wavlm-large

Maintainer: microsoft

Total Score

49

Last updated 9/6/2024

🛠️

PropertyValue
Run this modelRun on HuggingFace
API specView on HuggingFace
Github linkNo Github link provided
Paper linkNo paper link provided

Create account to get full access

or

If you already have an account, we'll log you in

Model overview

The wavlm-large model is a powerful pre-trained speech model created by Microsoft. It is based on the HuBERT framework and emphasizes both spoken content modeling and speaker identity preservation. The model was pre-trained on a massive dataset of 94,000 hours of speech from Libri-Light, GigaSpeech, and VoxPopuli. This large-scale training allows the model to exhibit strong performance on a variety of speech processing tasks.

The wavlm-large model is similar to other pre-trained speech models like Wav2Vec2-Large-960h-Lv60 + Self-Training and Whisper-Large, which are also trained on large speech datasets and demonstrate impressive generalization capabilities. However, the wavlm-large model uniquely focuses on preserving speaker identity, making it potentially well-suited for applications that require speaker-aware processing.

Model inputs and outputs

Inputs

  • Audio: The wavlm-large model takes 16kHz sampled speech audio as input.

Outputs

  • Latent speech representations: The model outputs a sequence of latent speech representations that capture the semantics and speaker identity of the input audio. These representations can be used for downstream tasks like speech recognition, speaker identification, or audio classification.

Capabilities

The wavlm-large model has been shown to achieve state-of-the-art performance on the SUPERB benchmark, which evaluates a model's ability to handle a variety of speech processing tasks. It exhibits strong capabilities in both spoken content modeling and speaker discrimination, making it a versatile model for applications that require understanding both the linguistic content and the speaker identity of audio.

What can I use it for?

The wavlm-large model is a powerful tool for building speech-based applications. It could be used for tasks like:

  • Automatic Speech Recognition (ASR): By fine-tuning the model on a labeled speech dataset, you can create a high-performing speech recognition system.
  • Speaker Identification: The model's ability to preserve speaker identity can be leveraged for speaker recognition and diarization.
  • Audio Classification: The model's latent representations can be used as features for classifying audio content, such as detecting specific keywords or events.
  • Multimodal Applications: The model's speech understanding capabilities can be combined with other modalities, such as text or visual information, for building multimodal systems.

Things to try

One interesting aspect of the wavlm-large model is its emphasis on preserving speaker identity. You could explore using the model's speaker-aware representations for tasks like speaker de-identification, where the goal is to remove speaker-specific information from audio while preserving the linguistic content. This could be useful for privacy-preserving applications or for creating synthetic speech that sounds more natural and less robotic.

Additionally, you could investigate how the model's performance varies across different accents, dialects, and demographic groups. Understanding the model's strengths and limitations in this area could help guide the development of more inclusive and equitable speech-based systems.



This summary was produced with help from an AI and may contain inaccuracies - check out the links to read the original source documents!

Related Models

🔎

wav2vec2-large-960h-lv60-self

facebook

Total Score

118

Facebook's Wav2Vec2 is a large model pretrained and fine-tuned on 960 hours of Libri-Light and Librispeech on 16kHz sampled speech audio. The model was trained with a Self-Training objective. wav2vec2-large-960h-lv60-self demonstrates state-of-the-art performance on speech recognition tasks, outperforming the previous best semi-supervised methods while using a simpler approach. Similar models include wav2vec2-base-960h, which is a smaller base model pretrained on the same Librispeech data, and wav2vec2-xls-r-300m, a large multilingual version of Wav2Vec2 pretrained on 436k hours of speech data across 128 languages. Model inputs and outputs Inputs Audio**: The model takes raw speech audio as input, which must be sampled at 16kHz. Outputs Transcription**: The model outputs a text transcription of the input speech audio. Capabilities The wav2vec2-large-960h-lv60-self model demonstrates state-of-the-art performance on speech recognition tasks, achieving 1.8/3.3 WER on the clean/other Librispeech test sets when using all labeled data. It can also achieve strong results with limited labeled data, outperforming previous methods on the 100 hour Librispeech subset while using 100 times less labeled data. What can I use it for? The wav2vec2-large-960h-lv60-self model is well-suited for building speech recognition systems, particularly for applications that require high accuracy on a variety of speech inputs. It can be used as a standalone acoustic model to transcribe audio files, or integrated into larger speech processing pipelines. Things to try One interesting aspect of the wav2vec2-large-960h-lv60-self model is its ability to perform well with limited labeled data. Developers could experiment with fine-tuning the model on domain-specific datasets to adapt it for specialized use cases, potentially achieving strong results even when only a small amount of labeled data is available.

Read more

Updated Invalid Date

🤯

whisper-large-v2

openai

Total Score

1.6K

The whisper-large-v2 model is a pre-trained Transformer-based encoder-decoder model for automatic speech recognition (ASR) and speech translation. Trained on 680k hours of labeled data by OpenAI, Whisper models demonstrate a strong ability to generalize to many datasets and domains without the need for fine-tuning. Compared to the original Whisper large model, the whisper-large-v2 model has been trained for 2.5x more epochs with added regularization for improved performance. Model inputs and outputs Inputs Audio samples**: The model takes audio samples as input and performs either speech recognition or speech translation. Outputs Text transcription**: The model outputs text transcriptions of the input audio. For speech recognition, the transcription is in the same language as the audio. For speech translation, the transcription is in a different language than the audio. Timestamps (optional)**: The model can optionally output timestamps for the transcribed text. Capabilities The whisper-large-v2 model exhibits improved robustness to accents, background noise, and technical language compared to many existing ASR systems. It also demonstrates strong zero-shot translation capabilities, allowing it to translate speech from multiple languages into English with high accuracy. What can I use it for? The whisper-large-v2 model can be a useful tool for developers building speech recognition and translation applications. Its strong generalization capabilities suggest it may be particularly valuable for tasks like improving accessibility through real-time captioning, language translation, and other speech-to-text use cases. However, the model's performance can vary across languages, accents, and demographics, so users should carefully evaluate its performance in their specific domain before deployment. Things to try One interesting aspect of the whisper-large-v2 model is its ability to perform long-form transcription of audio samples longer than 30 seconds. By using a chunking algorithm, the model can transcribe audio of arbitrary length, making it a useful tool for transcribing podcasts, lectures, and other long-form audio content. Users can also experiment with fine-tuning the model on their own data to further improve its performance for specific use cases.

Read more

Updated Invalid Date

🎲

whisper-large-v3

openai

Total Score

2.6K

The whisper-large-v3 model is a general-purpose speech recognition model developed by OpenAI. It is the latest version of the Whisper model, building on the previous Whisper large models. The whisper-large-v3 model has a few minor architectural differences from the previous large models, including using 128 Mel frequency bins instead of 80 and adding a new language token for Cantonese. The Whisper model was trained on a massive 680,000 hours of audio data, with 65% English data, 18% non-English data with English transcripts, and 17% non-English data with non-English transcripts covering 98 languages. This allows the model to perform well on a diverse range of speech recognition and translation tasks, without needing to fine-tune on specific datasets. Similar Whisper models include the Whisper medium, Whisper tiny, and the whisper-large-v3 model developed by Nate Raw. There is also an incredibly fast version of the Whisper large model by Vaibhav Srivastav. Model inputs and outputs The whisper-large-v3 model takes audio samples as input and generates text transcripts as output. The audio can be in any of the 98 languages covered by the training data. The model can also be used for speech translation, where it generates text in a different language than the audio. Inputs Audio samples in any of the 98 languages the model was trained on Outputs Text transcripts of the audio in the same language Translated text transcripts in a different language Capabilities The whisper-large-v3 model demonstrates strong performance on a variety of speech recognition and translation tasks, with 10-20% lower error rates compared to the previous Whisper large model. It is robust to accents, background noise, and technical language, and can perform zero-shot translation from multiple languages into English. However, the model's performance is uneven across languages, with lower accuracy on low-resource and low-discoverability languages where less training data was available. It also has a tendency to generate repetitive or hallucinated text that is not actually present in the audio input. What can I use it for? The primary intended use of the Whisper models is for AI researchers studying model capabilities, robustness, and limitations. However, the models can also be quite useful as a speech recognition solution for developers, especially for English transcription tasks. The Whisper models could be used to build applications that improve accessibility, such as closed captioning or voice-to-text transcription. While the models cannot be used for real-time transcription out of the box, their speed and size suggest that others may be able to build near-real-time applications on top of them. Things to try One interesting aspect of the Whisper models is their ability to perform speech translation, generating text transcripts in a different language than the audio input. Developers could experiment with using the model for tasks like simultaneous interpretation or multilingual subtitling. Another avenue to explore is fine-tuning the pre-trained Whisper model on specific datasets or domains. The blog post Fine-Tune Whisper with Transformers provides a guide on how to fine-tune the model with as little as 5 hours of labeled data, which can improve performance on particular languages or use cases.

Read more

Updated Invalid Date

🔎

whisper-large

openai

Total Score

438

The whisper-large model is a pre-trained AI model for automatic speech recognition (ASR) and speech translation, developed by OpenAI. Trained on 680k hours of labelled data, the Whisper models demonstrate a strong ability to generalize to many datasets and domains without the need for fine-tuning. The whisper-large-v2 model is a newer version that surpasses the performance of the original whisper-large model, with no architecture changes. The whisper-medium model is a slightly smaller version with 769M parameters, while the whisper-tiny model is the smallest at 39M parameters. All of these Whisper models are available on the Hugging Face Hub. Model inputs and outputs Inputs Audio samples, which the model converts to log-Mel spectrograms Outputs Textual transcriptions of the input audio, either in the same language as the audio (for speech recognition) or in a different language (for speech translation) The model can also output timestamps for the transcriptions Capabilities The Whisper models demonstrate strong performance on a variety of speech recognition and translation tasks, exhibiting improved robustness to accents, background noise, and technical language. They can also perform zero-shot translation from multiple languages into English. However, the models may occasionally produce text that is not actually spoken in the audio input, a phenomenon known as "hallucination". Their performance also varies across languages, with lower accuracy on low-resource and less common languages. What can I use it for? The Whisper models are primarily intended for use by AI researchers studying model robustness, generalization, capabilities, biases, and constraints. However, the models can also be useful for developers looking to build speech recognition or translation applications, especially for English speech. The models' speed and accuracy make them well-suited for applications that require transcription or translation of large volumes of audio data, such as accessibility tools, media production, and language learning. Developers can build applications on top of the models to enable near-real-time speech recognition and translation. Things to try One interesting aspect of the Whisper models is their ability to perform long-form transcription of audio samples longer than 30 seconds. This is achieved through a chunking algorithm that allows the model to process audio of arbitrary length. Another unique feature is the model's ability to automatically detect the language of the input audio and perform the appropriate speech recognition or translation task. Developers can leverage this by providing the model with "context tokens" that inform it of the desired task and language. Finally, the pre-trained Whisper models can be fine-tuned on smaller datasets to further improve their performance on specific languages or domains. The Fine-Tune Whisper with Transformers blog post provides a step-by-step guide on how to do this.

Read more

Updated Invalid Date