Files
LocalAI/docs/content/features/audio-diarization.md
Ettore Di Giacinto d8b615060d backend(audio-cpp): advertise the backend and document its options
Registers audio-cpp as preference-only in /backends/known: the family lives in
GGUF metadata that an importer cannot read from a remote repo, and one repo
hosts thirty families, so there is no honest auto-detect signal. Modality is a
single string and the import form chips on a fixed key set, so it registers as
tts with the other modalities named in the description rather than under an
invented key the UI would bucket as "other".

Adds a features page covering the option namespacing, the routing table per
endpoint, the RPCs this backend declines and why, the bundled VAD path, the
separation stem behaviour, and the family gotchas (supertonic needs the orig
package; chatterbox advertises cloning and no plain tts; nemotron_asr defers
its whole decode to finalize so live transcription emits nothing until the
client half-closes, unlike higgs_audio_stt and voxtral_realtime). Every option
name and family capability in it was read off the pinned upstream checkout.

Assisted-by: Claude:claude-opus-5 [Claude Code]
Signed-off-by: Ettore Di Giacinto <mudler@localai.io>
2026-07-27 05:21:23 +00:00

6.7 KiB

+++ disableToc = false title = "Speaker Diarization" weight = 33 url = "/features/audio-diarization/" +++

Diarization: segment, embed, and cluster (or a single ASR pass) into speaker-labelled segments

Speaker diarization answers the question "who spoke when?" - given an audio clip with multiple speakers, it returns time-stamped segments labelled with a stable speaker ID (SPEAKER_00, SPEAKER_01, …).

LocalAI exposes this through the /v1/audio/diarization endpoint, modelled after /v1/audio/transcriptions. Three backends are supported today:

  • sherpa-onnx - pyannote-3.0 segmentation + a speaker-embedding extractor (3D-Speaker, NeMo, WeSpeaker) + fast clustering. Pure diarization - no transcription cost. Recommended when you only need speaker turns.
  • vibevoice.cpp - produces speaker-labelled segments as a by-product of its long-form ASR pass, so you can optionally get a transcript per segment for free.
  • audio.cpp - the sortformer_diar family, served by the multi-modality [audio.cpp backend]({{%relref "features/audio-cpp" %}}).

Because diarization is exposed as a regular OpenAI-compatible endpoint, any HTTP client works. There is no Python dependency on pyannote or NeMo on the consumer side.

Endpoint

POST /v1/audio/diarization
Content-Type: multipart/form-data
Field Type Description
file file (required) audio file in any format ffmpeg accepts
model string (required) name of the diarization-capable model
num_speakers int exact speaker count when known (>0 forces; 0 = auto)
min_speakers int hint when auto-detecting
max_speakers int hint when auto-detecting
clustering_threshold float cosine distance threshold used when num_speakers is unknown
min_duration_on float discard segments shorter than this many seconds
min_duration_off float merge gaps shorter than this many seconds
language string only meaningful for backends that bundle ASR (e.g. vibevoice)
include_text bool when the backend can emit per-segment transcript for free, populate it
response_format string json (default), verbose_json, or rttm

Response - json (default)

Compact payload, no transcription, no per-speaker summary:

{
  "task": "diarize",
  "duration": 12.34,
  "num_speakers": 2,
  "segments": [
    {"id": 0, "speaker": "SPEAKER_00", "label": "0", "start": 0.00, "end": 2.34},
    {"id": 1, "speaker": "SPEAKER_01", "label": "1", "start": 2.34, "end": 4.10}
  ]
}

speaker is the normalized, zero-padded label clients should display. label preserves the raw backend-emitted ID for clients that maintain their own speaker dictionary.

Response - verbose_json

Adds per-speaker totals and (when the backend supports it and include_text=true) the per-segment transcript:

{
  "task": "diarize",
  "duration": 12.34,
  "language": "en",
  "num_speakers": 2,
  "segments": [
    {"id": 0, "speaker": "SPEAKER_00", "label": "0", "start": 0.00, "end": 2.34, "text": "Hello, world."},
    {"id": 1, "speaker": "SPEAKER_01", "label": "1", "start": 2.34, "end": 4.10, "text": "How are you?"}
  ],
  "speakers": [
    {"id": "SPEAKER_00", "label": "0", "total_speech_duration": 5.6, "segment_count": 3},
    {"id": "SPEAKER_01", "label": "1", "total_speech_duration": 1.76, "segment_count": 1}
  ]
}

Response - rttm

NIST RTTM, the standard interchange format used by pyannote.metrics / dscore:

SPEAKER audio 1 0.000 2.340 <NA> <NA> SPEAKER_00 <NA> <NA>
SPEAKER audio 1 2.340 1.760 <NA> <NA> SPEAKER_01 <NA> <NA>

Returned as Content-Type: text/plain; charset=utf-8.

Quick start

First install a diarization-capable model from the gallery. The example below uses vibevoice-cpp-asr, which serves the vibevoice.cpp backend and returns speaker-labelled segments (and, optionally, a transcript):

local-ai run vibevoice-cpp-asr
curl http://localhost:8080/v1/audio/diarization \
  -H "Content-Type: multipart/form-data" \
  -F file="@meeting.wav" \
  -F model="vibevoice-cpp-asr" \
  -F num_speakers=3

The sections below show how to configure the two supported backends by hand when you want full control over the segmentation and embedding models.

Backend setup - sherpa-onnx (pure diarization)

Sherpa-onnx needs two ONNX models: pyannote segmentation and a speaker-embedding extractor. Place them under your LocalAI models directory and reference them from the YAML:

name: pyannote-diarization
backend: sherpa-onnx
type: diarization
parameters:
  model: sherpa-onnx-pyannote-segmentation-3-0/model.onnx
options:
  - diarize.embedding_model=3dspeaker_speech_campplus_sv_zh-cn_16k-common.onnx
  # Optional clustering knobs (per-call DiarizeRequest fields override these):
  - diarize.threshold=0.5
  - diarize.min_duration_on=0.3
  - diarize.min_duration_off=0.5
known_usecases:
  - FLAG_DIARIZATION

Both model: and diarize.embedding_model= are resolved relative to the LocalAI models directory.

Backend setup - vibevoice.cpp (diarization + ASR)

vibevoice.cpp's ASR mode emits [{Start, End, Speaker, Content}] natively, so a single pass gives both diarization and transcription:

name: vibevoice-diarize
backend: vibevoice-cpp
parameters:
  model: vibevoice-asr.gguf
options:
  - type=asr
  - tokenizer=vibevoice-tokenizer.gguf
known_usecases:
  - FLAG_DIARIZATION
  - FLAG_TRANSCRIPT

Pass include_text=true on the request to populate the text field on each diarization segment.

curl http://localhost:8080/v1/audio/diarization \
  -H "Content-Type: multipart/form-data" \
  -F file="@interview.wav" \
  -F model="vibevoice-diarize" \
  -F include_text=true \
  -F response_format=verbose_json

Notes

  • Speaker identity across files: speaker IDs (SPEAKER_00, SPEAKER_01, …) are local to each request. To track the same person across multiple recordings, combine /v1/audio/diarization with /v1/voice/embed (speaker embedding) and maintain your own embedding store.
  • Hints vs. forces: num_speakers overrides clustering when set; min_speakers / max_speakers are advisory and only honored by backends that expose a range hint. vibevoice.cpp ignores them - its model picks the count itself.
  • Sample rate: input is automatically converted to 16 kHz mono via ffmpeg before the backend sees it; sherpa-onnx pyannote-3.0 requires 16 kHz.

See also

  • [Sound Classification]({{% relref "audio-classification" %}}) - tag non-speech sound events (alarms, glass breaking, baby cry) in a clip.