Update README.md

README.md

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----
-license: cc-by-4.0
----
+---
+license: cc-by-nc-sa-4.0
+library_name: nemo
+datasets:
+- fisher_english
+- NIST_SRE_2004-2010
+- librispeech
+- ami_meeting_corpus
+- voxconverse_v0.3
+- icsi
+- aishell4
+- dihard_challenge-3-dev
+- NIST_SRE_2000-Disc8_split1
+- Alimeeting-train
+- DipCo-dev
+thumbnail: null
+tags:
+- speaker-diarization
+- speaker-recognition
+- speech
+- audio
+- Transformer
+- FastConformer
+- Conformer
+- NEST
+- pytorch
+- NeMo
+widget:
+- example_title: Librispeech sample 1
+  src: https://cdn-media.huggingface.co/speech_samples/sample1.flac
+- example_title: Librispeech sample 2
+  src: https://cdn-media.huggingface.co/speech_samples/sample2.flac
+model-index:
+- name: diar_streaming_sortformer_4spk-v2
+  results:
+  - task:
+      name: Speaker Diarization
+      type: speaker-diarization-with-post-processing
+    dataset:
+      name: DIHARD3-eval 
+      type: dihard3-eval-1to4spks
+      config: with_overlap_collar_0.0s
+      split: eval
+    metrics:
+    - name: Test DER
+      type: der
+      value: 14.76
+  - task:
+      name: Speaker Diarization
+      type: speaker-diarization-with-post-processing
+    dataset:
+      name: CALLHOME (NIST-SRE-2000 Disc8)
+      type: CALLHOME-part2-2spk
+      config: with_overlap_collar_0.25s
+      split: part2-2spk
+    metrics:
+    - name: Test DER
+      type: der
+      value: 5.85
+  - task:
+      name: Speaker Diarization
+      type: speaker-diarization-with-post-processing
+    dataset:
+      name: CALLHOME (NIST-SRE-2000 Disc8)
+      type: CALLHOME-part2-3spk
+      config: with_overlap_collar_0.25s
+      split: part2-3spk
+    metrics:
+    - name: Test DER
+      type: der
+      value: 8.46
+  - task:
+      name: Speaker Diarization
+      type: speaker-diarization-with-post-processing
+    dataset:
+      name: CALLHOME (NIST-SRE-2000 Disc8)
+      type: CALLHOME-part2-4spk
+      config: with_overlap_collar_0.25s
+      split: part2-4spk
+    metrics:
+    - name: Test DER
+      type: der
+      value: 12.59
+  - task:
+      name: Speaker Diarization
+      type: speaker-diarization-with-post-processing
+    dataset:
+      name: call_home_american_english_speech
+      type: CHAES_2spk_109sessions
+      config: with_overlap_collar_0.25s
+      split: ch109
+    metrics:
+    - name: Test DER
+      type: der
+      value: 6.86
+metrics:
+- der
+pipeline_tag: audio-classification
+---
+
+
+# Sortformer Diarizer 4spk v1
+
+<style>
+img {
+ display: inline;
+}
+</style>
+
+[![Model architecture](https://img.shields.io/badge/Model_Arch-FastConformer--Transformer-lightgrey#model-badge)](#model-architecture)
+| [![Model size](https://img.shields.io/badge/Params-123M-lightgrey#model-badge)](#model-architecture)
+<!-- | [![Language](https://img.shields.io/badge/Language-multilingual-lightgrey#model-badge)](#datasets) -->
+
+[Sortformer](https://arxiv.org/abs/2409.06656)[1] is a novel end-to-end neural model for speaker diarization, trained with unconventional objectives compared to existing end-to-end diarization models.
+
+<div align="center">
+    <img src="sortformer_intro.png" width="750" />
+</div>
+
+Sortformer resolves permutation problem in diarization following the arrival-time order of the speech segments from each speaker. 
+
+## Model Architecture
+
+Sortformer consists of an L-size (18 layers) [NeMo Encoder for
+Speech Tasks (NEST)](https://arxiv.org/abs/2408.13106)[2] which is based on [Fast-Conformer](https://arxiv.org/abs/2305.05084)[3] encoder. Following that, an 18-layer Transformer[4] encoder with hidden size of 192, 
+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/2409.06656)[1].
+
+<div align="center">
+    <img src="sortformer-v1-model.png" width="450" />
+</div>
+
+## NVIDIA NeMo
+
+To train, fine-tune or perform diarization with Sortformer, you will need to install [NVIDIA NeMo](https://github.com/NVIDIA/NeMo)[5]. We recommend you install it after you've installed Cython and latest PyTorch version.
+```
+pip install git+https://github.com/NVIDIA/NeMo.git@main#egg=nemo_toolkit[asr]
+```
+
+## How to Use this Model
+
+The model is available for use in the NeMo Framework[5], and can be used as a pre-trained checkpoint for inference or for fine-tuning on another dataset.
+
+### Loading the Model
+
+```python
+from nemo.collections.asr.models import SortformerEncLabelModel
+
+# load model
+diar_model = SortformerEncLabelModel.restore_from(restore_path="diar_streaming_sortformer_4spk-v2", map_location=torch.device('cuda'), strict=False)
+```
+
+### Input Format
+Input to Sortformer can be either a list of paths to audio files or a jsonl manifest file.
+
+```python
+pred_outputs = diar_model.diarize(audio=["/path/to/multispeaker_audio1.wav", "/path/to/multispeaker_audio2.wav"], batch_size=1)
+```
+
+Individual audio file can be fed into Sortformer model as follows:
+```python
+pred_output1 = diar_model.diarize(audio="/path/to/multispeaker_audio1.wav", batch_size=1)
+```
+
+
+To use Sortformer for performing diarization on a multi-speaker audio recording, specify the input as jsonl manifest file, where each line in the file is a dictionary containing the following fields: 
+
+```yaml
+# Example of a line in `multispeaker_manifest.json`
+{
+    "audio_filepath": "/path/to/multispeaker_audio1.wav",  # path to the input audio file 
+    "offset": 0 # offset (start) time of the input audio
+    "duration": 600,  # duration of the audio, can be set to `null` if using NeMo main branch
+}
+{
+    "audio_filepath": "/path/to/multispeaker_audio2.wav",  
+    "offset": 0,
+    "duration": 580,  
+}
+```
+
+and then use:
+```python
+pred_outputs = diar_model.diarize(audio="/path/to/multispeaker_manifest.json", batch_size=1)
+```
+
+
+### Input
+
+This model accepts single-channel (mono) audio sampled at 16,000 Hz.
+- The actual input tensor is a Ns x 1 matrix for each audio clip, where Ns is the number of samples in the time-series signal. 
+- For instance, a 10-second audio clip sampled at 16,000 Hz (mono-channel WAV file) will form a 160,000 x 1 matrix.
+
+### Output
+
+The output of the model is an T x S matrix, where:  
+- S is the maximum number of speakers (in this model, S = 4).  
+- T is the total number of frames, including zero-padding. Each frame corresponds to a segment of 0.08 seconds of audio.  
+Each element of the T x S matrix represents the speaker activity probability in the [0, 1] range.  For example, a matrix element a(150, 2) = 0.95 indicates a 95% probability of activity for the second speaker during the time range [12.00, 12.08] seconds.
+
+
+## Train and evaluate Sortformer diarizer using NeMo
+### Training
+
+Sortformer diarizer models are trained on 8 nodes of 8×NVIDIA Tesla V100 GPUs. We use 90 second long training samples and batch size of 4.
+The model can be trained using this [example script](https://github.com/NVIDIA/NeMo/blob/main/examples/speaker_tasks/diarization/neural_diarizer/sortformer_diar_train.py) and [base config](https://github.com/NVIDIA/NeMo/blob/main/examples/speaker_tasks/diarization/conf/neural_diarizer/sortformer_diarizer_hybrid_loss_4spk-v1.yaml).
+
+### Inference
+
+Sortformer diarizer models can be performed with post-processing algorithms using inference [example script](https://github.com/NVIDIA/NeMo/blob/main/examples/speaker_tasks/diarization/neural_diarizer/e2e_diarize_speech.py). If you provide the post-processing YAML configs in [`post_processing` folder](https://github.com/NVIDIA/NeMo/tree/main/examples/speaker_tasks/diarization/conf/post_processing) to reproduce the optimized post-processing algorithm for each development dataset.
+
+### Technical Limitations
+
+- The model operates in a non-streaming mode (offline mode).
+- It can detect a maximum of 4 speakers; performance degrades on recordings with 5 and more speakers.
+- The maximum duration of a test recording depends on available GPU memory. For an RTX A6000 48GB model, the limit is around 12 minutes.
+- The model was trained on publicly available speech datasets, primarily in English. As a result:
+    * Performance may degrade on non-English speech.
+    * Performance may also degrade on out-of-domain data, such as recordings in noisy conditions.
+
+
+## Datasets
+
+Sortformer was trained on a combination of 2030 hours of real conversations and 5150 hours or simulated audio mixtures generated by [NeMo speech data simulator](https://arxiv.org/abs/2310.12371)[6].
+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.
+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.
+
+
+### Training Datasets (Real conversations)
+- Fisher English (LDC)
+- 2004-2010 NIST Speaker Recognition Evaluation (LDC)
+- Librispeech 
+- AMI Meeting Corpus
+- VoxConverse-v0.3
+- ICSI
+- AISHELL-4
+- Third DIHARD Challenge Development (LDC)
+- 2000 NIST Speaker Recognition Evaluation, split1 (LDC)
+
+### Training Datasets (Used to simulate audio mixtures)
+- 2004-2010 NIST Speaker Recognition Evaluation (LDC)
+- Librispeech
+
+## Performance
+
+
+### Evaluation dataset specifications
+
+| **Dataset**                   | **DIHARD3-Eval**   | **CALLHOME-part2**  | **CALLHOME-part2**  | **CALLHOME-part2**  | **CH109**          |
+|:------------------------------|:------------------:|:-------------------:|:-------------------:|:-------------------:|:------------------:|
+| **Number of Speakers**        | ≤ 4 speakers       | 2 speakers          | 3 speakers          | 4 speakers          | 2 speakers         |
+| **Collar (sec)**              | 0.0s               | 0.25s               | 0.25s               | 0.25s               | 0.25s              |
+| **Mean Audio Duration (sec)** | 453.0s             | 73.0s               | 135.7s              | 329.8s              | 552.9s             |
+
+### Diarization Error Rate (DER)
+
+* All evaluations include overlapping speech.  
+* Bolded and italicized numbers represent the best-performing Sortformer evaluations.
+* Post-Processing (PP) is optimized on two different held-out dataset splits. 
+    - [YAML file for DH3-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)    
+    - [YAML file for 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)    
+
+
+| **Dataset**                                                                   | **DIHARD3-Eval <= 4spk**   | **CALLHOME-2spk part2**  | **CALLHOME-3spk part2**  | **CALLHOME-4spk part2**  | **CH109**          |
+|:------------------------------------------------------------------------------|:--------------------------:|:------------------------:|:------------------------:|:------------------------:|:------------------:|
+| DER **diar_streaming_sortformer_4spk-v2 IBL=1.04s**                           | 14.57                      | 7.35                     | 11.57                    | 13.83                    | 5.59               |
+| DER **diar_streaming_sortformer_4spk-v2 IBL=1.04s + DH3-dev Opt. PP**         | **_13.32_**                | -                        | -                        | -                        | -                  |
+| DER **diar_streaming_sortformer_4spk-v2 IBL=1.04s + CallHome-part1 Opt. PP**  | -                          | **_6.43_**               | **_10.26_**              | **_12.40_**              | **_5.09_**         |
+
+* "IBL" stands for Input Buffer Latency which is identical to chunk length in the streaming implementation.
+
+### Real Time Factor (RTF)
+
+RTF is defined as the time taken to process a recording divided by its length.
+
+| **Latency [sec]** | **Chunk Size** | **Right Context** | **FIFO Queue [frame count]** | **Update Period** | **Speaker Cache** | **RTF** |
+|-------------------|----------------|-------------------|------------------------------|-------------------|-------------------|---------|
+| 10.0              | 124            | 1                 | 124                          | 124               | 188               | 0.005   |
+| 1.04              | 6              | 7                 | 188                          | 144               | 188               | 0.093   |
+| 0.32              | 3              | 1                 | 188                          | 144               | 188               | 0.180   |
+
+
+## NVIDIA Riva: Deployment
+
+[NVIDIA Riva](https://developer.nvidia.com/riva), is an accelerated speech AI SDK deployable on-prem, in all clouds, multi-cloud, hybrid, on edge, and embedded. 
+Additionally, Riva provides: 
+
+* World-class out-of-the-box accuracy for the most common languages with model checkpoints trained on proprietary data with hundreds of thousands of GPU-compute hours 
+* Best in class accuracy with run-time word boosting (e.g., brand and product names) and customization of acoustic model, language model, and inverse text normalization 
+* Streaming speech recognition, Kubernetes compatible scaling, and enterprise-grade support 
+
+Although this model isn’t supported yet by Riva, the [list of supported models](https://huggingface.co/models?other=Riva) is here.  
+Check out [Riva live demo](https://developer.nvidia.com/riva#demos). 
+
+
+## References
+[1] [Sortformer: Seamless Integration of Speaker Diarization and ASR by Bridging Timestamps and Tokens](https://arxiv.org/abs/2409.06656)
+
+[2] [NEST: Self-supervised Fast Conformer as All-purpose Seasoning to Speech Processing Tasks](https://arxiv.org/abs/2408.13106)
+
+[3] [Fast Conformer with Linearly Scalable Attention for Efficient Speech Recognition](https://arxiv.org/abs/2305.05084)
+
+[4] [Attention is all you need](https://arxiv.org/abs/1706.03762)
+
+[5] [NVIDIA NeMo Framework](https://github.com/NVIDIA/NeMo)
+
+[6] [NeMo speech data simulator](https://arxiv.org/abs/2310.12371)
+
+## Licence
+
+License to use this model is covered by the [CC-BY-NC-SA-4.0](https://creativecommons.org/licenses/by-nc-sa/4.0/legalcode). By downloading the public and release version of the model, you accept the terms and conditions of the CC-BY-NC-SA-4.0 license.