
## Introduction

In the world of audio processing and speech recognition, identifying when someone is speaking versus when there's silence or background noise is a critical first step. This process, known as Voice Activity Detection (VAD), serves as the foundation for many speech-related applications, from transcription services to voice assistants. While conceptually simple, implementing an efficient and accurate VAD system can significantly improve downstream tasks and reduce computational costs.

[PyAnnote Audio](https://github.com/pyannote/pyannote-audio), an open-source toolkit built on PyTorch, offers state-of-the-art models for VAD that are both accurate and accessible. Running these models on Vast.ai provides a cost-effective solution for processing large audio datasets without investing in expensive hardware. This combination gives developers and researchers the tools they need to build sophisticated audio processing pipelines at a fraction of the cost of traditional cloud providers.

### Why Voice Activity Detection Matters

VAD provides several key benefits for speech processing pipelines:

1. **Reduced Computation Load**: By filtering out non-speech segments before running speech-to-text (STT) models, we significantly reduce the computational resources needed for transcription.

2. **Improved Accuracy**: Many STT models perform better when processing only speech segments rather than trying to interpret silence or background noise.

3. **Efficient Storage**: Extracting only the speech segments can reduce storage requirements for large audio datasets.

4. **Better User Experience**: For applications like voice assistants or transcription services, VAD helps eliminate unnecessary processing of silence.

### What This Guide Covers

In this guide, we will:

-   Set up the Pyannote Audio VAD pipeline
-   Process audio files to detect speech segments
-   Extract and save only the speech portions of the input audio
-   Visualize the results

The output will be a collection of audio files containing only the detected speech segments from the original recording, making them ready for further processing in speech-to-text pipelines.

### Why VAST.ai

VAST.ai offers a marketplace approach to GPU rentals that provides significant advantages for audio processing tasks. Unlike traditional cloud providers, VAST.ai allows you to:

-   Rent precisely the GPU capacity needed for your workload
-   Access GPU's at more affordable rates
-   Access more types of SKU's of GPU's, particularly lower RAM/more affordable GPUs
-   Avoid long-term commitments for experimental projects

For VAD specifically, VAST.ai offers an ideal balance of performance and cost-effectiveness, as these models benefit from GPU acceleration without requiring the most expensive hardware tiers.

## Download This Notebook

To follow along with this tutorial, you can download the complete Jupyter notebook:

[Notebook](https://nbviewer.org/urls/bitbucket.org/%21api/2.0/snippets/jsbcannell/6q8akx/6b8dc846a00d27decad5a60a6159466813f6d445/files/pyannote_vad_vast.ipynb)

Having the notebook will allow you to execute the code blocks as you read through this guide, making it easier to understand and experiment with the VAD implementation.

## Choosing an Instance

For running the Pyannote Voice Activity Detection model on VAST.ai, you'll need a relatively modest GPU setup since VAD models are computationally efficient compared to larger AI tasks. Here are the recommended specifications:

-   GPU: A mid-range GPU like an RTX 3060 or 4060 would be sufficient.
-   VRAM: 6-8GB of VRAM should be adequate as the Pyannote VAD model is relatively small.
-   RAM: 8-16GB system RAM is recommended for processing audio files.
-   Storage: At least 10GB for the model, dependencies, and your audio files.
-   CUDA: Make sure the instance has CUDA installed (version 11.0+ recommended).
-   Python: Python 3.8+ with PyTorch installed.

## Selecting an Instance

Follow these steps to set up your environment:

1. Ensure that you have a Vast.ai account
2. Go to the Vast Templates in the Console https://cloud.vast.ai/templates/
3. Select the `PyTorch (CuDNN Runtime)` Template
4. Filter for an instance with:
    - 1 GPU
    - 6-8GB of VRAM
    - 8-16GB system RAM
    - 10GB of storage
5. Select an instance and click rent
6. Install the Vast TLS certificate in your browser to access the notebook server https://docs.vast.ai/instances/jupyter#1SmCz
7. Go to your Instances https://cloud.vast.ai/instances/ and click "Open" to access the jupyter server on your instance.
8. Upload the notebook to the server or create a new notebook

## Installing Dependencies

Let's start by installing the necessary Python packages:

```python
%%bash
pip install pyannote.audio
pip install pydub
pip install librosa
pip install yt-dlp
```

We also need to install FFmpeg for audio processing:

```python
%%bash
apt-get update && apt-get install -y ffmpeg
```

## Setting Up Your Hugging Face Token

Pyannote models are hosted on Hugging Face, so you'll need to set up authentication:

```python
# Make sure you've accepted the user conditions at:
# https://hf.co/pyannote/voice-activity-detection
# https://hf.co/pyannote/segmentation

HF_TOKEN = ""  # Add your token here
```

Ensure that you have accepted the terms for the models at the URLs above. The models are free to use, but you must agree to their terms of service.

## Generating Test Data

For this tutorial, we'll download a sample audio file from Vast.ai's YouTube channel. You can also use your own audio file if you prefer:

```bash
yt-dlp -f "bestaudio" --extract-audio --audio-format wav -o "test.wav" https://www.youtube.com/watch?v=542xENIxKFU
```

## Voice Activity Detection

First, let's set up the VAD pipeline:

```python
import torch

device = torch.device("cuda" if torch.cuda.is_available() else "cpu")

from pyannote.audio import Pipeline

pipeline = Pipeline.from_pretrained(
        "pyannote/voice-activity-detection",
        use_auth_token=HF_TOKEN
    )

# Move pipeline to appropriate device
pipeline = pipeline.to(device)
```

Next, we'll process our audio file to identify speech segments:

```python
# Process the audio file
audio_file = "test.wav"
output = pipeline(audio_file)

print(f"Processing {audio_file} on {device}")
print("Voice activity segments:")

# Get all speech segments
speech_segments = list(output.get_timeline().support())

for i, speech in enumerate(speech_segments):
    # active speech between speech.start and speech.end
    print(f"Segment {i+1}: Speech from {speech.start:.2f}s to {speech.end:.2f}s (duration: {speech.duration:.2f}s)")
```

The output will look like this when using the default audio file:

```
Processing test.wav on cuda
Voice activity segments:
Segment 1: Speech from 6.78s to 51.62s (duration: 44.84s)
Segment 2: Speech from 53.56s to 54.27s (duration: 0.71s)
Segment 3: Speech from 55.55s to 84.76s (duration: 29.21s)
Segment 4: Speech from 86.53s to 89.03s (duration: 2.50s)
...
```

## Saving Speech Segments

Now we'll create a function to extract the identified speech segments from our audio file:

```python
import os
import shutil
from pydub import AudioSegment


def split_audio_by_segments(audio_path, segments, output_dir="output_segments"):
    """
    Split an audio file into multiple files based on speech segments

    Parameters:
    -----------
    audio_path: str
        Path to the input audio file
    segments: list
        List of speech segments (with start and end attributes)
    output_dir: str
        Directory to save the output segments
    """
    # Clear the output directory if it exists
    if os.path.exists(output_dir):
        shutil.rmtree(output_dir)

    # Create output directory
    os.makedirs(output_dir, exist_ok=True)

    # Load the audio file
    audio = AudioSegment.from_file(audio_path)

    # Extract each segment
    for i, segment in enumerate(segments):
        # Convert seconds to milliseconds
        start_ms = int(segment.start * 1000)
        end_ms = int(segment.end * 1000)

        # Extract segment
        segment_audio = audio[start_ms:end_ms]

        # Generate output filename
        filename = os.path.basename(audio_path)
        name, ext = os.path.splitext(filename)
        output_path = os.path.join(output_dir, f"{name}_segment_{i+1:04d}_{start_ms:08d}ms-{end_ms:08d}ms{ext}")

        # Export segment
        segment_audio.export(output_path, format=ext.replace('.', ''))
        print(f"Saved segment {i+1} to {output_path}")
```

Let's apply this function to extract our speech segments:

```python
split_audio_by_segments(audio_file, speech_segments)
```

This will be the output when using the default audio file:

```
Saved segment 1 to output_segments/test_segment_0001_00006780ms-00051617ms.wav
Saved segment 2 to output_segments/test_segment_0002_00053558ms-00054267ms.wav
Saved segment 3 to output_segments/test_segment_0003_00055549ms-00084760ms.wav
Saved segment 4 to output_segments/test_segment_0004_00086532ms-00089029ms.wav
...
```

## Inspecting Results

To verify our results, we'll create a function to play audio files in our Jupyter environment:

```python
import librosa
from IPython.display import Audio, display

def play_audio(file_path, sr=None):
    """
    Play an audio file in a Jupyter notebook.

    Parameters:
    -----------
    file_path : str
        Path to the audio file to play
    sr : int, optional
        Sample rate to load the audio with. If None, uses the file's native sample rate.

    Returns:
    --------
    Audio widget that can be played in the notebook

    Example:
    --------
    >>> play_audio('path/to/audio.wav')
    """
    # Load the audio file
    y, sr = librosa.load(file_path, sr=sr)


    # Return an audio widget to play the sound
    audio_widget = Audio(data=y, rate=sr)
    display(audio_widget)
```

First, we'll play the original audio file. Listen to the first minute or so to get an idea of what it sounds like before VAD.

```python
play_audio(audio_file)
```

Next, we'll listen to the first three clips to verify that we have isolated the speech in our test file:

```python
import os
audio_dir = "./output_segments/"

audio_files = os.listdir(audio_dir)
audio_files.sort()

n_clips = 3

for fname in audio_files[0:n_clips]:
    play_audio(audio_dir + fname)
```

You'll see that the audio lengths match up with the `speech_segments` output:

```
Segment 1: Speech from 6.78s to 51.62s (duration: 44.84s)
Segment 2: Speech from 53.56s to 54.27s (duration: 0.71s)
Segment 3: Speech from 55.55s to 84.76s (duration: 29.21s)
...
```

Listening to the output files, we can see that we have effectively isolated the speech.

## Conclusion

With this implementation, you now have a working Voice Activity Detection system that can identify and extract speech segments from audio files. This forms an excellent foundation for more advanced audio processing tasks like speech recognition, speaker diarization, or audio content analysis.

Look out for more content from Vast about all these other types of tasks!


Run state-of-the-art Voice Activity Detection (VAD) with PyAnnote Audio on Vast.ai for cost-effective speech processing. Extract speech segments easily!

Voice Activity Detection (VAD) with Pyannote on VAST


# Transcribing Audio with Whisper Large V3 on Vast.ai

## Background

With the rise of Generative AI, there have been many advancements in the field of speech recognition. One of the most notable is the development of Whisper, a family of open-source speech recognition models from OpenAI that can transcribe audio in multiple languages. Companies use this technology to transcribe customer service calls, summarize meetings, and create interactive voice assistants. Other applications might be extracting structured data from audio or video about what is being said.

Whisper Large V3, the latest version of the Whisper models, offers improved accuracy and performance over previous versions. In this guide, we'll show you how to set up and run Whisper Large V3 for batch audio transcription. This guide is built for a backend data pipeline, but can be easily modified to interact with users directly.

With Vast, you can run this model on extremely affordable and powerful GPUs to increase the speed of your transcription pipeline. You can see the script that this is based upon [here](https://bitbucket.org/!api/2.0/snippets/jsbcannell/bqz47r/5b696af8939ef28b17138e8162479bb7c03e9705/files/transcriber.py).

## Setting Up the Environment

To deploy on Vast, we will use the Vast AI Template for Pytorch: [PyTorch (cuDNN Runtime)](https://cloud.vast.ai?template_id=e6330219a022c9511f290c3c2d44592d). It has a lot of the libraries that we need, and comes with ssh and jupyterlab out of the box for us. For GPU's the L40s are very cost effective for reasonable batch sizes and the model itself is pretty small. Finally, the dataset is fairly large and gets expanded when creating the dataset splits, so we recommend having 110GB or more of disk space just to be safe.

We will take the requirements.txt file from [here](https://bitbucket.org/!api/2.0/snippets/jsbcannell/k7b4L6/bf17c4b1248ce536e43fe54d85b8efebe59b7b69/files/whisper_vast_requirements.txt) to install the necessary libraries to the template so that we can run the script.

## Running the Script:

### Setting up Transformers

First, let's look at how to set up our transcription pipeline. We'll use Hugging Face's transformers library to load and run the model.

    ```python
    import os
    import torch
    from huggingface_hub import snapshot_download
    from transformers import (
        pipeline,
        AutoModelForSpeechSeq2Seq,
        AutoProcessor,
        logging
    )

    # Configure model details
    MODEL_DIR = "./model"
    MODEL_NAME = "openai/whisper-large-v3-turbo"
    MODEL_REVISION = "41f01f3fe87f28c78e2fbf8b568835947dd65ed9"

    # Create model directory and download model files
    os.makedirs(MODEL_DIR, exist_ok=True)
    snapshot_download(
        MODEL_NAME,
        local_dir=MODEL_DIR,
        ignore_patterns=["*.pt", "*.bin"],
        revision=MODEL_REVISION,
    )
    ```

### Creating the ASR Pipeline

Next, we'll set up our Automatic Speech Recognition (ASR) pipeline. This involves loading the model and processor, and configuring them for optimal performance:

    ```python
    def initialize_asr_pipeline():
        print("Setting up pipeline")

        processor = AutoProcessor.from_pretrained(MODEL_NAME)
        device = "cuda" if torch.cuda.is_available() else "cpu"

        model = AutoModelForSpeechSeq2Seq.from_pretrained(
            MODEL_NAME,
            torch_dtype=torch.float16,
            low_cpu_mem_usage=True,
            use_safetensors=True,
            device_map=device
        )
        model.generation_config.language = "<|en|>"

        # Configure feature extractor
        feature_extractor = processor.feature_extractor
        feature_extractor.sampling_rate = 16000
        feature_extractor.return_tensors = 'pt'

        # Create pipeline
        asr_pipeline = pipeline(
            "automatic-speech-recognition",
            model=model,
            tokenizer=processor.tokenizer,
            feature_extractor=feature_extractor,
            torch_dtype=torch.float16
        )

        return asr_pipeline
    ```

### Loading and Processing Audio Data

We'll use the Hugging Face datasets library to load our audio data. This example uses LibriSpeech, but you can modify it to use your own dataset:

    ```python
    def load_dataset(dataset_name):
        from datasets import load_dataset, Audio

        print("Loading dataset", dataset_name)
        dataset = load_dataset(dataset_name, "clean", split="validation")
        dataset = dataset.cast_column("audio", Audio(sampling_rate=16000))
        print("Dataset loaded")
        return dataset["audio"]

    def batch_audio(dataset, batch_size):
        for i in range(0, len(dataset), batch_size):
            print(f"Yielding batch {i}")
            yield dataset[i : i + batch_size]
    ```

### Running the Transcription

Finally, we'll put it all together to transcribe our audio files in batches. Because we defined the functions above, actually running the transcription is very simple.

    ```python
    # Initialize pipeline
    asr_pipeline = initialize_asr_pipeline()

    # Load dataset
    DATASET_NAME = "openslr/librispeech_asr"  # You can change this to your dataset
    dataset = load_dataset(DATASET_NAME)

    # Process batches
    results = []
    for batch in batch_audio(dataset, batch_size=32):
        print("Processing batch")
        with torch.no_grad():
            transcriptions = asr_pipeline(batch)
        print("Batch processed")
        results.extend(transcriptions)

    print(results)
    ```

## Results:

Some of our results are below:

```
'Mr. Quilter is the apostle of the middle classes, and we are glad to welcome his gospel.', " Nor is Mr. Quilter's manner less interesting than his matter.", ' he tells us that at this festive season of the year with christmas and roast beef looming before us similes drawn from eating and its results occur most readily to the mind'
```

These can be compared to the original audio dataset [here](https://huggingface.co/datasets/hf-internal-testing/librispeech_asr_dummy). The audio in this dataset is very clear and easy to transcribe.

We can see that the text that is returned from this pipeline is basically perfect for the samples that we show. We would expect that the model's performance would drift if it were run on a private dataset specific to your use case, or if the audio was not as clear as this dataset. Luckily since we have access to the weights, this model can be fine-tuned to your specific use case.

## Key Features

- **Batch Processing**: The code processes audio files in batches of 32 for improved efficiency. This is very important for running this model as it benefits a lot from batching on the GPU.
- **Transformers Pipeline**: This is a very easy way to run the model and handles a lot of boilerplate code to properly prepare the model and the individual samnples for inference. This is extra important as highly performant Whisper serving frameworks aren't released yet.
- **Flexible Dataset Support**: Can work with any dataset hosted on Hugging Face's datasets Hub, with minimal modification for specific dataset schema.

## Tips for Best Performance

1. Adjust the batch size based on your GPU memory to ensure to fully utilize the tensor cores on the GPU.
2. Consider preprocessing your audio files to remove noise or normalize volume
3. Process the audio outputs with an LLM to automatically see if the text is coherent, if it isn't, that might be a good sign that the audio isn't clear.
4. Use these outputs to speed up humans doing the transcription and get a faster transcription with humans in the loop.
5. Leverage the faster and accurately transcribed outputs to fine-tune Whisper to your specific use case, increasing the accuracy and speed of your data flywheel and value to your business.

This implementation provides a solid foundation for batch audio transcription using Whisper Large V3. You can modify the dataset loading code to work with your own audio files, and adjust the batch size based on your available GPU memory.

Now you can see how with Vast, you can save a meaningful amount of money on labor-intensive tasks like transcription with an affordable and powerful GPU. Happy Transcribing!


Transcribe audio efficiently with Whisper Large V3 on Vast.ai. Leverage powerful GPUs for fast, accurate batch processing using Hugging Face’s transformers library.

Posts about: Speech Recognition

Voice Activity Detection (VAD) with Pyannote on VAST

Transcribing Audio with Whisper Large V3 on Vast.ai

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