Discrimination of synthetic vowels by using tactile vocoder and acomparison to that of an eight-channel cochlear implant

A vowel discrimination test using a tactual vocoder was administered and the results were compared to that of an eight-channel cochlear implant. Both the tactile vocoder and the cochlear implant divided the speech signals into 16 frequency components using band-pass filters and lateral inhibition circuits. In the tactile vocoder, these 16 components were converted into a vibration with 200 Hz frequency and applied to a 3 x 16 element vibrator array using bimorph piezoelectric elements. The vibratory patterns were sensed on the fingertip. In the cochlear implant, the 16 components were reduced to eight current stimulation signals, consisting of biphasic pulses with 200 Hz frequency, which were applied to an eight-channel electrode array implanted in the scala tympani. The electrode array passed through the round window into the scala tympani to a depth of 23 mm. These psychophysical experiments investigate the ability of human subjects to discriminate synthetic vowels as a function of the number of channels employed. The results suggested that an eight-channel and a 16-channel tactile vocoder provided essentially the same discrimination scores. However, the ability to discriminate synthetic vowels decreased rapidly when less than eight channels were employed. The ability of an eight-channel tactile vocoder is expected to be better than that of the eight-channel cochlear implant because it is supposed that vowel discrimination is degraded by a phenomenon known as "current spreading" in the case of cochlear stimulation. However, the comparison between the two devices was not done on the cochlear implant subject.     

Estimation of Frequency for AM/FM Models Using the Phase Vocoder Framework

This paper proposes an extension of the applicability of phase-vocoder-based frequency estimators for generalized sinusoidal models, which include phase and amplitude modulations. A first approach, called phase corrected vocoder (PCV), takes into account the modification of the Fourier phases resulting from these modulations. Another approach is based on an adaptation of the principles of the time-frequency reassignment and is referred to as the reassigned vocoder (RV). The robustness of the estimation against noise is studied, both theoretically and experimentally, and the performance is assessed in comparison with two state-of-the-art algorithms: an unmodified version of the reassignment method and a quadratically interpolated fast Fourier transform method (QIFFT).     

语音增强技术在MELP声码器中的应用

为了使得MELP声码器在高噪声环境下仍然获得较好的语音效果,需对含噪声语音进行语音增强。本文采用谱减法和独立分量分析相结合方法,对语音进行增强。该方法可以在不增加语音采样硬件的条件下,满足独立分量分析中观测信号的数目不少于源信号数目的约束条件。结果表明,该方法能较好的分离出噪声和语音信号,增强输入到MELP声码器中的语音信号,提高MELP声码器在高噪声环境下应用的语音效果。     

Implementation of a Two-Channel LPC Vocoder

In this paper we describe hardware implementation of a 2400 bit/s two-channel linear predictive vocoder. The vocoder hardware uses the latest 2900 series "bit slice" microprocessor chips, and 2K RAM's and 1K ROM's of data memory. The system design is a two bus structure with a 208 ns cycle time. A significant feature of the vocoder is that it is capable of processing two voice channels simultaneously. Throughout the paper, emphasis is placed on details of firmware development of the vocoder system. Efficient design of the vocoder hardware is also discussed.     

改进型LPC声码器语音编码算法

本文给出了一种改进的ＬＰＣ语音编码算法，用于实现低速率声码器。与传统ＬＰＣ声码器算法相比，本算法在参数提取及合成等方面采取了一些改进措施，使得合成语音质量有很大的提高。本文在引言后概述了编码算法改进的考虑，然后给出编译码器的算法，重点讨论了本文提出的用动态规划法进行基音提取和平滑的新算法，以及合成端混合激励算法。本算法已经用ＴＭＳ３２０Ｃ２５实现单片编解码。     

极低码率语音编码—汉语识别声码器

本文给出了一个极低码率的６０ｂ／ｓ的主意编码系统－汉语识别声码器，以３２句话共２６７个音节作实验，其音节识别率平均为７４．１４％，句子平均可懂度为９１．９％，介绍了其系统结构，给出了实验结果。     

Voice Coding and Tree Encoding Speech Compression Systems Based Upon Inverse Filter Matching

Two speech compression systems based on codebooks of inverse filters produced by off-line linear predictive coding (LPC) and vector quantization (VQ) techniques are considered. The first system is a pitch excited vocoder that is a variation on a speech coding system based upon vector quantization. The encoder selects an LPC reverse filter from a finite codebook that best "matches" an observed frame of sampled speech. This filter is in turn used to determine the voicing and digitized pitch information. Unlike LPC systems, the digitization is performed in a single step on the data rather than separate modeling and digitization steps. The second system is a tree encoding system that uses the filter selected by an inverse filter matching vocoder to "color" a tree that is then searched for a minimum distortion path for the original sampled speech waveform. This system can be viewed as a hybrid between an adaptive predictive coder and a universal tree encoder. The two systems are described, simulated, and compared with other similar systems.     

ICA去除EEG中眼动伪差和工频干扰方法研究

眼动伪差和工频干扰是临床脑电图(EEG)中常见噪声,严重影响其有用信息提取.本文尝试采用独立分量分析(Independent Component Analysis,ICA)方法分离EEG中此类噪声.通过对早老性痴呆症(Alzheimer disease,AD)患者临床EEG信号(含眼动伪差和混入工频干扰,信噪比仅0dB)作ICA分析,比较了最大熵(Infomax)和扩展最大熵(Extended Infomax)ICA算法的分离效果,证实虽然最大熵算法可以分离出眼动慢波,但难以消除工频干扰,为此需采用扩展的最大熵算法;并知ICA方法在极低信噪比时也有较好的抗干扰性,且在处理非平稳信号时有好的鲁棒性;文中还结合近似熵(approximate entropy,ApEn)分析说明利用ICA去除干扰后有助于恢复和保持原始EEG信号的非线性特征.研究结果表明ICA方法在生物医学信号处理中具有潜在的重要应用价值,值得深入研究和推广.     

语音信号的线性预测编码技术

本文以一个双声道声码器为例,来说明线性预测编码技术在处理语音信号中的应用.一个2400b/s,双声道线性预测编码的声码器,使用高速、可编程序的位片型微处理机,字长可扩充.并使用2K的RAM和1K的ROM的数据存储器.该系统设计成循环时间为208毫微秒的双总线结构.该声码器能同时处理两个声道,用微处理机和线性预测编码技术开发出该系统,并叙述使用该技术设计声码器的要领.     

A Comparison of Digital Algorithms Used in Computing the Derivative of Left Ventricular Pressure

Five commonly used algorithms for digital differentiation are evaluated to determine how they perform in the presence of 8, 12, and 16 bit quantization noise. The algorithms are compared on the basis of rms error between a model derivative of the left ventricular pressure waveform and the approximate results of each algorithm. Algorithms based on interpolating techniques introduced the least amount of error when 16 bit data were used while algorithms based on least-squares data fit methods performed best on the less accurate 8 bit data. Some of the band-limiting characteristics of the algorithms are also discussed.     

A Novel Voice Coder At 4800BPS(HSEV)

ＡＮｏｖｅｌＶｏｉｃｅＣｏｄｅｒＡｔ４８００ＢＰＳ（ＨＳＥＶ）ＷａｎｇＸｉａｏｆｅｎｇＡＮＤＺｈａｏＥｒｙｕａｎ（ＤｅｐａｒｔｍｅｎｔｏｆＴｅｌｅｃｏｍｒｎｕｎｉｃａｔｉｏｎＥｎｇｉｎｅｅｒｉｎｇ，ＢｅｉｊｉｎｇＵｎｉｖｅｒｓｉｔｙｏｆＰｏｓｔｓ＆Ｔ...     

AMBE-2000TM在双路语音编解码器中的应用

AMBE-2000具有话音质量高、编码速率可变、配置灵活等特点,在卫星通信、安全通信、数字移动无线电领域得到了广泛应用。根据关键管脚功能以及时序要求使用4片ABME-2000设计实现了双路话音编解码器,每个AMBE-2000只完成编码或解码功能。为了保证话音质量,采用CD74HC4046A进行编解码时钟的同步设计。设计的双路话音编解码模块经试验验证,话音清晰,满足系统要求。     

AMR声码器研究及其基于C6000的多通道实现

对WCDMA的语音编码标准－AMR声码器中的算法进行了分析和研究，重点探讨了各个算法的特点及其理论依据，并实现了基于TMS320C6000系列芯片的多通道AMR声码器。仿真结果与3GPP提供的结果满足比特精确要求；实时处理的非正式主观测试表明，合成语音质量优于GSM的RPE－LTP的语音质量。     

A SINGLE PROCESSOR MULTI-RATE VOCODER

This paper presents the design of a full-duplex multi-rate vocoder which implements an LPC-10, CELPC and VSELPC algorithms in real time. A single commercially available digital signal processor IC, the TMS320C25, is used to perform the digital processing. The channel interfaces are configured with the design of ASIC, and including timing and control logic circuits.     

基于FPGA的ACELP声码器的设计与实现

根据ACELP语音编码算法的原理,设计出一种新的基于FPGA实现ACELP语音编码算法的声码器系统.对比传统的DSP实现方法,该声码器系统可以保证良好的语音质量和较高的实时性,并且价格低廉,功耗小.     

Channel and source considerations of a bit-rate reduction technique for a possible wireless communications system's performance enhancement

In wireless commercial and military communications systems, where bandwidth is at a premium, robust low-bit-rate speech coders are essential. They operate at fix bit rates and those bit rates cannot be altered without major modifications in the vocoder design. A novel approach to vocoders, in order to reduce the bit rate required to transmit speech signal, is proposed. While traditional low-bit-rate vocoders code original input speech, the proposed procedure operates on the time-scale modified signal. The proposed method offers any bit rate from 2400 b/s to downwards without modifying the principle vocoder structure, which is the new NATO standard, Stanag 4591, Mixed Excitation Linear Prediction (MELP) vocoder. We consider the application of transmitting MELP-encoded speech over noisy communication channels by applying different modulation techniques, after time-scale compression is applied. Three different time-scale modification algorithms have been evaluated and waveform similarity overlap and add (WSOLA) algorithm has been selected for time-scale modification purposes. Computer simulation results, both source and channel, are presented in terms of objective speech quality metrics and informal subjective listening tests. Design parameters such as codec complexity and delay are also investigated. Simulation results lead to a possible wireless communications system, whose performance might be enhanced by using the spared bits offered by the procedure.     

虚拟传感器在有源头靠系统的应用

主要对有源头靠系统进行了仿真研究,介绍了该系统的基本构成和技术特点。采用内模型控制（IMC）系统达到了有源头靠局部消声的目的。在内模型控制的基础上,为了将在误差传感器处形成的有源静区转移到入耳附近．介绍了2种基于虚拟传感器技术（VMC）的算法。仿真结果表明,2种基于虚拟传感器技术的算法对有源头靠系统都是有效的,且对于同样的窄带噪声,后一种算法较前一种算法能获得更好的消声效果。     

线性预测语音数字的编码技术

以一个双声道声码为例，说明一性预测编码技术在处理语音信号中的应用，一个２４００ｂｉｔ／ｓ双声道线性预测编码的声码器使用高速、可编程序的片型微处理机、字长可扩充。并使用２Ｋ的ＲＡＭ和１Ｋ的ＲＯＭ的数存储器。该系统设计成物理 ２０８毫微秒的双一结构。该声码器能同时处理两个声道，用微处理线性预测编码技术开发出该系统，并叙述使用该技术设计怕码器的要邻。     

Implementation of a Multirate Speech Digitizer

In this paper, implementation of a compact and efficient multirate speech digitizer with variable transmission rates of 2.4, 4.8, 9.6, and 14.96 kbits/s is presented. The multirate algorithm has been made based on the residual-excited linear prediction (RELP) vocoder with a transmission rate of 9.6 kbits/s. The residual encoder employed in the RELP vocoder uses hybrid companding delta modulation (HCDM). This HCDM is also used as a 14.96 kbit/s coder. If the residual in the RELP system is down-sampled before encoding, a 4.8 kbit/s coder can be realized. If the residual encoder is not used, a 2.4 kbit/s linear predictive coder (LPC) can be realized by incorporating a pitch extractor. In the 4.8 and 9.6 kbit/s coders the pitch-implanted residual excitation method has been used to generate the excitation signal to the synthesis filter. The multirate speech digitizer algorithm has been implemented using 2900 series bit-slice microprocessors. The external memory is composed of 2K RAM's and 2K ROM's. The system design is a two-bus structure with a 204 ns cycle time. With efficient hardware and software design, the multirate speech digitizer requires almost the same hardware complexity as compared with the conventional 2.4 kblt/s LPC vocoder.     

一种改进的2.4kb/s混合激励线性预测声码器方案

本文针对标准的2.4kb/s MELP声码器的不足之处提出了两项改进措施,一是提出了一种新的参数"能量-微分过零率比",用来对语音的过渡段和弱能量浊音段的清浊音判决进行调整;二是对线谱对的多级矢量量化(MSVQ)提出了一种多径搜索算法.实验和主观听觉测试表明,在同样2.4kb/s的码率下,改进MELP声码器的合成语音在可懂度和自然度方面都有一定的提高.