Update README.md (#1)
Browse files- Update README.md (3602167b6cdaf119e0768eb5ba539bf66eb0cac6)
Co-authored-by: Ivan Medennikov <[email protected]>
README.md
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@@ -12,7 +12,7 @@ datasets:
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- dihard_challenge-3-dev
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- NIST_SRE_2000-Disc8_split1
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- Alimeeting-train
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thumbnail: null
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tags:
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- speaker-diarization
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name:
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type: dihard3-eval-1to4spks
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config: with_overlap_collar_0.0s
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input_buffer_lenght: 1.04s
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split: eval
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metrics:
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- name: Test DER
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type: der
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name:
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type: CALLHOME-part2-2spk
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config: with_overlap_collar_0.25s
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input_buffer_lenght: 1.04s
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: CALLHOME (NIST-SRE-2000 Disc8)
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type: CALLHOME-part2-3spk
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config: with_overlap_collar_0.25s
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input_buffer_lenght: 1.04s
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: CALLHOME (NIST-SRE-2000 Disc8)
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type: CALLHOME-part2-4spk
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config: with_overlap_collar_0.25s
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input_buffer_lenght: 1.04s
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- name: Test DER
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type: der
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value: 12.40
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- task:
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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<img src="figures/sortformer_intro.png" width="750" />
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</div>
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Streaming Sortformer approach employs an Arrival-Order Speaker Cache (AOSC) to store frame-level acoustic embeddings of previously observed speakers.
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<div align="center">
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<img src="figures/streaming_sortformer_ani.gif" width="1400" />
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</div>
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</div>
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Aside from speaker-cache management part, streaming Sortformer follows the architecture of the offline version of Sortformer. Sortformer consists of an L-size (
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Speech Tasks (NEST)](https://arxiv.org/abs/2408.13106)[
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and two feedforward layers with 4 sigmoid outputs for each frame input at the top layer. More information can be found in the [Sortformer paper](https://arxiv.org/abs/
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<div align="center">
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<img src="figures/sortformer-v1-model.png" width="450" />
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## NVIDIA NeMo
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To train, fine-tune or perform diarization with Sortformer, you will need to install [NVIDIA NeMo](https://github.com/NVIDIA/NeMo)[
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```
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pip install git+https://github.com/NVIDIA/NeMo.git@main#egg=nemo_toolkit[asr]
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```
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## How to Use this Model
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The model is available for use in the NeMo Framework[
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### Loading the Model
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from nemo.collections.asr.models import SortformerEncLabelModel
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# load model
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diar_model = SortformerEncLabelModel.restore_from(restore_path="diar_streaming_sortformer_4spk-v2", map_location=torch.device('cuda'), strict=False)
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```
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### Input Format
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### Technical Limitations
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- The model operates in a
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- It can detect a maximum of 4 speakers; performance degrades on recordings with 5 and more speakers.
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- The model was trained on publicly available speech datasets, primarily in English. As a result:
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* Performance may degrade on non-English speech.
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* Performance may also degrade on out-of-domain data, such as recordings in noisy conditions.
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-
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## Datasets
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Sortformer was trained on a combination of
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All the datasets listed above are based on the same labeling method via [RTTM](https://web.archive.org/web/20100606092041if_/http://www.itl.nist.gov/iad/mig/tests/rt/2009/docs/rt09-meeting-eval-plan-v2.pdf) format. A subset of RTTM files used for model training are processed for the speaker diarization model training purposes.
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Data collection methods vary across individual datasets. For example, the above datasets include phone calls, interviews, web videos, and audiobook recordings. Please refer to the [Linguistic Data Consortium (LDC) website](https://www.ldc.upenn.edu/) or dataset webpage for detailed data collection methods.
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### Training Datasets (Real conversations)
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- Fisher English (LDC)
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- 2004-2010 NIST Speaker Recognition Evaluation (LDC)
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- Librispeech
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- AMI Meeting Corpus
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- VoxConverse-v0.3
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- ICSI
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- AISHELL-4
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- Third DIHARD Challenge Development (LDC)
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- 2000 NIST Speaker Recognition Evaluation, split1 (LDC)
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### Training Datasets (Used to simulate audio mixtures)
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- 2004-2010 NIST Speaker Recognition Evaluation (LDC)
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## Performance
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### Evaluation
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| **Dataset** | **DIHARD3-Eval** | **CALLHOME-part2** | **CALLHOME-part2** | **CALLHOME-part2** | **CH109** |
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|:------------------------------|:------------------:|:-------------------:|:-------------------:|:-------------------:|:------------------:|
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| **Number of Speakers** | ≤ 4 speakers | 2 speakers | 3 speakers | 4 speakers | 2 speakers |
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| **Collar (sec)** | 0.0s | 0.25s | 0.25s | 0.25s | 0.25s |
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| **Mean Audio Duration (sec)** | 453.0s | 73.0s | 135.7s | 329.8s | 552.9s |
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### Diarization Error Rate (DER)
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| DER **Input Buffer Length: 1.04s + DH3-dev Opt. PP** | **_13.32_** | - | - | - | - |
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| DER **Input Buffer Length: 1.04s + CallHome-part1 Opt. PP** | - | **_6.43_** | **_10.26_** | **_12.40_** | **_5.09_** |
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###
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| **Latency
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## NVIDIA Riva: Deployment
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## References
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[1] [Sortformer: Seamless Integration of Speaker Diarization and ASR by Bridging Timestamps and Tokens](https://arxiv.org/abs/2409.06656)
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[2] [
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## Licence
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License to use this model is covered by the [CC-BY-
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- dihard_challenge-3-dev
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- NIST_SRE_2000-Disc8_split1
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- Alimeeting-train
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- DiPCo
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thumbnail: null
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tags:
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- speaker-diarization
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: DIHARD III Eval (1-4 spk)
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type: dihard3-eval-1to4spks
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config: with_overlap_collar_0.0s
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input_buffer_lenght: 1.04s
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split: eval-1to4spks
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metrics:
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- name: Test DER
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type: der
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: DIHARD III Eval (5-9 spk)
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type: dihard3-eval-5to9spks
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config: with_overlap_collar_0.0s
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input_buffer_lenght: 1.04s
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split: eval-5to9spks
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metrics:
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- name: Test DER
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type: der
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value: 42.61
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- task:
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: DIHARD III Eval (full)
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type: dihard3-eval
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config: with_overlap_collar_0.0s
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input_buffer_lenght: 1.04s
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split: eval
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metrics:
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- name: Test DER
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type: der
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value: 18.97
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: CALLHOME (NIST-SRE-2000 Disc8) part2 (2 spk)
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type: CALLHOME-part2-2spk
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config: with_overlap_collar_0.25s
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input_buffer_lenght: 1.04s
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: CALLHOME (NIST-SRE-2000 Disc8) part2 (3 spk)
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type: CALLHOME-part2-3spk
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config: with_overlap_collar_0.25s
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input_buffer_lenght: 1.04s
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: CALLHOME (NIST-SRE-2000 Disc8) part2 (4 spk)
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type: CALLHOME-part2-4spk
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config: with_overlap_collar_0.25s
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input_buffer_lenght: 1.04s
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- name: Test DER
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type: der
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value: 12.40
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: CALLHOME (NIST-SRE-2000 Disc8) part2 (5 spk)
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type: CALLHOME-part2-5spk
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config: with_overlap_collar_0.25s
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input_buffer_lenght: 1.04s
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split: part2-5spk
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metrics:
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- name: Test DER
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type: der
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value: 24.41
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: CALLHOME (NIST-SRE-2000 Disc8) part2 (6 spk)
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type: CALLHOME-part2-6spk
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config: with_overlap_collar_0.25s
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input_buffer_lenght: 1.04s
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split: part2-6spk
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metrics:
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- name: Test DER
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type: der
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value: 27.78
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- task:
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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dataset:
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name: CALLHOME (NIST-SRE-2000 Disc8) part2 (full)
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type: CALLHOME-part2
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config: with_overlap_collar_0.25s
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input_buffer_lenght: 1.04s
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split: part2
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metrics:
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- name: Test DER
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type: der
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value: 10.79
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- task:
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name: Speaker Diarization
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type: speaker-diarization-with-post-processing
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<img src="figures/sortformer_intro.png" width="750" />
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</div>
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[Streaming Sortformer](https://arxiv.org/abs/25XX.XXXXX)[2] approach employs an Arrival-Order Speaker Cache (AOSC) to store frame-level acoustic embeddings of previously observed speakers.
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<div align="center">
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<img src="figures/streaming_sortformer_ani.gif" width="1400" />
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</div>
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</div>
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Aside from speaker-cache management part, streaming Sortformer follows the architecture of the offline version of Sortformer. Sortformer consists of an L-size (17 layers) [NeMo Encoder for
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Speech Tasks (NEST)](https://arxiv.org/abs/2408.13106)[3] which is based on [Fast-Conformer](https://arxiv.org/abs/2305.05084)[4] encoder. Following that, an 18-layer Transformer[5] encoder with hidden size of 192,
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and two feedforward layers with 4 sigmoid outputs for each frame input at the top layer. More information can be found in the [Streaming Sortformer paper](https://arxiv.org/abs/25XX.XXXXX)[2].
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<div align="center">
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<img src="figures/sortformer-v1-model.png" width="450" />
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## NVIDIA NeMo
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To train, fine-tune or perform diarization with Sortformer, you will need to install [NVIDIA NeMo](https://github.com/NVIDIA/NeMo)[6]. We recommend you install it after you've installed Cython and latest PyTorch version.
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```
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pip install git+https://github.com/NVIDIA/NeMo.git@main#egg=nemo_toolkit[asr]
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```
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## How to Use this Model
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The model is available for use in the NeMo Framework[6], and can be used as a pre-trained checkpoint for inference or for fine-tuning on another dataset.
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### Loading the Model
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from nemo.collections.asr.models import SortformerEncLabelModel
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# load model
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diar_model = SortformerEncLabelModel.restore_from(restore_path="/path/to/diar_streaming_sortformer_4spk-v2", map_location=torch.device('cuda'), strict=False)
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# switch to inference mode
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diar_model.eval()
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```
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### Input Format
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### Technical Limitations
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- The model operates in a streaming mode (online mode).
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- It can detect a maximum of 4 speakers; performance degrades on recordings with 5 and more speakers.
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- While the model is designed for long-form audio and can handle recordings that are several hours long, performance may degrade on very long recordings.
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- The model was trained on publicly available speech datasets, primarily in English. As a result:
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* Performance may degrade on non-English speech.
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* Performance may also degrade on out-of-domain data, such as recordings in noisy conditions.
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## Datasets
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Sortformer was trained on a combination of 2445 hours of real conversations and 5150 hours or simulated audio mixtures generated by [NeMo speech data simulator](https://arxiv.org/abs/2310.12371)[7].
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All the datasets listed above are based on the same labeling method via [RTTM](https://web.archive.org/web/20100606092041if_/http://www.itl.nist.gov/iad/mig/tests/rt/2009/docs/rt09-meeting-eval-plan-v2.pdf) format. A subset of RTTM files used for model training are processed for the speaker diarization model training purposes.
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Data collection methods vary across individual datasets. For example, the above datasets include phone calls, interviews, web videos, and audiobook recordings. Please refer to the [Linguistic Data Consortium (LDC) website](https://www.ldc.upenn.edu/) or dataset webpage for detailed data collection methods.
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### Training Datasets (Real conversations)
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- Fisher English (LDC)
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- AMI Meeting Corpus
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- VoxConverse-v0.3
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- ICSI
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- AISHELL-4
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- Third DIHARD Challenge Development (LDC)
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- 2000 NIST Speaker Recognition Evaluation, split1 (LDC)
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- DiPCo
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- AliMeeting
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### Training Datasets (Used to simulate audio mixtures)
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- 2004-2010 NIST Speaker Recognition Evaluation (LDC)
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## Performance
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### Evaluation data specifications
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| **Dataset** | **Number of speakers** | **Number of Sessions** |
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| 336 |
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|----------------------------|------------------------|------------------------|
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| 337 |
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| **DIHARD III Eval <=4spk** | 1-4 | 219 |
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| 338 |
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| **DIHARD III Eval >=5spk** | 5-9 | 40 |
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| 339 |
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| **DIHARD III Eval full** | 1-9 | 259 |
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| 340 |
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| **CALLHOME-part2 2spk** | 2 | 148 |
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| **CALLHOME-part2 3spk** | 3 | 74 |
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| 342 |
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| **CALLHOME-part2 4spk** | 4 | 20 |
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| **CALLHOME-part2 5spk** | 5 | 5 |
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| 344 |
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| **CALLHOME-part2 6spk** | 6 | 3 |
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| **CALLHOME-part2 full** | 2-6 | 250 |
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| 346 |
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| **CH109** | 2 | 109 |
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### Latency setups and Real Time Factor (RTF)
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* **Configuration Parameters**: Each setup is defined by its **Chunk Size**, **Right Context**, **FIFO Queue**, **Update Period**, and **Speaker Cache**. The value for each parameter represents the number of 80ms frames.
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* **Latency**: Refers to **Input Buffer Latency**, calculated as **Chunk Size** + **Right Context**. This value excludes computational processing time.
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* **Real-Time Factor (RTF)**: Characterizes processing speed, calculated as the time taken to process an audio file divided by its duration. RTF values are measured with a batch size of 1 on an NVIDIA RTX 6000 Ada Generation GPU.
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| **Latency** | **Chunk Size** | **Right Context** | **FIFO Queue** | **Update Period** | **Speaker Cache** | **RTF** |
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|-------------|----------------|-------------------|----------------|-------------------|-------------------|---------|
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| 10.0s | 124 | 1 | 124 | 124 | 188 | 0.005 |
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| 1.04s | 6 | 7 | 188 | 144 | 188 | 0.093 |
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| 0.32s | 3 | 1 | 188 | 144 | 188 | 0.180 |
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### Diarization Error Rate (DER)
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* All evaluations include overlapping speech.
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* Collar tolerance is 0s for DIHARD III Eval, and 0.25s for CALLHOME-part2 and CH109.
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* Post-Processing (PP) is optimized on two different held-out dataset splits.
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- [DIHARD III Dev Optimized Post-Processing](https://github.com/NVIDIA/NeMo/tree/main/examples/speaker_tasks/diarization/conf/post_processing/sortformer_diar_4spk-v1_dihard3-dev.yaml) for DIHARD III Eval
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- [CALLHOME-part1 Optimized Post-Processing](https://github.com/NVIDIA/NeMo/tree/main/examples/speaker_tasks/diarization/conf/post_processing/sortformer_diar_4spk-v1_callhome-part1.yaml) for CALLHOME-part2 and CH109
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| 367 |
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| 368 |
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| **Latency** | *PP* | **DIHARD III Eval <=4spk** | **DIHARD III Eval >=5spk** | **DIHARD III Eval full** | **CALLHOME-part2 2spk** | **CALLHOME-part2 3spk** | **CALLHOME-part2 4spk** | **CALLHOME-part2 5spk** | **CALLHOME-part2 6spk** | **CALLHOME-part2 full** | **CH109** |
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| 369 |
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|-------------|------|----------------------------|----------------------------|--------------------------|-------------------------|-------------------------|-------------------------|-------------------------|-------------------------|-------------------------|-----------|
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| 10.0s | no | 14.79 | 41.06 | 19.88 | 6.80 | 11.27 | 12.21 | 21.12 | 27.84 | 11.10 | 5.27 |
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| 10.0s | yes | 13.67 | 41.45 | 19.02 | 6.06 | 10.01 | 11.22 | 20.34 | 26.97 | 10.09 | 4.82 |
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| 1.04s | no | 14.57 | 42.12 | 19.89 | 7.35 | 11.57 | 13.83 | 25.81 | 29.06 | 12.00 | 5.59 |
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| 1.04s | yes | 13.32 | 42.61 | 18.97 | 6.43 | 10.26 | 12.40 | 24.41 | 27.78 | 10.79 | 5.09 |
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| 0.32s | no | 14.63 | 43.76 | 20.25 | 8.60 | 13.23 | 16.08 | 28.10 | 30.63 | 13.66 | 6.60 |
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| 0.32s | yes | 13.43 | 43.98 | 19.32 | 6.86 | 10.84 | 13.64 | 25.78 | 28.58 | 11.50 | 5.41 |
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| 376 |
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## NVIDIA Riva: Deployment
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| 391 |
## References
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| 392 |
[1] [Sortformer: Seamless Integration of Speaker Diarization and ASR by Bridging Timestamps and Tokens](https://arxiv.org/abs/2409.06656)
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| 393 |
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| 394 |
+
[2] [Streaming Sortformer: Speaker Cache-Based Online Speaker Diarization with Arrival-Time Ordering](https://arxiv.org/abs/25XX.XXXXX)
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| 395 |
+
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| 396 |
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[3] [NEST: Self-supervised Fast Conformer as All-purpose Seasoning to Speech Processing Tasks](https://arxiv.org/abs/2408.13106)
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| 397 |
|
| 398 |
+
[4] [Fast Conformer with Linearly Scalable Attention for Efficient Speech Recognition](https://arxiv.org/abs/2305.05084)
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| 399 |
|
| 400 |
+
[5] [Attention is all you need](https://arxiv.org/abs/1706.03762)
|
| 401 |
|
| 402 |
+
[6] [NVIDIA NeMo Framework](https://github.com/NVIDIA/NeMo)
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| 403 |
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| 404 |
+
[7] [NeMo speech data simulator](https://arxiv.org/abs/2310.12371)
|
| 405 |
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| 406 |
## Licence
|
| 407 |
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| 408 |
+
License to use this model is covered by the [CC-BY-4.0](https://creativecommons.org/licenses/by/4.0/legalcode). By downloading the public and release version of the model, you accept the terms and conditions of the CC-BY-4.0 license.
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