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RealSSA-Sim is derived from LRS3 (TED talks, CC BY-NC-ND 4.0), LibriSpeech (CC BY 4.0) and several noise collections with non-commercial or per-clip terms. Access is limited to non-commercial academic research, and requests are reviewed and approved manually. Read "Access and license" in the dataset card before submitting.

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RealSSA-Sim

RealSSA-Sim is the simulated counterpart of RealSSA-Real for audio-visual target-speaker extraction. Each example has a 4-channel 16 kHz mixture, the 4-channel target speech, the target speaker's mouth crops, and a JSON file describing how the mixture was simulated. All sources are static, in simulated rooms of four scene types: small_meeting, medium_indoor, large_public and outdoor.

Split Directory Examples Shards
train tr/ 100,826 202
validation cv/ 9,133 19
test tt/ 9,169 19

Four training examples (00001_01_87828, 00002_05_59973, 00020_00_46434, 50007_35934) have no JSON file or mouth crops and are not included.

Falcon project and resources

RealSSA accompanies Falcon: Efficient Streaming Audio-Visual Target Speaker Extraction for Real-World Acoustic Scenes (NeurIPS 2026).

Resource Link What it provides
Source code JusperLee/Falcon Training, evaluation, offline inference and Falcon Studio
Project page cslikai.cn/Falcon Method overview and demonstration samples
Paper model JusperLee/Falcon falcon.pt; 0.5 M trainable parameters, plus the frozen visual encoder
Larger inference model JusperLee/Falcon-Big falcon-big.pt; Studio's Big option, separate from the paper model
Browser demo Falcon Studio Space Record up to 60 s, then separate on ZeroGPU
Local / live inference Falcon Studio guide Local inference, live microphone separation and meeting routing
Space implementation space/ Source and deployment instructions for the ZeroGPU demo
Simulated benchmark RealSSA-Sim Paired multichannel mixtures, target speech and RGB mouth crops
Real-recorded benchmark RealSSA-Real Paired AV test inputs, enrollment audio and recording-level transcripts

The Space is a record-then-separate demo. Use the local Studio for live separation and the Python/CLI examples below for dataset inference. Dataset access is reviewed separately; a public model or demo does not grant access to either dataset. Code and weights use Apache 2.0; the datasets retain their own access and usage conditions.

Access and license

RealSSA-Sim contains the faces and voices of real people from TED talks. Access is gated on Hugging Face, and each request is reviewed and approved manually.

Permitted use: non-commercial academic research. Publishing aggregate results is allowed.

Prohibited:

  • Any commercial use.
  • Sharing access, or redistributing the data or derived files that contain the speakers' faces or voices, such as mouth crops or separated speech estimates.
  • Identifying, tracking or profiling speakers, or creating synthetic media of them.

The dataset is released under CC BY-NC-ND 4.0, the license of the TED talks it is built from. Every source below keeps its own terms, and users must follow them. Mixing, reverberation and cropping create adapted versions of the source material. Rights holders can ask for content to be removed at thucs.kaili@gmail.com, and we will remove it.

Component Source Mixtures containing it Terms
Target speech, mouth crops, transcripts LRS3 (TED and TEDx talks) 100% TED talks: CC BY-NC-ND 4.0; LRS3 terms apply
Interfering speech LibriSpeech train-clean-360 100% CC BY 4.0
Noise WHAM! noise 77.8% See original terms
Noise BBC Sound Effects 64.5% RemArc licence: non-commercial only, credit the BBC
Noise AudioSet 28.4% YouTube clips; copyright stays with the uploaders
Noise VGGSound 20.5% YouTube clips; copyright stays with the uploaders
Noise Clotho v2 9.2% Freesound clips; per-clip licenses
Noise FreeSound 6.3% Per-clip licenses
Noise DCASE (synthetic) 1.8% See original terms
Noise SoundBible 1.3% Per-clip licenses
Noise VoiceBank-DEMAND 1.2% CC BY 4.0
Noise ESC-50 1.0% CC BY-NC
Noise FSD50K 0.6% Per-clip Creative Commons licenses
Noise AVE 0.4% YouTube clips; copyright stays with the uploaders
Noise MUSIC21 0.2% YouTube clips; copyright stays with the uploaders
Noise DCASE (real) 0.1% See original terms

Layout

The data is stored as WebDataset tar shards of 500 examples, {tr,cv,tt}/{tr,cv,tt}-NNNNNN.tar. Train examples are shuffled across shards. Each example <id> is four consecutive tar members:

<id>.mix.wav      4-channel PCM16 mixture, 16 kHz
<id>.target.wav   4-channel PCM16 target speech, 16 kHz
<id>.json         scene, room (dimensions, mic center, T60), components (type, source, SIR, position), target position, transcript
<id>.mouths.pt    uint8 RGB mouth crops, torch tensor [T,96,96,3]

Component types are near and far (interfering speech) and env and sudden (noise). The path fields in the JSON files are the generation-time file paths and only identify the source clips.

Download a split or one shard

Request access on this page and wait for manual approval, then authenticate with the approved account:

python -m pip install -U huggingface_hub
hf auth login

Start with one test shard instead of downloading the complete dataset (about 522 GB):

from huggingface_hub import hf_hub_download, snapshot_download

shard = hf_hub_download(
    repo_id="JusperLee/RealSSA-Sim", repo_type="dataset",
    filename="tt/tt-000000.tar", local_dir="RealSSA-Sim",
)
# Optional: download all test shards. Use tr/* or cv/* for other splits.
# snapshot_download(
#     repo_id="JusperLee/RealSSA-Sim", repo_type="dataset",
#     allow_patterns=["README.md", "tt/*.tar"], local_dir="RealSSA-Sim",
# )

Read one example

Install NumPy, SoundFile, PyTorch and OpenCV in your inference environment. This example reads the four fields from one local shard without extracting the archive or using metadata source paths:

import io
import json
import tarfile
import numpy as np
import soundfile as sf
import torch

with tarfile.open(shard, "r:*") as archive:
    members = {m.name: m for m in archive.getmembers() if m.isfile()}
    key = next(name[:-len(".mix.wav")] for name in members if name.endswith(".mix.wav"))

    def read_field(suffix):
        with archive.extractfile(members[key + suffix]) as stream:
            return stream.read()

    mix, sr = sf.read(io.BytesIO(read_field(".mix.wav")), dtype="float32", always_2d=True)
    target, target_sr = sf.read(io.BytesIO(read_field(".target.wav")), dtype="float32", always_2d=True)
    metadata = json.loads(read_field(".json"))
    mouths = torch.load(io.BytesIO(read_field(".mouths.pt")), map_location="cpu", weights_only=True).numpy()

assert sr == target_sr == 16000
assert mix.shape[1] == target.shape[1] == 4
assert mouths.dtype == np.uint8 and mouths.shape[1:] == (96, 96, 3)
print(key, mix.shape, target.shape, mouths.shape)

The audio arrays are [samples, channels]; the mouth array is [frames, height, width, RGB]. The JSON transcript describes the target utterance. Source paths in metadata identify generation inputs and are not portable local paths. A shard contains at most 500 examples; the final shard of each split is smaller.

Run Falcon on the example

Clone the code repository and install the Studio environment:

git clone https://github.com/JusperLee/Falcon.git
cd Falcon
python -m pip install -r studio/requirements.txt

Continue the Python example above from the repository root. This explicitly selects microphone channel 0 and converts RGB mouth crops to grayscale. It is an inference example, not an assertion of the paper's channel protocol:

import sys
import cv2

sys.path.insert(0, "studio")
from falcon_runtime import FalconRuntime
from infer_file import extract

audio = np.ascontiguousarray(mix[:, 0], dtype=np.float32)
mouth = np.stack([cv2.cvtColor(frame, cv2.COLOR_RGB2GRAY) for frame in mouths])
frames = min(len(mouth), len(audio) // 640, len(target) // 640)
audio = audio[:frames * 640]
mouth = mouth[:frames]
clean_target = target[:frames * 640, 0]  # same channel and duration for paired metrics

model = FalconRuntime.from_pretrained("JusperLee/Falcon", device="auto")
estimate = extract(model, audio, mouth)
sf.write("falcon_estimate.wav", estimate, 16000, subtype="PCM_16")

Use JusperLee/Falcon-Big for the larger inference variant. Weights download automatically on first use. extract performs whole-clip inference and applies the model's validation transform (center crop to 88×88, divide by 255, normalize with mean 0.421/std 0.165). Do not normalize mouth pixels a second time. Whole-clip inference requires at least one second; FalconRuntime.infer instead accepts exactly 2 s of audio and 50 mouth frames.

Training and evaluation notes

  • Preserve the published train/validation/test split. Use the paired target on the same microphone channel and aligned duration for reference-based metrics. Report channel selection, visual conversion and segmentation with results.
  • The released TAR layout is not read directly by the current folder-based training loader. Adapt samples into tr/, cv/, tt/ with mix/, target/, mouths/, or implement a WebDataset loader. Convert multichannel audio and RGB mouth tensors explicitly. See data preprocessing and Falcon configuration.
  • The shards ship mixtures and target speech, not the original individual near-speaker signals. Generation-time components[].path entries are provenance only. The original speaker-contrastive loss needs valid near-speaker negatives; the indexing script's fallback to the mixture is not an equivalent reproduction of that setup.
  • Published data and model inference were not re-evaluated as part of this documentation update.

Citation

When using RealSSA or Falcon, cite the accompanying paper:

@inproceedings{sang2026falcon,
  title     = {Efficient Streaming Audio-Visual Target Speaker Extraction for Real-World Acoustic Scenes},
  author    = {Sang, Wendi and Li, Kai and Li, Yifan and Huang, Jianqiang and Hu, Xiaolin},
  booktitle = {Advances in Neural Information Processing Systems (NeurIPS)},
  year      = {2026}
}
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