From 2.1M Hours of Raw Broadcast Audio to Finnish Speech SSL: Confidence Filtering and Architectural Analysis at Scale
Authors: Yaroslav Getman, Tamás Grósz, Tommi Lehtonen, Mikko Kurimo
Organizations: Dept. lnformation and Communications Engineering Aalto University Espoo, Finland · Walton Institute, South East Technological University, Ireland Aalto University Espoo, Finland · Finnish National Audiovisual Institute (KAVI) Helsinki, Finland
Broadcast archives offer scale and diversity for self-supervised learning (SSL) but contain substantial noisy, non-speech, and multilingual content. Prior work typically relies on voice activity detection or language identification (LID) to extract speech, ensuring presence rather than quality. Whether and how to filter unlabeled speech by quality for SSL, particularly from noisy broadcasts, remains underexplored. We present a large-scale Finnish SSL study starting from 2.1M hours of unconstrained broadcast audio. We propose automatic speech recognition (ASR) confidence as a proxy for pretraining data quality and show 3-7% relative WER improvements over LID-based filtering despite a smaller training set. We further address the lack of controlled architectural comparisons in speech SSL by evaluating Transformer, Macaron-Net, Conformer, and E-Branchformer at matched parameter budgets. Our models improve upon prior Finnish SSL baselines, particularly at Base scale and out-of-domain.
TABLE I : Comparison of encoder architectures. Equations illustrate the sequence of sub-modules and residual connections within a single encoder block. Merge(⋅) denotes the depth-wise convolutional fusion module. Layer normalizations are omitted for clarity.
Fig. 1 : Impact of filtering criteria on downstream validation WER and pretraining data retention rate. All confidence thresholds ( Cchar≥τ ) are applied to deduplicated VAD segments.
Fig. 2 : Pretraining data preparation pipeline. We reduce 2.1M hours of raw broadcast archives to a high-quality 471K-hour subset through VAD, transcript-based deduplication, and the confidence-based filtering threshold τ=−0.1 optimized in Figure 1 .
Backend
L
Enc.
FFN
Params
In-domain
Out-of-domain
LP
FP’16
FP’20
CV
FLEURS
KAVI
Transformer
6
1024
4096
90.6M
28.30
14.08
11.58
15.60
17.13
31.49
12
768
3072
95.0M
22.28
10.72
8.70
9.74
13.02
24.85
18
768
1536
95.1M
21.33
10.64
8.66
8.99
12.71
23.91
24
576
2048
96.4M
21.66
10.69
8.73
8.77
12.85
24.57
Macaron-Net
12
768
1536
95.1M
21.80
10.58
9.10
8.62
12.54
25.22
TABLE II : WER (%) for wav2vec 2.0 Base variants across Finnish benchmarks. The best results within the same encoder backend are underlined , and the best overall results are shown in bold .
Backend
Config
Params
In-domain
Out-of-domain
(L / Enc. / FFN)
LP
FP’16
FP’20
CV
FLEURS
KAVI
Base models
Transformer [ 14 ] (baseline)
12 / 768 / 3072
95.0M
22.27
10.92
8.42
10.61
13.27
27.45
Transformer (ours)
12 / 768 / 3072
95.0M
22.28
10.72
8.70
9.74
13.02
24.85
18 / 768 / 1536
95.1M
21.33
10.64
8.66
8.99
12.71
23.91
Conformer
12 / 576 / 2048
96.7M
18.89
10.08
8.07
6.46
11.20
23.71
TABLE III : WER (%) across scales and encoder backends. Best results per scale are underlined and best overall in bold . LP = Lahjoita puhetta (colloquial Finnish), FP = Finnish Parliament, CV = Common Voice, KAVI = real-world broadcast speech.
Large-scale mined corpora provide abundant training data for end-to-end speech-to-speech translation (S2ST) but may contain noise, misalignment, and semantic errors. Filtering noisy data is crucial to maintain robust speech translation performance. We study how to train an audio-language model to make keep/drop decisions on paired speech directly from audio. To obtain reliable supervision without manual labels, we adopt a scalable two-stage Rank-to-Distill strategy. A lightweight ranker generates keep/drop pseudo-labels from noisy speech pairs, then trains an audio large language model to predict keep/drop directly from raw paired speech. The resulting model jointly captures acoustic fidelity and cross-lingual semantic consistency for the selection of speech-conditioned data. Experiments on CVSS-C and SpeechMatrix show consistent improvements over unfiltered training, yielding up to +1.4 ASR-BLEU for end-to-end S2ST.
Qixu Chen, Satoshi Nakamura
School of Data Science, The Chinese University of Hong Kong, Shenzhen, China · School of Artificial Intelligence The Chinese University of Hong Kong, Shenzhen, China
Speech language models (speech LLMs) can generate plausible outputs from audio that contains no usable speech evidence. We study this failure as a pre-generation support-estimation problem and present SURE-Voice, a training-free front end that decides whether an audio prompt contains intelligible speech evidence before calling a speech LLM. We build SURE-Challenge with a 640-example SURE-Core split and a 1,920-example SURE-Extended split derived from 120 LibriSpeech source utterances. Using one fixed operating point, an energy screen plus Whisper token confidence raises unsupported accuracy on the held-out Extended test from 0.000--0.133 to 0.919 for six non-degenerate speech LLM backbones, while supported accuracy remains 0.919--0.970 and downstream calls fall from 480 to 287. A 500-clip ESC-50 sanity set shows the same pattern on real environmental audio, with vocal non-speech as a residual failure mode. An overlap diagnostic shows that source attribution remains separate from speech-evidence filtering. The evidence supports a controlled benchmark baseline and a deployment-oriented analysis; it does not establish universal robustness to semantic answerability, gain variation or natural conversations.
Audio self-supervised learning (SSL) aims to learn general-purpose representations from large-scale unlabeled audio data. While recent advances have been driven mainly by generative reconstruction objectives, contrastive approaches remain less explored, partly due to the difficulty of designing effective audio augmentations and the large batch sizes required for contrastive pre-training. We introduce AudioMosaic, a contrastive learning-based audio encoder for general audio understanding. During pre-training, AudioMosaic constructs positive pairs by applying structured time-frequency masking to spectrogram patches, which reduces memory usage and enables efficient large-batch training. Compared with generative approaches, the AudioMosaic encoder learns more discriminative utterance-level representations that demonstrate strong transferability across datasets, domains, and acoustic conditions. Extensive experiments show that AudioMosaic achieves state-of-the-art performance on several standard audio benchmarks under both linear probing and fine-tuning. We further show that integrating the pretrained AudioMosaic encoder into audio-language models improves performance on audio-language tasks. The code is publicly available in our GitHub repository.
Hanxun Huang, Qizhou Wang, Xingjun Ma +3
School of Computing and Information Systems, The University of Melbourne, Australia · Institute of Trustworthy Embodied AI, Fudan University, China · Baskin School of Engineering, University of California, Santa Cruz, USA