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>
6.7 KiB
+++ disableToc = false title = "Speaker Diarization" weight = 33 url = "/features/audio-diarization/" +++
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_diarfamily, 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/diarizationwith/v1/voice/embed(speaker embedding) and maintain your own embedding store. - Hints vs. forces:
num_speakersoverrides clustering when set;min_speakers/max_speakersare 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.
