mirror of
https://github.com/mudler/LocalAI.git
synced 2026-05-17 04:56:52 -04:00
621c612b2d6169da9fcfe52e70853e1cdda67dda
4 Commits
| Author | SHA1 | Message | Date | |
|---|---|---|---|---|
LocalAI [bot]
|
2be07f61da |
feat(whisper): honor client cancellation via ggml abort_callback (#9710)
* 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 |
||
|
LocalAI [bot]
|
a8d7d37a3c |
fix: unbreak master CI (docs, kokoros, vibevoice-cpp ABI) (#9682)
* fix(docs): correct broken Hugo relrefs The Hugo build has been failing on master since the relevant pages landed: - text-generation.md:720 referenced `/docs/features/distributed-mode`, but Hugo `relref` paths are relative to the content root, not the rendered URL. Drop the `/docs/` prefix so the lookup matches the existing `features/...` form used elsewhere in the file. - audio-transform.md:144 referenced `tts.md`; the actual page is `text-to-audio.md`. Assisted-by: Claude:claude-opus-4-7[1m] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(kokoros): stub Diarize and AudioTransform Backend trait methods The recent backend.proto additions (Diarize, AudioTransform, AudioTransformStream) extended the gRPC Backend trait, breaking kokoros-grpc compilation with E0046 because the Rust implementation hadn't picked up the new methods. Add Unimplemented stubs matching the existing pattern for non-applicable RPCs in this TTS-only backend. Assisted-by: Claude:claude-opus-4-7[1m] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(vibevoice-cpp): track upstream ABI + wire 1.5B voice cloning Two recent commits in mudler/vibevoice.cpp reshaped the vv_capi_tts signature without a corresponding bump on the LocalAI side: 3bd759c "1.5b: unify into a single tts entry point" inserted a ref_audio_path parameter between voice_path and dst_wav_path. ad856bd "1.5b: multi-speaker dialog support" promoted that to a (const char* const* ref_audio_paths, int n_ref_audio_paths) pair for per-speaker conditioning. Because purego resolves symbols by name and not by signature, the build kept linking; at runtime the misaligned arguments turned the TTS->ASR closed-loop test into a SIGSEGV inside cgo. Track HEAD explicitly and bring the bridge in line with it: * Update the CppTTS purego binding to the 9-arg form. purego marshals []*byte as a **char by handing the C side the underlying array address; nil/empty maps to NULL, which matches the C contract for "no reference audio" on the realtime-0.5B path. * Add a `ref_audio` gallery option (comma-separated, repeatable) that the 1.5B path consumes for runtime voice cloning. Multiple entries are interpreted as one WAV per speaker (Speaker 0..n-1). * TTSRequest.Voice now routes by extension/shape: `.wav` or a comma-separated list goes to ref_audio_paths; anything else stays on voice_path (realtime-0.5B's pre-baked voice gguf). * Pin VIBEVOICE_CPP_VERSION to ad856bd and wire the Makefile into the existing bump_deps matrix so future upstream rolls land as reviewable PRs instead of a silent CI break. Assisted-by: Claude:claude-opus-4-7[1m] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * refactor(vibevoice-cpp): use ModelOptions.AudioPath for 1.5B ref audio Use the existing audio_path field from ModelOptions (already plumbed through config_file's `audio_path:` YAML and consumed by other audio backends like kokoros) instead of inventing a custom `ref_audio:` Options[] string. Multi-speaker setups stay on a single comma- separated value. No behavior change beyond the gallery key name; per-call routing via TTSRequest.Voice is unchanged. Assisted-by: Claude:claude-opus-4-7[1m] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> --------- Signed-off-by: Ettore Di Giacinto <mudler@localai.io> Co-authored-by: Ettore Di Giacinto <mudler@localai.io> |
||
|
Ettore Di Giacinto
|
e86ade54a6 |
feat(api): add /v1/audio/diarization endpoint with sherpa-onnx + vibevoice.cpp (#9654)
* 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 (
|
||
|
Ettore Di Giacinto
|
fe6eb57082 |
feat(vibevoice-cpp): add purego TTS+ASR backend (#9610)
* 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>
|