Model reference · open weights

speaker-diarization-3.1

Available as managed deployment Audio tensorlake Speech→text 1 variants 717 dl/mo

speaker-diarization-3.1 is an open-weight audio or speech model from tensorlake. AxForge deploys and operates it for you on dedicated EU-owned hardware — with the licence handled where one is required.

Available as managed deployment — configured and operated for you on dedicated EU hardware, quoted per deployment.

What it is

Released bytensorlake
TypeAudio & music
TaskSpeech→text
Runs withpyannote-audio
Released2024-07-25
Popularity717 downloads / month
LicenceOpen weights

About

What speaker-diarization-3.1 is

Using this open-source model in production? Consider switching to pyannoteAI for better and faster options.

Read the full model card

🎹 Speaker diarization 3.1

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.

Requirements

  1. Install pyannote.audio 3.1 with pip install pyannote.audio
  2. Accept pyannote/segmentation-3.0 user conditions
  3. Accept pyannote/speaker-diarization-3.1 user conditions
  4. Create access token at hf.co/settings/tokens.

Usage

# instantiate the pipeline
from pyannote.audio import Pipeline
pipeline = Pipeline.from_pretrained(
  "pyannote/speaker-diarization-3.1",
  use_auth_token="HUGGINGFACE_ACCESS_TOKEN_GOES_HERE")

# run the pipeline on an audio file
diarization = pipeline("audio.wav")

# dump the diarization output to disk using RTTM format
with open("audio.rttm", "w") as rttm:
    diarization.write_rttm(rttm)

Processing on GPU

pyannote.audio pipelines run on CPU by default. You can send them to GPU with the following lines:

import torch
pipeline.to(torch.device("cuda"))

Processing from memory

Pre-loading audio files in memory may result in faster processing:

waveform, sample_rate = torchaudio.load("audio.wav")
diarization = pipeline({"waveform": waveform, "sample_rate": sample_rate})

Monitoring progress

Hooks are available to monitor the progress of the pipeline:

from pyannote.audio.pipelines.utils.hook import ProgressHook
with ProgressHook() as hook:
    diarization = pipeline("audio.wav", hook=hook)

Controlling the number of speakers

In case the number of speakers is known in advance, one can use the num_speakers option:

diarization = pipeline("audio.wav", num_speakers=2)

One can also provide lower and/or upper bounds on the number of speakers using min_speakers and max_speakers options:

diarization = pipeline("audio.wav", min_speakers=2, max_speakers=5)

Benchmark

This pipeline has been benchmarked on a large collection of datasets.

Processing is fully automatic:

  • no manual voice activity detection (as is sometimes the case in the literature)
  • no manual number of speakers (though it is possible to provide it to the pipeline)
  • no fine-tuning of the internal models nor tuning of the pipeline hyper-parameters to each dataset

... with the least forgiving diarization error rate (DER) setup (named "Full" in this paper):

  • no forgiveness collar
  • evaluation of overlapped speech
BenchmarkDER%FA%Miss%Conf%Expected outputFile-level evaluation
AISHELL-412.23.84.44.0RTTMeval
AliMeeting (channel 1)24.44.410.010.0RTTMeval
AMI (headset mix, only_words)18.83.69.5

From the published model card. Full card on the HuggingFace links in the sidebar.

How it works

How audio & music work

Audio or textinputAudio modelrecognise / synthesiseText or audiooutputSpeech-to-text turns audio into text; text-to-speech and music models turn text into audio.

Using it via the API

Call it like any OpenAI endpoint

Once AxForge deploys tensorlake-speaker-diarization-3-1 for you, it answers on the OpenAI-compatible API — the same base URL and keys as every other model. (tensorlake-speaker-diarization-3-1 below is illustrative; you get the exact model name on deployment.)

$ curl -sS https://api.axforge.ai/v1/audio/transcriptions \
  -H "Authorization: Bearer $AXFORGE_API_KEY" \
  -F model="tensorlake-speaker-diarization-3-1" -F file=@audio.mp3

Create an account — your API key is available in the console. 3M free tokens every 30 days with every new account.

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