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REAL-T: Real Conversational Mixtures for Target Speaker Extraction

REAL-T Logo

Paper REAL-T Page Datasets on Hugging Face

Pipeline

Introduction

Target Speaker Extraction (TSE) models have demonstrated impressive performance on synthetic datasets such as LibriMix and WSJMix. However, these benchmarks lack the acoustic realism and conversational dynamics of actual human interactions — such as spontaneous speech, overlapping turns, and environmental noise — limiting their relevance to real-world scenarios.

Efforts like REAL-M and LibriCSS have attempted to bridge this gap. REAL-M collects simultaneous speech in shared environments through multi-speaker read-aloud sessions, while LibriCSS re-records isolated utterances via synchronized playback. Though valuable, these datasets still fall short of capturing the nuances of real conversations — including irregular speaker turns, sporadic utterances, and authentic background conditions.

To address these limitations, we introduce REAL-T, the first conversation-centric dataset specifically designed for TSE under real-world conditions. Built from speaker diarization corpora, REAL-T naturally includes overlapping speech, enrollment-ready segments, and complex conversational behaviors.

Key features of REAL-T include:

  • Multi-lingual: English and Mandarin recordings
  • Multi-genre: Covering diverse conversational scenarios
  • Multi-enrollment: Multiple enrollment utterance from different parts of the conversation

To support controlled evaluation, we define two test sets:

  • BASE: A filtered, balanced subset for initial testing
  • PRIMARY: A more realistic and challenging benchmark

Evaluations reveal that existing TSE models suffer significant performance degradation on REAL-T, highlighting the need for more robust approaches tailored to real conversational speech.

For more details, refer to our paper: REAL-T Paper

Installation

Datasets are at huggingface.

1. Clone the repository

git clone https://github.com/REAL-TSE/REAL-T.git
cd REAL-T

# install wesep
git submodule init
git submodule update

2. Create a Conda environment and install dependencies

conda create -n REAL-T python=3.9
conda activate REAL-T
pip install -r requirements.txt

3. Set up Linux PATH and PYTHONPATH

Please replace $PWD below with the absolute path to this project

$ export PATH=$PWD/FireRedASR/fireredasr/:$PWD/FireRedASR/fireredasr/utils/:$PATH
$ export PYTHONPATH=$PWD/FireRedASR/:$PYTHONPATH
$ export PYTHONPATH=$PWD/wesep/:$PYTHONPATH

4. Prepare Dataset and Checkpoints

Evaluation requires the REAL-T dataset and the ASR model checkpoint FireRedASR-AED-L and whisper-large-v2 from Hugging Face. The dataset must be prepared in a specific format before running evaluation. To automatically set up everything, run:

bash -i ./pre.sh

Evaluation

TSE Inference on REAL-T

The run_tse.sh script below demonstrates how to perform TSE inference with the Wesep toolkit using a BSRNN model trained on VoxCeleb1. You can adapt its input/output structure to suit your own TSE model.

➡️ See Bash Script Parameters: run_tse.sh for detailed setup.

cd REAL-T
bash -i run_tse.sh

ASR-based Evaluation of TSE

The transcribe_and_evaluation.sh script automates the evaluation of ASR outputs using Token Error Rate (TER) metrics. You can run transcription, evaluation, or both, depending on your needs. Before execution, make sure to review the script’s parameters—such as whether to include the Fisher corpus, the evaluation subset, input/output directories, and other configurable options. ➡️ See Bash Script Parameters: transcribe_and_evaluation.sh for detailed setup.

cd REAL-T

# Only ASR
bash -i ./transcribe_and_evaluation.sh 1

# Only Evaluation
bash -i ./transcribe_and_evaluation.sh 2

# Both ASR and Evaluation
bash -i ./transcribe_and_evaluation.sh 1 2

run_tse.sh

This script runs Target Speaker Extraction (TSE) inference for multiple datasets using a specified model. Each dataset will be processed individually, generating separated target speaker audio files.

Variable Name Description
MODEL_NAME 🚩 Name of the TSE model used for inference (e.g., bsrnn_vox1).
DATASETS 🚩 List of datasets to process (e.g., AliMeeting, AMI, CHiME6, AISHELL-4, DipCo). Fisher can also be included if needed.
TEST_SET 🚩 Test subset to use: PRIMARY or BASE.
DEVICE 🚩 Device on which to run inference (cuda for GPU, cpu for CPU).
BASE_META_PATH Base directory containing metadata CSV files for each dataset.
BASE_OUTPUT_DIR Directory where the separated audios will be saved.
TSE_SCRIPT Path to the TSE inference Python script (tse.py).
META_CSV_PATH Path to the CSV file containing mixture and enrolment utterance metadata.
UTTERANCE_MAP_CSV Path to the CSV mapping enrolment utterances to mixture utterances.
OUTPUT_DIR Directory where output audios for each dataset will be stored.

transcribe_and_evaluation.sh

This script automates two main stages for the REAL-T datasets:

  • ASR transcription using different models depending on dataset language.
  • ASR evaluation using TER (Token Error Rate) metrics.

The operation mode is controlled by passing one or two arguments:

  • 1 → only run ASR transcription
  • 2 → only run evaluation
  • 1 2 → run both transcription and evaluation sequentially

Typically, run 1 2 for a full process.

Step Function Description
1 run_asr Transcribe all audio files using the appropriate ASR model (Chinese or English) for each dataset.
2 run_asr_evaluation Evaluate the ASR outputs by calculating Token Error Rate (TER) against the ground-truth transcripts.
Variable Name Description
TEST_SET_DIR 🚩 (Required) Directory containing the ground-truth test set metadata (BASE or PRIMARY).
BASE_DIRS 🚩 (Required) List of base directories containing predicted TSE outputs to be transcribed and/or evaluated, as listed below.
INCLUDING_FISHER Whether to include the Fisher corpus in evaluation (True or False).
MODES Runtime arguments specifying which mode(s) to execute (1 for ASR, 2 for Evaluation).
ASR_SCRIPT Path to the Python script performing ASR inference (asr_inference.py).
EVAL_SCRIPT Path to the Python script performing ASR evaluation (asr_evaluation.py).
CHINESE_ASR_MODEL Name of the model used for Chinese datasets (e.g., FireRedASR-AED-L).
ENGLISH_ASR_MODEL Name of the model used for English datasets (e.g., whisper-large-v2).
MAPPING_CSV_NAME Name of the CSV file mapping mixture audio to clean references (default: tse_audio_mapping.csv).
BASE_DIRS=(
    "YourPath/output/PRIMARY/bsrnn_vox1"
    "YourPath/output/PRIMARY/bsrnn_hr_vox1"
)

Results

The table below compares the performance of several recently proposed TSE models on the simulated Libri2Mix and PRIMARY test sets.

Model Training Data Libri2Mix SI-SDR (dB) PRIMARY zh (%) PRIMARY en (%)
TSELM-L Libri2Mix-360 / 331.73 192.39
USEF-TFGridnet Libri2Mix-100 18.05 67.98 87.27
BSRNN Libri2Mix-100 12.95 81.74 91.20
Libri2Mix-360 16.57 69.80 73.61
VoxCeleb1 16.50 57.61 69.63
BSRNN_HR Libri2Mix-100 15.91 70.03 78.96
Libri2Mix-360 17.99 63.38 74.64
VoxCeleb1 16.38 58.77 66.46

Citation

@inproceedings{li25da_interspeech,
  title     = {{REAL-T: Real Conversational Mixtures for Target Speaker Extraction}},
  author    = {{Shaole Li and Shuai Wang and Jiangyu Han and Ke Zhang and Wupeng Wang and Haizhou Li}},
  year      = {{2025}},
  booktitle = {{Interspeech 2025}},
  pages     = {{1923--1927}},
  doi       = {{10.21437/Interspeech.2025-2662}},
  issn      = {{2958-1796}},
}

Contact

For any questions, please contact: shuaiwang@nju.edu.cn

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