dMel: Speech Tokenization made Simple
- URL: http://arxiv.org/abs/2407.15835v3
- Date: Wed, 21 May 2025 16:55:34 GMT
- Title: dMel: Speech Tokenization made Simple
- Authors: Richard He Bai, Tatiana Likhomanenko, Ruixiang Zhang, Zijin Gu, Zakaria Aldeneh, Navdeep Jaitly,
- Abstract summary: We introduce a novel speech representation (dmel) that discretizes mel-filterbank channels into intensity bins.<n>Our approach demonstrates superior performance in preserving audio content, robustness to out-of-domain data, and offers a training-free, natural, and streamable representation.
- Score: 16.679015298503593
- License: http://arxiv.org/licenses/nonexclusive-distrib/1.0/
- Abstract: Large language models have revolutionized natural language processing by leveraging self-supervised pretraining on vast textual data. Inspired by this success, researchers have investigated various compression-based speech tokenization methods to discretize continuous speech signals, enabling the application of language modeling techniques to discrete tokens. However, audio compressor introduces additional complexity and computational cost, and often fail on out-of-domain audio signals. In this work, we introduce a novel speech representation (dmel) that discretizes mel-filterbank channels into intensity bins, creating a simpler yet more effective representation compared to existing speech tokenization methods. Our approach demonstrates superior performance in preserving audio content, robustness to out-of-domain data, and offers a training-free, natural, and streamable representation. To address the high-dimensional nature of log-mel spectrograms, we propose an efficient parallel encoding and decoding method for high-dimensional tokens using an LM-style transformer architecture. This innovation enables us to develop RichTTS and RichASR, two models sharing the same architecture while achieving comparable or better results than specialized existing methods. Our results demonstrate the effectiveness of dmel in achieving high performance on both speech synthesis and recognition tasks within a unified framework, paving the way for efficient and effective joint modeling of speech and text.
Related papers
- LM-SPT: LM-Aligned Semantic Distillation for Speech Tokenization [8.365515332927444]
Recent speech tokenization approaches aim to isolate semantic information from low-level acoustics to better align with language models.<n>We propose LM-SPT, a speech tokenization method that introduces a novel semantic distillation.<n>We show that LM-SPT achieves superior reconstruction fidelity compared to baselines.
arXiv Detail & Related papers (2025-06-20T04:15:14Z) - Towards Efficient Speech-Text Jointly Decoding within One Speech Language Model [76.06585781346601]
Speech language models (Speech LMs) enable end-to-end speech-text modelling within a single model.<n>The choice of speech-text jointly decoding paradigm plays a critical role in performance, efficiency, and alignment quality.
arXiv Detail & Related papers (2025-06-04T23:53:49Z) - SEAL: Speech Embedding Alignment Learning for Speech Large Language Model with Retrieval-Augmented Generation [10.828717295018123]
We propose a unified embedding framework that eliminates the need for intermediate text representations.<n>Our model reduces pipeline latency by 50% while achieving higher retrieval accuracy compared to traditional two-stage methods.
arXiv Detail & Related papers (2025-01-26T15:04:02Z) - DM-Codec: Distilling Multimodal Representations for Speech Tokenization [11.433520275513803]
DM-Codec is a language model-guided distillation method that incorporates contextual information.
It significantly outperforms state-of-the-art speech tokenization models, reducing WER by up to 13.46%, WIL by 9.82%, and improving speech quality by 5.84% and intelligibility by 1.85% on the LibriSpeech benchmark dataset.
arXiv Detail & Related papers (2024-10-19T07:14:14Z) - Sylber: Syllabic Embedding Representation of Speech from Raw Audio [25.703703711031178]
We propose a new model, Sylber, that produces speech representations with clean and robust syllabic structure.
Specifically, we propose a self-supervised learning framework that bootstraps syllabic embeddings by distilling from its own initial unsupervised syllabic segmentation.
This results in a highly structured representation of speech features, offering three key benefits: 1) a fast, linear-time syllable segmentation algorithm, 2) efficient syllabic tokenization with an average of 4.27 tokens per second, and 3) novel phonological units suited for efficient spoken language modeling.
arXiv Detail & Related papers (2024-10-09T17:59:04Z) - SyllableLM: Learning Coarse Semantic Units for Speech Language Models [21.762112843104028]
We introduce a controllable self-supervised technique to merge speech representations into coarser syllable-like units.
Our method produces controllable-rate semantic units at as low as 5Hz and 60bps and SotA inc segmentation and clustering.
SyllableLM achieves significant improvements in efficiency with a 30x reduction in training compute and a 4x wall-clock inference speedup.
arXiv Detail & Related papers (2024-10-05T04:29:55Z) - Developing Instruction-Following Speech Language Model Without Speech Instruction-Tuning Data [84.01401439030265]
Recent end-to-end speech language models (SLMs) have expanded upon the capabilities of large language models (LLMs)
We present a simple yet effective automatic process for creating speech-text pair data.
Our model demonstrates general capabilities for speech-related tasks without the need for speech instruction-tuning data.
arXiv Detail & Related papers (2024-09-30T07:01:21Z) - LAST: Language Model Aware Speech Tokenization [24.185165710384997]
We propose a novel approach to training a speech tokenizer by leveraging objectives from pre-trained textual LMs.
Our aim is to transform features from a pre-trained speech model into a new feature space that enables better clustering for speech LMs.
arXiv Detail & Related papers (2024-09-05T16:57:39Z) - SpeechPrompt: Prompting Speech Language Models for Speech Processing Tasks [94.10497337235083]
We are first to explore the potential of prompting speech LMs in the domain of speech processing.
We reformulate speech processing tasks into speech-to-unit generation tasks.
We show that the prompting method can achieve competitive performance compared to the strong fine-tuning method.
arXiv Detail & Related papers (2024-08-23T13:00:10Z) - Codec-ASR: Training Performant Automatic Speech Recognition Systems with Discrete Speech Representations [16.577870835480585]
We present a comprehensive analysis on building ASR systems with discrete codes.
We investigate different methods for training such as quantization schemes and time-domain vs spectral feature encodings.
We introduce a pipeline that outperforms Encodec at similar bit-rate.
arXiv Detail & Related papers (2024-07-03T20:51:41Z) - Paralinguistics-Aware Speech-Empowered Large Language Models for Natural Conversation [46.93969003104427]
This paper introduces an extensive speech-text LLM framework, the Unified Spoken Dialog Model (USDM)
USDM is designed to generate coherent spoken responses with naturally occurring prosodic features relevant to the given input speech.
Our approach effectively generates natural-sounding spoken responses, surpassing previous and cascaded baselines.
arXiv Detail & Related papers (2024-02-08T14:35:09Z) - SpeechGPT-Gen: Scaling Chain-of-Information Speech Generation [56.913182262166316]
Chain-of-Information Generation (CoIG) is a method for decoupling semantic and perceptual information in large-scale speech generation.
SpeechGPT-Gen is efficient in semantic and perceptual information modeling.
It markedly excels in zero-shot text-to-speech, zero-shot voice conversion, and speech-to-speech dialogue.
arXiv Detail & Related papers (2024-01-24T15:25:01Z) - Learning Speech Representation From Contrastive Token-Acoustic
Pretraining [57.08426714676043]
We propose "Contrastive Token-Acoustic Pretraining (CTAP)", which uses two encoders to bring phoneme and speech into a joint multimodal space.
The proposed CTAP model is trained on 210k speech and phoneme pairs, achieving minimally-supervised TTS, VC, and ASR.
arXiv Detail & Related papers (2023-09-01T12:35:43Z) - On decoder-only architecture for speech-to-text and large language model
integration [59.49886892602309]
Speech-LLaMA is a novel approach that effectively incorporates acoustic information into text-based large language models.
We conduct experiments on multilingual speech-to-text translation tasks and demonstrate a significant improvement over strong baselines.
arXiv Detail & Related papers (2023-07-08T06:47:58Z) - Wav2Seq: Pre-training Speech-to-Text Encoder-Decoder Models Using Pseudo
Languages [58.43299730989809]
We introduce Wav2Seq, the first self-supervised approach to pre-train both parts of encoder-decoder models for speech data.
We induce a pseudo language as a compact discrete representation, and formulate a self-supervised pseudo speech recognition task.
This process stands on its own, or can be applied as low-cost second-stage pre-training.
arXiv Detail & Related papers (2022-05-02T17:59:02Z) - Towards Language Modelling in the Speech Domain Using Sub-word
Linguistic Units [56.52704348773307]
We propose a novel LSTM-based generative speech LM based on linguistic units including syllables and phonemes.
With a limited dataset, orders of magnitude smaller than that required by contemporary generative models, our model closely approximates babbling speech.
We show the effect of training with auxiliary text LMs, multitask learning objectives, and auxiliary articulatory features.
arXiv Detail & Related papers (2021-10-31T22:48:30Z) - An Effective Contextual Language Modeling Framework for Speech
Summarization with Augmented Features [13.97006782398121]
Bidirectional Representations from Transformers (BERT) model was proposed and has achieved record-breaking success on many natural language processing tasks.
We explore the incorporation of confidence scores into sentence representations to see if such an attempt could help alleviate the negative effects caused by imperfect automatic speech recognition.
We validate the effectiveness of our proposed method on a benchmark dataset.
arXiv Detail & Related papers (2020-06-01T18:27:48Z)
This list is automatically generated from the titles and abstracts of the papers in this site.
This site does not guarantee the quality of this site (including all information) and is not responsible for any consequences.