LocalAI/docs/content/features/audio-to-text.md

+++ disableToc = false title = "Audio to Text" weight = 30 url = "/features/audio-to-text/" +++

Audio to text models are models that can generate text from an audio file.

The transcription endpoint allows to convert audio files to text. The endpoint supports multiple backends:

• whisper.cpp: A C++ library for audio transcription (default)
• moonshine: Ultra-fast transcription engine optimized for low-end devices
• faster-whisper: Fast Whisper implementation with CTranslate2
• WhisperX: Whisper transcription with word alignment and optional speaker diarization. Set HF_TOKEN and pass diarize=true to load WhisperX's gated pyannote diarization pipeline.
• parakeet-cpp: A C++/ggml port of NVIDIA NeMo Parakeet (FastConformer TDT/CTC/RNNT/hybrid). Runs quantized GGUFs on CPU or GPU, emits word-level timestamps, and supports cache-aware streaming (the realtime_eou model surfaces end-of-utterance events). The same backend also loads Nemotron-3-Diarization (/v1/audio/diarization) and CED sound models (/v1/audio/classification), and can attach either as a companion to a transcription model.
• llama-cpp: Route transcription to any multimodal-audio GGUF model served by the llama-cpp backend (e.g. Qwen3-ASR, Voxtral, Qwen2-Audio). Under the hood the request is converted into a chat completion with the audio attached via the model's audio encoder - the same path the upstream llama.cpp server uses. Set backend: llama-cpp in the model YAML and point mmproj at the matching audio encoder.
• voxtral: Voxtral-family models served by a dedicated backend
• NeMo-Speech.cpp: NVIDIA's C++/ggml runtime for the Nemotron Speech models. Serves offline, streaming and live transcription, with VAD, punctuation, inverse text normalization and Sortformer speaker tags attached through model options, and covers diarization, speech synthesis and translation from the same backend. See the [NeMo-Speech.cpp backend]({{%relref "features/nemo-speech-cpp" %}}) page for the model options.
• audio.cpp: Multi-family GGML audio engine. Serves transcription and forced alignment from families such as nemotron_asr, qwen3_asr, citrinet_asr, higgs_audio_stt and voxtral_realtime, and covers diarization, VAD, TTS and source separation from the same backend. See the [audio.cpp backend]({{%relref "features/audio-cpp" %}}) page for the model options.

The endpoint input supports all the audio formats supported by ffmpeg.

Looking for "who spoke when" instead of a flat transcript? See Speaker Diarization - /v1/audio/diarization returns time-stamped speaker segments and supports the rttm format used by pyannote.metrics.

Usage

Once LocalAI is started and whisper models are installed, you can use the /v1/audio/transcriptions API endpoint.

For instance, with cURL:

curl http://localhost:8080/v1/audio/transcriptions -H "Content-Type: multipart/form-data" -F file="@<FILE_PATH>" -F model="<MODEL_NAME>"

When usage statistics are enabled, successful requests to /v1/audio/transcriptions and /audio/transcriptions contribute request counts and elapsed time for the requested model name, including an alias. This includes streaming responses. Token counts remain zero because transcription results do not provide canonical token usage. Failed transcriptions, including errors reported within a stream, do not count as successful usage.

When API tracing is enabled, multipart transcription requests appear in Traces with the model name, response, status, and elapsed time. The captured request body contains metadata marked multipart upload omitted; API traces do not store the uploaded audio. Response capture follows the configured trace size limit. Backend traces have their own payload capture behavior.

Example

Download one of the models from here in the models folder, and create a YAML file for your model:

name: whisper-1
backend: whisper
parameters:
  model: whisper-en

The transcriptions endpoint then can be tested like so:

## Get an example audio file
wget --quiet --show-progress -O gb1.ogg https://upload.wikimedia.org/wikipedia/commons/1/1f/George_W_Bush_Columbia_FINAL.ogg

## Send the example audio file to the transcriptions endpoint
curl http://localhost:8080/v1/audio/transcriptions -H "Content-Type: multipart/form-data" -F file="@$PWD/gb1.ogg" -F model="whisper-1"

Result:

{
  "segments":[{"id":0,"start":0,"end":9640000000,"text":" My fellow Americans, this day has brought terrible news and great sadness to our country.","tokens":[50364,1222,7177,6280,11,341,786,575,3038,6237,2583,293,869,22462,281,527,1941,13,50846]},{"id":1,"start":9640000000,"end":15960000000,"text":" At 9 o'clock this morning, Mission Control and Houston lost contact with our Space Shuttle","tokens":[1711,1722,277,6,9023,341,2446,11,20170,12912,293,18717,2731,3385,365,527,8705,13870,10972,51162]},{"id":2,"start":15960000000,"end":16960000000,"text":" Columbia.","tokens":[17339,13,51212]},{"id":3,"start":16960000000,"end":24640000000,"text":" A short time later, debris was seen falling from the skies above Texas.","tokens":[316,2099,565,1780,11,21942,390,1612,7440,490,264,25861,3673,7885,13,51596]},{"id":4,"start":24640000000,"end":27200000000,"text":" The Columbia's lost.","tokens":[440,17339,311,2731,13,51724]},{"id":5,"start":27200000000,"end":29920000000,"text":" There are no survivors.","tokens":[821,366,572,18369,13,51860]},{"id":6,"start":29920000000,"end":32920000000,"text":" And board was a crew of seven.","tokens":[50364,400,3150,390,257,7260,295,3407,13,50514]},{"id":7,"start":32920000000,"end":39780000000,"text":" Colonel Rick Husband, Lieutenant Colonel Michael Anderson, Commander Laurel Clark, Captain","tokens":[28478,11224,21282,4235,11,28412,28478,5116,18768,11,20857,27270,75,18572,11,10873,50857]},{"id":8,"start":39780000000,"end":50020000000,"text":" David Brown, Commander William McCool, Dr. Cooltna Chavla, and Elon Ramon, a Colonel","tokens":[4389,8030,11,20857,6740,4050,34,1092,11,2491,13,8561,83,629,761,706,875,11,293,28498,9078,266,11,257,28478,51369]},{"id":9,"start":50020000000,"end":52800000000,"text":" in the Israeli Air Force.","tokens":[294,264,19974,5774,10580,13,51508]},{"id":10,"start":52800000000,"end":58480000000,"text":" These men and women assumed great risk in the service to all humanity.","tokens":[1981,1706,293,2266,15895,869,3148,294,264,2643,281,439,10243,13,51792]},{"id":11,"start":58480000000,"end":63120000000,"text":" And an age when Space Flight has come to seem almost routine.","tokens":[50364,400,364,3205,562,8705,28954,575,808,281,1643,1920,9927,13,50596]},{"id":12,"start":63120000000,"end":68800000000,"text":" It is easy to overlook the dangers of travel by rocket and the difficulties of navigating","tokens":[467,307,1858,281,37826,264,27701,295,3147,538,13012,293,264,14399,295,32054,50880]},{"id":13,"start":68800000000,"end":72640000000,"text":" the fierce outer atmosphere of the Earth.","tokens":[264,25341,10847,8018,295,264,4755,13,51072]},{"id":14,"start":72640000000,"end":78040000000,"text":" These astronauts knew the dangers and they faced them willingly.","tokens":[1981,28273,2586,264,27701,293,436,11446,552,44675,13,51342]},{"id":15,"start":78040000000,"end":83040000000,"text":" Knowing they had a high and noble purpose in life.","tokens":[25499,436,632,257,1090,293,20171,4334,294,993,13,51592]},{"id":16,"start":83040000000,"end":90800000000,"text":" Because of their courage and daring and idealism, we will miss them all the more.","tokens":[50364,1436,295,641,9892,293,43128,293,7157,1434,11,321,486,1713,552,439,264,544,13,50752]},{"id":17,"start":90800000000,"end":96560000000,"text":" All Americans today are thinking as well of the families of these men and women who have","tokens":[1057,6280,965,366,1953,382,731,295,264,4466,295,613,1706,293,2266,567,362,51040]},{"id":18,"start":96560000000,"end":100440000000,"text":" been given this sudden shock in grief.","tokens":[668,2212,341,3990,5588,294,18998,13,51234]},{"id":19,"start":100440000000,"end":102400000000,"text":" You're not alone.","tokens":[509,434,406,3312,13,51332]},{"id":20,"start":102400000000,"end":105440000000,"text":" Our entire nation agrees with you.","tokens":[2621,2302,4790,26383,365,291,13,51484]},{"id":21,"start":105440000000,"end":112360000000,"text":" And those you loved will always have the respect and gratitude of this country.","tokens":[400,729,291,4333,486,1009,362,264,3104,293,16935,295,341,1941,13,51830]},{"id":22,"start":112360000000,"end":116600000000,"text":" The cause in which they died will continue.","tokens":[50364,440,3082,294,597,436,4539,486,2354,13,50576]},{"id":23,"start":116600000000,"end":124240000000,"text":" Man kind is led into the darkness beyond our world by the inspiration of discovery and the","tokens":[2458,733,307,4684,666,264,11262,4399,527,1002,538,264,10249,295,12114,293,264,50958]},{"id":24,"start":124240000000,"end":127000000000,"text":" longing to understand.","tokens":[35050,281,1223,13,51096]},{"id":25,"start":127000000000,"end":131160000000,"text":" Our journey into space will go on.","tokens":[2621,4671,666,1901,486,352,322,13,51304]},{"id":26,"start":131160000000,"end":136480000000,"text":" In the skies today, we saw destruction and tragedy.","tokens":[682,264,25861,965,11,321,1866,13563,293,18563,13,51570]},{"id":27,"start":136480000000,"end":142080000000,"text":" As farther than we can see, there is comfort and hope.","tokens":[1018,20344,813,321,393,536,11,456,307,3400,293,1454,13,51850]},{"id":28,"start":142080000000,"end":149800000000,"text":" In the words of the prophet Isaiah, lift your eyes and look to the heavens who created","tokens":[50364,682,264,2283,295,264,18566,27263,11,5533,428,2575,293,574,281,264,26011,567,2942,50750]},{"id":29,"start":149800000000,"end":151640000000,"text":" all these.","tokens":[439,613,13,50842]},{"id":30,"start":151640000000,"end":159960000000,"text":" He who brings out the story hosts one by one and calls them each by name because of his great","tokens":[634,567,5607,484,264,1657,21573,472,538,472,293,5498,552,1184,538,1315,570,295,702,869,51258]},{"id":31,"start":159960000000,"end":163400000000,"text":" power and mighty strength.","tokens":[1347,293,21556,3800,13,51430]},{"id":32,"start":163400000000,"end":166400000000,"text":" Not one of them is missing.","tokens":[1726,472,295,552,307,5361,13,51580]},{"id":33,"start":166400000000,"end":173600000000,"text":" The same creator who names the stars also knows the names of the seven souls we mourn","tokens":[50364,440,912,14181,567,5288,264,6105,611,3255,264,5288,295,264,3407,16588,321,22235,77,50724]},{"id":34,"start":173600000000,"end":175600000000,"text":" today.","tokens":[965,13,50824]},{"id":35,"start":175600000000,"end":183160000000,"text":" The crew of the shuttle Columbia did not return safely to earth yet we can pray that all","tokens":[440,7260,295,264,26728,17339,630,406,2736,11750,281,4120,1939,321,393,3690,300,439,51202]},{"id":36,"start":183160000000,"end":185840000000,"text":" are safely home.","tokens":[366,11750,1280,13,51336]},{"id":37,"start":185840000000,"end":192600000000,"text":" May God bless the grieving families and may God continue to bless America.","tokens":[1891,1265,5227,264,48454,4466,293,815,1265,2354,281,5227,3374,13,51674]},{"id":38,"start":196400000000,"end":206400000000,"text":" [BLANK_AUDIO]","tokens":[50364,542,37592,62,29937,60,50864]}],
  "text":"My fellow Americans, this day has brought terrible news and great sadness to our country. At 9 o'clock this morning, Mission Control and Houston lost contact with our Space Shuttle Columbia. A short time later, debris was seen falling from the skies above Texas. The Columbia's lost. There are no survivors. And board was a crew of seven. Colonel Rick Husband, Lieutenant Colonel Michael Anderson, Commander Laurel Clark, Captain David Brown, Commander William McCool, Dr. Cooltna Chavla, and Elon Ramon, a Colonel in the Israeli Air Force. These men and women assumed great risk in the service to all humanity. And an age when Space Flight has come to seem almost routine. It is easy to overlook the dangers of travel by rocket and the difficulties of navigating the fierce outer atmosphere of the Earth. These astronauts knew the dangers and they faced them willingly. Knowing they had a high and noble purpose in life. Because of their courage and daring and idealism, we will miss them all the more. All Americans today are thinking as well of the families of these men and women who have been given this sudden shock in grief. You're not alone. Our entire nation agrees with you. And those you loved will always have the respect and gratitude of this country. The cause in which they died will continue. Man kind is led into the darkness beyond our world by the inspiration of discovery and the longing to understand. Our journey into space will go on. In the skies today, we saw destruction and tragedy. As farther than we can see, there is comfort and hope. In the words of the prophet Isaiah, lift your eyes and look to the heavens who created all these. He who brings out the story hosts one by one and calls them each by name because of his great power and mighty strength. Not one of them is missing. The same creator who names the stars also knows the names of the seven souls we mourn today. The crew of the shuttle Columbia did not return safely to earth yet we can pray that all are safely home. May God bless the grieving families and may God continue to bless America. [BLANK_AUDIO]"
}

---

You can also specify the response_format parameter to be one of lrc, srt, vtt, text, json or verbose_json (default):

## Send the example audio file to the transcriptions endpoint
curl http://localhost:8080/v1/audio/transcriptions -H "Content-Type: multipart/form-data" -F file="@$PWD/gb1.ogg" -F model="whisper-1" -F response_format="srt"

Result (first few lines):

1
00:00:00,000 --> 00:00:09,640
My fellow Americans, this day has brought terrible news and great sadness to our country.

2
00:00:09,640 --> 00:00:15,960
At 9 o'clock this morning, Mission Control and Houston lost contact with our Space Shuttle

3
00:00:15,960 --> 00:00:16,960
Columbia.

4
00:00:16,960 --> 00:00:24,640
A short time later, debris was seen falling from the skies above Texas.

5
00:00:24,640 --> 00:00:27,200
The Columbia's lost.

6
00:00:27,200 --> 00:00:29,920
There are no survivors.

Supported request parameters

In addition to file and model, the endpoint accepts the following multipart form fields, matching the OpenAI audio transcription API:

Field	Description
language	ISO-639-1 language hint (e.g. en). Passed through to the backend.
prompt	Optional context hint to bias the decoder.
temperature	Sampling temperature (float). Honored by backends that support it.
timestamp_granularities[]	Multi-value form field: word and/or segment. Honored when the backend produces the requested granularity.
response_format	One of json (default for backwards-compat), verbose_json, text, srt, vtt, lrc.
stream	When true, the endpoint emits an SSE stream of transcript.text.delta events followed by a final transcript.text.done event.
diarize	LocalAI extension - speaker diarization. WhisperX requires HF_TOKEN; requests fail with FailedPrecondition when it is missing.

If speaker diarization fails after transcription succeeded, the WhisperX backend logs the error and returns the transcript without speaker labels. Other transcription failures return an error instead of an empty transcript. Diarization still requires HF_TOKEN.

The response body for verbose_json includes text, language, duration, and segments[] (with speaker populated when diarization is enabled).

Streaming transcriptions

Set -F stream=true to receive token-by-token SSE events as the backend produces them. The event shape matches the OpenAI streaming transcription format:

curl -N http://localhost:8080/v1/audio/transcriptions \
  -H "Content-Type: multipart/form-data" \
  -F file="@sample.wav" \
  -F model="whisper-1" \
  -F stream=true

data: {"type":"transcript.text.delta","delta":"And so, my"}

data: {"type":"transcript.text.delta","delta":" fellow Americans..."}

data: {"type":"transcript.text.done","text":"And so, my fellow Americans..."}

data: [DONE]

Backends that do not natively stream tokens fall back to emitting one delta plus a done event with the full text - the SSE contract is identical either way.

Using the llama-cpp backend with an audio-capable model

Any GGUF model whose mmproj contains an audio encoder can be used for transcription via the llama-cpp backend. This reuses the model's own audio front-end rather than shelling out to whisper.cpp, which is useful when you want a single backend serving both chat-with-audio and transcription.

Example using ggml-org/Qwen3-ASR-0.6B-GGUF:

name: qwen3-asr
backend: llama-cpp
parameters:
  model: Qwen3-ASR-0.6B-Q8_0.gguf
mmproj: mmproj-Qwen3-ASR-0.6B-Q8_0.gguf

Then call /v1/audio/transcriptions as usual:

curl http://localhost:8080/v1/audio/transcriptions \
  -H "Content-Type: multipart/form-data" \
  -F file="@jfk.wav" \
  -F model="qwen3-asr"

LFM3-Audio (experimental)

The gallery entries lfm3-audio-2b and lfm3-audio-2b-q8 also transcribe through the llama-cpp backend, using the same model as speech-to-speech and TTS. When a language is given, the request asks for transcription in that language with the system prompt Perform ASR in <Language>.. Without one, the model is asked to infer the language. Transcription uses the second slot (parallel:2), so it can run next to a realtime session. A realtime session does not use it for the input transcript by default; see [Realtime]({{% relref "openai-realtime" %}}). This needs a llama.cpp build with LFM3-Audio support.

Using the parakeet-cpp backend

parakeet.cpp is a C++/ggml port of NVIDIA NeMo Parakeet that matches the upstream PyTorch models on CPU. GGUF weights for every model and quant are published in a single repo, mudler/parakeet-cpp-gguf. F16 is the recommended default, and Q4_K stays near-lossless on the small models. The easiest path is to import directly (the GGUFs auto-detect to this backend):

local-ai models import https://huggingface.co/mudler/parakeet-cpp-gguf/resolve/main/tdt_ctc-110m-f16.gguf

Or write a model YAML:

name: parakeet-110m
backend: parakeet-cpp
parameters:
  model: tdt_ctc-110m-f16.gguf

Then call /v1/audio/transcriptions as usual. Pass timestamp_granularities[]=word for per-word timings:

curl http://localhost:8080/v1/audio/transcriptions \
  -H "Content-Type: multipart/form-data" \
  -F file="@jfk.wav" \
  -F model="parakeet-110m" \
  -F "timestamp_granularities[]=word"

For real-time use, load a cache-aware streaming model (e.g. realtime_eou_120m-v1-*.gguf) and pass -F stream=true. Deltas are emitted as the audio is decoded, with end-of-utterance events closing each segment.

Diarization and sound classification

The same backend also serves the /v1/audio/diarization and /v1/audio/classification endpoints, and can attach a diarization or sound model to a live transcription session. options: accepts paths relative to the models directory, or absolute:

Option	Allowed on	Used for
asr_model:<path>	a diarization model	include_text on /v1/audio/diarization
diarization_model:<path>	an ASR model	a speaker on transcript segments (and words), and speaker segments during realtime live transcription
sound_model:<path>	an ASR or diarization model	sound events during realtime live transcription, sounds on /v1/audio/diarization with include_sounds=true, and /v1/audio/classification on the loaded model
diarization_latency:<model|low|very_low|ultra_low>	a model with a diarization companion	latency mode for the live speaker stream; default low
speaker_model:<path>	a model with a diarization model	names registered speakers; a bundle can use speaker_component:<name> instead (see Bundle GGUF files) (see [Voice Recognition]({{% relref "voice-recognition" %}}#naming-speakers-in-diarization-and-live-transcription))
speaker_tag:<tag>	a model with speaker_component	extra encoder tag for registered voices that carry only a file-name tag (see [Voice Recognition]({{% relref "voice-recognition" %}}#naming-speakers-from-a-bundle))
speaker_threshold:<float>	a model with speaker_model	distance (1 minus cosine similarity) under which a speaker is named, in (0, 2); default 0.5
speaker_margin:<float>	a model with speaker_model	how much the best match must beat the runner-up, in [0, 1); default 0.05
speaker_strict:<bool>	a model with speaker_model	do not use registered voices that have no encoder fingerprint (see [Voice Recognition]({{% relref "voice-recognition" %}}#encoder-fingerprint)); default false

With a diarization_model companion, /v1/audio/transcriptions labels each segment with its speaker ("0", "1", ... in order of first appearance) and splits segments where the speaker changes; with timestamp_granularities[]=word each word carries its speaker too. With stream=true the closing transcript.text.done event lists the segments with their speakers. Pass -F diarize=false to skip diarization for one request. The diarization GGUF can also be imported directly: local-ai models import https://huggingface.co/mudler/parakeet-cpp-gguf/resolve/main/nemotron-3-diarization-f16.gguf.

speaker_model: needs libparakeet with C-API v10. A wrong setup fails at load time with one of these errors: parakeet-cpp: speaker_model needs libparakeet.so ABI 10 (parakeet_capi_speaker_registry_add_embedding); the loaded library is older, parakeet-cpp: speaker_model needs a diarization model (the primary or diarization_model:), parakeet-cpp: a speaker model cannot be the primary model; use it as speaker_model: next to a diarization model, parakeet-cpp: speaker_model "<path>" is a <kind> model, expected a speaker model (the file is not a speaker encoder GGUF), or parakeet-cpp: speaker_threshold "<value>" must be a distance in (0, 2) (1 minus cosine similarity) / parakeet-cpp: speaker_margin "<value>" must be a number in [0, 1) for a bad number.

The loader rejects a companion whose role duplicates the primary's own (for example asr_model: on an already-ASR primary, or sound_model: on a CED primary), and rejects a companion GGUF that does not match the role its option names (for example sound_model: pointing at an ASR GGUF fails to load, naming the kind it expected). See [Speaker Diarization]({{% relref "audio-diarization" %}}) for the Diarize RPC and [Sound Classification]({{% relref "audio-classification" %}}) for SoundDetection, and [Realtime API]({{% relref "openai-realtime" %}}) for the live speaker/sound events emitted during a realtime session.

Segment timestamps

Transcriptions are split into segments the same way NVIDIA NeMo does: a new segment starts after sentence-ending punctuation (., ?, !), and each segment carries start/end times. This is the default (NeMo's punctuation-only segmentation) and needs no configuration. While streaming, each end-of-utterance closes a segment, now with timestamps.

You can additionally split on silence by setting segment_gap_threshold (NeMo's segment_gap_threshold, in encoder frames; off by default). When set, a gap between two words wider than the threshold also starts a new segment. The value is in frames to match NeMo exactly; the backend converts it to seconds using the model's frame stride (frame_sec, reported by the engine):

name: parakeet-110m
backend: parakeet-cpp
parameters:
  model: tdt_ctc-110m-f16.gguf
options:
- segment_gap_threshold:12   # split on silence > 12 encoder frames (default 0 = off, punctuation-only)

Dynamic batching

The backend can coalesce concurrent transcription requests into a single batched engine call, which improves throughput on GPU when many requests arrive at once. Batching is off by default (batch_max_size:1, one request at a time); raise it to opt in. Two options: knobs control it:

name: parakeet-110m
backend: parakeet-cpp
parameters:
  model: tdt_ctc-110m-f16.gguf
options:
- batch_max_size:8      # max requests coalesced into one batch (default 1 = off)
- batch_max_wait_ms:15  # how long to wait to fill a batch, in ms (default 15)

By default each request runs on its own. Raise batch_max_size (for example 4 to 16) to enable batching; it pays off on GPU under concurrent load, where coalescing the per-step decode GEMMs across requests is a large throughput win. Leave it at 1 on CPU and for low-concurrency setups, where batching only adds latency. Batching only affects concurrent unary requests; streaming sessions always run on their own.

Moondream Ultra and Redux

Moondream publishes two derivatives of NVIDIA parakeet-tdt-0.6b-v3, Ultra and Redux. Both have a voice-activity-detection (VAD) head. The gallery has five entries, built from the GGUFs in mudler/parakeet-cpp-gguf:

Gallery entry	File	Runs on
parakeet-cpp-moondream-ultra-f16	ultra-f16.gguf	CPU and GPU
parakeet-cpp-moondream-ultra-q8_0	ultra-q8_0.gguf	CPU and GPU
parakeet-cpp-moondream-redux-packed	redux-packed.gguf	CPU only, offline only
parakeet-cpp-moondream-redux-f16	redux-f16.gguf	any backend, can stream
parakeet-cpp-moondream-redux-q8_0	redux-q8_0.gguf	any backend

The packed Redux file stores the encoder as ternary weights (213 MB). It cannot load on a GPU backend and cannot stream. If a GPU build fails to load it, check the backend log for the library message and use the redux-f16 or redux-q8_0 entry instead. The weights are CC-BY-4.0: credit Moondream and NVIDIA.

VAD-only slices

If you only need the VAD head, for the [VAD endpoint]({{%relref "features/voice-activity-detection" %}}) or to cut audio before transcription, the same repository has two small files with the head cut out of the full model. The weights are not retrained, and the files cannot transcribe:

Gallery entry	File	Size	Cut from
parakeet-cpp-vad-moondream-redux	redux-vad.gguf	9.9 MB	Redux (213 MB packed to 1.4 GB)
parakeet-cpp-vad-moondream-ultra	ultra-vad-q8_0.gguf	6.0 MB	Ultra Q8_0

Measured by the parakeet.cpp author against loading a whole Redux or Ultra model: the files are 6 to 10 MB instead of 213 MB to 1.4 GB, load in a few milliseconds instead of 0.1 to 0.7 s, and use about 245 MiB peak memory for a 33 s clip instead of 0.6 to 1.6 GiB. The output is byte-identical to the full parent model, and the speed is the same as the parent's head. A transcription request on a slice fails with an error. The slices load only with a parakeet.cpp build that includes VAD-only GGUF support, so an older backend build fails to load them. The weights are CC-BY-4.0: credit Moondream and NVIDIA.

With vad:true, long audio is cut at pauses found by the model's VAD head into pieces of at most 30 seconds, and each piece is transcribed in turn. Word timestamps stay relative to the whole file. Audio of 30 seconds or less gives the same result as without the option. The gallery entries set it. Add it to your own model YAML like this:

name: moondream-ultra
backend: parakeet-cpp
parameters:
  model: ultra-q8_0.gguf
options:
- vad:true   # cut long audio at pauses (default false); needs a model with a VAD head

vad:true applies to offline transcription only and bypasses dynamic batching, because the batched entry point has no VAD variant. Streaming is not affected. A model without a VAD head fails each request with model has no VAD head, and a libparakeet.so that is too old to export the VAD entry point fails the load. Remove the option for models that have no VAD head.

Cutting long audio with Silero (vad_model)

A model without a VAD head, such as parakeet-cpp-tdt-0.6b-v3 or a Nemotron model, can cut long audio with Silero VAD instead. Name a Silero GGUF in the vad_model option. The path is resolved against the models directory, like the other companion files. vad_model implies vad:

name: parakeet-v3-silero
backend: parakeet-cpp
parameters:
  model: parakeet-cpp/tdt-0.6b-v3-f16.gguf
options:
- vad_model:parakeet-cpp/silero-vad-f16.gguf   # Silero GGUF that cuts long audio at pauses
- vad_min_pause:0.3                            # optional, seconds

The gallery entry parakeet-cpp-tdt-0.6b-v3-silero-vad installs both files with this configuration. Audio of 30 seconds or less is transcribed whole and the VAD does not run. vad:true alone keeps meaning "use the model's own head". With vad_model set, the Silero model is used even if the ASR model has a head.

The segmenter options below apply to both vad:true and vad_model. Each is optional; an unset value keeps the default of the detector in use, and a bad value fails the load:

Option	Unit	Meaning
vad_threshold	0 to 1	A frame is speech when its probability is at least this
vad_min_pause	seconds	A silence this long separates two pieces
vad_min_speech	seconds	Shorter speech runs are dropped
vad_max_segment	seconds	Cap on the length of a piece (default 30)
vad_trim	seconds	Each piece shrinks to its first and last speech frame plus this much. Default 0.3; 0 keeps the whole cuts, as before this option existed

vad_speech_pad (seconds) pads each region and only affects the [VAD endpoint]({{%relref "features/voice-activity-detection" %}}). vad_model needs a libparakeet.so that exports parakeet_capi_transcribe_path_json_vad_with; an older library fails the load with a message that names it.

Dropping noise words (guard_*)

A decode of noise or silence can contain words that no one said. An opt-in filter removes the words that stand alone or sit among low-confidence words, and the words that are only punctuation. It runs on the finished decode. It is off unless one of these options is set, and a bad value fails the load:

Option	Unit	Meaning
guard_min_local_conf	0 to 1	A word is dropped when the mean confidence of the words that start within guard_local_radius seconds of it, itself included, is below this. 0 is off. 0.5 is a good start; higher values also drop real words on some models
guard_local_radius	seconds, above 0	The window of that mean. Default 5
guard_drop_punct_only	true or false	Drop words that are only punctuation. A CTC model can emit a lone . on noise. Default false

options:
- guard_min_local_conf:0.5
- guard_drop_punct_only:true

The filter applies to offline transcription, and with vad:true or vad_model to each piece on its own. It bypasses dynamic batching, like vad:true. Streaming is not affected. Speech with confident words comes out the same as without the filter. The number of dropped words is written to the backend log at debug level; the transcription response has no field for it. The options need a libparakeet.so that exports parakeet_capi_transcribe_path_json_with; an older library fails the load with a message that names it.

Bundle GGUF files (several models in one file)

A bundle is one GGUF file that holds several models, called components. Each component keeps its own licence. The backend opens the components it needs from the one file, so a single model YAML can serve transcription, VAD, diarization, speaker naming and sound events. A bundle needs a libparakeet.so from parakeet.cpp with bundle support (pin 781a973 or newer); the format is described in the parakeet.cpp bundle documentation. Single-model files and every existing option work as before.

The gallery has three bundles, built from mudler/parakeet-cpp-gguf:

Gallery entry	Size	Components	Serves
parakeet-cpp-bundle-small	338 MB	Parakeet TDT+CTC 110M (Q8_0), Nemotron-3-Diarization (Q8_0), CED-Small (Q8_0), WeSpeaker ResNet34-LM (F32), Silero VAD (F16)	transcription, VAD, diarization, speaker naming, sound events
parakeet-cpp-bundle-standard	1.1 GB	Parakeet TDT 0.6B v3 (Q8_0), plus the same four components	the same, with the multilingual 0.6B model
parakeet-cpp-bundle-moondream-redux	215 MB	Moondream Redux (packed ternary), Silero VAD (F16)	transcription and VAD; CPU only and offline only

The component that each role uses, and the option that picks another one:

Role	Component used	Option
Transcription	the only asr component	bundle_asr:<name> picks one when the bundle has several
VAD (/v1/vad, and vad:true for long audio)	the vad (Silero) component, loaded with no option; without one the VAD head of the ASR model	vad_component:<name> picks one, and implies vad:true
Diarization	the diar component, only when asked for	diar_component:<name>
Sound events	the ced component, only when asked for	sound_component:<name>
Speaker naming	the voice component, only when asked for	speaker_component:<name> (needs a diarization component)
Voice embedding and verification (/v1/voice/*, the realtime voice_recognition stage)	the voice component, only when asked for	speaker_component:<name>; declare speaker_recognition in known_usecases. Needs a libparakeet.so with parakeet_capi_speaker_embed_pcm. See [Voice Recognition]({{% relref "voice-recognition#a-parakeet-cpp-bundle-as-the-embedding-model" %}})

speaker_component: also names registered speakers: LocalAI sends the voices from /v1/voice/register to the bundle's speaker component, as it does for speaker_model:. This works in /v1/audio/diarization and in realtime live transcription, and it needs no speaker_model:. speaker_threshold, speaker_margin and speaker_strict apply too. Only voices that carry an encoder fingerprint (voices enrolled from speaker_profiles) and voices with no tag at all are used by default. A voice registered with a tag only matches through speaker_tag:. See [Voice Recognition]({{% relref "voice-recognition" %}}#naming-speakers-from-a-bundle).

A *_component option without the matching companion option takes the component from the model file itself. The companion options (diarization_model:, sound_model:, speaker_model:, vad_model:, asr_model:) can also name a bundle file, even the same file as the model: the only component of the wanted kind is used, and the *_component option picks one when there are several. This YAML loads the same file for four roles:

name: parakeet-bundle
backend: parakeet-cpp
parameters:
  model: parakeet-cpp/parakeet-bundle-small.gguf
options:
- vad:true                  # cut long audio at pauses, with the Silero component
- diar_component:diar       # same as diarization_model:parakeet-cpp/parakeet-bundle-small.gguf
- sound_component:ced
- speaker_component:voice

A role that the bundle cannot fill fails with a message that lists the components, for example parakeet-cpp: diarization_model needs a "diar" component, but the bundle "<path>" has none (components: asr (asr), vad (vad)). A request for a role the loaded model does not have (diarization from a bundle without a diar component) returns parakeet-cpp: model is not a diarization model (the model file is a bundle without a "diar" component; ...). A *_component option on a file that is not a bundle, or on a library without bundle support, fails the load.

Licences: a bundle has no single licence, so the gallery entries use license: other. The licence, source and credit of every component are in the file header, and the NOTICE file next to each bundle in the repository (NOTICE-<bundle>.txt) has the credits and the full licence texts. Keep it with any copy of the file you pass on.

Component	Licence	Credit
Parakeet TDT+CTC 110M, Parakeet TDT 0.6B v3	CC-BY-4.0	NVIDIA
Moondream Redux	CC-BY-4.0	Moondream, derived from Parakeet TDT 0.6B v3 by NVIDIA
Nemotron-3-Diarization	OpenMDW-1.1	NVIDIA
CED-Small	Apache-2.0, as stated on the model card	Heinrich Dinkel et al., Xiaomi (mispeech)
WeSpeaker ResNet34-LM	CC-BY-4.0	the WeSpeaker project
Silero VAD	MIT	Silero Team

The licence of the CED weights is not consistent upstream: the model card says Apache-2.0, the upstream code repository is GPL-3.0 and the original checkpoint records say CC-BY-4.0. The file here is converted, not trained, and follows the model card. The weights of all components were converted to GGUF, and quantised where the table shows it; nothing was retrained.

See also

• [Audio Transform]({{< relref "audio-transform.md" >}}) - clean up the audio (echo cancellation, noise suppression, dereverberation) before passing it to a transcription model.