We propose TQCodec, a neural audio codec designed for high-bitrate, high-fidelity music streaming. Unlike existing neural codecs that primarily target ultra-low bitrates (<= 16kbps), TQCodec operates at 44.1 kHz and supports bitrates from 32 kbps to 128 kbps, aligning with the standard quality of modern music streaming platforms. The model adopts an encoder-decoder architecture based on SEANet for efficient on-device computation and introduces several enhancements: an imbalanced network design for improved quality with low overhead, SimVQ for mid-frequency detail preservation, and a phase-aware waveform loss. Additionally, we introduce a perception-driven band-wise bit allocation strategy to prioritize perceptually critical lower frequencies. Evaluations on diverse music datasets demonstrate that TQCodec achieves superior audio quality at target bitrates, making it well-suited for high-quality audio applications.
Figures & tables
Codec
SR (khz)
BR (kbps)
CB
FR
TQCodec
44.1
32, 64, 128
5/ 10/ 20
689
DAC [ 6 ]
44.1
16
18
86
Encodec [ 3 ]
24/ 48
24
16
75 /150
Table 1: Comparison of compression approaches (SR: sample rate, BR: bit rate, CB: codebook, FR: frame rate).
Figure 1: Illustration of our subband modeling.
Name
Number of tracks
MusDBHQ [ 12 ]
150
Jingju [ 4 ]
120
Jamendo [ 1 ]
55,609
Fma [ 2 ]
106,574
Private dataset
100000+
Table 2: The dataset information.
Figure 2: Spectrogram visualization, where the RVQ fails to reconstruct the mid-frequency clearly (above 4000Hz).
Name
LSD ↓
LSD-L ↓
LSD-H ↓
MACs-E ↓
MACs-D ↓
Receptive field ↓
DAC (16kbps)
0.927
0.877
1.016
25G
365G
17706
DAC-small (32kbps)
0.833
0.781
0.936
25G
365G
17706
Ablation on TQCodec (32kbps)
Encodec + DAC RVQ
0.854
0.790
0.983
2.4G
6.31G
2410
+ RSimVQ
0.850
0.772
0.997
2.4G
6.31G
2410
+ Waveform loss
0.844
0.769
0.995
2.4G
6.31G
2410
Table 3: Ablation study results of the proposed codec, where LSD-L refers to frequency below 16000Hz and LSD-H refers to frequency above 16000Hz.
Model
Bitrates
LSD-L ↓
LSD-H ↓
SNR ↑
Ogg-Vorbis [ 11 ]
48kbps
0.763
2.847
16.998
TQCodec
43kbps
0.718
0.954
16.074
+ Subband
43kbps
0.702
1.163
16.88
Table 4: Comparison with baselines to verify the subband modeling.
MOS
Average
4.18
Pass Rate
100%
Excellent Rate
77.8%
Preference
HQ (Ogg) Better
39.4%
AICodec Better
39.1%
Table 5: Subjective Listening Test: AICodec-64kbps vs. Ogg-96kbps (HQ)
Neural audio codecs are a fundamental component of modern speech generation systems. While recent codecs achieve increasingly low bitrates, reducing frame rate remains challenging, as each token must preserve more information while maintaining reconstruction quality. We present ZipCodec, a streaming neural speech codec operating at 6.25 Hz and 0.80 kbps with a theoretical latency of 160 ms. Our approach combines large-scale WavLM distillation with a redesigned transformer-based architecture, a scalar spherical quantizer, and a latency-aware streaming decoder. Experiments show that ZipCodec substantially outperforms existing streaming codecs at comparable bitrates in both reconstruction and downstream tasks, while operating at a significantly lower frame rate. Despite its 842M parameters, ZipCodec achieves real-time single-stream inference on a consumer-grade CPU. Demo samples, code and checkpoints are available at https://lucadellalib.github.io/zipcodec-web/.
Luca Della Libera, Cem Subakan, Mirco Ravanelli
Concordia University · Mila-Quebec AI Institute · Universit´e Laval
High-fidelity audio coding requires faithful reconstruction of perceptually important high-frequency components, which are often lost due to the aggressive temporal decimation and quantization of neural codecs. State-of-the-art neural audio coding methods rely on adversarial learning to hallucinate these missing components, yielding perceptually plausible but imprecise reconstructions; a limitation for high-fidelity music applications. As a remedy, we propose high-efficiency neural audio coding (HE-NAC), which provides a multi-band extension of the popular residual vector quantization (RVQ) scheme. HE-NAC introduces an effective training mechanism that harmonizes a bitrate-free blind upsampling process, residual learning to compensate it with minimal bitrate, and a multi-head decoder architecture for band-specific reconstruction. As a result, HE-NAC achieves a balanced trade-off, reliably reconstructing the core-band (≤ 3 kHz) component while preserving overall sound fidelity. Subjective listening tests and objective evaluations demonstrate that HE-NAC outperforms both the DAC baseline and MPEG HE-AAC at equal or lower bitrates.
Darius Pétermann, Wootaek Lim, Seungkwon Beack +1
Indiana University, Bloomington, IN, USA · Electronics and Telecommunications Research Institute, Daejeon, South Korea · University of Illinois at Urbana-Champaign, Urbana, IL, USA
Speech Bandwidth Extension improves clarity and intelligibility by restoring/inferring appropriate high-frequency content for low-bandwidth speech. Existing methods often rely on spectrogram or waveform modeling, which can incur higher computational cost and have limited high-frequency fidelity. Neural audio codecs offer compact latent representations that better preserve acoustic detail, yet accurately recovering high-resolution latent information remains challenging due to representation mismatch. We present CodecFlow, a neural codec-based BWE framework that performs efficient speech reconstruction in a compact latent space. CodecFlow employs a voicing-aware conditional flow converter on continuous codec embeddings and a structure-constrained residual vector quantizer to improve latent alignment stability. Optimized end-to-end, CodecFlow achieves strong spectral fidelity and enhanced perceptual quality on 8 kHz to 16 kHz and 44.1 kHz speech BWE tasks.
Bowen Zhang, Junchuan Zhao, Ian McLoughlin +2
Singapore Institute of Technology, Singapore · Nanyang Technological University, Singapore · National University of Singapore, Singapore