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In today's technologically integrated world, speech synthesis has become a pivotal tool in human-computer interaction. This technology, also known as text-to-speech (TTS), involves the artificial production of human speech. It enables the conversion of textual data into audible output, facilitating communication across various platforms and devices.
A significant application of speech synthesis is within speech-to-speech translation systems. In these systems, spoken language is first converted into text, translated into the target language, and then synthesized back into speech. This process allows for seamless communication between individuals speaking different languages, breaking down linguistic barriers and fostering global interaction.
Speech synthesis technology is now integral to various applications, including:
The quality of the generated speech depends on factors including the sophistication of the TTS engine, the accuracy of linguistic analysis, and the naturalness of the voice model used.
Modern speech synthesis systems aim to produce output that closely resembles natural human speech, with appropriate intonation, rhythm, and emotional nuances, enhancing human-computer interaction across various applications.
Speech synthesis complements speech recognition by forming a bidirectional communication bridge. This enhances applications like voice assistants, real-time translation, and educational tools.
Concatenative TTS relies on high-quality audio clips recordings, which are combined together to form the speech. At the first step voice actors are recorded saying a range of speech units, from whole sentences to syllables that are further labeled and segmented by linguistic units from phones to phrases and sentences forming a huge database. During speech synthesis, a Text-to-Speech engine searches such database for speech units that match the input text, concatenates them together and produces an audio file.
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Formant synthesis technique is a rule-based TTS technique. It produces speech segments by generating artificial signals based on a set of specified rules mimicking the formant structure and other spectral properties of natural speech. The synthesized speech is produced using an additive synthesis and an acoustic model. The acoustic model uses parameters like, voicing, fundamental frequency, noise levels, etc that varied over time. Formant-based systems can control all aspects of the output speech, producing a wide variety of emotions and different tone voices with the help of some prosodic and intonation modeling techniques.
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HMM based Speech Synthesis
To overcome the limitations inherent in concatenative text-to-speech (TTS) systems, a statistical parametric approach was developed. This method operates on the premise that by approximating the parameters constituting speech, a model can be trained to generate diverse speech outputs. It integrates parameters such as fundamental frequency and magnitude spectrum, processing them to synthesize speech.
The process begins with text analysis to extract linguistic features, including phonemes and duration. Subsequently, vocoder features—such as cepstra, spectrogram, and fundamental frequency—that encapsulate intrinsic characteristics of human speech are extracted for audio processing. These hand-engineered features, along with the linguistic attributes, are input into a mathematical model known as a vocoder. During waveform generation, the vocoder transforms these features and estimates speech parameters like phase, speech rate, and intonation. This technique employs Hidden Semi-Markov Models (HSMMs), which, while maintaining state transitions characteristic of Markov models, incorporate explicit duration modeling within each state.
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Deep Neural Network (DNN)-based speech synthesis represents a significant advancement in statistical synthesis methods, addressing the limitations of Hidden Markov Models (HMMs) that utilize decision trees for modeling complex contextual dependencies. A notable innovation in this approach is the automation of feature design, allowing machines to learn feature representations without human intervention. Unlike manually crafted features based on human understanding of speech—which may not always be accurate—DNN techniques establish relationships between input text and their acoustic realizations through deep learning models. These models generate acoustic features using maximum likelihood parameter generation with trajectory smoothing.
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WaveNet, the foundation of Google Cloud Text-to-Speech, is a fully convolutional neural network that accepts raw audio waveforms as input. These waveforms traverse multiple convolutional layers, producing output waveforms. Despite achieving near-perfect intelligibility and naturalness, WaveNet's initial implementation was notably slow, reportedly requiring approximately four minutes to generate one second of audio.
Advancements in end-to-end training led to Google's development of the Tacotron model, which learns to synthesize speech directly from text-audio pairs. Tacotron processes text characters through various neural network submodules to generate corresponding audio spectrograms. This approach streamlines the speech synthesis process by integrating text analysis and audio generation into a unified framework.
In summary, the human voice remains the benchmark for speech synthesis technology. Continuous advancements in understanding and replicating human speech patterns are bringing us closer to creating synthesized voices that authentically capture the expressiveness and diversity of natural human communication.
Narris has developed advanced AI-powered speech synthesis models that facilitate natural, human-like conversations across multiple languages. Our platform supports high-performance, robust, and customizable deep learning models tailored to the complexities of Automatic Speech Recognition (ASR) and speech synthesis. By integrating real-time translation and voice technology, Narris enables various businesses to produce high-quality and compelling content.
Reference:https://www.ijert.org/research/speech-synthesis-a-review-IJERTV2IS60087.pdf
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