Search arXivSearch

arXiv · 2407.09346

A Preliminary Investigation on Flexible Singing Voice Synthesis Through Decomposed Framework with Inferrable Features

Abstract

We investigate the feasibility of a singing voice synthesis (SVS) system by using a decomposed framework to improve flexibility in generating singing voices. Due to data-driven approaches, SVS performs a music score-to-waveform mapping; however, the direct mapping limits control, such as being able to only synthesize in the language or the singers present in the labeled singing datasets. As collecting large singing datasets labeled with music scores is an expensive task, we investigate an alternative approach by decomposing the SVS system and inferring different singing voice features. We decompose the SVS system into three-stage modules of linguistic, pitch contour, and synthesis, in which singing voice features such as linguistic content, F0, voiced/unvoiced, singer embeddings, and loudness are directly inferred from audio. Through this decomposed framework, we show that we can alleviate the labeled dataset requirements, adapt to different languages or singers, and inpaint the lyrical content of singing voices. Our investigations show that the framework has the potential to reach state-of-the-art in SVS, even though the model has additional functionality and improved flexibility. The comprehensive analysis of our investigated framework's current capabilities sheds light on the ways the research community can achieve a flexible and multifunctional SVS system.

Explore related subjects

Keep this discovery

Explore connections, maps & timelines

BibTeXRIS

Lester Phillip Violeta, Taketo Akama. 2024-07-12. A Preliminary Investigation on Flexible Singing Voice Synthesis Through Decomposed Framework with Inferrable Features. https://arxiv.org/abs/2407.09346

Cite the original work for its findings. Save a collection to share your selection of sources.

KEEP EXPLORING

Related papers

Text-only adaptation in LLM-based ASR through text denoising

Adapting large language model (LLM)-based automatic speech recognition (ASR) systems to new domains using text-only data is a significant yet underexplored challenge. Standard fine-tuning of the LLM on the target domain text often disrupts the critical alignment between the speech and text modality learned by the projector, degrading performance. We introduce a novel text-only adaptation method that frames this process as a text denoising task. Our approach trains the LLM to recover clean transcripts from noisy inputs. This process effectively adapts the model to a target domain while preserving cross-modal alignment. Our solution is lightweight, requiring no architectural changes or additional parameters. Extensive evaluation on two datasets demonstrates up to 22.1% relative improvement, outperforming recent state-of-the-art text-only adaptation methods.

cs.SD

Discrete vs. Continuous: A Comprehensive Study of Unified Audio Understanding in LALMs

Large Audio Language Models (LALMs) utilize either continuous features or discrete tokens, yet the optimal representation paradigm for general audio understanding remains debated. Existing benchmarks often focus on narrow domains or evaluate encoders outside LALM contexts. To address these gaps, we systematically evaluate continuous and discrete representations across speech, sound and music. Utilizing our UniARC framework with dual evaluation strategies across model scales from SmolLM2-135M to Llama-3-8B, we analyze the dynamic relationships of data volume, model capacity, and computational efficiency. Our results reveal the pivotal role of semantic constraints in tokenization for audio understanding and demonstrate that scaling backbones fail to compensate for information loss in audio representation, especially in data-limited tasks. These findings offer practical guidance for balancing semantic density, fidelity, and efficiency in future LALMs.

cs.SD

Synthesis and editing of multi-instrument audio mixtures using scalar-quantised latents with MIDI Span conditioning

Music creation often involves iterative refinement, changing selected musical details while retaining the rest. To support such refinement, we introduce SpanSynth-Edit, a flow-matching model for MIDI-guided synthesis and editing of multi-instrument audio mixtures using low-frame-rate scalar-quantised latents. MIDI Span encodes instrument-labelled note lifecycles as unordered event sets with continuous-valued attributes and pools each set into one conditioning vector per audio-latent frame. The model uses contextual audio for instrument-specific timbre guidance and supports editing by resynthesising the target region from revised MIDI. Experiments on single- and multi-instrument benchmarks show competitive performance and demonstrate within-frame onset control. We also discuss limitations of transcription-based note-adherence evaluation.

cs.SD