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Text-To-Speech Synthesis

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하위 태스크
2
평가표
6
보존된 평가 기록
21

Text-To-Speech Synthesis is a machine learning task that involves converting written text into spoken words. The goal is to generate synthetic speech that sounds natural and resembles human speech as closely as possible.

데이터셋

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LJSpeech

This is a public domain speech dataset consisting of 13,100 short audio clips of a single speaker reading passages from 7 non-fiction books. A transcription is provided for each clip. Clips vary in length from 1 to 10 seconds and have a total length of approximately 24 hours. The texts were published between 1884 and 1964, and are in the public domain. The audio was recorded in 2016-17 by the LibriVox project and is also in the public domain. Source: The LJ Speech Dataset Image Source: https://keithito.com/LJ-Speech-Dataset/ Audio Source: https://keithito.com/LJ-Speech-Dataset/

LibriTTS

LibriTTS is a multi-speaker English corpus of approximately 585 hours of read English speech at 24kHz sampling rate, prepared by Heiga Zen with the assistance of Google Speech and Google Brain team members. The LibriTTS corpus is designed for TTS research. It is derived from the original materials (mp3 audio files from LibriVox and text files from Project Gutenberg) of the LibriSpeech corpus. The main differences from the LibriSpeech corpus are listed below: - The audio files are at 24kHz sampling rate. - The speech is split at sentence breaks. - Both original and normalized texts are included. - Contextual information (e.g., neighbouring sentences) can be extracted. - Utterances with significant background noise are excluded.

AISHELL-3

AISHELL-3 is a large-scale and high-fidelity multi-speaker Mandarin speech corpus which could be used to train multi-speaker Text-to-Speech (TTS) systems. The corpus contains roughly 85 hours of emotion-neutral recordings spoken by 218 native Chinese mandarin speakers and total 88035 utterances. Their auxiliary attributes such as gender, age group and native accents are explicitly marked and provided in the corpus. Accordingly, transcripts in Chinese character-level and pinyin-level are provided along with the recordings. The word & tone transcription accuracy rate is above 98%, through professional speech annotation and strict quality inspection for tone and prosody.

CVSS

CVSS is a massively multilingual-to-English speech to speech translation (S2ST) corpus, covering sentence-level parallel S2ST pairs from 21 languages into English. CVSS is derived from the Common Voice speech corpus and the CoVoST 2 speech-to-text translation (ST) corpus, by synthesizing the translation text from CoVoST 2 into speech using state-of-the-art TTS systems

SOMOS

The SOMOS dataset is a large-scale mean opinion scores (MOS) dataset consisting of solely neural text-to-speech (TTS) samples. It can be employed to train automatic MOS prediction systems focused on the assessment of modern synthesizers, and can stimulate advancements in acoustic model evaluation. It consists of 20K synthetic utterances of the LJ Speech voice, a public domain speech dataset which is a common benchmark for building neural acoustic models and vocoders. Utterances are generated from 200 TTS systems including vanilla neural acoustic models as well as models which allow prosodic variations.

Emilia Dataset

Recent advancements in speech generation models have been significantly driven by the use of large-scale training data. However, producing highly spontaneous, human-like speech remains a challenge due to the scarcity of large, diverse, and spontaneous speech datasets. In response, we introduce Emilia, the first large-scale, multilingual, and diverse speech generation dataset. Emilia starts with over 101k hours of speech across six languages, covering a wide range of speaking styles to enable more natural and spontaneous speech generation. To facilitate the scale-up of Emilia, we also present Emilia-Pipe, the first open-source preprocessing pipeline designed to efficiently transform raw, in-the-wild speech data into high-quality training data with speech annotations. Experimental results demonstrate the effectiveness of both Emilia and Emilia-Pipe. Demos are available at: https://emilia-dataset.github.io/Emilia-Demo-Page/.

Gumar Corpus

A large-scale corpus of Gulf Arabic consisting of 110 million words from 1,200 forum novels. Source: A Large Scale Corpus of Gulf Arabic

KazakhTTS

KazakhTTS is an open-source speech synthesis dataset for Kazakh, a low-resource language spoken by over 13 million people worldwide. The dataset consists of about 91 hours of transcribed audio recordings spoken by two professional speakers (female and male). It is the first publicly available large-scale dataset developed to promote Kazakh text-to-speech (TTS) applications in both academia and industry.

EMOVIE

EMOVIE is a Mandarin emotion speech dataset including 9,724 samples with audio files and its emotion human-labeled annotation.

HUI speech corpus

The data set contains several speakers. The 5 largest are listed individually, the rest are summarized as other. All audio files have a sampling rate of 44.1kHz. For each speaker, there is a clean variant in addition to the full data set, where the quality is even higher. Furthermore, there are various statistics. The dataset can also be used for automatic speech recognition (ASR) if audio files are converted to 16 kHz.

OpenSLR

OpenSLR is a repository of open speech and language resources, including large-scale transcribed audio corpora and related software. It serves as a central platform for researchers and practitioners to access and share datasets used in speech recognition (ASR), text-to-speech (TTS), and linguistic research. The OpenSLR collection includes over 30 diverse datasets spanning more than 25 languages, such as Javanese, Nepali, Malayalam, Yoruba, and various English and Spanish dialects. These datasets are contributed by institutions including Google, North West University, and others. Most contain audio recordings along with transcriptions, covering both crowd-sourced and professionally recorded material. Many of the datasets are high-quality multi-speaker corpora intended for use in building ASR and TTS models, particularly for under-resourced languages. Use cases include multilingual speech recognition, dialect modeling, language technology research, and building open-source voice applications. The OpenSLR site also acts as a mirror for widely-used tools and models to ensure continued availability. Website: OpenSLR website Source: openslr/openslr

RyanSpeech

RyanSpeech is a speech corpus for research on automated text-to-speech (TTS) systems. This dataset contains textual materials from real-world conversational settings. These materials contain over 10 hours of a professional male voice actor's speech recorded at 44.1 kHz.

Trinity Speech-Gesture Dataset

Trinity Gesture Dataset includes 23 takes, totalling 244 minutes of motion capture and audio of a male native English speaker producing spontaneous speech on different topics. The actor’s motion was captured with 20 Viconcameras at 59.94 frames per second(fps), and the skeleton includes 69 joints. Source: https://trinityspeechgesture.scss.tcd.ie/data/

ArVoice

We introduce ArVoice, a multi-speaker Modern Standard Arabic (MSA) speech corpus with diacritized transcriptions, intended for multi-speaker speech synthesis, and can be useful for other tasks such as speech-based diacritic restoration, voice conversion, and deepfake detection. ArVoice comprises: (1) a new professionally recorded set from six voice talents with diverse demographics, (2) a modified subset of the Arabic Speech Corpus; and (3) high-quality synthetic speech from two commercial systems. The complete corpus consists of a total of 83.52 hours of speech across 11 voices; around 10 hours consist of human voices from 7 speakers. We train three open-source TTS and two voice conversion systems to illustrate the use cases of the dataset. The corpus is available for research use.

CANDOR Corpus

The CANDOR corpus is a large, novel, multimodal corpus of 1,656 recorded conversations in spoken English. This 7+ million word, 850 hour corpus totals over 1TB of audio, video, and transcripts, with moment-to-moment measures of vocal, facial, and semantic expression, along with an extensive survey of speaker post conversation reflections.

GneutralSpeech Female

A Brazilian Portuguese TTS dataset featuring a female voice recorded with high quality in a controlled environment, with neutral emotion and more than 20 hours of recordings. with neutral emotion and more than 20 hours of recordings. Our dataset aims to facilitate transfer learning for researchers and developers working on TTS applications: a highly professional neutral female voice can serve as a good warm-up stage for learning language-specific structures, pronunciation and other non-individual characteristics of speech, leaving to further training procedures only to learn the specific adaptations needed (e.g. timbre, emotion and prosody). This can surely help enabling the accommodation of a more diverse range of female voices in Brazilian Portuguese. By doing so, we also hope to contribute to the development of accessible and high-quality TTS systems for several use cases such as virtual assistants, audiobooks, language learning tools and accessibility solutions. Possible use cases: TTS; Voice Conversion; ASR; Speech Enhancement

GneutralSpeech Male

A database containing high sampling rate recordings of a single speaker reading sentences in Brazilian Portuguese with neutral voice, along with the corresponding text corpus. Intended for speech synthesis and automatic speech recognition applications, the dataset contains text extracted from a popular Brazilian news TV program, totalling roughly 20 h of audio spoken by a trained individual in a controlled environment. The text was normalized in the recording process and special textual occurrences (e.g. acronyms, numbers, foreign names etc.) were replaced by their phonetic translation to a readable text in Portuguese. There are no noticeable accidental sounds and background noise has been kept to a minimum in all audio samples. Intended for: TTS ASR Speech Enhancement Voice Conversion

Helsinki Prosody Corpus

The Helsinki Prosody Corpus is a dataset for predicting prosodic prominence from written text. The prosodic annotations are automatically generated, high quality prosodic for the 'clean' subsets of LibriTTS corpus (Zen et al., 2019), comprising of 262.5 hours of read speech from 1230 speakers. The transcribed sentences were aligned and then prosodically annotated with word-level acoustic prominence labels. Source: Predicting Prosodic Prominence from Text with Pre-trained Contextualized Word Representations

IMaSC

IMaSC is a Malayalam text and speech corpus made available by ICFOSS for the purpose of developing speech technology for Malayalam, particularly text-to-speech. The corpus contains 34,473 text-audio pairs of Malayalam sentences spoken by 8 speakers, totalling in approximately 50 hours of audio.

Thorsten voice 21.02 neutral

Thorsten-Voice (Thorsten-21.02-neutral) is a neutrally spoken voice dataset recorded by Thorsten Müller, audio optimized by Dominik Kreutz and licenced under CC0 to provide it for anybody without any financial or licence struggle. It is intended to be used for speech synthesis in German as a single speaker dataset. It contains about 23 hours of high quality audio

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