hosted 'bigger-runner' label that GPU image builds in
backend.yml use, plus the documented Free-disk-space prep step (purge
dotnet / ghc / android / CodeQL caches) the disabled vllm/sglang
entries in this file describe. That gives the 7B-param Q4_K ASR
model the disk + CPU runway it needs.
Keep timeout-minutes: 150 - even on a beefier runner the 30 s JFK
decode plus 10 GB download has to fit comfortably.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* ci(vibevoice-cpp): apt-get install make on bigger-runner before transcription e2e
bigger-runner is a self-hosted bare runner without the standard
ubuntu image's preinstalled build tools, so the previous job died at
the very first command with 'make: command not found' (exit 127).
Add the Dependencies step that the disabled vllm/sglang entries in
this file already document - apt-get installs make + build-essential
+ curl + unzip + ca-certificates + git + tar before the make target
runs. Mirrors how every other 'runs-on: bigger-runner' entry in
backend.yml prepares the runner.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
---------
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* chore: add golangci-lint with new-from-merge-base baseline
Configure golangci-lint v2 with the standard linter set (errcheck, govet,
ineffassign, unused) plus forbidigo, which enforces the Ginkgo/Gomega-only
test convention from .agents/coding-style.md by rejecting stdlib testing
calls (t.Errorf, t.Fatalf, t.Run, ...). staticcheck is disabled — the
codebase has many pre-existing QF-style suggestions not worth gating on.
issues.new-from-merge-base = master makes the lint job a gate for new
issues only; the ~1300 pre-existing baseline stays visible via
'make lint-all' for incremental cleanup. CI runs 'make lint'.
Backends needing C/C++ headers we don't install in the lint runner are
excluded via a deny list in the Makefile (backend/go/{piper,silero-vad,
llm}, cmd/launcher). Discovery still flows through 'go list ./...', so
new packages are scanned automatically.
To make backend/go/{sam3-cpp,stablediffusion-ggml,whisper} typecheckable,
move their .cpp/.h sources into cpp/ subdirs (matching qwen3-tts-cpp /
acestep-cpp). Without this 'go list' rejects the package because Go does
not allow .cpp alongside .go without cgo.
Fix two real bugs found by lint in tests/integration/ (run only via
'make test-stores', not default CI): a stale zerolog reference left over
from the slog migration (c37785b7) and an unused 'os' import.
Assisted-by: Claude Code:Opus 4.7 (1M) [Bash] [Read] [Edit] [Write]
Signed-off-by: Richard Palethorpe <io@richiejp.com>
* ci(lint): generate proto sources and fetch full history
The lint job was failing for two reasons:
- pkg/grpc/proto/*.go is generated, not checked in. Several packages
import it, so without 'make protogen-go' typecheck fails project-wide
with "no required module provides package github.com/mudler/LocalAI/
pkg/grpc/proto".
- golangci-lint's new-from-merge-base needs to git-merge-base the PR
against master, but actions/checkout's default shallow clone doesn't
fetch master. fetch-depth: 0 brings full history; the config now
references origin/master (the remote-tracking branch that survives
the shallow checkout) instead of bare master (which doesn't exist
locally after checkout).
Assisted-by: Claude Code:Opus 4.7 (1M) [Bash] [Read] [Edit] [Write]
Signed-off-by: Richard Palethorpe <io@richiejp.com>
* ci(lint): stub react-ui/dist for go:embed glob
core/http/app.go has //go:embed react-ui/dist/*. The glob must match at
least one non-hidden entry or typecheck fails the whole core/http
package. We don't need the real React bundle to lint Go code, so just
touch an empty index.html to satisfy the embed.
Assisted-by: Claude Code:Opus 4.7 (1M) [Bash] [Read] [Edit] [Write]
Signed-off-by: Richard Palethorpe <io@richiejp.com>
---------
Signed-off-by: Richard Palethorpe <io@richiejp.com>
whisper.cpp can emit bytes that are not valid UTF-8 — typically a
multibyte codepoint split across token boundaries. protobuf string
fields reject those at marshal time, which would surface as a transcribe
failure. Run strings.ToValidUTF8 on the segment text before it leaves
the cgo boundary so the bad byte gets replaced with U+FFFD.
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
Assisted-by: Claude:claude-opus-4-7 [Claude Code]
feat(backend): Add Sherpa ONNX backend and Omnilingual ASR
Adds a new Go backend wrapping sherpa-onnx via purego (no cgo). Same
approach as opus/stablediffusion-ggml/whisper — a thin C shim
(csrc/shim.c + shim.h → libsherpa-shim.so) wraps the bits purego
can't reach directly: nested struct config writes, result-struct field
reads, and the streaming TTS callback trampoline. The Go side uses
opaque uintptr handles and purego.NewCallback for the TTS callback.
Supports:
- VAD via sherpa-onnx's Silero VAD
- Offline ASR: Whisper, Paraformer, SenseVoice, Omnilingual CTC
- Online/streaming ASR: zipformer transducer with endpoint detection
(AudioTranscriptionStream emits delta events during decode)
- Offline TTS: VITS (LJS, etc.)
- Streaming TTS: sherpa-onnx's callback API → PCM chunks on a channel,
prefixed by a streaming WAV header
Gallery entries: omnilingual-0.3b-ctc-q8-sherpa (1600-language offline
ASR), streaming-zipformer-en-sherpa (low-latency streaming ASR),
silero-vad-sherpa, vits-ljs-sherpa.
E2E coverage: tests/e2e-backends for offline + streaming ASR,
tests/e2e for the full realtime pipeline (VAD + STT + TTS).
Assisted-by: claude-opus-4-7-1M [Claude Code]
Signed-off-by: Richard Palethorpe <io@richiejp.com>
* feat(react-ui): add Face & Voice Recognition pages
Expose the face and voice biometrics endpoints
(/v1/face/*, /v1/voice/*) through the React UI. Each page has four
tabs driving the six endpoints per modality: Analyze (demographics
with bounding boxes / waveform segments), Compare (verify with a
match gauge and live threshold slider), Enrollment (register /
identify / forget with a top-K matches view), Embedding (raw
vector inspector with sparkline + copy).
MediaInput supports file upload plus live capture: webcam
snap-to-canvas for face, MediaRecorder -> AudioContext ->
16-bit PCM mono WAV transcode for voice (libsndfile on the
backend only handles WAV/FLAC/OGG natively).
Sidebar gets a new Biometrics section feature-gated on
face_recognition / voice_recognition; routes are wrapped in
<RequireFeature>. No new dependencies -- Font Awesome icons
picked from the Free set.
Assisted-by: Claude:Opus 4.7
* fix(localai): accept data URI prefixes with codec/charset params
Browser MediaRecorder produces data URIs like
data:audio/webm;codecs=opus;base64,...
so the pre-';base64,' section can carry multiple parameter
segments. The `^data:([^;]+);base64,` regex in pkg/utils/base64.go
and core/http/endpoints/localai/audio.go only matched exactly one
segment, so recordings straight from the React UI's live-capture
tab failed the strip and then tripped the base64 decoder on the
leading 'data:' literal, surfacing as
"invalid audio base64: illegal base64 data at input byte 4"
Widened both regexes to `^data:[^,]+?;base64,` so any number of
';param=value' segments between the mime type and ';base64,' are
tolerated. Added a regression test covering the MediaRecorder
shape.
Assisted-by: Claude:Opus 4.7
* fix(insightface): scope pack ONNX loading to known manifests
LocalAI's gallery extracts buffalo_* zips flat into the models
directory, which inevitably mixes with ONNX files from other
backends (opencv face engine, MiniFASNet antispoof, WeSpeaker
voice embedding) and older buffalo pack installs. Feeding those
foreign files into insightface's model_zoo.get_model() blows up
inside the router -- it assumes a 4-D NCHW input and indexes
`input_shape[2]` on tensors that aren't shaped like a face model,
raising IndexError mid-load and leaving the backend unusable.
The router's dispatch isn't amenable to per-file try/except alone
(first-file-wins picks det_10g.onnx from buffalo_l even when the
user asked for buffalo_sc -- alphabetical order happens to favour
the wrong pack). Instead, ship an explicit manifest of the
upstream v0.7 pack contents and scope the glob to that when the
requested pack is known. The manifest is small and stable; future
packs can be added alongside or fall through to the tolerance
loop, which also swallows any remaining IndexError / ValueError
from foreign files with a clear `[insightface] skipped` stderr
line for diagnostics.
Assisted-by: Claude:Opus 4.7
* fix(speaker-recognition): extract FBank features for rank-3 ONNX encoders
Pre-exported speaker-encoder ONNX graphs come in two shapes:
rank-2 [batch, samples] -- some 3D-Speaker exports,
take raw waveform directly.
rank-3 [batch, frames, n_mels] -- WeSpeaker and most Kaldi-
lineage encoders, expect
pre-computed Kaldi FBank.
OnnxDirectEngine unconditionally fed `audio.reshape(1, -1)` --
correct for rank-2, IndexError-on-input_shape[3] on rank-3, which
surfaced to the UI as
"Invalid rank for input: feats Got: 2 Expected: 3"
Detect the input rank at session init and run Kaldi FBank
(80-dim, 25ms/10ms frames, dither=0.0, per-utterance CMN) before
the forward pass when rank>=3. All knobs are configurable via
backend options for encoders that deviate from defaults.
torchaudio.compliance.kaldi is already in the backend's
requirements (SpeechBrain pulls torchaudio in), so no new
dependency.
Assisted-by: Claude:Opus 4.7
* fix(biometrics): isolate face and voice vector stores
Face (ArcFace, 512-D) and voice (ECAPA-TDNN 192-D / WeSpeaker
256-D) biometric embeddings were colliding inside a single
in-memory local-store instance. Enrolling one after the other
failed with
"Try to add key with length N when existing length is M"
because local-store correctly refuses to mix dimensions in one
keyspace.
The registries were constructed with `storeName=""`, which in
StoreBackend() is just a WithModel() call. But ModelLoader's
cache is keyed on `modelID`, not `model` -- so both registries
collapsed to the same `modelID=""` slot and reused the same
backend process despite looking isolated on paper.
Three complementary fixes:
1. application.go -- give each registry a distinct default
namespace ("localai-face-biometrics" /
"localai-voice-biometrics"). The comment claimed
isolation, now it's actually enforced.
2. stores.go -- pass the storeName as both WithModelID and
WithModel so the ModelLoader cache key separates
namespaces and the loader spawns distinct processes.
3. local-store/store.go -- drop the Load() `opts.Model != ""`
guard. It was there to prevent generic model-loading loops
from picking up local-store by accident, but that auto-load
path is being retired; the guard now just blocks legitimate
namespace isolation. opts.Model is treated as a tag; the
per-tuple process isolation upstream handles discrimination.
Assisted-by: Claude:Opus 4.7
* fix(gallery): stale-file cleanup and upgrade-tmp directory safety
Two related robustness fixes for backend install/upgrade:
pkg/downloader/uri.go
OCI downloads passed through
if filepath.Ext(filePath) != "" ...
filePath = filepath.Dir(filePath)
which was intended to redirect file-shaped download targets
into their parent directory for OCI extraction. The heuristic
misfires on directory-shaped paths with a dot-suffix --
gallery.UpgradeBackend uses
tmpPath = "<backendsPath>/<name>.upgrade-tmp"
and Go's filepath.Ext treats ".upgrade-tmp" as an extension.
The rewrite landed the extraction at "<backendsPath>/", which
then **overwrote the real install** (backends/<name>/) with a
flat-layout file and left a stray run.sh at the top level. The
tmp dir itself stayed empty, so the validation step that
checked "<tmpPath>/run.sh" predictably failed with
"upgrade validation failed: run.sh not found in new backend"
Every manual upgrade silently corrupted the backends tree this
way. Guard the rewrite behind "target isn't already an existing
directory" -- InstallBackend / UpgradeBackend both pre-create
the target as a directory, so they get the correct behaviour;
existing file-path callers with a genuine dot-extension still
get the parent redirect.
core/gallery/backends.go
InstallBackend's MkdirAll returned ENOTDIR when something at
the target path was already a file (legacy dev builds dropped
golang backend binaries directly at `<backendsPath>/<name>`
instead of nesting them under their own subdir). That
permanently blocked reinstall and upgrade for anyone carrying
that state, since every retry hit the same error. Detect a
pre-existing non-directory, warn, and remove it before the
MkdirAll so the fresh install can write the correct nested
layout with metadata.json + run.sh.
Assisted-by: Claude:Opus 4.7
* fix(galleryop): refresh upgrade cache after backend ops
UpgradeChecker caches the last upgrade-check result and only
refreshes on the 6-hour tick or after an auto-upgrade cycle.
Manual upgrades (POST /api/backends/upgrade/:name) go through
the async galleryop worker, which completes the upgrade
correctly but never tells UpgradeChecker to re-check -- so
/api/backends/upgrades continued to list a just-upgraded backend
as upgradeable, indistinguishable from a failed upgrade, for up
to six hours.
Add an optional `OnBackendOpCompleted func()` hook on
GalleryService that fires after every successful install /
upgrade / delete on the backend channel (async, so a slow
callback doesn't stall the queue). startup.go wires it to
UpgradeChecker.TriggerCheck after both services exist. Result:
the upgrade banner clears within milliseconds of the worker
finishing.
Assisted-by: Claude:Opus 4.7
* build: prepend GOPATH/bin to PATH for protogen-go
install-go-tools runs `go install` for protoc-gen-go and
protoc-gen-go-grpc, which writes them into `go env GOPATH`/bin.
That directory isn't on every dev's PATH, and protoc resolves
its code-gen plugins via PATH, so the immediately-following
protoc invocation fails with
"protoc-gen-go: program not found"
which in turn blocks `make build` and any
`make backends/%` target that depends on build.
Prepend `go env GOPATH`/bin to PATH for the protoc invocation
so the freshly-installed plugins are found without requiring a
shell-profile change.
Assisted-by: Claude:Opus 4.7
* refactor(ui-api): non-blocking backend upgrade handler with opcache
POST /api/backends/upgrade/:name used to send the ManagementOp
directly onto the unbuffered BackendGalleryChannel, which blocked
the HTTP request whenever the galleryop worker was busy with a
prior operation. The op also didn't show up in /api/operations,
so the Backends UI couldn't reflect upgrade progress on the
affected row.
Register the op in opcache immediately, wrap it in a cancellable
context, store the cancellation function on the GalleryService,
and push onto the channel from a goroutine so the handler
returns right away. Response gains a `jobID` field and a
`message` string so clients have a consistent handle regardless
of whether the op is queued or running.
Pairs with the OnBackendOpCompleted hook added in the galleryop
commit — together the UI sees the upgrade start, watches
progress via /api/operations, and drops the "upgradeable" flag
the moment the worker finishes.
Assisted-by: Claude:Opus 4.7
gen_video's ffmpeg subprocess was relying on the filename extension to
choose the output container. Distributed LocalAI hands the backend a
staging path (e.g. /staging/localai-output-NNN.tmp) that is renamed to
.mp4 only after the backend returns, so ffmpeg saw a .tmp extension and
bailed with "Unable to choose an output format". Inference had already
completed and the frames were piped in, producing the cryptic
"video inference failed (code 1)" at the API layer.
Pass -f mp4 explicitly so the container is selected by flag instead of
by filename suffix.
Co-authored-by: Claude Opus 4.7 (1M context) <noreply@anthropic.com>
* feat: add distributed mode (experimental)
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix data races, mutexes, transactions
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactorings
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fixups
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix events and tool stream in agent chat
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* use ginkgo
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactoring and consolidation
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactoring and consolidation
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactoring and consolidation
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactoring and consolidation
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactoring and consolidation
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactoring and consolidation
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactoring and consolidation
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactoring and consolidation
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* fix(cron): compute correctly time boundaries avoiding re-triggering
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* enhancements, refactorings
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* do not flood of healthy checks
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* do not list obvious backends as text backends
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* tests fixups
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* refactoring and consolidation
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* Drop redundant healthcheck
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* enhancements, refactorings
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
---------
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
