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13 Commits
| Author | SHA1 | Message | Date | |
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LocalAI [bot]
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a7cb587d96 |
feat(parakeet-cpp): real segment timestamps (NeMo-faithful) (#10207)
* feat(parakeet-cpp): real segment timestamps (NeMo-faithful)
Offline: replace the single synthetic whole-clip segment with multiple
segments grouped exactly like NeMo's get_segment_offsets - a new segment
after sentence-ending punctuation ('. ? !'), each carrying start/end and
its time-window token ids. The optional model option segment_gap_threshold
(NeMo's unit: encoder FRAMES, default 0=off) adds NeMo's silence-gap split,
converted to seconds via the JSON frame_sec the engine now reports.
Per-segment words are still gated behind timestamp_granularities=["word"];
a zero-word document falls back to a single text segment.
Streaming: when libparakeet.so exposes the ABI v4 JSON entry points
(probed), drive parakeet_capi_stream_feed_json / _finalize_json and
accumulate the streamed per-word timestamps into per-utterance segments
(EOU stays the boundary), so streaming FinalResult segments now carry
start/end. Falls back to the text-only feed against an older library.
Pure-Go specs cover splitWordsIntoSegments (punctuation + gap rules, NeMo
elif order, fallback), transcriptResultFromDoc (multi-segment, token
windows, word-granularity gate), and the streaming segmenter.
Assisted-by: Claude:claude-opus-4-8 [Claude Code]
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* docs(audio): document parakeet-cpp segment timestamps + segment_gap_threshold
Assisted-by: Claude:claude-opus-4-8 [Claude Code]
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
* test(parakeet-cpp): update model-gated specs for multi-segment output
The offline AudioTranscription specs asserted the old single synthetic
segment (Segments HaveLen(1), Segments[0].Text == res.Text). With
NeMo-faithful segmentation a multi-sentence clip now yields multiple
punctuation-delimited segments, so assert the new contract instead:
one-or-more time-ordered segments, each with text and (under word
granularity) per-segment words whose span tracks the segment start/end.
Caught by running the model-gated suite on the dgx (GB10) against the
real tdt_ctc-110m + realtime_eou models.
Assisted-by: Claude:claude-opus-4-8 [Claude Code]
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>
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LocalAI [bot]
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860f9d63ad |
feat(parakeet-cpp): dynamic batching for concurrent transcription requests (#10112)
* feat(parakeet-cpp): dynamic-batching scheduler (queue + dispatcher) Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(parakeet-cpp): dynamic batching for AudioTranscription via batched JSON C-API Drop SingleThread; route unary transcription through the in-process batcher which coalesces concurrent requests into one batched engine call. Streaming stays mutually exclusive via engineMu. Adds batch_max_size / batch_max_wait_ms options (size=1 disables; recommended on CPU). Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(parakeet-cpp): tear down dispatcher in Free; log batch config; preallocate; clarify stream lock Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(parakeet-cpp): Ginkgo batcher tests; optional batch C-API binding with per-request fallback The batched JSON C-API symbol exists only in newer libparakeet.so (ABI >= 2); probe it with Dlsym and register optionally so the backend still loads against an older library, falling back to per-request transcription. Rewrites the batcher unit tests as Ginkgo/Gomega specs (forbidigo bans t.Fatal in tests). Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(parakeet-cpp): debug-log coalesced batch size in runBatch Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(parakeet-cpp): default batch_max_size to 1 (batching opt-in) Dynamic batching now defaults off (batch_max_size:1, one request at a time). Raise batch_max_size to opt in: it is a large throughput win on GPU under concurrent load, but on CPU and low-concurrency setups it only adds latency, so off is the safer default. The startup log now states whether batching is on or off, and the audio-to-text docs are updated to match. Signed-off-by: Ettore Di Giacinto <mudler@localai.io> Assisted-by: Claude:claude-opus-4-8 [Claude Code] * chore(parakeet-cpp): bump parakeet.cpp to 8a7c482 (batched decode + B=1 fast-path) parakeet.cpp PR #1 merged the batched encoder/decode and the B=1 encoder fast-path to master. Point PARAKEET_VERSION at that commit so the backend builds the batched C-API (parakeet_capi_transcribe_pcm_batch_json) that the dynamic batcher calls; the prior pin (30a3075) predated it, so only the per-request fallback path was exercised. Verified the shared lib builds with the backend's CMake flags and exports the batch symbol. Signed-off-by: Ettore Di Giacinto <mudler@localai.io> Assisted-by: Claude:claude-opus-4-8 [Claude Code] --------- Signed-off-by: Ettore Di Giacinto <mudler@localai.io> Co-authored-by: Ettore Di Giacinto <mudler@localai.io> |
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LocalAI [bot]
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4912c9b73a |
feat(parakeet-cpp): add NVIDIA NeMo Parakeet ASR backend (parakeet.cpp) (#10084)
* feat(parakeet-cpp): L0 backend scaffold, LoadModel + AudioTranscription (text) Add a Go gRPC backend that bridges LocalAI to parakeet.cpp via the flat C-API (parakeet_capi.h), loaded with purego (cgo-less, mirrors the whisper / vibevoice-cpp backends). L0 scope: - main.go: dlopen libparakeet.so (override via PARAKEET_LIBRARY), register the C-API entry points, start the gRPC server. - goparakeetcpp.go: Load (parakeet_capi_load), AudioTranscription (parakeet_capi_transcribe_path, decoder=0 = per-arch default head), Free, serialized through base.SingleThread since the C engine is a thread-unsafe singleton. char* returns are bound as uintptr so the malloc'd buffer is freed via parakeet_capi_free_string after copy. - AudioTranscriptionStream returns a clear "not implemented in L0" error (closes the channel so the server doesn't hang), wired in L2. - Makefile: clone-at-pin + cmake (PARAKEET_VERSION for bump_deps.sh), with a local-symlink dev shortcut; run.sh / package.sh mirror whisper. - Test auto-skips without PARAKEET_BACKEND_TEST_MODEL/_WAV fixtures. Builds clean (CGO_ENABLED=0), gofmt clean, test passes. The single unsafeptr vet note in goStringFromCPtr is documented and matches the whisper backend's tolerated pattern. Word/segment timestamps (L1) and cache-aware streaming (L2) follow. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(parakeet-cpp): L1 word/segment timestamps via transcribe_path_json AudioTranscription now calls parakeet_capi_transcribe_path_json and shapes the per-word / per-token timestamps into the TranscriptResult: - Bind parakeet_capi_transcribe_path_json (purego, char* as uintptr like the other returns) and register it in main.go + the test loader. - Parse the JSON document ({"text","words":[{w,start,end,conf}], "tokens":[{id,t,conf}]}) into typed structs. - Synthesise a single whole-clip segment (parakeet emits no native segment boundaries) spanning the first word start to the last word end; token ids populate Segment.Tokens. - Attach word-level timings only when timestamp_granularities=["word"], matching the OpenAI API (segment-level default). secondsToNanos mirrors the whisper backend's nanosecond convention. Verified end-to-end against tdt_ctc-110m (f16): both the default and word-granularity specs pass; builds clean, gofmt clean, vet shows only the one documented unsafeptr note shared with the whisper backend. Cache-aware streaming (L2) follows. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(parakeet-cpp): L2 cache-aware streaming with EOU segmentation Wire AudioTranscriptionStream to the streaming RNN-T C-API: - Bind parakeet_capi_stream_{begin,feed,finalize,free}; feed takes 16 kHz mono float PCM ([]float32 via purego) and writes *eou_out on <EOU>/<EOB>. - Decode opts.Dst to 16 kHz mono PCM (utils.AudioToWav + go-audio, same as the whisper backend), feed it in 1 s chunks, and emit each newly-finalized text run as a TranscriptStreamResponse delta. - <EOU>/<EOB> events close the current segment; a closing FinalResult carries the full transcript plus the per-utterance segments (with a whole-clip fallback segment when no EOU fired). - stream_begin returns 0 for non-streaming models, surfaced as a clear error instead of an empty stream. Honours context cancellation between chunks. Frees every malloc'd delta and the session. Verified end-to-end against realtime_eou_120m-v1 (f16): the streamed transcript matches the offline 110m reference word-for-word, deltas reconstruct the final text, and the spec passes alongside the offline specs. Builds clean, gofmt clean, vet shows only the shared documented unsafeptr note. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(parakeet-cpp): L3 register backend in build/CI/gallery (whisper parity) Wire the new Go gRPC parakeet-cpp backend (parakeet.cpp ggml port of NVIDIA NeMo Parakeet ASR) into LocalAI's build/CI/gallery surfaces, matching the existing ggml whisper Go backend 1:1. - .github/backend-matrix.yml: add 11 linux entries + 1 darwin entry mirroring every whisper build (cpu amd64/arm64, intel sycl f32/f16, vulkan amd64/arm64, nvidia cuda-12, nvidia cuda-13, nvidia-l4t-arm64, nvidia-l4t-cuda-13-arm64, rocm hipblas, metal-darwin-arm64), all on ./backend/Dockerfile.golang with backend: "parakeet-cpp" and -*-parakeet-cpp tag-suffixes. - scripts/changed-backends.js: explicit inferBackendPath branch resolving parakeet-cpp to backend/go/parakeet-cpp/ before the generic golang branch. - .github/workflows/bump_deps.yaml: track the PARAKEET_VERSION pin in backend/go/parakeet-cpp/Makefile (repo mudler/parakeet.cpp, branch master). - backend/index.yaml: add ¶keetcpp meta + latest/development image entries for every matrix tag-suffix. - Makefile: add backends/parakeet-cpp to .NOTPARALLEL, BACKEND_PARAKEET_CPP definition, docker-build target eval, and test-extra-backend-parakeet-cpp- transcription target (mirrors test-extra-backend-whisper-transcription). Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(parakeet-cpp): L4 gallery importer for parakeet GGUFs Add ParakeetCppImporter so parakeet.cpp GGUFs auto-detect on /import-model and route to the parakeet-cpp backend (it also surfaces in /backends/known, which drives the import dropdown). - Match is narrow: a .gguf whose name carries a parakeet architecture token (<arch>-<size>-<quant>.gguf, e.g. tdt_ctc-110m-f16.gguf, rnnt-0.6b-q4_k.gguf, realtime_eou_120m-v1-q8_0.gguf), a direct URL to one, or preferences.backend="parakeet-cpp". It deliberately does NOT claim arbitrary llama-style GGUFs, nor the upstream nvidia/parakeet-* NeMo repos (.nemo, not runnable here). - Registered in the ASR batch BEFORE LlamaCPPImporter so its GGUFs aren't swallowed by the generic .gguf importer. - Import nests files under parakeet-cpp/models/<name>/, defaults to the smallest quant (q4_k, near-lossless on parakeet) with a size-ladder fallback, and honours preferences.quantizations / name / description. Tested with synthetic HF details (no network): metadata, positive matches (HF repo, direct URL, preference), narrowness negatives (llama GGUF, NeMo repo), and import (default quant, override, direct URL), 9 specs pass, build/vet/gofmt clean. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * docs(parakeet-cpp): document the parakeet-cpp transcription backend Add parakeet-cpp to the audio-to-text backend list and a dedicated usage section: direct GGUF import (auto-detects to the backend), model YAML, word-level timestamps via timestamp_granularities[]=word, and cache-aware streaming with the realtime_eou model. Points at the mudler/parakeet-cpp-gguf collection repo. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * ci(parakeet-cpp): wire transcription gRPC e2e test into test-extra The L3 commit added the test-extra-backend-parakeet-cpp-transcription Makefile target but never invoked it in CI. Mirror the whisper job: - Add a parakeet-cpp output to detect-changes (emitted by changed-backends.js from the matrix entry). - Add tests-parakeet-cpp-grpc-transcription, gated on the parakeet-cpp path filter / run-all, building the backend image and running the transcription e2e against tdt_ctc-110m + the JFK clip. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * style(parakeet-cpp): drop em dashes from comments and docs Replace em dashes with plain punctuation in the backend comments, the importer, package.sh, and the audio-to-text docs section (and use "and" instead of the multiplication sign). No behaviour change. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * feat(gallery): add parakeet-cpp f16 models to the model gallery Add the 10 NVIDIA Parakeet models (f16, the recommended quality/speed default) as gallery entries that install on the parakeet-cpp backend from mudler/parakeet-cpp-gguf: tdt_ctc-110m/1.1b, tdt-0.6b-v2/v3, tdt-1.1b, ctc-0.6b/1.1b, rnnt-0.6b/1.1b, and the cache-aware streaming realtime_eou_120m-v1. Each pins the file sha256 and routes transcript usecases to the backend. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(parakeet-cpp): satisfy govet lint + bump PARAKEET_VERSION - goparakeetcpp.go: //nolint:govet on the C-owned-pointer unsafe.Pointer conversion (golangci-lint reports new-only issues, so unlike the whisper backend's identical line this one is flagged). - Makefile: bump PARAKEET_VERSION to the current parakeet.cpp master commit (the previous pin's commit no longer exists after upstream history was squashed), so the backend image clone/build resolves again. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(parakeet-cpp): pin PARAKEET_VERSION to a tag-stable commit The previous SHA pin was orphaned when parakeet.cpp's single-commit master was amended/force-pushed, so the backend image clone (git fetch <sha>) failed across every build variant. Repoint to 845c29e, which upstream now keeps permanently fetchable via the `localai-backend-pin` tag, so future upstream amends no longer break the backend build. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(parakeet-cpp): init the ggml submodule in the backend image clone The backend Dockerfile clones parakeet.cpp at PARAKEET_VERSION with a shallow fetch + checkout but never initialised submodules, so third_party/ggml was empty and the parakeet.cpp cmake build failed at `add_subdirectory(third_party/ggml)` (CMakeLists.txt:53) on every build variant. Add `git submodule update --init --recursive --depth 1 --single-branch` after checkout, mirroring the whisper backend. Verified locally: clone + submodule + cmake configure now succeeds. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(parakeet-cpp): statically link ggml into libparakeet.so The shared libparakeet.so linked ggml's shared libs (libggml*.so), but the package only ships libparakeet.so, so at runtime dlopen failed with "libggml.so.0: cannot open shared object file" (the e2e transcription test panicked on load). Build ggml static + PIC (BUILD_SHARED_LIBS=OFF, CMAKE_POSITION_INDEPENDENT_CODE=ON) so libparakeet.so embeds ggml and depends only on system libs already present in the runtime image. Verified locally: ldd shows no libggml dependency. Assisted-by: Claude:claude-opus-4-8 [Claude Code] Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fix(parakeet-cpp): non-streaming fallback in AudioTranscriptionStream The e2e streaming test ran AudioTranscriptionStream against tdt_ctc-110m (not a cache-aware streaming model), so stream_begin returned 0 and the call errored. Per LocalAI's streaming contract (and the whisper backend), a non-streaming model should fall back to a single offline transcription emitted as one delta plus a closing FinalResult. Do that instead of erroring, so the streaming endpoint works for every parakeet model. Verified locally: the streaming spec passes against the non-streaming 110m model via fallback. Assisted-by: Claude:claude-opus-4-8 [Claude Code] 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> |
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Ettore Di Giacinto
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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 (
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Richard Palethorpe
|
bb033b16a9 |
feat: add LocalVQE backend and audio transformations UI (#9640)
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>
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Ettore Di Giacinto
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833b7e8557 |
chore(docs): update transcription endpoint
Signed-off-by: Ettore Di Giacinto <mudler@localai.io> |
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Ettore Di Giacinto
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7e0b73deaa |
fix(docs): fix broken references to distributed mode
Signed-off-by: Ettore Di Giacinto <mudler@localai.io> |
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Lukas Schaefer
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ed0bfb8732 |
fix: rename json_verbose to verbose_json (#8627)
Signed-off-by: Lukas Schaefer <lukas@lschaefer.xyz> |
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Ettore Di Giacinto
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53276d28e7 |
feat(musicgen): add ace-step and UI interface (#8396)
* feat(musicgen): add ace-step and UI interface Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * Correctly handle model dir Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * Drop auto-download Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * Fixups Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * Add to models, fixup UIs icons Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fixups Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * Update docs Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * l4t13 is incompatbile Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * avoid pinning version for cuda12 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * Drop l4t12 Signed-off-by: Ettore Di Giacinto <mudler@localai.io> --------- Signed-off-by: Ettore Di Giacinto <mudler@localai.io> |
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Andres
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b6459ddd57 |
feat(api): Add transcribe response format request parameter & adjust STT backends (#8318)
* WIP response format implementation for audio transcriptions (cherry picked from commit e271dd764bbc13846accf3beb8b6522153aa276f) Signed-off-by: Andres Smith <andressmithdev@pm.me> * Rework transcript response_format and add more formats (cherry picked from commit 6a93a8f63e2ee5726bca2980b0c9cf4ef8b7aeb8) Signed-off-by: Andres Smith <andressmithdev@pm.me> * Add test and replace go-openai package with official openai go client (cherry picked from commit f25d1a04e46526429c89db4c739e1e65942ca893) Signed-off-by: Andres Smith <andressmithdev@pm.me> * Fix faster-whisper backend and refactor transcription formatting to also work on CLI Signed-off-by: Andres Smith <andressmithdev@pm.me> (cherry picked from commit 69a93977d5e113eb7172bd85a0f918592d3d2168) Signed-off-by: Andres Smith <andressmithdev@pm.me> --------- Signed-off-by: Andres Smith <andressmithdev@pm.me> Co-authored-by: nanoandrew4 <nanoandrew4@gmail.com> Co-authored-by: Ettore Di Giacinto <mudler@users.noreply.github.com> |
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Ettore Di Giacinto
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2cc4809b0d |
feat: docs revamp (#7313)
* docs Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * Small enhancements Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * Enhancements * Default to zen-dark Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * fixups Signed-off-by: Ettore Di Giacinto <mudler@localai.io> --------- Signed-off-by: Ettore Di Giacinto <mudler@localai.io> |
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Ettore Di Giacinto
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6ca4d38a01 |
docs/examples: enhancements (#1572)
* docs: re-order sections * fix references * Add mixtral-instruct, tinyllama-chat, dolphin-2.5-mixtral-8x7b * Fix link * Minor corrections * fix: models is a StringSlice, not a String Signed-off-by: Ettore Di Giacinto <mudler@localai.io> * WIP: switch docs theme * content * Fix GH link * enhancements * enhancements * Fixed how to link Signed-off-by: lunamidori5 <118759930+lunamidori5@users.noreply.github.com> * fixups * logo fix * more fixups * final touches --------- Signed-off-by: Ettore Di Giacinto <mudler@localai.io> Signed-off-by: lunamidori5 <118759930+lunamidori5@users.noreply.github.com> Co-authored-by: lunamidori5 <118759930+lunamidori5@users.noreply.github.com> |
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Ettore Di Giacinto
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c5c77d2b0d |
docs: Initial import from localai-website (#1312)
Signed-off-by: Ettore Di Giacinto <mudler@localai.io> |