Speech Technology
1.59K subscribers
122 photos
4 videos
1 file
2.12K links
Download Telegram
12m hours of speech data

https://arxiv.org/abs/2303.01037

Google USM: Scaling Automatic Speech Recognition Beyond 100 Languages

Yu Zhang, Wei Han, James Qin, Yongqiang Wang, Ankur Bapna, Zhehuai Chen, Nanxin Chen, Bo Li, Vera Axelrod, Gary Wang, Zhong Meng, Ke Hu, Andrew Rosenberg, Rohit Prabhavalkar, Daniel S. Park, Parisa Haghani, Jason Riesa, Ginger Perng, Hagen Soltau, Trevor Strohman, Bhuvana Ramabhadran, Tara Sainath, Pedro Moreno, Chung-Cheng Chiu, Johan Schalkwyk, Françoise Beaufays, Yonghui Wu

We introduce the Universal Speech Model (USM), a single large model that performs automatic speech recognition (ASR) across 100+ languages. This is achieved by pre-training the encoder of the model on a large unlabeled multilingual dataset of 12 million (M) hours spanning over 300 languages, and fine-tuning on a smaller labeled dataset. We use multilingual pre-training with random-projection quantization and speech-text modality matching to achieve state-of-the-art performance on downstream multilingual ASR and speech-to-text translation tasks. We also demonstrate that despite using a labeled training set 1/7-th the size of that used for the Whisper model, our model exhibits comparable or better performance on both in-domain and out-of-domain speech recognition tasks across many languages.
Tried a popular https://github.com/Kyubyong/g2p. As usual, networks are very bad for unseen cases. Missing letters, extra letters, etc. Watch outputs carefully. Example:

bio-sand B AY1 OW0 S T AE2 N D
How can we make inference faster when using big #speech #selfsupervised models?

Check out @salah_zaiem 's paper that compares various approaches, revealing some pretty interesting insights.

https://arxiv.org/abs/2303.06740

These techniques will be soon available in #SpeechBrain

https://twitter.com/mirco_ravanelli/status/1635678132731518976
New model from Assembly AI. Definitely improved from before, but not as great as Speechmatics.

On a toy test WER 10.89, previous assemblyAI (version 9) was at 11.04, version before 11.89. Speechmatics 6.88. Whisper large 8.94

https://twitter.com/AssemblyAI/status/1636050346240884744

Introducing Conformer-1: our latest state-of-the-art speech recognition model.

Built on top of the Conformer architecture and trained on 650K hours of audio data, it achieves near-human-level performance, making up to 43% fewer errors on noisy data than other ASR models.

We use a modified version of the conformer neural net published by Google Brain.

It's built on top of an Efficient Conformer (Orange Labs, 2021), that introduces the following technical modifications:

- Progressive Downsampling to reduce the length of the encoded sequence
- Grouped Attention: A modified version of the attention mechanism that makes it agnostic to sequence-length

These changes yield speedups of 29% at inference time and 36% at training time.

To further improve our model’s accuracy on noisy audio, we implemented a modified version of Sparse Attention, a pruning method for achieving sparsity of the model’s weights in order to achieve regularization.

We took inspiration from the data scaling laws described in DeepMind's Chinchilla paper and adapted them to the ASR domain.

Our team curated a dataset of 650K hours of English audio - making our model the largest-trained supervised model for English available today.

Based on our results, Conformer-1 is more robust on real-world data than popular commercial and open-source ASR models, making up to 43% fewer errors on average on noisy data:

The biggest improvement with this new release is in our robustness to a wide variety of data domains and noisy audio.
Kincaid46 WER from Ursa announcement:

AssemblyAI: 8.6
Speechmatics: 7.88
Microsoft: 9.70
Whisper Large-v2: 8.7
Vosk 0.42 Gigaspeech 15.8
Google 12.52
Amazon 10.94
The amount of models this guy trained is quite outstanding

https://malaya-speech.readthedocs.io/en/latest/index.html
12th ISCA Speech Synthesis Workshop (SSW) is now open for submissions!
Final submission deadline: May, 3 2023
Late breaking reports submission deadline : June, 28 2023

The Speech Synthesis Workshop will be held in Grenoble, France and is organized as a satellite event of the Interspeech conference in Dublin, Ireland
Come and join the SSW community and the people who creates machines that talk!

Visit the official site for more information
https://ssw2023.org/
Forwarded from Machinelearning
WavCaps: A ChatGPT-Assisted Weakly-Labelled Audio Captioning Dataset for Audio-Language Multimodal Research

Propose a three-stage processing pipeline for filtering noisy data and generating high-quality captions, where ChatGPT.

Конвейер обработки для фильтрации зашумленных данных и создания высококачественных титров
.

🖥 Github: https://github.com/xinhaomei/wavcaps

Paper: https://arxiv.org/abs/2303.17395v1

💨 Dataset: https://paperswithcode.com/dataset/sounddescs

ai_machinelearning_big_data
Please open Telegram to view this post
VIEW IN TELEGRAM