FOAdapter: A Plug-and-Play Spatial Tokenizer for Low-Bitrate FOA Speech Coding
Authors: Kirak Kim, Yoonjeong Park, Juhan Nam, Sungyoung Kim, Minje Kim
Organizations: Graduate School of Culture Technology, KAIST, Daejeon, Republic of Korea · Kim Jaechul Graduate School of AI, KAIST, Daejeon, Republic of Korea · Siebel School of Computing and Data Science, University of Illinois Urbana-Champaign, USA
Spatially encoded speech conveys not only linguistic information but also spatial cues. This enables listeners to distinguish and localize speakers, contributing to more natural and realistic auditory experiences in applications such as remote conferencing and immersive media. Neural speech codecs (NSCs) enable high-quality reconstruction of monaural speech at increasingly low bitrates while preserving linguistic information, but their ability to effectively preserve spatial cues remains limited. We propose FOAdapter, a plug-and-play spatial tokenizer that leverages a pretrained monaural speech codec to efficiently represent both linguistic and spatial information in first-order ambisonic (FOA) speech at low bitrates. Comprehensive evaluation demonstrates that FOAdapter consistently preserves spatial information across all evaluated codecs. Source code and audio examples are available on our demo website.
Figures & tables
Figure 1: DOA error of reconstructed FOA signal versus total bitrate on the Spatial LibriSpeech test set. FOAdapter achieves approximately 7∘ error across the evaluated monaural codecs using a 275-bit/s spatial stream.
Figure 2: Overview of the proposed FOAdapter framework.
Total bitrate
Spatial fidelity
Speech quality
Intelligibility
Codec
Coding
(kbps)
Az. ↓
Ele. ↓
DOA ↓
RTF ↓
UTMOS ↑
PESQ ↑
SNR ↑
STOI ↑
WER ↓
Ground Truth
–
–
–
–
–
2.127
4.644
∞
1.000
4.47
DAC-1
CI
3.000
12.95
6.65
15.69
1.5271
1.277
1.207
1.31
0.716
51.95
+ FOAdapter
1.025
4.45
4.26
6.86
1.4893
1.269
1.146
−0.19
0.656
76.10
DAC-2
CI
6.000
4.70
3.39
6.42
1.521
1.369
1.560
3.34
0.802
23.22
+ FOAdapter
1.775
4.43
4.18
6.78
1.4879
1.319
1.338
1.23
0.732
42.03
Table 1: Reconstruction on the Spatial LibriSpeech test set. FOAdapter framework bitrates include a 0.275-kbps spatial stream.
Neural speech codecs provide discrete tokens for speech generation. Lower frame rates shorten token sequences, but each token must then carry more linguistic and acoustic information. Variable-frame-rate codecs address this trade-off by adapting token allocation to speech content, but threshold-based allocation lets content determine the token count, so the sequence length a speech generator must produce varies across utterances and datasets at a fixed setting. We present RubatoCodec, a variable-frame-rate speech codec built on X-Codec-2 with an exact utterance-level event budget and streaming decoding. Its Adaptive Firing mechanism uses changes in codec features and waveform spectra to divide speech into variable-duration events without a learned boundary predictor. Discrete codes and an explicit duration represent each event, allowing a speech generator to predict and decode one event at a time. Controlled experiments show that adaptive allocation outperforms fixed-rate coding in reconstruction fidelity and intelligibility at low event rates. A model trained on large-scale multilingual speech achieves competitive reconstruction quality across English, Japanese, and Mandarin, with the best intelligibility preservation among the compared codecs on the evaluated English and Japanese benchmarks at comparable bitrates. In a matched text-to-speech comparison, the adaptive representation yields lower WER and higher UTMOS and UTMOSv2 with comparable speaker similarity. The speech generator also learns the codec's utterance-level event budget without an explicit generation-time rate constraint. These results demonstrate that budgeted adaptive speech representations support both high-quality reconstruction and streaming text-to-speech generation. Code and weights are available at https://github.com/third-intelligence/RubatoCodec and https://huggingface.co/third-intelligence/RubatoCodec.
Neural audio codecs provide compact discrete representations for speech generation and manipulation. However, most codecs organize tokens as frame-level sequences, making it difficult to study or intervene on global factors of variation. In this work, we propose the Latent Audio Tokenizer for Token-space Editing (LATTE) that appends a fixed set of learnable latent tokens to the audio feature sequence and retains only these tokens for quantization and decoding. This design produces a compact, non-temporally aligned bottleneck in which each token can aggregate global information across the full utterance. We show that the resulting tokenizer preserves competitive reconstruction quality in low-bitrate speech coding settings while enabling simple token-space interventions. In particular, we find that swapping selected latent token positions between utterances can modify global attributes, such as speaker identity and background noise, and we evaluate these interventions on voice conversion and denoising tasks. Our results suggest that compact latent audio tokenizers can support controllable audio manipulation without supervision in task-specific editing models.
Francesco Paissan, Luca Della Libera, Mirco Ravanelli +1
1Mila – Québec AI Institute · 2Université Laval · 3Concordia University
Pretrained spatial audio encoders are increasingly used as general-purpose representations for perceptual tasks, yet their spatial encoding capabilities remain poorly understood. We introduce the Spatial Audio Representation Learning (SARL) benchmark, a controlled framework for evaluating spatial information in pretrained audio models. SARL probes source-level factors (azimuth, elevation, distance, class) and room-level factors (RT60, volume, shape). Experiments across diverse encoders reveal three patterns: input configuration and training paradigm shape spatial encoding; source factors are consistently easier to decode than room factors; and sensitivity analysis under controlled perturbations shows heterogeneous responses to source and room variation. These results reveal systematic biases in current pretrained audio representations. SARL is released as an open-source benchmark for reproducible evaluation of spatial audio representations.
Chuyang Chen, Sivan Ding, Adrian S. Roman +1
Music and Audio Research Laboratory, New York University, USA