How to Create Your Own Automatic Speech Recognition System with TensorFlow

Mar 11, 2021 | Data Science

Welcome to the fascinating world of Automatic Speech Recognition (ASR)! If you’ve ever dreamed of building a system that can understand spoken language, you’re in the right place. In this guide, you’ll learn how to set up an end-to-end ASR system implemented in TensorFlow. We will also touch upon recent updates, installation, usage, and some helpful troubleshooting tips. Let’s get started!

Recent Updates

  • Support TensorFlow r1.0
  • Support dropout for dynamic RNN
  • Support running in shell file
  • Automatic evaluation every few training epochs
  • Bug fixes for character-level ASR
  • API improvements for reusability
  • Data preprocessing enhancements
  • Support for LibriSpeech training
  • Added n-gram model for random generation
  • Reorganization of the project structure
  • Additions of DeepSpeech2 implementation
  • Support for Mandarin Speech Recognition
  • Release 1.0.0 version!
  • Will support TF1.12 soon

Installation and Usage

Before diving into the code, ensure you have Python 3.5 installed on your system, as this is the only version currently supported. You’ll also need to install libsndfile to execute the audio processing functions properly.

To install the required dependencies, clone the repository and run the following commands:

sudo pip3 install -r requirements.txt
sudo python3 setup.py install

Once installed, you can start the training process with the command:

python maintimit_train.py [options]

Various options can be set, including the model you want to use, RNN cell types, and more. You could also set these arguments in the timit_train.py.

Understanding the Code with an Analogy

Think of building an automatic speech recognition system as setting up a restaurant. The raw audio files are like the ingredients you’ll use to create dishes (words and sentences). Here’s a breakdown:

  • Data Preprocessing: Just as you need to prepare your ingredients (wash, chop, and marinate), the audio files must be converted into feature vectors for the system to understand.
  • Acoustic Modeling: This is akin to the cook in the kitchen who combines the ingredients to craft a dish; the model learns how to map these features into phonemes (the building blocks of speech).
  • CTC Decoding: Like presenting a completed dish to customers, CTC decoding takes the cooked results (phonemes) and structures them into understandable words.

Troubleshooting Tips

If you encounter issues while setting up the system, try the following suggestions:

  • Ensure all dependencies are installed correctly.
  • Check if TensorFlow is compatible with your Python version.
  • Inspect your data preprocessing steps; make sure they align with the format expected by the model.
  • For configurations, double-check command line arguments or refer to the timit_train.py.

For more insights, updates, or to collaborate on AI development projects, stay connected with fxis.ai.

Core Features and Future Work

This powerful library not only supports dynamic RNN models like GRU and LSTM but also boasts features such as CTC Decoding and allows for mini-batch training. Future enhancements include:

  • Release of pretrained English ASR model
  • Incorporating attention mechanisms
  • Implementing speaker verification
  • Expanding to include Text-to-Speech (TTS) capabilities

Conclusion

The journey to create an Automatic Speech Recognition system might seem challenging, but with the right tools and guidance, it becomes an engaging and fulfilling experience. At fxis.ai, we believe that such advancements are crucial for the future of AI, as they enable more comprehensive and effective solutions. Our team is continually exploring new methodologies to push the envelope in artificial intelligence, ensuring that our clients benefit from the latest technological innovations.

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