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# Autodiarize |
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This repository provides a comprehensive set of tools for audio diarization, transcription, and dataset management. It leverages state-of-the-art models like Whisper, NeMo, and wav2vec2 to achieve accurate results. |
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## Table of Contents |
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- [Installation](#installation) |
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- [Usage](#usage) |
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- [Diarization and Transcription](#diarization-and-transcription) |
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- [Bulk Transcription](#bulk-transcription) |
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- [Audio Cleaning](#audio-cleaning) |
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- [Dataset Management](#dataset-management) |
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- [YouTube to WAV Conversion](#youtube-to-wav-conversion) |
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- [LJSpeech Dataset Structure](#ljspeech-dataset-structure) |
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- [Contributing](#contributing) |
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- [License](#license) |
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## Installation |
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### 1. Clone the repository: |
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```bash |
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git clone https://github.com/your-username/whisper-diarization.git |
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cd whisper-diarization |
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``` |
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### 2. Create a Python virtual environment and activate it: |
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```bash |
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./create-env.sh |
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source autodiarize/bin/activate |
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``` |
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or if you want to ruin your python env |
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### Install the required packages: |
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```bash |
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pip install -r requirements.txt |
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``` |
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## Usage |
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### Diarization and Transcription |
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The `diarize.py` script performs audio diarization and transcription on a single audio file. It uses the Whisper model for transcription and the NeMo MSDD model for diarization. |
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```bash |
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python diarize.py -a <audio_file> [--no-stem] [--suppress_numerals] [--whisper-model <model_name>] [--batch-size <batch_size>] [--language <language>] [--device <device>] |
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``` |
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- `-a`, `--audio`: Path to the target audio file (required). |
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- `--no-stem`: Disables source separation. This helps with long files that don't contain a lot of music. |
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- `--suppress_numerals`: Suppresses numerical digits. This helps the diarization accuracy but converts all digits into written text. |
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- `--whisper-model`: Name of the Whisper model to use (default: "medium.en"). |
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- `--batch-size`: Batch size for batched inference. Reduce if you run out of memory. Set to 0 for non-batched inference (default: 8). |
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- `--language`: Language spoken in the audio. Specify None to perform language detection (default: None). |
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- `--device`: Device to use for inference. Use "cuda" if you have a GPU, otherwise "cpu" (default: "cuda" if available, else "cpu"). |
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### Bulk Transcription |
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The `bulktranscript.py` script performs diarization and transcription on multiple audio files in a directory. |
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```bash |
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python bulktranscript.py -d <directory> [--no-stem] [--suppress_numerals] [--whisper-model <model_name>] [--batch-size <batch_size>] [--language <language>] [--device <device>] |
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``` |
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- `-d`, `--directory`: Path to the directory containing the target files (required). |
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- Other arguments are the same as in `diarize.py`. |
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### Audio Cleaning |
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The `audio_clean.py` script cleans an audio file by removing silence and applying EQ and compression. |
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```bash |
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python audio_clean.py <audio_path> <output_path> |
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``` |
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- `<audio_path>`: Path to the input audio file. |
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- `<output_path>`: Path to save the cleaned audio file. |
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### Dataset Management |
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The repository includes several scripts for managing datasets in the LJSpeech format. |
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#### Merging Folders |
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The `mergefolders.py` script allows you to merge two LJSpeech-like datasets into one. |
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```bash |
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python mergefolders.py |
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``` |
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Follow the interactive prompts to select the directories to merge and specify the output directory. |
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#### Consolidating Datasets |
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The `consolidate_datasets.py` script consolidates multiple LJSpeech-like datasets into a single dataset. |
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```bash |
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python consolidate_datasets.py |
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``` |
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Modify the `base_folder` and `output_base_folder` variables in the script to specify the input and output directories. |
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#### Combining Sets |
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The `combinesets.py` script combines multiple enumerated folders within an LJSpeech-like dataset into a chosen folder. |
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```bash |
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python combinesets.py |
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``` |
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Enter the name of the chosen folder when prompted. The script will merge the enumerated folders into the chosen folder. |
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### YouTube to WAV Conversion |
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The `youtube_to_wav.py` script downloads a YouTube video and converts it to a WAV file. |
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```bash |
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python youtube_to_wav.py [<youtube_url>] |
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``` |
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- `<youtube_url>`: (Optional) URL of the YouTube video to download and convert. If not provided, the script will prompt for the URL. |
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## LJSpeech Dataset Structure |
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The `autodiarize.py` script generates an LJSpeech-like dataset structure for each input audio file. Here's an example of how the dataset structure looks: |
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``` |
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autodiarization/ |
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βββ 0/ |
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β βββ speaker_0/ |
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β β βββ speaker_0_001.wav |
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β β βββ speaker_0_002.wav |
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β β βββ ... |
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β β βββ metadata.csv |
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β βββ speaker_1/ |
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β β βββ speaker_1_001.wav |
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β β βββ speaker_1_002.wav |
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β β βββ ... |
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β β βββ metadata.csv |
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β βββ ... |
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βββ 1/ |
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β βββ speaker_0/ |
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β β βββ speaker_0_001.wav |
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β β βββ speaker_0_002.wav |
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β β βββ ... |
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β β βββ metadata.csv |
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β βββ speaker_1/ |
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β β βββ speaker_1_001.wav |
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β β βββ speaker_1_002.wav |
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β β βββ ... |
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β β βββ metadata.csv |
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β βββ ... |
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βββ ... |
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``` |
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Each input audio file is processed and assigned an enumerated directory (e.g., `0/`, `1/`, etc.). Within each enumerated directory, there are subdirectories for each speaker (e.g., `speaker_0/`, `speaker_1/`, etc.). |
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Inside each speaker's directory, the audio segments corresponding to that speaker are saved as individual WAV files (e.g., `speaker_0_001.wav`, `speaker_0_002.wav`, etc.). Additionally, a `metadata.csv` file is generated for each speaker, containing the metadata for each audio segment. |
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The `metadata.csv` file has the following format: |
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``` |
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filename|speaker|text |
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speaker_0_001|Speaker 0|Transcribed text for speaker_0_001 |
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speaker_0_002|Speaker 0|Transcribed text for speaker_0_002 |
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... |
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``` |
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Each line in the `metadata.csv` file represents an audio segment, with the filename (without extension), speaker label, and transcribed text separated by a pipe character (`|`). |
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