Papers for

game audio designers

Papers whose findings have a practical use for this group, as judged from the abstract. Open a paper to read what it means in practice.

Face images control emotion intensity in expressive text to speech

EmoFI-TTS: Face-Conditioned Expressive Voice Synthesis with Intensity Modulation

Abstract: Controlling emotional intensity remains challenging in expressive text-to-speech synthesis, particularly when emotion must be inferred without explicit labels. We propose EmoFI-TTS, a face-conditioned framework that derives emotion category and intensity from a single facial image while extracting speaker characteristics from reference audio. Modality-specific representations are fused to condition a pretrained speech synthesizer, with cross-speaker training encouraging separation of affect and vocal identity. An intensity-conditioned refinement module enhances the generated mel-spectrogram, while interpolation of learned intensity embeddings enables continuous control without additional training. On speaker-disjoint MEAD evaluation, EmoFI-TTS achieves on average 64.4% and 75.2% objective and subjective emotion intensity accuracies, while retaining low word error rate and high naturalness.

Wed 21 OctSound
The gist
Expressing emotions at different intensities is hard for text-to-speech systems, especially when the system doesn't have explicit emotion labels to guide it. The authors propose a model called EmoFI-TTS that uses a single face image to figure out both the type and strength of an emotion, while also extracting who is speaking from an audio sample. They combine these clues to generate speech that sounds both emotional and natural, and let users control how strong the emotion is without retraining the model. Their tests showed that EmoFI-TTS accurately matches the emotion intensity and keeps the speech understandable and natural sounding.
Open → 2610.30477v1

Text to music models improve by aligning with symbolic MIDI data

MIDI-REPA: Improving Text-to-Music Generation with Symbolic Representation Alignment

Abstract: State-of-the-art text-to-music models increasingly rely on scaling model capacity, data, and compute, while still learning musical structure directly from dense audio. Yet melody, rhythm, harmony, and structure can be represented compactly in symbolic form. We introduce MIDI-REPA, a representation-alignment framework that derives MIDI supervision from automatic music transcription (AMT) to align intermediate representations of a flow-matching DiT. Building on the 122M-parameter FluxAudio-S baseline from the 2026 ICME Academic Text-To-Music Challenge, we investigate what, where, and when to align. Aligning a MuScriptor-derived piano roll with an early DiT layer throughout the full flow trajectory performs the best. Without additional training data or techniques, MIDI-REPA remains competitive on the ATTM objective metrics while remaining trainable on a single RTX 3090 in about 31 hours. Our results highlight symbolic supervision as a simple and inexpensive learning signal for improving text-to-music generation. We release our transcription pipeline, JamendoMIDI, training code, and audio samples at https://midi-repa.vaclis.net.

Wed 14 OctSoundMachine Learning
The gist
Generating music from text is tricky because machines usually learn directly from raw audio, which is complex. The authors show that teaching the model using simpler symbolic music data, like MIDI, can improve its understanding of melody, rhythm, and harmony. They created a method called MIDI-REPA that combines this symbolic information during training without needing extra data or expensive hardware. Their results suggest that using symbolic music as guidance is an easy and effective way to make text-to-music generation better.
Open → 2610.20967v1

Prime-EVC converts emotional voices in one fast and natural step

Prime-EVC: One-Step Expressive Emotional Voice Conversion via Emotion-Primed Flow Matching

Abstract: Recent diffusion- and conditional flow matching (CFM)-based emotional voice conversion (EVC) systems achieve strong conversion quality. However, they typically rely on multiple sampling steps, because the conditioning specifies which emotion to impose while its acoustic realization is left to the sampler. In this paper, we propose Prime-EVC, a one-step expressive EVC architecture based on emotion-primed flow matching. Our key idea is to reallocate the emotional conversion between the conditioning and the sampler: a FiLM-fused speaker-emotion style, an emotional variance adaptor, and direct emotion and spectro-temporal supervision assemble an emotion-rich conditioning upstream, rendering the residual conversion left to the CFM decoder attainable within one Euler step. Experiments on the ESD corpus demonstrate that single-step Prime-EVC surpasses five representative EVC systems on every quality metric and exceeds 4- and 100-step DurFlex-EVC in subjective naturalness and emotion while running 3.4--50 times faster than both.

Mon 12 OctSound
The gist
Emotional voice conversion means changing how a voice sounds to express different feelings. Previous systems needed many steps to get good results, which made them slow. The authors propose Prime-EVC, a new method that combines emotional cues in a way that lets it convert voices in a single quick step. Tests show it makes more natural and expressive voices than other popular methods, while being much faster.
Open → 2610.16036v1

Framework improves piano music creation from detailed text descriptions

TASTE: Taxonomy-Aware Symbolic Piano Generation from Textual Description

Abstract: Text-conditioned symbolic music generation has attracted growing interest with the development of generative models. Yet two questions remain underexplored: what aspects of music the conditioning text should describe, and how the corresponding data should be collected and trained to improve controllability and text-music alignment. We address these challenges with TASTE, a taxonomy-aware framework for symbolic music generation with a main focus on piano pieces. First, we introduce a taxonomy of musical elements that specifies how a piece can be captured through textual description in a musically grounded way. The taxonomy comprises a set of metadata tags (genre, emotion, section plan, etc.) together with a free-form description of music imagery. This structure decomposes textual input to match the generation goal more coherently and precisely. Second, we build a data pipeline for collecting, annotating, and retrieving public symbolic piano resources, yielding a corpus of 432K MIDI files labeled at multiple levels of granularity. Finally, we design a staged training procedure that leverages this multi-granularity data across two model families: an autoregressive model and a non-autoregressive diffusion model. Both objective and subjective evaluations show that our models achieve superior musical-element control, text-description alignment, and overall music quality. We will release the full pipeline, all annotated datasets, and the model checkpoints.

Sun 11 OctSound
The gist
Creating music from text is hard because the text can describe many parts of music, and it's unclear how to train models well. The authors designed a way to break down music descriptions into clear categories like genre and emotion, making it easier to turn text into piano music. They built a large dataset of piano music files tagged in detail, and trained models that generate music matching the text more closely and with better quality. Their method shows improvements in controlling music elements and aligning music with text.
Open → 2610.15960v1

Sound morphing preserves timing and changes instruments smoothly

Smorph: Playable Sound Morphing with Diffusion Models

Abstract: Sound morphing, generating intermediate sounds that transition from one sonic identity to another, can be a powerful tool for musical sound design. Existing diffusion-based morphing approaches entangle temporal structure and timbral identity, offering no mechanism to hold one fixed while transforming the other. We present smorph, a training-free guidance framework that preserves how a sound behaves over time while transforming what the sound is, allowing users to morph, for instance from brass to strings at a fixed pitch. We demonstrate across three morphing modes: prompt-to-prompt, audio-to-prompt, and audio-to-audio. Evaluations across diverse datasets show that smorph effectively produces smooth morph trajectories while substantially improving temporal-structure and source preservation over baselines, albeit with more conservative target-ward transformation in some settings. In an exploratory case study, musicians found smorph trajectories to be expressive and playable, suggesting structural anchoring can serve as a productive constraint for instrumental interaction.

Mon 5 OctSound
The gist
Sound morphing is the process of creating sounds that gradually change from one instrument to another. The authors found that existing methods mix up the timing and the unique qualities of the sound, making it hard to control each separately. Their new method, smorph, keeps the timing fixed and changes only the sound’s character, like morphing a trumpet into a violin note without changing the rhythm. Musicians using smorph found it expressive and felt it could help make musical instruments easier to control.
Open → 2610.06478v1

Game sound effects completed with event-level redesign guidance

Game Sound-Effect Completion with Event-Level Transformation Hints

Abstract: Creating sound effects for a new game-character skin requires a distinct acoustic identity while preserving gameplay-event roles. The challenge is to complete a coherent set of related sounds whose required degrees of redesign differ. We formulate this task as completion conditioned on base-skin audio, completed target assets, and a textual design description. We develop a pipeline to collect, process, and align corresponding events across League of Legends skins. Building on Stable Audio 3's pretrained audio prior, we fine-tune a latent inpainting model to jointly complete missing events. A signed soft retention mask encodes available audio and an adjustable transformation hint for each missing event, specifying the requested balance between retention and redesign. Experiments on held-out skins show improved reconstruction over the evaluated general-purpose audio editors. Target-derived hints further improve paired similarity, with three-level hints retaining most of the benefit of continuous guidance.

Wed 30 SeptSound
The gist
When making new sound effects for game characters, it’s important that the sounds match the game’s actions but still sound unique. The authors created a method that fills in missing sounds for a character’s skin by using existing sounds, completed sounds, and design instructions. Their model can choose how much to keep from the original sound versus redesign it, making the new sounds fit well with gameplay. Tests showed their method works better than general sound editing tools.
Open → 2609.39044v1

Large language model enables step-by-step editing of music scores

From One-Shot Generation to Incremental Music Composition: Adapting a General-Purpose Instruction LLM for Persistent Symbolic Editing

Abstract: Most music-generation systems are still framed and evaluated primarily as producers of complete outputs, whereas composition often proceeds through successive revisions to a shared musical artifact. This paper studies a different use of a general-purpose instruction-following large language model: not as a one-shot music generator, but as a reusable operator over an evolving symbolic score. We formulate incremental composition as a sequence of operation-aware state transitions over persistent ABC notation, with explicit requirements on what each operation may change and what it must preserve. The interaction includes two artifact-initialization variants and three editing operations -- chord addition, inpainting, and transposition. We instantiate the formulation by adapting Llama 3.1 8B Instruct with Low-Rank Adaptation (LoRA) on 496,038 operation-aware dialogue records derived from Irish traditional music. The comparison with the unadapted model is used to test the feasibility of learning this interaction contract, not to claim novelty for fine-tuning itself. Across 500 dialogues per model (1,750 attempted output states), checker admission rises from 29.37% to 99.37%, while compliance conditional on admission rises from 0.7205 to 0.9798. Strict eligibility for reference-relative musical-feature analysis increases from 14 to 1,548 outputs, and Longest Common Subsequence analysis does not show a systematic increase in high-overlap sequences relative to held-out baselines under the specified protocol. The results support the technical feasibility of persistent, operation-aware symbolic editing with a general-purpose instruction LLM. They do not establish superior musical quality or human-AI co-creativity, which remain questions for musician-centered evaluation.

Mon 28 SeptArtificial IntelligenceMachine Learning
The gist
Most music generation tools create a whole piece at once, but real music composition often happens through many small changes over time. The authors show that a type of large language model, originally designed to follow instructions in general, can be adapted to edit and improve music scores incrementally. They trained the model to understand specific editing actions like adding chords or changing keys, which lets it work on music step-by-step while keeping track of previous edits. Their results show this approach is technically possible, though it’s not yet clear if it makes better music or helps musicians work with AI.
Open → 2609.34994v1

Controllable multi-speaker dialogue speech improves with iterative refinement

From Script to Drama: An Agentic Framework for Controllable Multi-Speaker Dialogue TTS

Abstract: Multi-speaker dialogue TTS requires natural speech generation, consistent speaker identity, coherent cross-turn transitions, and fine-grained control of expressive attributes such as emotion, speaking rate, and loudness. These requirements are difficult to satisfy reliably with one-shot generation, especially in long-form dialogue. We propose a controllable multi-speaker dialogue TTS framework that formulates synthesis as critique-driven iterative refinement. Its speech backbone, ControlEdit-TTS, unifies instruction-following synthesis and natural-language-guided attribute editing, enabling correction of expressive errors without full regeneration. The framework further performs hierarchical utterance-level and scene-level critique, routing detected issues to editing, resynthesis, or timing adjustment. Experiments on a bilingual Chinese--English dialogue benchmark show improved utterance-level instruction following, better dialogue-level preference than direct dialogue models and agentic baselines, and more effective refinement than regeneration-only alternatives while preserving speaker identity. Ablations further confirm the benefits of scene-level critique and edit-based correction.

Sun 27 SeptComputation and LanguageSound
The gist
Producing realistic conversations with different speakers in text-to-speech (TTS) systems is challenging because it needs natural voices, consistent speaker sound, smooth turn-taking, and control over emotions and speech styles. The authors developed a method where the system listens to its own output and makes improvements step-by-step instead of generating everything once. Their approach can fix mistakes in tone, speed, or loudness without redoing the whole conversation, leading to more natural and coherent multi-speaker dialogues. Tests on Chinese-English dialogues showed better quality and control compared to other methods.
Open → 2609.33362v1

SSE model enhances and remixes audio using video and text guidance

Spot, Separate, and Enhance: Fully Generative Approach for Audio Mixing

Abstract: We introduce Spot, Separate, and Enhance (SSE), the first multimodal, user-guided generative model for audio remixing and enhancement. SSE enhances video content by rebalancing the audio, removing unwanted audio sources, and reducing reverberation, guided by both video and textual descriptions. To support its training and evaluation, we propose DegradedMix, a new dataset built on the audio remixing benchmark MuddyMix. We also adopt evaluation metrics from generative modeling, which better capture the creative nature of remixing than standard reconstruction-based metrics. SSE outperforms existing baselines in both controllability and remixing quality, as shown by extensive experiments. Project page: https://sse-ai.notion.site

Thu 24 SeptSoundArtificial Intelligence
The gist
Mixing audio in videos is hard when unwanted sounds or echo get in the way. The authors created a new system called SSE that uses the video and text descriptions to find, separate, and improve sounds in a video’s audio track. This lets people remove unwanted noise, balance sound levels, and reduce echoes with simple instructions. They also made a new dataset to help train and test their system and showed it works better than existing methods.
Open → 2609.29169v1

ReaFlow TTS improves speech synthesis with mood based control

ReaFlow-TTS: Realization-Conditioned Flow Matching for High-Quality and Controllable Speech Synthesis

Abstract: In flow-matching text-to-speech (TTS), different speech realizations can induce different target velocities under the same generation conditions. A deterministic velocity field trained with squared error predicts their conditional mean, thereby marginalizing realization-dependent variation. Meanwhile, modeling such variation does not inherently provide a semantically interpretable interface for attribute manipulation. We propose ReaFlow-TTS, a realization-conditioned flow-matching framework that introduces an utterance-level stochastic realization latent and uses it to condition velocity prediction throughout the generation trajectory. We further impose valence-arousal-dominance (VAD) semantics on the realization space, enabling direct and graded attribute manipulation without target speech at inference. Experiments demonstrate improved synthesis quality over a matched full-mask baseline and reproducible latent-induced pitch, energy, and timing tendencies across initial-noise samples, providing behavioral evidence that the latent is used as a reusable realization condition. Subjective evaluation further demonstrates graded VAD manipulation across generation contexts with only modest changes in naturalness.

Thu 24 SeptSound
The gist
Making computer-generated speech sound natural is tricky because the same words can be spoken in many different ways. The authors created a system called ReaFlow-TTS that lets the computer learn different ways to say the same phrase and control the mood of the speech by adjusting feelings like excitement or calmness. This system can change pitch, energy, and timing in speech according to these moods, making it more expressive without needing real examples at the time of use. Listening tests showed people could tell the mood changes without losing much naturalness in the voice.
Open → 2609.28906v1

Dance to music generation improves with hierarchical expert supervision

CMA-OT: Hierarchical Expert Supervision for Dance-to-Music Generation

Abstract: Dance-to-music (D2M) generation aims to synthesize music that is rhythmically and stylistically aligned with dance videos. A key challenge arises from the semantic mismatch between sparse dance cues, such as rhythm and style, and the dense information required for music composition, including structure, instrumentation, and expressive dynamics. Existing methods typically rely on these sparse cues and supervise only the final audio output, resulting in poorly learned music representations and generated music with limited musicality and structural coherence. To address these issues, we propose Curriculum-guided Multi-scale representation Alignment with scale-aware Optimal Transport (CMA-OT), a novel paradigm that leverages an external music expert to provide hierarchical supervision for the generator's latent features, bridging the semantic gap and enhancing representation learning. To effectively incorporate hierarchical supervision, we introduce a curriculum-guided multi-scale learning strategy that progressively transfers musical knowledge from the expert to the music generator, enabling stable and effective representation learning. Moreover, to accommodate the semantic and structural variations across different expert scales and achieve fine-grained alignment under temporal mismatch, we propose a scale-aware optimal transport alignment mechanism, which models soft correspondences between hierarchical expert representations and the generator's latent features. Extensive experiments on two datasets demonstrate that CMA-OT achieves state-of-the-art performance in rhythmic synchronization, perceptual quality, and overall music generation.

Fri 11 SeptArtificial IntelligenceSound
The gist
Generating music that matches a dance video is tricky because dance gives limited clues, while music includes many complex features. The authors developed a new method that uses a music expert model to guide the music generation process at multiple levels, helping bridge the gap between dance and music. They also teach the system step-by-step and align its understanding of music features carefully to improve quality and rhythm matching. Their experiments show better music that fits the dance style and timing well.
Open → 2609.13118v1

PhaseGAN improves vocoders by separating amplitude and phase reconstruction

PhaseGAN: High-Fidelity Vocoder via Decoupled Amplitude and GAN-Driven Phase Reconstruction

Abstract: A vocoder is a pivotal component of modern text-to-speech (TTS) systems. Despite the significant progress of neural network-based vocoders, accurate phase reconstruction remains the main challenge limiting both audio quality and modeling efficiency. We introduce PhaseGAN, a lightweight vocoder that addresses this limitation through a "mel $\rightarrow$ Amplitude $\rightarrow$ Phase" reconstruction pipeline. By reconstructing amplitude and phase spectra via distinct methodologies, the proposed PhaseGAN outperforms state-of-the-art baselines while utilizing fewer model parameters and reduced computational requirements. The compact version generates high-fidelity audio with approximately 500K parameters and 1 GMAC computational load, making it highly suitable for real-time applications on edge devices. In addition, our approach exhibits exceptional musical audio synthesis capabilities despite no training on musical data, illustrating unprecedented cross-domain generalization. See https://github.com/phasegan/phasegan-audio-demo for demos of our work.

Fri 11 SeptSound
The gist
Creating realistic computer-generated speech involves accurately rebuilding sounds from data, which requires recovering both loudness and timing details. The authors identified that reconstructing the timing part, called phase, is a major challenge that affects sound quality and speed. They designed PhaseGAN, a small and fast system that builds sounds first by focusing on volume patterns, then separately reconstructs timing patterns using a special method. This approach makes the system produce better-sounding audio with fewer resources and even works for musical sounds it wasn’t trained on.
Open → 2609.12918v1

Lempel Ziv method adjusts music note copying for better synthesis

Copying Versus Randomization in Lempel-Ziv Music Synthesis

Abstract: We utilize Lempel-Ziv universal compression for music note generation. We control the algorithm's tendency to over-copy or under-copy training data by manipulating the average sequence length saved in the dictionary.

Thu 10 SeptSoundInformation Theory
The gist
Creating new music notes from existing songs can be tricky because sometimes the computer copies too much or too little from the original. This paper uses a known method called Lempel-Ziv compression to generate music notes. The authors found ways to control how much it copies by changing how long sequences it saves during this process. This helps balance between repeating exact parts and making new random notes.
Open → 2609.11353v1