Skip to page content
SAVRN
Search Plan a project

Open-weight model · Speech recognition

parakeet-ultra-glade

by Josh Leverette coder543/parakeet-ultra-glade

parakeet-ultra-glade is an open-weight model for speech recognition from Josh Leverette, released under Creative Commons Attribution 4.0. Its published files total 1.6 GB.

Glade is a native Swift library for on-device speech inference on Apple silicon. This repository supplies converted model assets and configuration for Glade's model-specific host runtime. Glade's runtime source is currently private.

Parameters—
Context—
Weights1.6 GB
Licensecc-by-4.0
AccessOpen weights
Monthly Downloads—

Model Card

By Josh Leverette, published under cc-by-4.0, revision 3343192ceabf.

Glade is a native Swift library for on-device speech inference on Apple silicon. This repository supplies converted model assets and configuration for Glade's model-specific host runtime. Glade's runtime source is currently private. Glade conversion of Moondream Parakeet Ultra, a post-trained NVIDIA Parakeet TDT 0.6B V3 model. Source revision: 73175eb7aeb0d82f1e2a6b53b3aabc10a90bcd0b. The weights retain the upstream CC-BY-4.0 license. See LICENSE and NOTICE. This conversion is not endorsed by Moondream or NVIDIA. The repository contains one self-contained bundle. It preserves the model's native speech head to find pauses, selecting chunks of at most 30 seconds. All audio is retained…

Read Josh Leverette's full model card

Parakeet Ultra for Glade

Glade is a native Swift library for on-device speech inference on Apple silicon. This repository supplies converted model assets and configuration for Glade's model-specific host runtime. Glade's runtime source is currently private.

Glade conversion of Moondream Parakeet Ultra, a post-trained NVIDIA Parakeet TDT 0.6B V3 model. Source revision: 73175eb7aeb0d82f1e2a6b53b3aabc10a90bcd0b. The weights retain the upstream CC-BY-4.0 license. See LICENSE and NOTICE. This conversion is not endorsed by Moondream or NVIDIA.

The repository contains one self-contained bundle. It preserves the model's native speech head to find pauses, selecting chunks of at most 30 seconds. All audio is retained, including silence. The speech head runs on CPU over ANE subsampling features; the acoustic encoder and batched TDT decoder prefer ANE. No separate VAD model is needed.

Requires physical Apple silicon with macOS 27 or iOS 27 and Glade's model-specific host runtime. These graphs accept features and recurrent states, not audio files. The host implements the frontend, chunk planning, greedy TDT loop and tokenizer using the supplied metadata and sidecars.

Runtime configuration

  • W8A16 encoder with selected sensitive projections retained in FP16; FP16 batched decoder. Original attention and subsampling masks are preserved.
  • One 384-position encoder shape, four concurrent encoder requests and a decoder batch capacity of 128 independent chunks.
  • The native speech head uses bounded 120-second normalization blocks. The chunker selects pauses within the 30-second limit.
  • metadata.json describes graph conventions; runtime.json owns chunking and scheduling. Its internal fast profile selects the ANE decoder; it does not identify a second downloadable variant.
  • 8,193 vocabulary entries, blank ID 8,192, durations 0/1/2/3/4 and at most ten symbols per acoustic frame. Batched token IDs are represented as an exact FP16 partition/local pair: partition * 2048 + local.

The model retains V3's 25-language coverage. Native token/word timing uses TDT emission frames and predicted durations at an 80 ms frame step. The host must join subwords and attach punctuation without stretching word spans across silence. These are decoder alignments, not forced alignment.

Measured performance

JFK's “We choose to go to the Moon” speech: a 20-second excerpt and the 18-minute 15-second recording. Three-run warmed medians, excluding model preparation. Measurements were made September 30, 2026.

Device Audio Processing Audio / elapsed
M3 MacBook Air, 16 GB 20 seconds 0.192 s 104.4×
M3 MacBook Air, 16 GB 18 min 15 s 2.462 s 444.9×
iPhone 15 Pro Max 20 seconds 0.218 s 91.6×
iPhone 15 Pro Max 18 min 15 s 3.447 s 317.7×

The Mac timing includes frontend, encoder and decoding; file reading and chunk planning are excluded. Native VAD analysis took another 0.325 s for the full recording, yielding about 2.79 s / 393× including that analysis. Phone timing includes bounded file reading, VAD, inference and text delivery. Phone runs stayed in nominal thermal state and foreground throughout. This is one recording, not a general speed or accuracy ranking.

Peak measured full-recording memory was 1.27 GB on Mac (combined client and separately attributed ANE allocations) and 1.14 GB on iPhone (client footprint, which already includes its ANE allocation).

Device Observed initial preparation Subsequent cached preparation
M3 MacBook Air 7.26 s 0.094 s
iPhone 15 Pro Max 65.88 s 0.68–0.72 s

Preparation includes the runtime's required functions, not inference. The Mac already had related experimental assets cached; its 7.26-second result is not a clean-device compilation estimate. Core AI's underlying cache state is not fully observable, and OS updates or eviction can cause recompilation.

Qualification

  • The packaged Mac bundle and all three phone runs returned identical tokens and native-VAD boundaries for the complete recording, consuming every sample.
  • A separate complete phone trace recorded 278 ANE prediction intervals and zero app GPU intervals. All 39 encoder and 39 transcription subsampling scopes contained ANE work. Overlapping calls make per-stage hardware counts non-additive; this establishes placement, not ALU utilization.
  • On this recording, Whisper-normalized WER was 54/2,220 errors (2.43%), matching the source model's error count under the same frozen cut plan.
  • Seven public multilingual regression samples matched source token sequences, with native timing bounds checked. This is conversion coverage, not a comprehensive multilingual accuracy evaluation.
  • The quantized speech head is approximate: a 120-second comparison against the FP32 source had 0.00247 mean absolute probability error, with six of 1,500 frames crossing the 0.5 threshold. Native-VAD cuts are not claimed to reproduce every FP32 source boundary.

Download the complete repository snapshot. Source locations, evaluation audio, reference tensors and compiled device caches are excluded. Hugging Face's snapshot metadata supplies file hashes for integrity verification.

Experimental Core ML bundle for Apple Watch

coreml-watch/bundle.tar contains compiled watchOS 27 Core ML graphs, the host frontend and TDT decoder, runtime metadata, license files and loading.json with measured per-file loading estimates. Extract this uncompressed POSIX ustar archive into one model directory. The Core AI bundle remains available separately.

The public Core ML runtime requests CPU and Neural Engine execution with the default specialization strategy. Six encoder stages remain loaded; the frontend and TDT decoder run on CPU. The native speech head selects pauses for chunks up to 30 seconds, retaining all audio including silence. Recording replay uses up to eight independent chunks per decoder batch. Native word timings are available; they are not forced-aligned boundaries.

Measured on Apple Watch Ultra 4, watchOS 27.0.1, Release runtime, resident graphs, one warmup and three uninterrupted foreground runs without real-time pauses. The input is JFK’s “We choose to go to the Moon” speech.

Measurement Apple Watch Ultra 4
20-second excerpt 0.98 s · 20.3× real time
Complete 18.3-minute speech 42.7 s · 25.6× real time
Observed first preparation 85.4 s
Cached load in a new process 2.3 s
Download size 789.5 MB
Peak app footprint, full speech 197 MiB

Processing includes audio reads, the frontend, native VAD/chunk planning, encoder, decoder and transcript delivery; it excludes preparation, warmup and result serialization. Preparation is an observed first strategy-specific load with existing caches preserved, not a guaranteed globally cold measurement. Cache reuse is demonstrated across these two launches, not guaranteed after eviction or OS updates.

The full recording’s 39 native VAD cuts match the independent FP32 source. Three Watch runs produce identical tokens, and their transcripts match the source after lowercasing and removing punctuation. This single recording does not establish a general accuracy ranking. Full-recording hardware tracing confirmed ANE predictions for subsampling, native VAD and all six encoder stages. The separate Neural Engine allocation ledger reports about 1 GiB during inference; that accounting does not prove all bytes are physically resident at once.

Identity and Version

Repository
coder543/parakeet-ultra-glade
Publisher
Josh Leverette
Task
Speech recognition
Modality
Audio
Library
Not stated by the source
Parameters
Not stated by the source
Languages
Not stated by the source
Revision
3343192ceabf9298087e583f7264f1a99afac83b
First published
2026-09-30
Last updated
2026-10-09

Files and Weights

28 files, 1.6 GB in total.

Configuration11 files · 82.5 KB
Documentation3 files · 26.9 KB
Other13 files · 1.6 GB
Repository1 file · 1.9 KB
Every file
FileTypeSizeSHA-256
coreml-watch/distribution.jsonConfiguration151 B —
coreml-watch/loading.jsonConfiguration565 B —
decoder.aimodel/metadata.jsonConfiguration105 B —
decoder.jsonConfiguration931 B —
decoder_batch.aimodel/metadata.jsonConfiguration105 B —
encoder.aimodel/metadata.jsonConfiguration105 B —
metadata.jsonConfiguration972 B —
runtime.jsonConfiguration222 B —
subsampling.aimodel/metadata.jsonConfiguration105 B —
vad.jsonConfiguration611 B —
vocabulary.jsonConfiguration78.6 KB —
LICENSEDocumentation18.7 KB —
NOTICEDocumentation389 B —
README.mdDocumentation7.8 KB —
coreml-watch/bundle.tarOther789.5 MB ef60e55043f3
decoder.aimodel/main.hashOther32 B —
decoder.aimodel/main.mlirbOther42.4 MB 8a5f5f530ee9
decoder.f32Other126.2 MB 5c1c569b7e72
decoder_batch.aimodel/main.hashOther32 B —
decoder_batch.aimodel/main.mlirbOther21.2 MB 3b4d8816660c
encoder.aimodel/main.hashOther32 B —
encoder.aimodel/main.mlirbOther646.1 MB d389411b43e4
mel-filter.f32Other131.6 KB bce5ec5f194a
subsampling.aimodel/main.hashOther32 B —
subsampling.aimodel/main.mlirbOther4.3 MB 9ce3e21da992
vad.f32Other853.5 KB 32640ebd64d7
window.f32Other1.6 KB —
.gitattributesRepository1.9 KB —

License and Download

License
cc-by-4.0
Access
Open weights, no gate
Download from Josh Leverette

Released by Josh Leverette through its official repository on Hugging Face. Read the license.

Built From

  • Derived from moondream/parakeet-ultra
  • Quantized from moondream/parakeet-ultra

Questions About parakeet-ultra-glade

Can I use parakeet-ultra-glade commercially?

Yes. parakeet-ultra-glade is released under Creative Commons Attribution 4.0. CC BY 4.0 permits sharing and adapting the work, including commercially, provided the creator is credited and changes are indicated.

Similar Models

Model · Speech recognition

wav2vec2-large-xlsr-53-japanese

Jonatas Grosman

Fine-tuned facebook/wav2vec2-large-xlsr-53 on Japanese using the train and validation splits of Common Voice 6.1, CSS10 and JSUT. When using this model, make sure that your speech input is sampled at 16kHz. This model has been fine-tuned thanks to the GPU credits generously given by the OVHcloud:) The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint The model can be used directly (without a language model) as follows... Using the HuggingSound library: The model can be evaluated as follows on the Japanese test data of Common Voice. In the table below I report the Word Error Rate (WER) and the Character Error Rate (CER) of the model. I ran the…

Open weights apache-2.0 transformers

Model · Speech recognition

whisperkit-coreml

Argmax

WhisperKit is part of Argmax OSS, an On-device Speech AI SDK for Apple Silicon: https://github.com/argmaxinc/argmax-oss-swift Check out the WhisperKit paper and presentation from ICML 2025: https://icml.cc/virtual/2025/47854 For real-time transcription with speakers and custom vocabulary, check out Argmax Pro SDK: https://www.argmaxinc.com/blog/argmax-sdk-2

Open weights mit whisperkit

Model · Speech recognition

speaker-diarization-3.1

Pyannote

Using this open-source model in production? Consider switching to pyannoteAI for better and faster options. This pipeline is the same as pyannote/speaker-diarization-3.0 except it removes the problematic use of onnxruntime. Both speaker segmentation and embedding now run in pure PyTorch. This should ease deployment and possibly speed up inference. It requires pyannote.audio version 3.1 or higher. It ingests mono audio sampled at 16kHz and outputs speaker diarization as an Annotation instance: - stereo or multi-channel audio files are automatically downmixed to mono by averaging the channels. - audio files sampled at a different rate are resampled to 16kHz automatically upon loading. 1.…

Access requested at publisher mit pyannote-audio

Model · Speech recognition

speaker-diarization-community-1

Pyannote

This pipeline ingests mono audio sampled at 16kHz and outputs speaker diarization. - stereo or multi-channel audio files are automatically downmixed to mono by averaging the channels. - audio files sampled at a different rate are resampled to 16kHz automatically upon loading. The main improvements brought by Community-1 are: - improved speaker assignment and counting - simpler reconciliation with transcription timestamps with exclusive speaker diarization - easy offline use (i.e. without internet connection) - (optionally) hosted on pyannoteAI cloud 1. pip install pyannote.audio 3. Create access token at hf.co/settings/tokens. Out of the box, Community-1 is much better than…

Access requested at publisher cc-by-4.0 pyannote-audio

Model · Speech recognition

faster-whisper-small

Systran

This repository contains the conversion of openai/whisper-small to the CTranslate2 model format. This model can be used in CTranslate2 or projects based on CTranslate2 such as faster-whisper. The original model was converted with the following command: Note that the model weights are saved in FP16. This type can be changed when the model is loaded using the computetype option in CTranslate2. For more information about the original model, see its model card.

Open weights mit ctranslate2

Fine-tuned facebook/wav2vec2-large-xlsr-53 on Portuguese using the train and validation splits of Common Voice 6.1. When using this model, make sure that your speech input is sampled at 16kHz. This model has been fine-tuned thanks to the GPU credits generously given by the OVHcloud:) The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint The model can be used directly (without a language model) as follows... Using the HuggingSound library: 1. To evaluate on mozilla-foundation/commonvoice60 with split test 2. To evaluate on speech-recognition-community-v2/devdata If you want to cite this model you can use this

Open weights apache-2.0 transformers