LibriConvo: Simulating Conversations from Read Literature for ASR and Diarization
- URL: http://arxiv.org/abs/2510.23320v1
- Date: Mon, 27 Oct 2025 13:35:22 GMT
- Title: LibriConvo: Simulating Conversations from Read Literature for ASR and Diarization
- Authors: Máté Gedeon, Péter Mihajlik,
- Abstract summary: We introduce LibriConvo, a simulated multi-speaker conversational dataset based on speaker-aware conversation simulation (SASC)<n>Unlike prior resources that mostly rely on semantically disconnected utterances, LibriConvo ensures semantic coherence and realistic conversational timing.<n>The dataset comprises 240.1 hours across 1,496 dialogues with 830 unique speakers, split in a speaker-disjoint manner for robust evaluation.
- Score: 1.0251581485267474
- License: http://creativecommons.org/licenses/by/4.0/
- Abstract: We introduce LibriConvo, a simulated multi-speaker conversational dataset based on speaker-aware conversation simulation (SASC), designed to support training and evaluation of speaker diarization and automatic speech recognition (ASR) systems. Unlike prior resources that mostly rely on semantically disconnected utterances and implausible temporal gaps, LibriConvo ensures semantic coherence and realistic conversational timing. Our pipeline leverages CallHome with external VAD for reliable boundaries, applies compression to reduce unnaturally long silences, and organizes LibriTTS utterances by book to maintain contextual consistency. Acoustic realism is enhanced via a novel room impulse response selection procedure that ranks speaker-microphone configurations by spatial plausibility, balancing realism and diversity. The dataset comprises 240.1 hours across 1,496 dialogues with 830 unique speakers, split in a speaker-disjoint manner for robust evaluation. Baselines show that the sortformer model outperforms the pyannote pipeline in diarization, while a fine-tuned Fast Conformer-CTC XLarge with Serialized Output Training achieves 7.29\% WER for ASR, surpassing zero-shot Whisper-large-v3. LibriConvo provides a valuable resource for advancing multi-speaker speech processing research with realistic conversational dynamics and controlled experimental conditions.
Related papers
- Covo-Audio Technical Report [61.09708870154148]
Covo-Audio, a 7B-end LALM, directly processes continuous audio inputs and generates audio outputs within a single unified architecture.<n>Covo-Audio-Chat, a dialogue-oriented variant, demonstrates semantic strong spoken conversational abilities.
arXiv Detail & Related papers (2026-02-10T14:31:11Z) - Speaker-Aware Simulation Improves Conversational Speech Recognition [1.0251581485267474]
We adapt and implement the SASC framework for Hungarian conversational ASR.<n>We propose C-SASC, an extended variant that incorporates pause modeling conditioned on utterance duration.<n>We generate synthetic Hungarian dialogues from the BEA-Large corpus and combine them with real conversational data for ASR training.
arXiv Detail & Related papers (2026-02-04T17:12:09Z) - MSA-ASR: Efficient Multilingual Speaker Attribution with frozen ASR Models [59.80042864360884]
Speaker-attributed automatic speech recognition (SA-ASR) aims to transcribe speech while assigning transcripts to the corresponding speakers accurately.<n>This paper introduces a novel approach, leveraging a frozen multilingual ASR model to incorporate speaker attribution into the transcriptions.
arXiv Detail & Related papers (2024-11-27T09:01:08Z) - Speech Rhythm-Based Speaker Embeddings Extraction from Phonemes and
Phoneme Duration for Multi-Speaker Speech Synthesis [16.497022070614236]
This paper proposes a speech rhythm-based method for speaker embeddings to model phoneme duration using a few utterances by the target speaker.
A novel feature of the proposed method is the rhythm-based embeddings extracted from phonemes and their durations, which are known to be related to speaking rhythm.
arXiv Detail & Related papers (2024-02-11T02:26:43Z) - Extending Whisper with prompt tuning to target-speaker ASR [18.31992429200396]
Target-speaker automatic speech recognition (ASR) aims to transcribe the desired speech of a target speaker from overlapped utterances.
Most of the existing target-speaker ASR (TS-ASR) methods involve either training from scratch or fully fine-tuning a pre-trained model.
This work leverages prompt tuning, a parameter-efficient fine-tuning approach, to extend Whisper, a large-scale single-talker ASR model, to TS-ASR.
arXiv Detail & Related papers (2023-12-13T11:49:16Z) - Disentangling Voice and Content with Self-Supervision for Speaker
Recognition [57.446013973449645]
This paper proposes a disentanglement framework that simultaneously models speaker traits and content variability in speech.
It is validated with experiments conducted on the VoxCeleb and SITW datasets with 9.56% and 8.24% average reductions in EER and minDCF.
arXiv Detail & Related papers (2023-10-02T12:02:07Z) - Zero-shot text-to-speech synthesis conditioned using self-supervised
speech representation model [13.572330725278066]
A novel point of the proposed method is the direct use of the SSL model to obtain embedding vectors from speech representations trained with a large amount of data.
The disentangled embeddings will enable us to achieve better reproduction performance for unseen speakers and rhythm transfer conditioned by different speeches.
arXiv Detail & Related papers (2023-04-24T10:15:58Z) - Continual Learning for On-Device Speech Recognition using Disentangled
Conformers [54.32320258055716]
We introduce a continual learning benchmark for speaker-specific domain adaptation derived from LibriVox audiobooks.
We propose a novel compute-efficient continual learning algorithm called DisentangledCL.
Our experiments show that the DisConformer models significantly outperform baselines on general ASR.
arXiv Detail & Related papers (2022-12-02T18:58:51Z) - Streaming Speaker-Attributed ASR with Token-Level Speaker Embeddings [53.11450530896623]
This paper presents a streaming speaker-attributed automatic speech recognition (SA-ASR) model that can recognize "who spoke what"
Our model is based on token-level serialized output training (t-SOT) which was recently proposed to transcribe multi-talker speech in a streaming fashion.
The proposed model achieves substantially better accuracy than a prior streaming model and shows comparable or sometimes even superior results to the state-of-the-art offline SA-ASR model.
arXiv Detail & Related papers (2022-03-30T21:42:00Z) - Continuous speech separation: dataset and analysis [52.10378896407332]
In natural conversations, a speech signal is continuous, containing both overlapped and overlap-free components.
This paper describes a dataset and protocols for evaluating continuous speech separation algorithms.
arXiv Detail & Related papers (2020-01-30T18:01:31Z)
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.