CN103714823B - A kind of self adaptation subsurface communication method based on integrated voice coding - Google Patents
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Abstract
本发明涉及一种基于综合语音编码的自适应水下通信方法,水声通信系统信源端输入的语音由双路混合编码和判决器处理后经信道编码、调制、功率放大器和水声换能器发射出去;经水声信道传输后,信宿端通过水声换能器接收到信号后经滤波放大器放大、信道均衡、信道估计、解调、信道解码后送入双路混合译码;在进行语音编码/语音识别的水声通信之前,由水下工作者甲的信源端的判决器发送一组训练码,经信道到达水下工作者乙的信宿端;水下工作者乙的信宿端将获得的信道特性进行处理后,将结果转发给水下工作者乙的信源端,再经水声信道到达水下工作者甲的信宿端,水下工作者甲的判决器根据所收到的信道特性能数进行判决。本发明可实现与移动通信相当的通话质量。
The invention relates to an adaptive underwater communication method based on comprehensive speech coding. The speech input by the source end of the underwater acoustic communication system is processed by a two-way hybrid coding and decision device, and then undergoes channel coding, modulation, power amplifier and underwater acoustic transduction. After being transmitted through the underwater acoustic channel, the sink end receives the signal through the underwater acoustic transducer and sends it to the dual-channel hybrid decoding after being amplified by the filter amplifier, channel equalization, channel estimation, demodulation, and channel decoding; Before the underwater acoustic communication of speech coding/speech recognition, a group of training codes are sent by the decision device at the source end of underwater worker A, and arrive at the sink end of underwater worker B through the channel; the sink end of underwater worker B will After the obtained channel characteristics are processed, the result is forwarded to the source end of underwater worker B, and then to the sink end of underwater worker A through the underwater acoustic channel. Feature performance is judged. The invention can realize the communication quality equivalent to that of mobile communication.
Description
技术领域 technical field
本发明属于一种语音通信系统。特别涉及一种基于综合语音编码的自适应水下通信方法。 The invention belongs to a voice communication system. In particular, it relates to an adaptive underwater communication method based on integrated speech coding.
背景技术 Background technique
所有的水声通信方式一般分为模拟和数字两种,其中模拟通信方式多以单边带调幅调制方式为主,当信道属于严格带限时,通常采用模拟通信方式就可以达到语音通信的带宽要求,但是模拟通信方式的质量较差,语音清晰度时好时坏,不能适应于复杂而多变的恶劣信道。通常海洋信道具有高噪声背景、低载波频率、极为有限的带宽以及时间-空间-频率变化的特性,这些特性使得水声信道成为迄今最为恶劣的通信信道之一。国内外众多研究机构在水声通信的信道稳定性、调制方法上做了很多研究,但传统方法的水声通信很难突破稳定信道带宽和实时可靠通信的技术难题。为此,从信源角度考虑,对信源进行编码,大幅度地压缩语音信号的数据量,实现与水声信道所提供的速率相匹配的稳定可靠的语音通信系统。 All underwater acoustic communication methods are generally divided into two types: analog and digital. Among them, the analog communication method is mostly based on single-sideband amplitude modulation modulation. When the channel is strictly band-limited, the analog communication method can usually meet the bandwidth requirements of voice communication. , but the quality of the analog communication method is poor, the voice clarity is up and down, and it cannot adapt to complex and changeable bad channels. Generally, oceanic channels have the characteristics of high noise background, low carrier frequency, extremely limited bandwidth, and time-space-frequency variation, which make underwater acoustic channels one of the worst communication channels so far. Many research institutions at home and abroad have done a lot of research on channel stability and modulation methods of underwater acoustic communication, but it is difficult for traditional underwater acoustic communication to break through the technical problems of stable channel bandwidth and real-time reliable communication. Therefore, from the point of view of the information source, the information source is coded to greatly compress the data volume of the voice signal, so as to realize a stable and reliable voice communication system that matches the rate provided by the underwater acoustic channel.
在实际的水声通信中,部分海域只能以0.8kbps以下的速率进行通信,由于2.4kbps速率以下的语音编码速度过低,激励信息相对不足,使其语音质量与环境噪声的相关性大,不容易构建具有适用性较大的编码算法,目前语音编码低于2.4kbps速率的语音编码方案目前尚无国际标准。 In actual underwater acoustic communication, some sea areas can only communicate at a rate below 0.8kbps. Since the speech coding speed below 2.4kbps is too low and the excitation information is relatively insufficient, the correlation between speech quality and environmental noise is large. It is not easy to construct a coding algorithm with great applicability. Currently, there is no international standard for a speech coding scheme with a speech coding rate lower than 2.4kbps.
低速率语音编码器采用了比传统编码器更为复杂的方法对语音的激励源进行划分和重构,最早提出的是线性预测编码模型,在此基础上不断有新的编码算法提出。目前国内外主要有使用码本作为激励源的码激励线性预测编码技术、多带激励模型、混合激励线性预测语音编码技术和原形波形内插方法这四种主要的低速率编码算法,研究者们在此基础上展开了积极的探索和研究,但是低速率编解码后的语音信号,其语音的自然度和容忍环境噪声的鲁棒能力有很大的缺陷,无法较好地满足水声信道的通信要求。在众多的语音信源压缩编码技术中,语音识别无疑是压缩比最高的算法。 The low-rate speech coder uses a more complex method than the traditional coder to divide and reconstruct the speech excitation source. The linear predictive coding model was first proposed, and new coding algorithms have been continuously proposed on this basis. At present, there are mainly four main low-rate coding algorithms at home and abroad: code-excited linear predictive coding technology using codebook as the excitation source, multi-band excitation model, mixed-excitation linear predictive speech coding technology and prototype waveform interpolation method. On this basis, active exploration and research have been carried out, but the voice signal after low-rate codec has great defects in the naturalness of voice and the robustness of tolerance to environmental noise, and cannot better meet the needs of underwater acoustic channels. Communication requirements. Among the numerous speech source compression coding technologies, speech recognition is undoubtedly the algorithm with the highest compression ratio.
目前世界上关于水下语音通信技术研究的还不多,一般都是按照传统的无线网络通信的方式进行压缩编码、调制、传输以及解调解码的。目前查阅到有关于语音识别技术运用于水下通信系统的专利和文献2篇:中国专利申请号(200810064290.0)一种低码率的水下语音通信方法中应用了语音识别技术和语音的拼音合成技术,先通过语音识别系统将发送端的语音信号转换成汉语拼音或者汉语拼音对应的ASCII码,然后进行信道编码后发送出去,接收端进行解调和解码,再根据解调出来的拼音代码或者汉字发音代码,调用语音库相对应的拼音或者汉字发音信息。利用语音合成算法将发音合成为语音信号输出;文献“语音识别于合成技术在水声通信中的运用”,2003年通信理论与信号处理年会论文集,2008年3月,P434~439,程恩等,其思路是将语音信号转换成文本信号,利用现成的ViaVoice识别软件和语音合成软件,实现水下语音通信,选用的ViaVoice软件只能运用于通用计算机,发送的是文字代码,而语音合成方法未有说明,不适合将其运用于水下微小型载体和水下工作者之间的语音通话。 At present, there are not many researches on underwater voice communication technology in the world. Generally, compression coding, modulation, transmission, demodulation and decoding are performed according to the traditional wireless network communication method. At present, I have found 2 patents and documents about the application of speech recognition technology in underwater communication systems: Chinese Patent Application No. (200810064290.0), a low-bit-rate underwater speech communication method that uses speech recognition technology and speech pinyin synthesis Technology, first convert the voice signal at the sending end into Chinese Pinyin or the ASCII code corresponding to Chinese Pinyin through the speech recognition system, then perform channel coding and send it out, and the receiving end performs demodulation and decoding, and then according to the demodulated Pinyin code or Chinese characters Pronunciation code, calling the corresponding pinyin or Chinese character pronunciation information of the voice library. Use speech synthesis algorithm to synthesize pronunciation into speech signal output; Literature "Speech Recognition and Synthesis Technology Application in Underwater Acoustic Communication", Proceedings of the 2003 Communication Theory and Signal Processing Annual Conference, March 2008, P434~439, Cheng En et al., the idea is to convert voice signals into text signals, and use the ready-made ViaVoice recognition software and speech synthesis software to realize underwater voice communication. The synthesis method is not described, and it is not suitable for the voice communication between the underwater micro-carrier and the underwater workers.
发明内容 Contents of the invention
本发明的目的在于提供一种基于综合语音编码的自适应水下通信方法。 The purpose of the present invention is to provide an adaptive underwater communication method based on integrated speech coding.
在信道恶劣的水下环境中进行语音通信时,采用语音识别与合成技术,在发射端将语音信号转化为极低速率的语音文本数据通过水声信道进行传输,接收端将接收到的语音文本数据合成为语音,由于识别后的语音文本数据的数据率极低,可以利用低码率纠错编码技术来改善信道特性,达到稳定可靠的传输,这是实现水声语音低速率数字通信的一种新思路。由于水声环境下水下工作者通信设备硬件的限制,业界性能良好的语音识别系统都是采用隐马尔科夫模型(HMM)的连续语音识别建模技术框架,而在水下通信环境下,由于受到环境噪声,运算复杂度等方面的限制,直接使用传统的算法和模型远远满足不了实用的需求。 When conducting voice communication in the underwater environment with harsh channels, the voice recognition and synthesis technology is used to convert the voice signal into extremely low-speed voice text data at the transmitter for transmission through the underwater acoustic channel, and the voice text received at the receiver The data is synthesized into speech. Since the data rate of the recognized speech and text data is extremely low, low bit rate error correction coding technology can be used to improve channel characteristics and achieve stable and reliable transmission. A new way of thinking. Due to the limitations of the communication equipment hardware of underwater workers in the underwater acoustic environment, the speech recognition systems with good performance in the industry all use the Hidden Markov Model (HMM) continuous speech recognition modeling technology framework, but in the underwater communication environment, due to Restricted by environmental noise and computational complexity, direct use of traditional algorithms and models is far from meeting practical needs.
根据水声通信和海洋信道的特性,本发明利用语音识别/语音编码技术,将语音转化为带宽要求极低的语音识别数据和较低速率的语音编码数据,并根据信道特性自适应地选择语音识别数据/语音编码数据进行双路混合编码或语音识别数据单独编码两种方式进行传输,灵活实现适应于不同通信距离、不同通信方式、不同通信需求条件下的语音通信,数据通信的系统体系结构和方式。 According to the characteristics of underwater acoustic communication and ocean channel, the present invention utilizes speech recognition/speech coding technology to convert speech into speech recognition data with extremely low bandwidth requirements and speech coding data at a lower rate, and adaptively select speech according to channel characteristics Recognition data/speech coded data are transmitted in two ways mixed coding or voice recognition data is coded separately, flexibly realizing the system architecture of voice communication and data communication suitable for different communication distances, different communication methods, and different communication requirements and way.
本发明提出的基于综合语音编码的自适应水下通信方法,通过自适应水声通信系统实现,所述通信系统由双路混合编码1、判决器2、信道编码3、调制器4、功率放大器5、第一水声换能器6、水声信道7、第二水声换能器8、滤波放大器9、信道均衡10、信道估计11、解调器12、信道解码13和双路混合译码14组成,其中:双路混合编码1的输入端连接语音输入,输出端连接判决器2,判决器2的输出端连接信道编码3,信道编码3的输出端连接调制器4,调制器4的输出端连接功率放大器5,第一水声换能器6的输出端分别连接功率放大器5和水声信道7,水声信道7的输出端连接第二水声换能器8的输入端,第二水声换能器8的输出端连接滤波放大器9,滤波放大器9的输出端分别连接信道均衡10和信道估计11,信道估计11的输出端分别连接解调器12和信道均衡10,解调器12输出端连接信道解码13,信道解码13的输出端连接双路混合译码14,双路混合译码14进行语音输出;具体步骤如下: The adaptive underwater communication method based on integrated speech coding proposed by the present invention is realized through an adaptive underwater acoustic communication system. The communication system consists of a two-way hybrid code 1, a decision device 2, a channel code 3, a modulator 4, and a power amplifier. 5. First underwater acoustic transducer 6, underwater acoustic channel 7, second underwater acoustic transducer 8, filter amplifier 9, channel equalization 10, channel estimation 11, demodulator 12, channel decoding 13 and two-way hybrid translation code 14, wherein: the input end of the two-way mixed code 1 is connected to the voice input, the output end is connected to the decision device 2, the output end of the decision device 2 is connected to the channel code 3, the output end of the channel code 3 is connected to the modulator 4, and the modulator 4 The output end of the first underwater acoustic transducer 6 is connected to the power amplifier 5, the output end of the first underwater acoustic transducer 6 is respectively connected to the power amplifier 5 and the underwater acoustic channel 7, and the output end of the underwater acoustic channel 7 is connected to the input end of the second underwater acoustic transducer 8, The output end of the second underwater acoustic transducer 8 is connected to the filter amplifier 9, and the output end of the filter amplifier 9 is connected to the channel equalizer 10 and the channel estimation 11 respectively, and the output end of the channel estimation 11 is respectively connected to the demodulator 12 and the channel equalization 10, and the solution The output terminal of tuner 12 is connected to channel decoding 13, and the output end of channel decoding 13 is connected to two-way mixed decoding 14, and two-way mixed decoding 14 carries out voice output; concrete steps are as follows:
(1)水声通信系统信源端输入的语音由双路混合编码和判决器处理后经信道编码、调制、功率放大器和水声换能器发射出去; (1) The voice input from the source end of the underwater acoustic communication system is processed by a dual-channel hybrid coder and judger, and then transmitted through channel coding, modulation, power amplifier and underwater acoustic transducer;
(2)经水声信道传输后,信宿端通过水声换能器接收到信号后经滤波放大器放大、信道均衡、信道估计、解调、信道解码后送入双路混合译码; (2) After being transmitted through the underwater acoustic channel, the sink end receives the signal through the underwater acoustic transducer, and after the signal is amplified by the filter amplifier, channel equalization, channel estimation, demodulation, and channel decoding, it is sent to dual-channel hybrid decoding;
(3)在进行语音编码/语音识别的水声通信之前,由水下工作者甲的信源端的判决器发送一组训练码,经信道到达水下工作者乙的信宿端; (3) Before carrying out the underwater acoustic communication of speech coding/speech recognition, a group of training codes are sent by the decision device at the source end of the underwater worker A, and arrive at the sink end of the underwater worker B through the channel;
(4)水下工作者乙的信宿端将获得的信道特性进行处理后,将结果转发给水下工作者乙的信源端,再经水声信道到达水下工作者甲的信宿端,水下工作者甲的判决器根据所收到的信道特性能数进行判决。 (4) The sink end of underwater worker B processes the obtained channel characteristics, forwards the result to the source end of underwater worker B, and then reaches the sink end of underwater worker A through the underwater acoustic channel. The decider of worker A makes a decision according to the received channel characteristic performance.
本发明中,信源端采用语音识别和低速率语音编码两种方式来压缩信源;语音信号经预处理后分为两路,分别送入数据支路和语音支路。在数据支路中,语音信号特征提取后进行语音识别,输出语音识别数据信号。在语音支路中,语音信号参数提取后进行低速率语音编码,输出语音编码数据。语音识别数据和语音编码数据根据判决器对当前信道特性进行判断,根据判断结果选择语音识别数据/语音编码数据进行双路混合编码或语音识别数据单独编码方式传输。 In the present invention, the information source adopts voice recognition and low-rate voice coding to compress the information source; the voice signal is divided into two channels after preprocessing, and sent to the data branch and the voice branch respectively. In the data branch, speech recognition is performed after speech signal feature extraction, and speech recognition data signals are output. In the speech branch, after the speech signal parameters are extracted, low-rate speech coding is performed, and the speech coded data is output. The voice recognition data and the voice coding data are judged according to the characteristics of the current channel by the judging device, and the voice recognition data/speech coded data are selected according to the judgment result for two-way mixed coding or the voice recognition data is separately coded for transmission.
本发明中,信宿端双路同传系统模块接收端信道解码后的数据,判断接收到的编码方式,如果是语音识别数据/语音编码数据双路混合编码,以解码后的编码语音信号为主,语音识别数据合成语音信号为辅进行输出;如果是语音识别数据单独编码,则以语音识别数据合成后的语音信号进行输出。 In the present invention, the two-way simultaneous interpretation system module at the sink end receives the channel-decoded data at the end, and judges the received coding method. If it is two-way mixed coding of speech recognition data/speech coded data, the decoded coded speech signal is mainly used , the voice recognition data is synthesized into a voice signal as an auxiliary output; if the voice recognition data is encoded separately, the voice signal after the voice recognition data synthesis is used for output.
本发明拟以水下语音通信为应用目标,开展对基于语音编码/语音识别的水声通信体系研究。其基本构思是低速率语音编码与语音识别和合成技术相结合,在通信信道状况良好时提供语音编码通信,可实现与移动通信相当的通话质量;当通信信道状况差时,提供可适应更低数据率的语音识别与合成的语音通信(此时,语音的自然度、与说话人的语音相符程度要降低)。这样的通信架构可以保证在目前的水声通信技术的基础上,实现更远距离的通信。 The present invention intends to take underwater voice communication as the application target, and conduct research on the underwater acoustic communication system based on voice coding/voice recognition. Its basic idea is to combine low-rate speech coding with speech recognition and synthesis technology to provide speech coding communication when the communication channel is in good condition, which can achieve the same call quality as mobile communication; when the communication channel is in poor condition, it can provide lower adaptability Data rate speech recognition and synthetic speech communication (at this time, the naturalness of the speech and the degree of conformity with the speaker's speech will be reduced). Such a communication architecture can ensure longer-distance communication based on the current underwater acoustic communication technology.
附图说明 Description of drawings
图1是本发明所采用的语音编码/语音识别双路同传的自适应水声通信系统总体框图。 Fig. 1 is the overall block diagram of the self-adaptive underwater acoustic communication system of two-way simultaneous interpretation of speech coding/speech recognition adopted in the present invention.
图2是本发明信源端双路同传系统模块框图。 Fig. 2 is a module block diagram of the two-channel simultaneous interpretation system at the source end of the present invention.
图3是本发明信宿端双路同传系统模块框图。 Fig. 3 is a module block diagram of the dual-channel simultaneous interpretation system at the sink end of the present invention.
图中标号:1为双路混合编码,101为预处理模块,102为特征提取,103为语音识别模块,104为数据编码模块,105为参数提取模块,106为语音编码模块,2为判决器,3为信道编码,4为调制器,5为功率放大器,6为第一水声换能器,7为水声信道,8为第二水声换能器,9为滤波放大器,10为信道均衡器,11为信道估计器,12为解调器,13为信道解码,14为双路混合译码,1401为编码方式判决器,1402为数据解码模块,1403为语言合成模块,1404为语音解码模块。 In the figure: 1 is a two-way mixed coding, 101 is a preprocessing module, 102 is a feature extraction, 103 is a speech recognition module, 104 is a data coding module, 105 is a parameter extraction module, 106 is a speech coding module, and 2 is a decision device , 3 is the channel coding, 4 is the modulator, 5 is the power amplifier, 6 is the first underwater acoustic transducer, 7 is the underwater acoustic channel, 8 is the second underwater acoustic transducer, 9 is the filter amplifier, 10 is the channel Equalizer, 11 is a channel estimator, 12 is a demodulator, 13 is a channel decoding, 14 is a two-way hybrid decoding, 1401 is a coding mode decision device, 1402 is a data decoding module, 1403 is a language synthesis module, 1404 is a voice decoding module.
具体实施方式 detailed description
下面通过附图进一步说明本发明。 Further illustrate the present invention below by accompanying drawing.
实施例1:首先在进行语音编码/语音识别的水声通信之前,由水下工作者甲的信源端的判决器发送一组训练码,经信道到达水下工作者乙的信宿端。水下工作者乙信宿端将获得的信道特性进行处理后,将结果转发给水下工作者乙的信源端,再经信道到达水下工作者甲的信宿端,水下工作者甲的判决器2根据所收到的信道特性能数进行判决。 Embodiment 1: First, before the underwater acoustic communication of speech coding/speech recognition, a group of training codes are sent by the decision device at the source end of underwater worker A to the sink end of underwater worker B through the channel. The sink end of underwater worker B processes the channel characteristics obtained, and forwards the result to the source end of underwater worker B, and then reaches the sink end of underwater worker A through the channel, and the decision device of underwater worker A 2 Make a decision based on the received channel characteristic performance.
附图1中水声通信系统信源端输入的语音由信源端双路同传系统模块(双路混合编码1和判决器2)处理后经信道编码3、调制4、功率放大器5和水声换能器6发射出去,经水声信道7传输后,信宿端通过水声换能器8接收到信号后经滤波放大9、信道均衡10、信道估计11、解调12、信道解码13后送信宿端双路同传系统模块(双路混合译码14)。 The voice input from the source end of the underwater acoustic communication system in Figure 1 is processed by the dual-channel simultaneous interpretation system module (dual-channel hybrid code 1 and decision device 2) at the source end, and then channel coded 3, modulated 4, power amplifier 5 and water The acoustic transducer 6 emits and transmits through the underwater acoustic channel 7. After receiving the signal through the underwater acoustic transducer 8, the sink end undergoes filtering and amplification 9, channel equalization 10, channel estimation 11, demodulation 12, and channel decoding 13. The dual-channel simultaneous interpretation system module at the sink end (two-channel mixed decoding 14).
附图2中信源端采用语音识别和低速率语音编码两种方式来压缩信源;语音信号经预处理101后分为两路,分别送入数据支路和语音支路。在数据支路中,语音信号特征提取105后进行语音识别106,输出语音识别数据信号。在语音支路中,语音信号参数提取102后进行语音合成103,然后对合成后的语音进行低速率语音编码104,输出语音编码数据。语音识别数据和语音编码数据根据判决器2对当前信道特性进行判断,根据判断结果选择语音识别数据/语音编码数据进行双路混合编码或语音识别数据单独编码方式传输。 In Fig. 2, the information source adopts speech recognition and low-rate speech coding to compress the information source; the speech signal is divided into two paths after preprocessing 101, and sent to the data branch and the speech branch respectively. In the data branch, speech signal feature extraction 105 is followed by speech recognition 106, and a speech recognition data signal is output. In the speech branch, after speech signal parameter extraction 102, speech synthesis 103 is performed, and then the synthesized speech is subjected to low-rate speech coding 104, and speech coded data is output. The speech recognition data and the speech coding data are judged according to the current channel characteristics by the judging unit 2, and the speech recognition data/speech coding data are selected according to the judgment result for two-way mixed coding or the speech recognition data is separately coded for transmission.
附图3中信宿端双路同传系统模块接收端信道解码后的数据,通过接收到的数据进行编码方式判决1401,如果是语音识别数据/语音编码数据双路混合编码,语音信号以语音解码1404后为主,数据解码1402后合成的语音信号1403为辅进行输出;如果是语音识别数据单独编码,语音信号则以数据解码1402后进行语音合成1403后的信号进行输出。 In the accompanying drawing 3, the receiver-side two-way simultaneous interpretation system module receives the channel-decoded data, and performs a coding mode decision 1401 based on the received data. If it is two-way mixed coding of speech recognition data/speech coding data, the speech signal is decoded by speech After 1404, it is main, and the voice signal 1403 synthesized after data decoding 1402 is output as an auxiliary; if the voice recognition data is encoded separately, the voice signal is output with the signal after data decoding 1402 and voice synthesis 1403.
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