WO2022227190A1 - Speech synthesis method and apparatus, and electronic device and storage medium - Google Patents

Speech synthesis method and apparatus, and electronic device and storage medium Download PDF

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Publication number
WO2022227190A1
WO2022227190A1 PCT/CN2021/097075 CN2021097075W WO2022227190A1 WO 2022227190 A1 WO2022227190 A1 WO 2022227190A1 CN 2021097075 W CN2021097075 W CN 2021097075W WO 2022227190 A1 WO2022227190 A1 WO 2022227190A1
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speech synthesis
training text
autoregressive
sound spectrum
training
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PCT/CN2021/097075
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French (fr)
Chinese (zh)
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孙奥兰
王健宗
程宁
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平安科技(深圳)有限公司
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L13/00Speech synthesis; Text to speech systems
    • G10L13/02Methods for producing synthetic speech; Speech synthesisers
    • G10L13/04Details of speech synthesis systems, e.g. synthesiser structure or memory management
    • G10L13/047Architecture of speech synthesisers
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/27Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the analysis technique
    • G10L25/30Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the analysis technique using neural networks

Definitions

  • the present application relates to the field of artificial intelligence, and in particular, to a speech synthesis method, apparatus, electronic device, and computer-readable storage medium.
  • Speech synthesis refers to the technology that converts text information generated by the computer itself or input from external sources into intelligible and fluent speech output.
  • speech synthesis is one of the key technical links of modern speech artificial intelligence products.
  • the speed and quality of synthesized speech are the two main evaluation indicators for judging the pros and cons of speech synthesis systems, and are also important factors that affect the user experience of speech products.
  • the autoregressive type is based on the generated speech and can effectively learn acoustics.
  • the temporal change of features can generate high-quality speech, but due to the nature of speech sequence generation, the reasoning speed of speech in the generation process is slow, so that the speed of speech synthesis is slow; non-autoregressive type can generate speech audio.
  • Parallel generation of speech makes speech inference faster during generation, but ignores temporal changes in acoustic features, resulting in relatively poor synthesized speech quality.
  • a speech synthesis method provided by this application includes:
  • the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model
  • the present application also provides a speech synthesis device, the device comprising:
  • an acquisition module used for acquiring training text, and using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
  • an extraction module for extracting linguistic features from the first sound spectrum to obtain first linguistic features
  • a training module for training a pre-built non-autoregressive speech synthesis model according to the training text, the first sound spectrum and the first linguistic feature, to obtain a trained non-autoregressive speech synthesis model;
  • the synthesis module is configured to perform speech synthesis on the speech text to be synthesized by using the non-autoregressive speech synthesis model completed by the training to obtain a speech synthesis result.
  • the present application also provides an electronic device, the electronic device comprising:
  • the memory stores a computer program executable by the at least one processor, the computer program being executed by the at least one processor to implement a speech synthesis method as described below:
  • the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model
  • the present application also provides a computer-readable storage medium, where at least one computer program is stored, and the at least one computer program is executed by a processor in an electronic device to implement the speech synthesis method described below :
  • the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model
  • FIG. 1 is a schematic flowchart of a speech synthesis method provided by an embodiment of the present application.
  • FIG. 2 is a detailed schematic flowchart of one of the steps of the speech synthesis method provided in FIG. 1 in the first embodiment of the application;
  • FIG. 3 is a schematic block diagram of a speech synthesis apparatus provided by an embodiment of the present application.
  • FIG. 4 is a schematic diagram of an internal structure of an electronic device for implementing a speech synthesis method provided by an embodiment of the present application
  • the embodiment of the present application provides a speech synthesis method.
  • the executive body of the speech synthesis method includes, but is not limited to, at least one of electronic devices that can be configured to execute the method provided by the embodiments of the present application, such as a server, a terminal, and the like.
  • the speech synthesis method can be executed by software or hardware installed in a terminal device or a server device, and the software can be a blockchain platform.
  • the server includes but is not limited to: a single server, a server cluster, a cloud server or a cloud server cluster, and the like.
  • the speech synthesis method includes:
  • the training text includes different types of text, such as news text, financial text, and medical text.
  • the pre-trained autoregressive speech synthesis includes an encoder, a decoder and a vocoder for synthesizing text into speech.
  • the autoregressive speech synthesis model can be understood as an end-to-end speech synthesis model, that is, in the speech synthesis process, the speech data needs to be processed step by step.
  • the generation process of a sentence may include: first, generate the first word according to the probability distribution.
  • the second word is generated according to the first word
  • the third word is generated according to the first two words, and so on, until the entire sentence is generated, because the autoregressive speech synthesis model performs speech synthesis on the training text
  • the process is serial, so the synthesis quality of the speech synthesis result of the training text is guaranteed.
  • the pre-trained autoregressive speech synthesis model is used to perform speech synthesis on the training text to obtain the first sound spectrum of the training text, including:
  • the encoder is generated by connecting a convolution layer and a gated recurrent unit (GRU) residual to extract the acoustic features in the training text, and the decoder is connected to a long short-term memory network unit (LSTM) residual connection generation, the vocoder is built with multiple layers of causal convolutional layers and fully connected layers.
  • the residual connection is to ensure that the problem of model gradient dispersion is suppressed during the processing of the training text.
  • the acoustic characteristics refer to the physical quantities that represent the acoustic characteristics of speech, and are also the general term for the acoustic performance of various elements of the sound, such as the energy concentration area, formant frequency, formant intensity and bandwidth that represent the timbre, as well as the duration and fundamental frequency that represent the prosody characteristics of speech. , average speech power, etc.
  • the S20 includes: using a convolution layer in the encoder to perform acoustic feature convolution on the training text to obtain initial acoustic features, and using a gated loop unit in the encoder to perform a convolution on the initial acoustic features The features are up-sampled to obtain the acoustic features of the training text.
  • the convolution of the acoustic features of the training text is implemented through the convolution kernel of the convolution layer in the encoder, and the upsampling of the initial acoustic features is implemented through the update in the gated recurrent unit.
  • the gate and the reset gate are implemented, the update gate is used to discard and add the information of the initial acoustic features, and the reset gate is used to selectively filter the information of the initial acoustic features processed by the update gate.
  • the S21 includes: using the convolution layer in the decoder to perform feature extraction on the acoustic features, and using the long short-term memory network unit in the decoder to perform acoustic sequence recognition on the feature-extracted acoustic features , to obtain the acoustic feature sequence of the training text.
  • the acoustic sequence refers to the discrete time signal in the acoustic feature.
  • the feature extraction of the acoustic features is realized by the convolution kernel of the convolution layer in the decoder, and the acoustic sequence recognition is realized by the input gate and the forget gate in the long short-term memory network unit. and output gate implementation, the input gate is used to split the acoustic sequence of the acoustic features after feature extraction, the forget gate is used to identify the acoustic sequence of the acoustic features after feature extraction, and the output gate An acoustic sequence for outputting the feature-extracted acoustic features.
  • the S22 includes: using the causal convolution layer in the vocoder to perform sound frequency extraction on the acoustic feature sequence to obtain the sound frequency, and using the fully connected layer in the vocoder to extract the sound frequency
  • the frequency is subjected to matrix mapping to obtain the first sound spectrum of the training text.
  • Mel(f) represents the sound frequency
  • ln represents the filter function
  • f represents the acoustic feature sequence.
  • the linguistic features include the duration of pronunciation of words and the duration of sentence pauses. For example, there are 10 words and 2 sentences in a sound spectrum, and the pronunciation duration between the 10 words is 0.89s, 2 The pause time of each sentence is 1.2s. Based on the linguistic features, the phoneme duration of the generated sound spectrum can be identified, so that the generation quality of the sound spectrum can be judged.
  • the extracting linguistic features from the first sound spectrum to obtain the first linguistic features includes: acquiring word vectors and sentence vectors in the first sound spectrum, The vector and the sentence vector are smoothed, and the smoothed word vector and the sentence vector are sampled for the pronunciation duration to obtain the word pronunciation duration and the sentence pronunciation duration. According to the word pronunciation duration and the sentence pronunciation duration, the first linguistics is generated. feature.
  • the word vector is smoothed using the following formula:
  • Y(n) represents the word vector after smoothing
  • Y(n-1) represents the word vector
  • x 1 (n) and x 2 (n) represent the sound frequency and sound signal of the word vector
  • a 1 and a 2 represents the filter coefficient.
  • the smoothing process of the sentence vector is the same as the smoothing process of the word vector, which will not be further described here.
  • the pronunciation duration sampling of the word vector and the sentence vector may be implemented by a sampler in the autoregressive speech synthesis model, such as a KONTAKT sampler.
  • the non-autoregressive speech synthesis model does not consider the context dependencies between words in the training text, so as to directly perform speech decoding on the words in the training text, and realize the parallel decoding of words, thereby greatly improving the The speech synthesis speed of the training text, therefore, the embodiment of the present application trains the pre-built non-autoregressive speech synthesis model by using the training text and the first linguistic feature to supervise the training text in the The accuracy of speech synthesis points in the non-autoregressive model, thereby improving the speech synthesis accuracy rate of the non-autoregressive speech synthesis model, so that the non-autoregressive model has a faster speech synthesis speed and also has a higher voice synthetic accuracy.
  • the pre-built non-autoregressive speech synthesis model is trained according to the training text, the first sound spectrum and the first linguistic feature, and a trained non-autoregressive speech synthesis model is obtained.
  • a regression speech synthesis model comprising: inputting the training text into the non-autoregressive speech synthesis model to output a second sound spectrum of the training text, and extracting linguistic features from the second sound spectrum to obtain the second linguistic feature; according to the first sound spectrum, the second sound spectrum, the first linguistic feature and the second linguistic feature, calculate the training loss of the non-autoregressive speech synthesis model; if the training loss is not When a preset condition is met, adjust the parameters of the non-autoregressive model, and return to the step of inputting the training text into the non-autoregressive speech synthesis model; if the training loss meets the preset condition, get The trained non-autoregressive speech synthesis model.
  • calculating the training loss of the non-autoregressive speech synthesis model according to the first sound spectrum, the second sound spectrum, the first linguistic feature, and the second linguistic feature includes: For the first sound spectrum and the second sound spectrum, calculate the first training loss of the non-autoregressive speech synthesis model; according to the first linguistic feature and the second linguistic feature, calculate the non-autoregressive speech synthesis The second training loss of the model, the training loss of the non-autoregressive speech synthesis model is calculated according to the first training loss and the second training loss.
  • the first training loss is calculated according to the following formula:
  • LC represents the first training loss
  • m g represents the first sound spectrum
  • mp represents the second sound spectrum.
  • the second training loss is calculated according to the following formula:
  • L1 represents the second training loss
  • ⁇ g represents the first linguistic feature
  • ⁇ p represents the second linguistic feature
  • the second training loss is obtained by adding the word pronunciation duration loss and the sentence pronunciation duration loss in the linguistic feature.
  • the preset condition includes that the training loss is less than a loss threshold. That is, when the training loss is less than the loss threshold, it means that the training loss satisfies the preset condition, and when the training loss is greater than or equal to the loss threshold, it means that the training loss does not meet the predetermined condition. when the preset conditions are stated.
  • the loss threshold may be set to 0.1, or may be set according to actual scenarios.
  • the parameter adjustment of the non-autoregressive speech synthesis model may be implemented by a gradient descent algorithm, such as a stochastic descent algorithm.
  • the use of the trained non-autoregressive speech synthesis model to perform speech synthesis on to-be-synthesized speech text to obtain a speech synthesis result includes: inputting the to-be-synthesized speech text into the trained non-autoregressive speech text
  • the non-autoregressive speech synthesis model completed by the training is used to identify the sound spectrum of the speech text to be synthesized, and the speech synthesis result of the speech text to be synthesized is obtained.
  • the recognition of the sound spectrum of the speech text to be synthesized is realized by the encoder and the decoder in the non-autoregressive speech synthesis model completed by the training.
  • the speech synthesis result can also be stored in a blockchain node.
  • the training text is first synthesized by using a pre-trained autoregressive speech synthesis model to obtain the first sound spectrum of the training text, and linguistic features are extracted from the first sound spectrum to obtain the first sound spectrum of the training text.
  • the autoregressive speech synthesis model is trained to obtain the trained non-autoregressive speech synthesis model, so that the linguistic features can be added to the model training data, which can enhance the quality of the training data, and combine the first sound spectrum generated by the autoregressive model to supervise
  • the training results of the non-autoregressive model so that the non-autoregressive model can better learn the effect of speech synthesis of the autoregressive model, thereby improving the speech synthesis quality of the non-autoregressive model, and combined with the non-autoregressive model in the process of speech synthesis.
  • Parallel decoding of audio sequences is implemented, so that the non-autoregressive model has a faster speech synthesis speed and higher speech synthesis quality. Therefore, a speech synthesis method proposed in this application can improve the
  • FIG. 3 it is a functional block diagram of the speech synthesis apparatus of the present application.
  • the speech synthesis apparatus 100 described in this application can be installed in an electronic device.
  • the speech synthesis apparatus may include an acquisition module 101 , an extraction module 102 , a training module 103 and a synthesis module 104 .
  • the modules described in the present invention can also be called units, which refer to a series of computer program segments that can be executed by the electronic device processor and can perform fixed functions, and are stored in the memory of the electronic device.
  • each module/unit is as follows:
  • the acquisition module 101 is used for acquiring training text, and using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain a first sound spectrum of the training text;
  • the extraction module 102 is configured to extract linguistic features from the first sound spectrum to obtain first linguistic features
  • the training module 103 is configured to train a pre-built non-autoregressive speech synthesis model according to the training text, the first sound spectrum and the first linguistic feature, to obtain a trained non-autoregressive speech. synthetic model;
  • the synthesis module 104 is configured to use the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
  • the modules in the speech synthesis apparatus 100 in the embodiments of the present application use the same technical means as the speech synthesis methods described in the above-mentioned FIG. 1 and FIG. 2, and can generate the same technology The effect will not be repeated here.
  • FIG. 4 it is a schematic structural diagram of an electronic device implementing the speech synthesis method of the present application.
  • the electronic device 1 may include a processor 10, a memory 11 and a bus, and may also include a computer program, such as a speech synthesis program 12, stored in the memory 11 and executable on the processor 10.
  • a computer program such as a speech synthesis program 12 stored in the memory 11 and executable on the processor 10.
  • the memory 11 includes at least one type of readable storage medium, and the readable storage medium may be volatile or non-volatile.
  • the readable storage medium includes a flash memory, a mobile hard disk, a multimedia card, a card-type memory (eg, SD or DX memory, etc.), a magnetic memory, a magnetic disk, an optical disk, and the like.
  • the memory 11 may be an internal storage unit of the electronic device 1 in some embodiments, such as a mobile hard disk of the electronic device 1 . In other embodiments, the memory 11 may also be an external storage device of the electronic device 1, such as a pluggable mobile hard disk, a smart memory card (Smart Media Card, SMC), a secure digital (Secure Digital) equipped on the electronic device 1.
  • the memory 11 may also include both an internal storage unit of the electronic device 1 and an external storage device.
  • the memory 11 can not only be used to store application software installed in the electronic device 1 and various types of data, such as the code of the speech synthesis program 12, etc., but also can be used to temporarily store data that has been output or will be output.
  • the processor 10 may be composed of integrated circuits, for example, may be composed of a single packaged integrated circuit, or may be composed of multiple integrated circuits packaged with the same function or different functions, including one or more integrated circuits.
  • Central Processing Unit CPU
  • microprocessor digital processing chip
  • graphics processor and combination of various control chips, etc.
  • the processor 10 is the control core (Control Unit) of the electronic device, and uses various interfaces and lines to connect the various components of the entire electronic device, by running or executing the program or module (for example, executing the program) stored in the memory 11. speech synthesis program 12, etc.), and call data stored in the memory 11 to execute various functions of the electronic device 1 and process data.
  • the bus may be a peripheral component interconnect (PCI for short) bus or an extended industry standard architecture (Extended industry standard architecture, EISA for short) bus or the like.
  • PCI peripheral component interconnect
  • EISA Extended industry standard architecture
  • the bus can be divided into address bus, data bus, control bus and so on.
  • the bus is configured to implement connection communication between the memory 11 and at least one processor 10 and the like.
  • FIG. 4 only shows an electronic device with components. Those skilled in the art can understand that the structure shown in FIG. 4 does not constitute a limitation on the electronic device 1, and may include fewer or more components than those shown in the drawings. components, or a combination of certain components, or a different arrangement of components.
  • the electronic device 1 may also include a power supply (such as a battery) for powering the various components, preferably, the power supply may be logically connected to the at least one processor 10 through a power management device, so that the power management
  • the device implements functions such as charge management, discharge management, and power consumption management.
  • the power source may also include one or more DC or AC power sources, recharging devices, power failure detection circuits, power converters or inverters, power status indicators, and any other components.
  • the electronic device 1 may further include various sensors, Bluetooth modules, Wi-Fi modules, etc., which will not be repeated here.
  • the electronic device 1 may also include a network interface, optionally, the network interface may include a wired interface and/or a wireless interface (such as a WI-FI interface, a Bluetooth interface, etc.), which is usually used in the electronic device 1 Establish a communication connection with other electronic devices.
  • a network interface optionally, the network interface may include a wired interface and/or a wireless interface (such as a WI-FI interface, a Bluetooth interface, etc.), which is usually used in the electronic device 1 Establish a communication connection with other electronic devices.
  • the electronic device 1 may further include a user interface, and the user interface may be a display (Display), an input unit (eg, a keyboard (Keyboard)), optionally, the user interface may also be a standard wired interface or a wireless interface.
  • the display may be an LED display, a liquid crystal display, a touch-sensitive liquid crystal display, an OLED (Organic Light-Emitting Diode, organic light-emitting diode) touch panel, and the like.
  • the display may also be appropriately called a display screen or a display unit, which is used for displaying information processed in the electronic device 1 and for displaying a visualized user interface.
  • the speech synthesis 12 stored in the memory 11 in the electronic device 1 is a combination of multiple computer programs. When running in the processor 10, it can realize:
  • the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model
  • the modules/units integrated in the electronic device 1 may be stored in a non-volatile computer-readable storage medium.
  • the computer-readable storage medium may be volatile or non-volatile.
  • the computer-readable medium may include: any entity or device capable of carrying the computer program code, a recording medium, a USB flash drive, a removable hard disk, a magnetic disk, an optical disc, a computer memory, a read-only memory (ROM, Read-Only). Memory).
  • the present application also provides a computer-readable storage medium, where the readable storage medium stores a computer program, and when executed by a processor of an electronic device, the computer program can realize:
  • the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model
  • modules described as separate components may or may not be physically separated, and the components shown as modules may or may not be physical units, that is, may be located in one place, or may be distributed to multiple network units. Some or all of the modules may be selected according to actual needs to achieve the purpose of the solution in this embodiment.
  • each functional module in each embodiment of the present application may be integrated into one processing unit, or each unit may exist physically alone, or two or more units may be integrated into one unit.
  • the above-mentioned integrated units can be implemented in the form of hardware, or can be implemented in the form of hardware plus software function modules.
  • the blockchain referred to in this application is a new application mode of computer technologies such as distributed data storage, point-to-point transmission, consensus mechanism, and encryption algorithm.
  • Blockchain essentially a decentralized database, is a series of data blocks associated with cryptographic methods. Each data block contains a batch of network transaction information to verify its Validity of information (anti-counterfeiting) and generation of the next block.
  • the blockchain can include the underlying platform of the blockchain, the platform product service layer, and the application service layer.

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Abstract

The present application relates to the field of artificial intelligence. Disclosed is a speech synthesis method, comprising: acquiring training text, and performing speech synthesis on the training text by using a pre-trained autoregressive speech synthesis model, so as to obtain a first sound spectrum of the training text; extracting a linguistic feature from the first sound spectrum, so as to obtain a first linguistic feature; training a pre-constructed non-autoregressive speech synthesis model according to the training text, the first sound spectrum and the first linguistic feature, so as to obtain a non-autoregressive speech synthesis model, the training of which is completed; and by using the non-autoregressive speech synthesis model, the training of which is completed, performing, speech synthesis on speech text to be subjected to synthesis, so as to obtain a speech synthesis result. In addition, the present application further relates to blockchain technology. The speech synthesis result can be stored in a blockchain. Further disclosed in the present application are a speech synthesis apparatus, an electronic device and a storage medium By means of the present application, the speed and quality of speech synthesis can be improved.

Description

语音合成方法、装置、电子设备及存储介质Speech synthesis method, device, electronic device and storage medium
本申请要求于2021年04月25日提交中国专利局、申请号为202110450578.7,发明名称为“语音合成方法、装置、电子设备及存储介质”的中国专利申请的优先权,其全部内容通过引用结合在本申请中。This application claims the priority of the Chinese patent application with the application number 202110450578.7 and the invention titled "Speech Synthesis Method, Apparatus, Electronic Device and Storage Medium" filed with the China Patent Office on April 25, 2021, the entire contents of which are incorporated by reference in this application.
技术领域technical field
本申请涉及人工智能领域,尤其涉及一种语音合成方法、装置、电子设备及计算机可读存储介质。The present application relates to the field of artificial intelligence, and in particular, to a speech synthesis method, apparatus, electronic device, and computer-readable storage medium.
背景技术Background technique
语音合成指的是将计算机自己产生的、或外部输入的文字信息转变为可以听得懂的、流利的语音输出的技术,现如今语音合成是现代语音人工智能产品的关键技术环节之一,其合成语音的速度和质量是评判语音合成系统优劣的两个主要评价指标,也是影响语音产品用户体验的重要影响因素。Speech synthesis refers to the technology that converts text information generated by the computer itself or input from external sources into intelligible and fluent speech output. Nowadays, speech synthesis is one of the key technical links of modern speech artificial intelligence products. The speed and quality of synthesized speech are the two main evaluation indicators for judging the pros and cons of speech synthesis systems, and are also important factors that affect the user experience of speech products.
发明人意识到,目前语音合成中的声学模型通常有两种实现方法:自回归型和非自回归型,自回归型在生成语音音频时,是以已生成的语音为基础,能有效学习声学特征的时间变化,从而生成高质量的语音,但由于语音序列生成的性质,导致语音在生成过程中推理速度较慢,从而使得语音合成的速度较慢;非自回归型在生成语音音频时可以并列生成语音,使得语音在生成过程中推理速度更快,但忽略了声学特征的时间变化,导致合成的语音质量相对较差。The inventor realized that there are usually two implementation methods for acoustic models in speech synthesis: autoregressive type and non-autoregressive type. When generating speech audio, the autoregressive type is based on the generated speech and can effectively learn acoustics. The temporal change of features can generate high-quality speech, but due to the nature of speech sequence generation, the reasoning speed of speech in the generation process is slow, so that the speed of speech synthesis is slow; non-autoregressive type can generate speech audio. Parallel generation of speech makes speech inference faster during generation, but ignores temporal changes in acoustic features, resulting in relatively poor synthesized speech quality.
发明内容SUMMARY OF THE INVENTION
本申请提供的一种语音合成方法,包括:A speech synthesis method provided by this application includes:
获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;Acquiring training text, using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
从所述第一声音频谱中提取语言学特征,得到第一语言学特征;Extracting linguistic features from the first sound spectrum to obtain first linguistic features;
根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;According to the training text, the first sound spectrum and the first linguistic feature, the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model;
利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。Using the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
本申请还提供一种语音合成装置,所述装置包括:The present application also provides a speech synthesis device, the device comprising:
获取模块,用于获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;an acquisition module, used for acquiring training text, and using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
提取模块,用于从所述第一声音频谱中提取语言学特征,得到第一语言学特征;an extraction module for extracting linguistic features from the first sound spectrum to obtain first linguistic features;
训练模块,用于根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;A training module for training a pre-built non-autoregressive speech synthesis model according to the training text, the first sound spectrum and the first linguistic feature, to obtain a trained non-autoregressive speech synthesis model;
合成模块,用于利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。The synthesis module is configured to perform speech synthesis on the speech text to be synthesized by using the non-autoregressive speech synthesis model completed by the training to obtain a speech synthesis result.
本申请还提供一种电子设备,所述电子设备包括:The present application also provides an electronic device, the electronic device comprising:
至少一个处理器;以及,at least one processor; and,
与所述至少一个处理器通信连接的存储器;其中,a memory communicatively coupled to the at least one processor; wherein,
所述存储器存储有可被所述至少一个处理器执行的计算机程序,所述计算机程序被所 述至少一个处理器执行,以实现如下所述的语音合成方法:The memory stores a computer program executable by the at least one processor, the computer program being executed by the at least one processor to implement a speech synthesis method as described below:
获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;Acquiring training text, using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
从所述第一声音频谱中提取语言学特征,得到第一语言学特征;Extracting linguistic features from the first sound spectrum to obtain first linguistic features;
根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;According to the training text, the first sound spectrum and the first linguistic feature, the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model;
利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。Using the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
本申请还提供一种计算机可读存储介质,所述计算机可读存储介质中存储有至少一个计算机程序,所述至少一个计算机程序被电子设备中的处理器执行以实现如下所述的语音合成方法:The present application also provides a computer-readable storage medium, where at least one computer program is stored, and the at least one computer program is executed by a processor in an electronic device to implement the speech synthesis method described below :
获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;Acquiring training text, using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
从所述第一声音频谱中提取语言学特征,得到第一语言学特征;Extracting linguistic features from the first sound spectrum to obtain first linguistic features;
根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;According to the training text, the first sound spectrum and the first linguistic feature, the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model;
利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。Using the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
附图说明Description of drawings
图1为本申请一实施例提供的语音合成方法的流程示意图;1 is a schematic flowchart of a speech synthesis method provided by an embodiment of the present application;
图2为本申请第一实施例中图1提供的语音合成方法其中一个步骤的详细流程示意图;2 is a detailed schematic flowchart of one of the steps of the speech synthesis method provided in FIG. 1 in the first embodiment of the application;
图3为本申请一实施例提供的语音合成装置的模块示意图;3 is a schematic block diagram of a speech synthesis apparatus provided by an embodiment of the present application;
图4为本申请一实施例提供的实现语音合成方法的电子设备的内部结构示意图;4 is a schematic diagram of an internal structure of an electronic device for implementing a speech synthesis method provided by an embodiment of the present application;
本申请目的的实现、功能特点及优点将结合实施例,参照附图做进一步说明。The realization, functional characteristics and advantages of the purpose of the present application will be further described with reference to the accompanying drawings in conjunction with the embodiments.
具体实施方式Detailed ways
应当理解,此处所描述的具体实施例仅仅用以解释本申请,并不用于限定本申请。It should be understood that the specific embodiments described herein are only used to explain the present application, but not to limit the present application.
本申请实施例提供一种语音合成方法。所述语音合成方法的执行主体包括但不限于服务端、终端等能够被配置为执行本申请实施例提供的该方法的电子设备中的至少一种。换言之,所述语音合成方法可以由安装在终端设备或服务端设备的软件或硬件来执行,所述软件可以是区块链平台。所述服务端包括但不限于:单台服务器、服务器集群、云端服务器或云端服务器集群等。The embodiment of the present application provides a speech synthesis method. The executive body of the speech synthesis method includes, but is not limited to, at least one of electronic devices that can be configured to execute the method provided by the embodiments of the present application, such as a server, a terminal, and the like. In other words, the speech synthesis method can be executed by software or hardware installed in a terminal device or a server device, and the software can be a blockchain platform. The server includes but is not limited to: a single server, a server cluster, a cloud server or a cloud server cluster, and the like.
参照图1所示,为本申请一实施例提供的语音合成方法的流程示意图。在本申请实施例中,所述语音合成方法包括:Referring to FIG. 1 , a schematic flowchart of a speech synthesis method provided by an embodiment of the present application is shown. In the embodiment of the present application, the speech synthesis method includes:
S1、获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱。S1. Acquire training text, and use a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain a first sound spectrum of the training text.
本申请实施例中,所述训练文本包括不同类型的文本,比如:新闻文本、金融文本以及医疗文本等。所述预先训练好的自回归语音合成包括编码器、解码器以及声码器,用于将文本合成为语音。所述自回归语音合成模型可以理解为端到端的语音合成模型,即在语音合成过程,需要一步一步的处理语音数据,例如对于一个句子的生成过程可以包括:首先根据概率分布生成第一个词,其次根据第一个词生成第二个词,再根据前两个词生成第三个词,以此类推,直到生成整个句子,由于所述自回归语音合成模型对训练文本的语音合成的执行过程为串行,因此保障训练文本的语音合成结果的合成质量。In the embodiment of the present application, the training text includes different types of text, such as news text, financial text, and medical text. The pre-trained autoregressive speech synthesis includes an encoder, a decoder and a vocoder for synthesizing text into speech. The autoregressive speech synthesis model can be understood as an end-to-end speech synthesis model, that is, in the speech synthesis process, the speech data needs to be processed step by step. For example, the generation process of a sentence may include: first, generate the first word according to the probability distribution. , then the second word is generated according to the first word, the third word is generated according to the first two words, and so on, until the entire sentence is generated, because the autoregressive speech synthesis model performs speech synthesis on the training text The process is serial, so the synthesis quality of the speech synthesis result of the training text is guaranteed.
本申请的其中一个实施例,参阅图2所示,所述利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱,包括:In one embodiment of the present application, referring to FIG. 2 , the pre-trained autoregressive speech synthesis model is used to perform speech synthesis on the training text to obtain the first sound spectrum of the training text, including:
S20、利用所述自回归语音合成模型中的编码器对所述训练文本进行声学特征提取,得到所述训练文本的声学特征;S20, using the encoder in the autoregressive speech synthesis model to perform acoustic feature extraction on the training text to obtain the acoustic features of the training text;
S21、利用所述自回归语音合成模型中的解码器对所述声学特征进行序列识别,得到所述训练文本的声学特征序列;S21, using the decoder in the autoregressive speech synthesis model to perform sequence recognition on the acoustic features, to obtain the acoustic feature sequence of the training text;
S22、利用所述自回归语音合成模型中的声码器对所述声学特征序列进行频谱转换,得到所述训练文本的第一声音频谱。S22. Use the vocoder in the autoregressive speech synthesis model to perform spectrum conversion on the acoustic feature sequence to obtain a first sound spectrum of the training text.
其中,所述编码器通过卷积层与门控循环单元(GRU)残差连接生成,用于提取所述训练文本中的声学特征,所述解码器通过卷积层与长短期记忆网络单元(LSTM)残差连接生成,所述声码器通过多层因果卷积层和全连接层构建。所述残差连接是为了保证在对训练文本处理过程中抑制模型梯度弥散的问题。所述声学特征是指表示语音声学特性的物理量,也是声音诸要素声学表现的统称,如表示音色的能量集中区、共振峰频率、共振峰强度和带宽,以及表示语音韵律特性的时长、基频、平均语声功率等。Wherein, the encoder is generated by connecting a convolution layer and a gated recurrent unit (GRU) residual to extract the acoustic features in the training text, and the decoder is connected to a long short-term memory network unit ( LSTM) residual connection generation, the vocoder is built with multiple layers of causal convolutional layers and fully connected layers. The residual connection is to ensure that the problem of model gradient dispersion is suppressed during the processing of the training text. The acoustic characteristics refer to the physical quantities that represent the acoustic characteristics of speech, and are also the general term for the acoustic performance of various elements of the sound, such as the energy concentration area, formant frequency, formant intensity and bandwidth that represent the timbre, as well as the duration and fundamental frequency that represent the prosody characteristics of speech. , average speech power, etc.
进一步地,所述S20包括:利用所述编码器中的卷积层对所述训练文本进行声学特征卷积,得到初始声学特征,利用所述编码器中的门控循环单元对所述初始声学特征进行上采样,得到所述训练文本的声学特征。Further, the S20 includes: using a convolution layer in the encoder to perform acoustic feature convolution on the training text to obtain initial acoustic features, and using a gated loop unit in the encoder to perform a convolution on the initial acoustic features The features are up-sampled to obtain the acoustic features of the training text.
一个可选实施例中,所述训练文本的声学特征卷积通过所述编码器中的卷积层的卷积核实现,所述初始声学特征的上采样通过所述门控循环单元中的更新门和重置门实现,所述更新门用于舍去和增添所述初始声学特征的信息,所述重置门用于对更新门处理后的初始声学特征的信息进行选择性过滤。In an optional embodiment, the convolution of the acoustic features of the training text is implemented through the convolution kernel of the convolution layer in the encoder, and the upsampling of the initial acoustic features is implemented through the update in the gated recurrent unit. The gate and the reset gate are implemented, the update gate is used to discard and add the information of the initial acoustic features, and the reset gate is used to selectively filter the information of the initial acoustic features processed by the update gate.
进一步地,所述S21包括:利用所述解码器中的卷积层对所述声学特征进行特征提取,利用所述解码器中的长短期记忆网络单元对特征提取后的声学特征进行声学序列识别,得到所述训练文本的声学特征序列。其中,所述声学序列是指所述声学特征中的离散时间信号。Further, the S21 includes: using the convolution layer in the decoder to perform feature extraction on the acoustic features, and using the long short-term memory network unit in the decoder to perform acoustic sequence recognition on the feature-extracted acoustic features , to obtain the acoustic feature sequence of the training text. Wherein, the acoustic sequence refers to the discrete time signal in the acoustic feature.
一个可选实施例中,所述声学特征的特征提取通过所述解码器中的卷积层的卷积核实现,所述声学序列识别通过所述长短期记忆网络单元中的输入门、遗忘门和输出门实现,所述输入门用于对所述特征提取后的声学特征进行声学序列拆分,所述遗忘门用于对所述特征提取后的声学特征进行声学序列识别,所述输出门用于输出所述特征提取后的声学特征的声学序列。In an optional embodiment, the feature extraction of the acoustic features is realized by the convolution kernel of the convolution layer in the decoder, and the acoustic sequence recognition is realized by the input gate and the forget gate in the long short-term memory network unit. and output gate implementation, the input gate is used to split the acoustic sequence of the acoustic features after feature extraction, the forget gate is used to identify the acoustic sequence of the acoustic features after feature extraction, and the output gate An acoustic sequence for outputting the feature-extracted acoustic features.
进一步地,所述S22包括:利用所述声码器中的因果卷积层对所述声学特征序列进行声音频率提取,得到声音频率,利用所述声码器中的全连接层对所述声音频率进行矩阵映射,得到所述训练文本的第一声音频谱。可选的,利用下述公式对所述声学特征序列进行声音频率提取:Further, the S22 includes: using the causal convolution layer in the vocoder to perform sound frequency extraction on the acoustic feature sequence to obtain the sound frequency, and using the fully connected layer in the vocoder to extract the sound frequency The frequency is subjected to matrix mapping to obtain the first sound spectrum of the training text. Optionally, use the following formula to perform sound frequency extraction on the acoustic feature sequence:
Figure PCTCN2021097075-appb-000001
Figure PCTCN2021097075-appb-000001
其中,Mel(f)表示声音频率,ln表示滤波函数,f表示声学特征序列。Among them, Mel(f) represents the sound frequency, ln represents the filter function, and f represents the acoustic feature sequence.
S2、从所述第一声音频谱中提取语言学特征,得到第一语言学特征。S2. Extract linguistic features from the first sound spectrum to obtain first linguistic features.
本申请实施例中,所述语言学特征包括词语发音时长和句子停顿时长,比如在一段声音频谱中存在10个单词和2个句子,所述10个单词之间的发音时长为0.89s,2个句子的停顿时长为1.2s。基于所述语言学特征,可以识别出生成的声音频谱的音素持续时间,从而可以判断出声音频谱的生成质量。In the embodiment of the present application, the linguistic features include the duration of pronunciation of words and the duration of sentence pauses. For example, there are 10 words and 2 sentences in a sound spectrum, and the pronunciation duration between the 10 words is 0.89s, 2 The pause time of each sentence is 1.2s. Based on the linguistic features, the phoneme duration of the generated sound spectrum can be identified, so that the generation quality of the sound spectrum can be judged.
本申请的其中一个实施例,所述从所述第一声音频谱中提取语言学特征,得到第一语言学特征包括:获取所述第一声音频谱中的词语向量及句子向量,对所述词语向量及句子向量进行平滑处理,将平滑处理后的词语向量与句子向量进行发音时长采样,得到词语发音时长和句子发音时长,根据所述词语发音时长和句子发音时长,生成所述第一语言学特 征。In one of the embodiments of the present application, the extracting linguistic features from the first sound spectrum to obtain the first linguistic features includes: acquiring word vectors and sentence vectors in the first sound spectrum, The vector and the sentence vector are smoothed, and the smoothed word vector and the sentence vector are sampled for the pronunciation duration to obtain the word pronunciation duration and the sentence pronunciation duration. According to the word pronunciation duration and the sentence pronunciation duration, the first linguistics is generated. feature.
一个可选实施例中,利用下述公式对所述词语向量进行平滑处理:In an optional embodiment, the word vector is smoothed using the following formula:
Y(n)=a 1x 1(n)+a 2x 2(n)-Y(n-1) Y(n)=a 1 x 1 (n)+a 2 x 2 (n)-Y(n-1)
其中,Y(n)表示平滑处理后的词语向量,Y(n-1)表示词语向量,x 1(n)和x 2(n)表示词语向量的声音频率和声音信号,a 1和a 2表示滤波系数。本申请实施例中,所述句子向量的平滑处理与所述词语向量的平滑处理原理相同,在此不做进一步地赘述。 Among them, Y(n) represents the word vector after smoothing, Y(n-1) represents the word vector, x 1 (n) and x 2 (n) represent the sound frequency and sound signal of the word vector, a 1 and a 2 represents the filter coefficient. In the embodiment of the present application, the smoothing process of the sentence vector is the same as the smoothing process of the word vector, which will not be further described here.
一个可选实施例中,所述词语向量与句子向量的发音时长采样可以通过所述自回归语音合成模型中的采样器实现,如KONTAKT采样器。In an optional embodiment, the pronunciation duration sampling of the word vector and the sentence vector may be implemented by a sampler in the autoregressive speech synthesis model, such as a KONTAKT sampler.
S3、根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型。S3. Train the pre-built non-autoregressive speech synthesis model according to the training text, the first sound spectrum and the first linguistic feature, to obtain a trained non-autoregressive speech synthesis model.
本申请实施例中,所述非自回归语音合成模型不考虑训练文本中词语之间上下文的依赖关系,以直接对所述训练文本中词语进行语音解码,实现了词语的并行解码,从而大大提高训练文本的语音合成速度,因此,本申请实施例通过所述训练文本及所述第一语言学特征对所述预先构建的非自回归语音合成模型进行训练,以监督所述训练文本在所述非自回归模型中语音合成点的准确性,从而提高所述非自回归语音合成模型的语音合成准确率,使得所述非自回归模型具有较快的语音合成速度的同时也具有较高的语音合成准确率。In the embodiment of the present application, the non-autoregressive speech synthesis model does not consider the context dependencies between words in the training text, so as to directly perform speech decoding on the words in the training text, and realize the parallel decoding of words, thereby greatly improving the The speech synthesis speed of the training text, therefore, the embodiment of the present application trains the pre-built non-autoregressive speech synthesis model by using the training text and the first linguistic feature to supervise the training text in the The accuracy of speech synthesis points in the non-autoregressive model, thereby improving the speech synthesis accuracy rate of the non-autoregressive speech synthesis model, so that the non-autoregressive model has a faster speech synthesis speed and also has a higher voice synthetic accuracy.
本申请的其中一个实施例,所述根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型,包括:将所述训练文本输入所述非自回归语音合成模型中,以输出所述训练文本的第二声音频谱,并从所述第二声音频谱中提取语言学特征,得到第二语言学特征;根据所述第一声音频谱、第二声音频谱、第一语言学特征以及第二语言学特征,计算所述非自回归语音合成模型的训练损失;若所述训练损失不满足预设条件时,调整所述非自回归模型的参数,并返回所述将所述训练文本输入所述非自回归语音合成模型中的步骤;若所述训练损失满足预设条件时,得到训练完成的非自回归语音合成模型。In one of the embodiments of the present application, the pre-built non-autoregressive speech synthesis model is trained according to the training text, the first sound spectrum and the first linguistic feature, and a trained non-autoregressive speech synthesis model is obtained. A regression speech synthesis model, comprising: inputting the training text into the non-autoregressive speech synthesis model to output a second sound spectrum of the training text, and extracting linguistic features from the second sound spectrum to obtain the second linguistic feature; according to the first sound spectrum, the second sound spectrum, the first linguistic feature and the second linguistic feature, calculate the training loss of the non-autoregressive speech synthesis model; if the training loss is not When a preset condition is met, adjust the parameters of the non-autoregressive model, and return to the step of inputting the training text into the non-autoregressive speech synthesis model; if the training loss meets the preset condition, get The trained non-autoregressive speech synthesis model.
一个可选实施例中,所述第二声音频谱的语言学特征提取原理可以参阅上述第一声音频谱的语言学特征,再次不做进一步赘述。In an optional embodiment, for the extraction principle of the linguistic feature of the second sound spectrum, reference may be made to the linguistic feature of the first sound spectrum, which will not be further described again.
一个可选实施例中,所述根据所述第一声音频谱、第二声音频谱、第一语言学特征以及第二语言学特征,计算所述非自回归语音合成模型的训练损失,包括:根据所述第一声音频谱及第二声音频谱,计算所述非自回归语音合成模型的第一训练损失;根据所述第一语言学特征以及第二语言学特征,计算所述非自回归语音合成模型的第二训练损失,根据所述第一训练损失和第二训练损失,计算所述非自回归语音合成模型的训练损失。In an optional embodiment, calculating the training loss of the non-autoregressive speech synthesis model according to the first sound spectrum, the second sound spectrum, the first linguistic feature, and the second linguistic feature includes: For the first sound spectrum and the second sound spectrum, calculate the first training loss of the non-autoregressive speech synthesis model; according to the first linguistic feature and the second linguistic feature, calculate the non-autoregressive speech synthesis The second training loss of the model, the training loss of the non-autoregressive speech synthesis model is calculated according to the first training loss and the second training loss.
在本申请的一个可选实施例中,根据下述公式计算所述第一训练损失:In an optional embodiment of the present application, the first training loss is calculated according to the following formula:
LC=m glogm p+(1-m g)log(1-m p) LC=mg logm p +(1-m g ) log(1-m p )
其中,LC表示第一训练损失,m g表示第一声音频谱,m p表示第二声音频谱。 Among them, LC represents the first training loss, m g represents the first sound spectrum, and mp represents the second sound spectrum.
在本申请的一个可选实施例中,根据下述公式计算所述第二训练损失:In an optional embodiment of the present application, the second training loss is calculated according to the following formula:
L1=|α pg| L1=|α pg |
其中,L1表示第二训练损失,α g表示第一语言学特征,α p表示第二语言学特征。 Among them, L1 represents the second training loss, α g represents the first linguistic feature, and α p represents the second linguistic feature.
需要说明的是,所述第二训练损失是指所述语言学特征中词语发音时长损失和句子发音时长损失相加得到。It should be noted that the second training loss is obtained by adding the word pronunciation duration loss and the sentence pronunciation duration loss in the linguistic feature.
在本申请的一个可选实施例中,将所述第一训练损失和所述第二训练损失进行相加,得到所述非自回归语音合成模型的训练损失即L=L1+LC。In an optional embodiment of the present application, the first training loss and the second training loss are added to obtain the training loss of the non-autoregressive speech synthesis model, that is, L=L1+LC.
在本申请的一个可选实施例中,所述预设条件包括所述训练损失小于损失阈值。即当 所述训练损失小于所述损失阈值时,则表示所述训练损失满足所述预设条件时,当所述训练损失大于或者等于所述损失阈值时,则表示所述训练损失不满足所述预设条件时。其中,所述损失阈值可以设置为0.1,也可以根据实际场景设置。进一步地,所述非自回归语音合成模型的参数调整可以通过梯度下降算法实现,如随机下降算法。In an optional embodiment of the present application, the preset condition includes that the training loss is less than a loss threshold. That is, when the training loss is less than the loss threshold, it means that the training loss satisfies the preset condition, and when the training loss is greater than or equal to the loss threshold, it means that the training loss does not meet the predetermined condition. when the preset conditions are stated. Wherein, the loss threshold may be set to 0.1, or may be set according to actual scenarios. Further, the parameter adjustment of the non-autoregressive speech synthesis model may be implemented by a gradient descent algorithm, such as a stochastic descent algorithm.
S4、利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。S4, using the non-autoregressive speech synthesis model completed by the training to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
本申请实施例中,所述利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果,包括:将所述待合成语音文本输入所述训练完成的非自回归语音合成模型中,利用所述训练完成的非自回归语音合成模型识别所述待合成语音文本的声音频谱,得到所述待合成语音文本的语音合成结果。其中,所述待合成语音文本的声音频谱的识别通过所述训练完成的非自回归语音合成模型中的编码器和解码器实现。In the embodiment of the present application, the use of the trained non-autoregressive speech synthesis model to perform speech synthesis on to-be-synthesized speech text to obtain a speech synthesis result includes: inputting the to-be-synthesized speech text into the trained non-autoregressive speech text In the regression speech synthesis model, the non-autoregressive speech synthesis model completed by the training is used to identify the sound spectrum of the speech text to be synthesized, and the speech synthesis result of the speech text to be synthesized is obtained. Wherein, the recognition of the sound spectrum of the speech text to be synthesized is realized by the encoder and the decoder in the non-autoregressive speech synthesis model completed by the training.
进一步地,为保障所述语音合成结果的安全性和隐私性,所述语音合成结果还可存储于一区块链节点中。Further, in order to ensure the security and privacy of the speech synthesis result, the speech synthesis result can also be stored in a blockchain node.
本申请实施例首先利用预先训练好的自回归语音合成模型对训练文本进行语音合成,得到所述训练文本的第一声音频谱,并从所述第一声音频谱中提取语言学特征,得到第一语言学特征,以获取后续非自回归语音合成模型的训练数据;其次,本申请实施例将根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型,以将语言学特征加入到模型训练数据中,可以增强训练数据的质量,并结合自回归模型生成的第一声音频谱监督非自回归模型的训练结果,以使非自回归模型可以更好的学习自回归模型语音合成的效果,从而提高非自回归模型的语音合成质量,再结合非自回归模型在语音合成过程中可以实现音频序列并行解码,从而使得所述非自回归模型具有较快的语音合成速度的同时又具有较高的语音合成质量。因此,本申请提出的一种语音合成方法可以提高语音合成的速度和质量。In the embodiment of the present application, the training text is first synthesized by using a pre-trained autoregressive speech synthesis model to obtain the first sound spectrum of the training text, and linguistic features are extracted from the first sound spectrum to obtain the first sound spectrum of the training text. linguistic features to obtain the training data of the subsequent non-autoregressive speech synthesis model; secondly, in this embodiment of the present application, according to the training text, the first sound spectrum and the first linguistic feature, the pre-built non-autoregressive speech The autoregressive speech synthesis model is trained to obtain the trained non-autoregressive speech synthesis model, so that the linguistic features can be added to the model training data, which can enhance the quality of the training data, and combine the first sound spectrum generated by the autoregressive model to supervise The training results of the non-autoregressive model, so that the non-autoregressive model can better learn the effect of speech synthesis of the autoregressive model, thereby improving the speech synthesis quality of the non-autoregressive model, and combined with the non-autoregressive model in the process of speech synthesis. Parallel decoding of audio sequences is implemented, so that the non-autoregressive model has a faster speech synthesis speed and higher speech synthesis quality. Therefore, a speech synthesis method proposed in this application can improve the speed and quality of speech synthesis.
如图3所示,是本申请语音合成装置的功能模块图。As shown in FIG. 3 , it is a functional block diagram of the speech synthesis apparatus of the present application.
本申请所述语音合成装置100可以安装于电子设备中。根据实现的功能,所述语音合成装置可以包括获取模块101、提取模块102、训练模块103以及合成模块104、。本发所述模块也可以称之为单元,是指一种能够被电子设备处理器所执行,并且能够完成固定功能的一系列计算机程序段,其存储在电子设备的存储器中。The speech synthesis apparatus 100 described in this application can be installed in an electronic device. According to the realized functions, the speech synthesis apparatus may include an acquisition module 101 , an extraction module 102 , a training module 103 and a synthesis module 104 . The modules described in the present invention can also be called units, which refer to a series of computer program segments that can be executed by the electronic device processor and can perform fixed functions, and are stored in the memory of the electronic device.
在本实施例中,关于各模块/单元的功能如下:In this embodiment, the functions of each module/unit are as follows:
所述获取模块101,用于获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;The acquisition module 101 is used for acquiring training text, and using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain a first sound spectrum of the training text;
所述提取模块102,用于从所述第一声音频谱中提取语言学特征,得到第一语言学特征;The extraction module 102 is configured to extract linguistic features from the first sound spectrum to obtain first linguistic features;
所述训练模块103,用于根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;The training module 103 is configured to train a pre-built non-autoregressive speech synthesis model according to the training text, the first sound spectrum and the first linguistic feature, to obtain a trained non-autoregressive speech. synthetic model;
所述合成模块104,用于利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。The synthesis module 104 is configured to use the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
详细地,本申请实施例中所述语音合成装置100中的所述各模块在使用时采用与上述的图1和图2中所述的语音合成方法一样的技术手段,并能够产生相同的技术效果,这里不再赘述。In detail, the modules in the speech synthesis apparatus 100 in the embodiments of the present application use the same technical means as the speech synthesis methods described in the above-mentioned FIG. 1 and FIG. 2, and can generate the same technology The effect will not be repeated here.
如图4所示,是本申请实现语音合成方法的电子设备的结构示意图。As shown in FIG. 4 , it is a schematic structural diagram of an electronic device implementing the speech synthesis method of the present application.
所述电子设备1可以包括处理器10、存储器11和总线,还可以包括存储在所述存储 器11中并可在所述处理器10上运行的计算机程序,如语音合成程序12。The electronic device 1 may include a processor 10, a memory 11 and a bus, and may also include a computer program, such as a speech synthesis program 12, stored in the memory 11 and executable on the processor 10.
其中,所述存储器11至少包括一种类型的可读存储介质,所述可读存储介质可以是易失性的,也可以是非易失性的。具体的,所述可读存储介质包括闪存、移动硬盘、多媒体卡、卡型存储器(例如:SD或DX存储器等)、磁性存储器、磁盘、光盘等。所述存储器11在一些实施例中可以是电子设备1的内部存储单元,例如该电子设备1的移动硬盘。所述存储器11在另一些实施例中也可以是电子设备1的外部存储设备,例如电子设备1上配备的插接式移动硬盘、智能存储卡(Smart Media Card,SMC)、安全数字(Secure Digital,SD)卡、闪存卡(Flash Card)等。进一步地,所述存储器11还可以既包括电子设备1的内部存储单元也包括外部存储设备。所述存储器11不仅可以用于存储安装于电子设备1的应用软件及各类数据,例如语音合成程序12的代码等,还可以用于暂时地存储已经输出或者将要输出的数据。Wherein, the memory 11 includes at least one type of readable storage medium, and the readable storage medium may be volatile or non-volatile. Specifically, the readable storage medium includes a flash memory, a mobile hard disk, a multimedia card, a card-type memory (eg, SD or DX memory, etc.), a magnetic memory, a magnetic disk, an optical disk, and the like. The memory 11 may be an internal storage unit of the electronic device 1 in some embodiments, such as a mobile hard disk of the electronic device 1 . In other embodiments, the memory 11 may also be an external storage device of the electronic device 1, such as a pluggable mobile hard disk, a smart memory card (Smart Media Card, SMC), a secure digital (Secure Digital) equipped on the electronic device 1. , SD) card, flash memory card (Flash Card), etc. Further, the memory 11 may also include both an internal storage unit of the electronic device 1 and an external storage device. The memory 11 can not only be used to store application software installed in the electronic device 1 and various types of data, such as the code of the speech synthesis program 12, etc., but also can be used to temporarily store data that has been output or will be output.
所述处理器10在一些实施例中可以由集成电路组成,例如可以由单个封装的集成电路所组成,也可以是由多个相同功能或不同功能封装的集成电路所组成,包括一个或者多个中央处理器(Central Processing unit,CPU)、微处理器、数字处理芯片、图形处理器及各种控制芯片的组合等。所述处理器10是所述电子设备的控制核心(Control Unit),利用各种接口和线路连接整个电子设备的各个部件,通过运行或执行存储在所述存储器11内的程序或者模块(例如执行语音合成程序12等),以及调用存储在所述存储器11内的数据,以执行电子设备1的各种功能和处理数据。In some embodiments, the processor 10 may be composed of integrated circuits, for example, may be composed of a single packaged integrated circuit, or may be composed of multiple integrated circuits packaged with the same function or different functions, including one or more integrated circuits. Central Processing Unit (CPU), microprocessor, digital processing chip, graphics processor and combination of various control chips, etc. The processor 10 is the control core (Control Unit) of the electronic device, and uses various interfaces and lines to connect the various components of the entire electronic device, by running or executing the program or module (for example, executing the program) stored in the memory 11. speech synthesis program 12, etc.), and call data stored in the memory 11 to execute various functions of the electronic device 1 and process data.
所述总线可以是外设部件互连标准(peripheral component interconnect,简称PCI)总线或扩展工业标准结构(extended industry standard architecture,简称EISA)总线等。该总线可以分为地址总线、数据总线、控制总线等。所述总线被设置为实现所述存储器11以及至少一个处理器10等之间的连接通信。The bus may be a peripheral component interconnect (PCI for short) bus or an extended industry standard architecture (Extended industry standard architecture, EISA for short) bus or the like. The bus can be divided into address bus, data bus, control bus and so on. The bus is configured to implement connection communication between the memory 11 and at least one processor 10 and the like.
图4仅示出了具有部件的电子设备,本领域技术人员可以理解的是,图4示出的结构并不构成对所述电子设备1的限定,可以包括比图示更少或者更多的部件,或者组合某些部件,或者不同的部件布置。FIG. 4 only shows an electronic device with components. Those skilled in the art can understand that the structure shown in FIG. 4 does not constitute a limitation on the electronic device 1, and may include fewer or more components than those shown in the drawings. components, or a combination of certain components, or a different arrangement of components.
例如,尽管未示出,所述电子设备1还可以包括给各个部件供电的电源(比如电池),优选地,电源可以通过电源管理装置与所述至少一个处理器10逻辑相连,从而通过电源管理装置实现充电管理、放电管理、以及功耗管理等功能。电源还可以包括一个或一个以上的直流或交流电源、再充电装置、电源故障检测电路、电源转换器或者逆变器、电源状态指示器等任意组件。所述电子设备1还可以包括多种传感器、蓝牙模块、Wi-Fi模块等,在此不再赘述。For example, although not shown, the electronic device 1 may also include a power supply (such as a battery) for powering the various components, preferably, the power supply may be logically connected to the at least one processor 10 through a power management device, so that the power management The device implements functions such as charge management, discharge management, and power consumption management. The power source may also include one or more DC or AC power sources, recharging devices, power failure detection circuits, power converters or inverters, power status indicators, and any other components. The electronic device 1 may further include various sensors, Bluetooth modules, Wi-Fi modules, etc., which will not be repeated here.
进一步地,所述电子设备1还可以包括网络接口,可选地,所述网络接口可以包括有线接口和/或无线接口(如WI-FI接口、蓝牙接口等),通常用于在该电子设备1与其他电子设备之间建立通信连接。Further, the electronic device 1 may also include a network interface, optionally, the network interface may include a wired interface and/or a wireless interface (such as a WI-FI interface, a Bluetooth interface, etc.), which is usually used in the electronic device 1 Establish a communication connection with other electronic devices.
可选地,该电子设备1还可以包括用户接口,用户接口可以是显示器(Display)、输入单元(比如键盘(Keyboard)),可选地,用户接口还可以是标准的有线接口、无线接口。可选地,在一些实施例中,显示器可以是LED显示器、液晶显示器、触控式液晶显示器以及OLED(Organic Light-Emitting Diode,有机发光二极管)触摸器等。其中,显示器也可以适当的称为显示屏或显示单元,用于显示在电子设备1中处理的信息以及用于显示可视化的用户界面。Optionally, the electronic device 1 may further include a user interface, and the user interface may be a display (Display), an input unit (eg, a keyboard (Keyboard)), optionally, the user interface may also be a standard wired interface or a wireless interface. Optionally, in some embodiments, the display may be an LED display, a liquid crystal display, a touch-sensitive liquid crystal display, an OLED (Organic Light-Emitting Diode, organic light-emitting diode) touch panel, and the like. The display may also be appropriately called a display screen or a display unit, which is used for displaying information processed in the electronic device 1 and for displaying a visualized user interface.
应该了解,所述实施例仅为说明之用,在专利申请范围上并不受此结构的限制。It should be understood that the embodiments are only used for illustration, and are not limited by this structure in the scope of the patent application.
所述电子设备1中的所述存储器11存储的语音合成12是多个计算机程序的组合,在所述处理器10中运行时,可以实现:The speech synthesis 12 stored in the memory 11 in the electronic device 1 is a combination of multiple computer programs. When running in the processor 10, it can realize:
获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;Acquiring training text, using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
从所述第一声音频谱中提取语言学特征,得到第一语言学特征;Extracting linguistic features from the first sound spectrum to obtain first linguistic features;
根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;According to the training text, the first sound spectrum and the first linguistic feature, the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model;
利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。Using the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
具体地,所述处理器10对上述计算机程序的具体实现方法可参考图1对应实施例中相关步骤的描述,在此不赘述。Specifically, for the specific implementation method of the above-mentioned computer program by the processor 10, reference may be made to the description of the relevant steps in the corresponding embodiment of FIG. 1, and details are not described herein.
进一步地,所述电子设备1集成的模块/单元如果以软件功能单元的形式实现并作为独立的产品销售或使用时,可以存储在一个非易失性计算机可读取存储介质中。所述计算机可读存储介质可以是易失性的,也可以是非易失性的。例如,所述计算机可读介质可以包括:能够携带所述计算机程序代码的任何实体或装置、记录介质、U盘、移动硬盘、磁碟、光盘、计算机存储器、只读存储器(ROM,Read-Only Memory)。Further, if the modules/units integrated in the electronic device 1 are implemented in the form of software functional units and sold or used as independent products, they may be stored in a non-volatile computer-readable storage medium. The computer-readable storage medium may be volatile or non-volatile. For example, the computer-readable medium may include: any entity or device capable of carrying the computer program code, a recording medium, a USB flash drive, a removable hard disk, a magnetic disk, an optical disc, a computer memory, a read-only memory (ROM, Read-Only). Memory).
本申请还提供一种计算机可读存储介质,所述可读存储介质存储有计算机程序,所述计算机程序在被电子设备的处理器所执行时,可以实现:The present application also provides a computer-readable storage medium, where the readable storage medium stores a computer program, and when executed by a processor of an electronic device, the computer program can realize:
获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;Acquiring training text, using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
从所述第一声音频谱中提取语言学特征,得到第一语言学特征;Extracting linguistic features from the first sound spectrum to obtain first linguistic features;
根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;According to the training text, the first sound spectrum and the first linguistic feature, the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model;
利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。Using the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
在本申请所提供的几个实施例中,应该理解到,所揭露的设备,装置和方法,可以通过其它的方式实现。例如,以上所描述的装置实施例仅仅是示意性的,例如,所述模块的划分,仅仅为一种逻辑功能划分,实际实现时可以有另外的划分方式。In the several embodiments provided in this application, it should be understood that the disclosed apparatus, apparatus and method may be implemented in other manners. For example, the apparatus embodiments described above are only illustrative. For example, the division of the modules is only a logical function division, and there may be other division manners in actual implementation.
所述作为分离部件说明的模块可以是或者也可以不是物理上分开的,作为模块显示的部件可以是或者也可以不是物理单元,即可以位于一个地方,或者也可以分布到多个网络单元上。可以根据实际的需要选择其中的部分或者全部模块来实现本实施例方案的目的。The modules described as separate components may or may not be physically separated, and the components shown as modules may or may not be physical units, that is, may be located in one place, or may be distributed to multiple network units. Some or all of the modules may be selected according to actual needs to achieve the purpose of the solution in this embodiment.
另外,在本申请各个实施例中的各功能模块可以集成在一个处理单元中,也可以是各个单元单独物理存在,也可以两个或两个以上单元集成在一个单元中。上述集成的单元既可以采用硬件的形式实现,也可以采用硬件加软件功能模块的形式实现。In addition, each functional module in each embodiment of the present application may be integrated into one processing unit, or each unit may exist physically alone, or two or more units may be integrated into one unit. The above-mentioned integrated units can be implemented in the form of hardware, or can be implemented in the form of hardware plus software function modules.
对于本领域技术人员而言,显然本申请不限于上述示范性实施例的细节,而且在不背离本申请的精神或基本特征的情况下,能够以其他的具体形式实现本申请。It will be apparent to those skilled in the art that the present application is not limited to the details of the above-described exemplary embodiments, but that the present application can be implemented in other specific forms without departing from the spirit or essential characteristics of the present application.
因此,无论从哪一点来看,均应将实施例看作是示范性的,而且是非限制性的,本申请的范围由所附权利要求而不是上述说明限定,因此旨在将落在权利要求的等同要件的含义和范围内的所有变化涵括在本申请内。不应将权利要求中的任何附关联图标记视为限制所涉及的权利要求。Accordingly, the embodiments are to be regarded in all respects as illustrative and not restrictive, and the scope of the application is to be defined by the appended claims rather than the foregoing description, which is therefore intended to fall within the scope of the claims. All changes within the meaning and scope of the equivalents of , are included in this application. Any reference signs in the claims shall not be construed as limiting the involved claim.
本申请所指区块链是分布式数据存储、点对点传输、共识机制、加密算法等计算机技术的新型应用模式。区块链(Blockchain),本质上是一个去中心化的数据库,是一串使用密码学方法相关联产生的数据块,每一个数据块中包含了一批次网络交易的信息,用于验证其信息的有效性(防伪)和生成下一个区块。区块链可以包括区块链底层平台、平台产品服务层以及应用服务层等。The blockchain referred to in this application is a new application mode of computer technologies such as distributed data storage, point-to-point transmission, consensus mechanism, and encryption algorithm. Blockchain, essentially a decentralized database, is a series of data blocks associated with cryptographic methods. Each data block contains a batch of network transaction information to verify its Validity of information (anti-counterfeiting) and generation of the next block. The blockchain can include the underlying platform of the blockchain, the platform product service layer, and the application service layer.
此外,显然“包括”一词不排除其他单元或步骤,单数不排除复数。系统权利要求中陈述的多个单元或装置也可以由一个单元或装置通过软件或者硬件来实现。第二等词语用来表示名称,而并不表示任何特定的顺序。Furthermore, it is clear that the word "comprising" does not exclude other units or steps and the singular does not exclude the plural. Several units or means recited in the system claims can also be realized by one unit or means by means of software or hardware. Second-class terms are used to denote names and do not denote any particular order.
最后应说明的是,以上实施例仅用以说明本申请的技术方案而非限制,尽管参照较佳 实施例对本申请进行了详细说明,本领域的普通技术人员应当理解,可以对本申请的技术方案进行修改或等同替换,而不脱离本申请技术方案的精神和范围。Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present application rather than limitations. Although the present application has been described in detail with reference to the preferred embodiments, those of ordinary skill in the art should understand that the technical solutions of the present application can be Modifications or equivalent substitutions can be made without departing from the spirit and scope of the technical solutions of the present application.

Claims (20)

  1. 一种语音合成方法,其中,所述方法包括:A speech synthesis method, wherein the method comprises:
    获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;Acquiring training text, using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
    从所述第一声音频谱中提取语言学特征,得到第一语言学特征;Extracting linguistic features from the first sound spectrum to obtain first linguistic features;
    根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;According to the training text, the first sound spectrum and the first linguistic feature, the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model;
    利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。Using the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
  2. 如权利要求1所述的语音合成方法,其中,所述利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱,包括:The speech synthesis method according to claim 1, wherein said using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text, comprising:
    利用所述自回归语音合成模型中的编码器对所述训练文本进行声学特征提取,得到所述训练文本的声学特征;Use the encoder in the autoregressive speech synthesis model to perform acoustic feature extraction on the training text to obtain the acoustic features of the training text;
    利用所述自回归语音合成模型中的解码器对所述声学特征进行序列识别,得到所述训练文本的声学特征序列;Use the decoder in the autoregressive speech synthesis model to perform sequence recognition on the acoustic features to obtain the acoustic feature sequence of the training text;
    利用所述自回归语音合成模型中的声码器对所述声学特征序列进行频谱转换,得到所述训练文本的第一声音频谱。The vocoder in the autoregressive speech synthesis model is used to perform spectrum conversion on the acoustic feature sequence to obtain the first sound spectrum of the training text.
  3. 如权利要求1所述的语音合成方法,其中,所述利用所述自回归语音合成模型中的编码器对所述训练文本进行声学特征提取,得到所述训练文本的声学特征,包括:The speech synthesis method according to claim 1, wherein, the use of the encoder in the autoregressive speech synthesis model to perform acoustic feature extraction on the training text to obtain the acoustic features of the training text, comprising:
    利用所述编码器中的卷积层对所述训练文本进行声学特征卷积,得到初始声学特征;Use the convolution layer in the encoder to perform acoustic feature convolution on the training text to obtain initial acoustic features;
    利用所述编码器中的门控循环单元对所述初始声学特征进行上采样,得到所述训练文本的声学特征。The initial acoustic features are up-sampled by the gated loop unit in the encoder to obtain the acoustic features of the training text.
  4. 如权利要求2所述的语音合成方法,其中,所述利用所述自回归语音合成模型中的解码器对所述声学特征进行序列识别,得到所述训练文本的声学特征序列,包括:The speech synthesis method according to claim 2, wherein the sequence recognition of the acoustic features by the decoder in the autoregressive speech synthesis model to obtain the acoustic feature sequence of the training text, comprising:
    利用所述解码器中的卷积层对所述声学特征进行特征提取;Feature extraction is performed on the acoustic features by using a convolutional layer in the decoder;
    利用所述解码器中的长短期记忆网络单元对特征提取后的声学特征进行声学序列识别,得到所述训练文本的声学特征序列。The long-short-term memory network unit in the decoder is used to perform acoustic sequence recognition on the acoustic features after feature extraction, so as to obtain the acoustic feature sequence of the training text.
  5. 如权利要求1所述的语音合成方法,其中,所述从所述第一声音频谱中提取语言学特征,得到第一语言学特征,包括:The speech synthesis method according to claim 1, wherein the extracting linguistic features from the first sound spectrum to obtain the first linguistic features comprises:
    获取所述第一声音频谱中的词语向量及句子向量,对所述词语向量及句子向量进行平滑处理;Acquiring word vectors and sentence vectors in the first sound spectrum, and smoothing the word vectors and sentence vectors;
    将平滑处理后的词语向量与句子向量进行发音时长采样,得到词语发音时长和句子发音时长;Sampling the pronunciation duration of the smoothed word vector and the sentence vector to obtain the pronunciation duration of the word and the pronunciation duration of the sentence;
    根据所述词语发音时长和句子发音时长,生成所述第一语言学特征。The first linguistic feature is generated according to the pronunciation duration of the word and the pronunciation duration of the sentence.
  6. 如权利要求1至5中任意一项所述的语音合成方法,其中,所述根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型,包括:The speech synthesis method according to any one of claims 1 to 5, wherein the pre-built non-autoregressive speech is performed according to the training text, the first sound spectrum and the first linguistic feature. The synthetic model is trained to obtain a trained non-autoregressive speech synthesis model, including:
    将所述训练文本输入所述非自回归语音合成模型中,以输出所述训练文本的第二声音频谱,并从所述第二声音频谱中提取语言学特征,得到第二语言学特征;Inputting the training text into the non-autoregressive speech synthesis model, to output the second sound spectrum of the training text, and extracting linguistic features from the second sound spectrum to obtain second linguistic features;
    根据所述第一声音频谱、第二声音频谱、第一语言学特征以及第二语言学特征,计算所述非自回归语音合成模型的训练损失;Calculate the training loss of the non-autoregressive speech synthesis model according to the first sound spectrum, the second sound spectrum, the first linguistic feature, and the second linguistic feature;
    若所述训练损失不满足预设条件时,调整所述非自回归模型的参数,并返回所述将所述训练文本输入所述非自回归语音合成模型中的步骤;If the training loss does not meet the preset condition, adjust the parameters of the non-autoregressive model, and return to the step of inputting the training text into the non-autoregressive speech synthesis model;
    若所述训练损失满足预设条件时,得到训练完成的非自回归语音合成模型。If the training loss satisfies the preset condition, a trained non-autoregressive speech synthesis model is obtained.
  7. 如权利要求6所述的语音合成方法,其中,所述根据所述第一声音频谱、第二声音频谱、第一语言学特征以及第二语言学特征,计算所述非自回归语音合成模型的训练损失,包括:The speech synthesis method according to claim 6, wherein calculating the non-autoregressive speech synthesis model according to the first sound spectrum, the second sound spectrum, the first linguistic feature and the second linguistic feature Training losses, including:
    根据所述第一声音频谱及第二声音频谱,计算所述非自回归语音合成模型的第一训练损失;calculating the first training loss of the non-autoregressive speech synthesis model according to the first sound spectrum and the second sound spectrum;
    根据所述第一语言学特征以及第二语言学特征,计算所述非自回归语音合成模型的第二训练损失;calculating a second training loss of the non-autoregressive speech synthesis model according to the first linguistic feature and the second linguistic feature;
    根据所述第一训练损失和第二训练损失,计算所述非自回归语音合成模型的训练损失。According to the first training loss and the second training loss, the training loss of the non-autoregressive speech synthesis model is calculated.
  8. 一种语音合成装置,其中,所述装置包括:A speech synthesis device, wherein the device comprises:
    获取模块,用于获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;an acquisition module, used for acquiring training text, and using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
    提取模块,用于从所述第一声音频谱中提取语言学特征,得到第一语言学特征;an extraction module for extracting linguistic features from the first sound spectrum to obtain first linguistic features;
    训练模块,用于根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;A training module for training a pre-built non-autoregressive speech synthesis model according to the training text, the first sound spectrum and the first linguistic feature, to obtain a trained non-autoregressive speech synthesis model;
    合成模块,用于利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。The synthesis module is configured to perform speech synthesis on the speech text to be synthesized by using the non-autoregressive speech synthesis model completed by the training to obtain a speech synthesis result.
  9. 一种电子设备,其中,所述电子设备包括:An electronic device, wherein the electronic device comprises:
    至少一个处理器;以及,at least one processor; and,
    与所述至少一个处理器通信连接的存储器;其中,a memory communicatively coupled to the at least one processor; wherein,
    所述存储器存储有可被所述至少一个处理器执行的计算机程序,所述计算机程序被所述至少一个处理器执行,以使所述至少一个处理器能够执行如下所述的语音合成方法:The memory stores a computer program executable by the at least one processor, the computer program being executed by the at least one processor to enable the at least one processor to perform a speech synthesis method as described below:
    获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;Acquiring training text, using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
    从所述第一声音频谱中提取语言学特征,得到第一语言学特征;Extracting linguistic features from the first sound spectrum to obtain first linguistic features;
    根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;According to the training text, the first sound spectrum and the first linguistic feature, the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model;
    利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。Using the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
  10. 如权利要求9所述的电子设备,其中,所述利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱,包括:The electronic device as claimed in claim 9, wherein, performing speech synthesis on the training text by using a pre-trained autoregressive speech synthesis model to obtain the first sound spectrum of the training text, comprising:
    利用所述自回归语音合成模型中的编码器对所述训练文本进行声学特征提取,得到所述训练文本的声学特征;Use the encoder in the autoregressive speech synthesis model to perform acoustic feature extraction on the training text to obtain the acoustic features of the training text;
    利用所述自回归语音合成模型中的解码器对所述声学特征进行序列识别,得到所述训练文本的声学特征序列;Use the decoder in the autoregressive speech synthesis model to perform sequence recognition on the acoustic features to obtain the acoustic feature sequence of the training text;
    利用所述自回归语音合成模型中的声码器对所述声学特征序列进行频谱转换,得到所述训练文本的第一声音频谱。The acoustic feature sequence is spectrally converted by the vocoder in the autoregressive speech synthesis model to obtain the first sound spectrum of the training text.
  11. 如权利要求9所述的电子设备,其中,所述利用所述自回归语音合成模型中的编码器对所述训练文本进行声学特征提取,得到所述训练文本的声学特征,包括:The electronic device according to claim 9, wherein, the use of the encoder in the autoregressive speech synthesis model to perform acoustic feature extraction on the training text to obtain the acoustic features of the training text, comprising:
    利用所述编码器中的卷积层对所述训练文本进行声学特征卷积,得到初始声学特征;Use the convolution layer in the encoder to perform acoustic feature convolution on the training text to obtain initial acoustic features;
    利用所述编码器中的门控循环单元对所述初始声学特征进行上采样,得到所述训练文本的声学特征。The initial acoustic features are up-sampled by the gated loop unit in the encoder to obtain the acoustic features of the training text.
  12. 如权利要求10所述的电子设备,其中,所述利用所述自回归语音合成模型中的解码器对所述声学特征进行序列识别,得到所述训练文本的声学特征序列,包括:The electronic device according to claim 10, wherein the sequence recognition of the acoustic features by using the decoder in the autoregressive speech synthesis model, to obtain the acoustic feature sequence of the training text, comprises:
    利用所述解码器中的卷积层对所述声学特征进行特征提取;Feature extraction is performed on the acoustic features by using a convolutional layer in the decoder;
    利用所述解码器中的长短期记忆网络单元对特征提取后的声学特征进行声学序列识别,得到所述训练文本的声学特征序列。The long-short-term memory network unit in the decoder is used to perform acoustic sequence recognition on the acoustic features after feature extraction, so as to obtain the acoustic feature sequence of the training text.
  13. 如权利要求9所述的电子设备,其中,所述从所述第一声音频谱中提取语言学特征,得到第一语言学特征,包括:The electronic device according to claim 9, wherein the extracting linguistic features from the first sound spectrum to obtain the first linguistic features comprises:
    获取所述第一声音频谱中的词语向量及句子向量,对所述词语向量及句子向量进行平滑处理;Acquiring word vectors and sentence vectors in the first sound spectrum, and smoothing the word vectors and sentence vectors;
    将平滑处理后的词语向量与句子向量进行发音时长采样,得到词语发音时长和句子发音时长;Sampling the pronunciation duration of the smoothed word vector and the sentence vector to obtain the pronunciation duration of the word and the pronunciation duration of the sentence;
    根据所述词语发音时长和句子发音时长,生成所述第一语言学特征。The first linguistic feature is generated according to the pronunciation duration of the word and the pronunciation duration of the sentence.
  14. 如权利要求9至13中任意一项所述的电子设备,其中,所述根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型,包括:The electronic device according to any one of claims 9 to 13, wherein the pre-built non-autoregressive speech synthesis is performed according to the training text, the first sound spectrum and the first linguistic feature The model is trained to obtain a trained non-autoregressive speech synthesis model, including:
    将所述训练文本输入所述非自回归语音合成模型中,以输出所述训练文本的第二声音频谱,并从所述第二声音频谱中提取语言学特征,得到第二语言学特征;Inputting the training text into the non-autoregressive speech synthesis model, to output the second sound spectrum of the training text, and extracting linguistic features from the second sound spectrum to obtain second linguistic features;
    根据所述第一声音频谱、第二声音频谱、第一语言学特征以及第二语言学特征,计算所述非自回归语音合成模型的训练损失;Calculate the training loss of the non-autoregressive speech synthesis model according to the first sound spectrum, the second sound spectrum, the first linguistic feature, and the second linguistic feature;
    若所述训练损失不满足预设条件时,调整所述非自回归模型的参数,并返回所述将所述训练文本输入所述非自回归语音合成模型中的步骤;If the training loss does not meet the preset condition, adjust the parameters of the non-autoregressive model, and return to the step of inputting the training text into the non-autoregressive speech synthesis model;
    若所述训练损失满足预设条件时,得到训练完成的非自回归语音合成模型。If the training loss satisfies the preset condition, a trained non-autoregressive speech synthesis model is obtained.
  15. 一种计算机可读存储介质,存储有计算机程序,其中,所述计算机程序被处理器执行时实现如下所述的语音合成方法:A computer-readable storage medium storing a computer program, wherein the computer program implements the following speech synthesis method when the computer program is executed by a processor:
    获取训练文本,利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱;Acquiring training text, using a pre-trained autoregressive speech synthesis model to perform speech synthesis on the training text to obtain the first sound spectrum of the training text;
    从所述第一声音频谱中提取语言学特征,得到第一语言学特征;Extracting linguistic features from the first sound spectrum to obtain first linguistic features;
    根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型;According to the training text, the first sound spectrum and the first linguistic feature, the pre-built non-autoregressive speech synthesis model is trained to obtain a trained non-autoregressive speech synthesis model;
    利用所述训练完成的非自回归语音合成模型对待合成语音文本进行语音合成,得到语音合成结果。Using the trained non-autoregressive speech synthesis model to perform speech synthesis on the speech text to be synthesized to obtain a speech synthesis result.
  16. 如权利要求15所述的计算机可读存储介质,其中,所述利用预先训练好的自回归语音合成模型对所述训练文本进行语音合成,得到所述训练文本的第一声音频谱,包括:The computer-readable storage medium according to claim 15 , wherein, performing speech synthesis on the training text by using a pre-trained autoregressive speech synthesis model to obtain the first sound spectrum of the training text, comprising:
    利用所述自回归语音合成模型中的编码器对所述训练文本进行声学特征提取,得到所述训练文本的声学特征;Use the encoder in the autoregressive speech synthesis model to perform acoustic feature extraction on the training text to obtain the acoustic features of the training text;
    利用所述自回归语音合成模型中的解码器对所述声学特征进行序列识别,得到所述训练文本的声学特征序列;Use the decoder in the autoregressive speech synthesis model to perform sequence recognition on the acoustic features to obtain the acoustic feature sequence of the training text;
    利用所述自回归语音合成模型中的声码器对所述声学特征序列进行频谱转换,得到所述训练文本的第一声音频谱。The acoustic feature sequence is spectrally converted by the vocoder in the autoregressive speech synthesis model to obtain the first sound spectrum of the training text.
  17. 如权利要求15所述的计算机可读存储介质,其中,所述利用所述自回归语音合成模型中的编码器对所述训练文本进行声学特征提取,得到所述训练文本的声学特征,包括:The computer-readable storage medium according to claim 15 , wherein, using the encoder in the autoregressive speech synthesis model to perform acoustic feature extraction on the training text to obtain the acoustic features of the training text, comprising:
    利用所述编码器中的卷积层对所述训练文本进行声学特征卷积,得到初始声学特征;Use the convolution layer in the encoder to perform acoustic feature convolution on the training text to obtain initial acoustic features;
    利用所述编码器中的门控循环单元对所述初始声学特征进行上采样,得到所述训练文本的声学特征。The initial acoustic features are up-sampled by the gated loop unit in the encoder to obtain the acoustic features of the training text.
  18. 如权利要求16所述的计算机可读存储介质,其中,所述利用所述自回归语音合成模型中的解码器对所述声学特征进行序列识别,得到所述训练文本的声学特征序列,包括:The computer-readable storage medium according to claim 16, wherein the sequence identification of the acoustic features by the decoder in the autoregressive speech synthesis model to obtain the acoustic feature sequence of the training text comprises:
    利用所述解码器中的卷积层对所述声学特征进行特征提取;Feature extraction is performed on the acoustic features by using a convolutional layer in the decoder;
    利用所述解码器中的长短期记忆网络单元对特征提取后的声学特征进行声学序列识 别,得到所述训练文本的声学特征序列。The long-short-term memory network unit in the decoder is used to perform acoustic sequence recognition on the acoustic features after feature extraction, so as to obtain the acoustic feature sequence of the training text.
  19. 如权利要求15所述的计算机可读存储介质,其中,所述从所述第一声音频谱中提取语言学特征,得到第一语言学特征,包括:The computer-readable storage medium of claim 15, wherein the extracting linguistic features from the first sound spectrum to obtain the first linguistic features comprises:
    获取所述第一声音频谱中的词语向量及句子向量,对所述词语向量及句子向量进行平滑处理;Acquiring word vectors and sentence vectors in the first sound spectrum, and smoothing the word vectors and sentence vectors;
    将平滑处理后的词语向量与句子向量进行发音时长采样,得到词语发音时长和句子发音时长;Sampling the pronunciation duration of the smoothed word vector and the sentence vector to obtain the pronunciation duration of the word and the pronunciation duration of the sentence;
    根据所述词语发音时长和句子发音时长,生成所述第一语言学特征。The first linguistic feature is generated according to the pronunciation duration of the word and the pronunciation duration of the sentence.
  20. 如权利要求15至19中任意一项所述的计算机可读存储介质,其中,所述根据所述训练文本、所述第一声音频谱以及所述第一语言学特征,对预先构建的非自回归语音合成模型进行训练,得到训练完成的非自回归语音合成模型,包括:The computer-readable storage medium according to any one of claims 15 to 19, wherein the pre-built non-self The regression speech synthesis model is trained, and the trained non-autoregressive speech synthesis model is obtained, including:
    将所述训练文本输入所述非自回归语音合成模型中,以输出所述训练文本的第二声音频谱,并从所述第二声音频谱中提取语言学特征,得到第二语言学特征;Inputting the training text into the non-autoregressive speech synthesis model, to output the second sound spectrum of the training text, and extracting linguistic features from the second sound spectrum to obtain second linguistic features;
    根据所述第一声音频谱、第二声音频谱、第一语言学特征以及第二语言学特征,计算所述非自回归语音合成模型的训练损失;Calculate the training loss of the non-autoregressive speech synthesis model according to the first sound spectrum, the second sound spectrum, the first linguistic feature, and the second linguistic feature;
    若所述训练损失不满足预设条件时,调整所述非自回归模型的参数,并返回所述将所述训练文本输入所述非自回归语音合成模型中的步骤;If the training loss does not meet the preset condition, adjust the parameters of the non-autoregressive model, and return to the step of inputting the training text into the non-autoregressive speech synthesis model;
    若所述训练损失满足预设条件时,得到训练完成的非自回归语音合成模型。If the training loss satisfies the preset condition, a trained non-autoregressive speech synthesis model is obtained.
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