--- language: - en license: cc-by-4.0 library_name: nemo tags: - speaker-recognition - speech - audio - speaker-verification - titanet - speaker-diarization - NeMo - pytorch datasets: - librispeech_asr - VOXCCELEB-1 - VOXCCELEB-2 - FISHER - Switchboard - SRE(2004-2010) model-index: - name: speakerverification_en results: - task: type: automatic-speech-recognition dataset: name: Librispeech (clean) type: librispeech_asr config: other split: test args: language: en metrics: - type: wer value: 8.1 name: WER --- ## Model Overview This model extracts speaker embeddings from given speech, which is the backbone for speaker verification and diarization tasks. It is a "large" version of TitaNet (around 23M parameters) models. See the [model architecture](#model-architecture) section and [NeMo documentation](https://docs.nvidia.com/deeplearning/nemo/user ## How to Use this Model The model is available for use in the NeMo toolkit [3] and can be used as a pre-trained checkpoint for inference or for fine-tuning on another dataset. ### Automatically instantiate the model ```python import nemo.collections.asr as nemo_asr speaker_model = nemo_asr.models.EncDecSpeakerLabelModel.from_pretrained("nvidia/speakerverification_en_titanet_large") ``` ### Embedding Extraction Using ```python emb = speaker_model.get_embedding("an255-fash-b.wav") ``` ### Verifying two utterances (Speaker Verification) Now to check if two audio files are from the same speaker or not, simply do: ```python speaker_model.verify_speakers("an255-fash-b.wav","cen7-fash-b.wav") ``` ### Extracting Embeddings for more audio files To extract embeddings from a bunch of audio files: Write audio files to a `manifest.json` file with lines as in format: ```json {"audio_filepath": "/audio_file.wav", "duration": "duration of file in sec", "label": "speaker_id"} ``` Then running following script will extract embeddings and writes to current working directory: ```shell python /examples/speaker_tasks/recognition/extract_speaker_embeddings.py --manifest=manifest.json ``` ### Input This model accepts 16000 KHz Mono-channel Audio (wav files) as input. ### Output This model provides speaker embeddings for an audio file. ## Model Architecture TitaNet model is a depth-wise separable conv1D model [1] for Speaker Verification and diarization tasks. You may find more info on the detail of this model here: [TitaNet-Model](https://docs.nvidia.com/deeplearning/nemo/user-guide/docs/en/main/asr/speaker_recognition/models.html). ## Training The NeMo toolkit [3] was used for training the models for over several hundred epochs. These model are trained with this [example script](https://github.com/NVIDIA/NeMo/blob/main/examples/speaker_tasks/recognition/speaker_reco.py) and this [base config](https://github.com/NVIDIA/NeMo/blob/main/examples/speaker_tasks/recognition/conf/titanet-large.yaml). ### Datasets All the models in this collection are trained on a composite dataset comprising several thousand hours of English speech: - Voxceleb-1 - Voxceleb-2 - Fisher - Switchboard - Librispeech - SRE (2004-2010) ## Performance Performances of the these models are reported in terms of Equal Error Rate (EER%) on speaker verification evaluation trial files and as Diarization Error Rate (DER%) on diarization test sessions. * Speaker Verification (EER%) | Version | Model | Model Size | VoxCeleb1 (Cleaned trial file) | |---------|--------------|-----|---------------| | 1.10.0 | TitaNet-Large | 23M | 0.66 | * Speaker Diarization (DER%) | Version | Model | Model Size | Evaluation Condition | NIST SRE 2000 | AMI (Lapel) | AMI (MixHeadset) | CH109 | |---------|--------------|-----|----------------------|---------------|-------------|------------------|-------| | 1.10.0 | TitaNet-Large | 23M | Oracle VAD KNOWN # of Speakers | 6.73 | 2.03 | 1.73 | 1.19 | | 1.10.0 | TitaNet-Large | 23M | Oracle VAD UNKNOWN # of Speakers | 5.38 | 2.03 | 1.89 | 1.63 | ## Limitations This model is trained on both telephonic and non-telephonic speech from voxceleb datasets, Fisher and switch board. If your domain of data differs from trained data or doesnot show relatively good performance consider finetuning for that speech domain. ## 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 is here](https://huggingface.co/models?other=Riva). Check out [Riva live demo](https://developer.nvidia.com/riva#demos). ## References [1] [TitaNet: Neural Model for Speaker Representation with 1D Depth-wise Separable convolutions and global context](https://ieeexplore.ieee.org/stamp/stamp.jsp?arnumber=9746806) [2] [NVIDIA NeMo Toolkit](https://github.com/NVIDIA/NeMo) ## Licence License to use this model is covered by the [CC-BY-4.0](https://creativecommons.org/licenses/by/4.0/). By downloading the public and release version of the model, you accept the terms and conditions of the [CC-BY-4.0](https://creativecommons.org/licenses/by/4.0/) license.