共查询到20条相似文献,搜索用时 734 毫秒
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基音检测是河南方言语音信号处理中的一个重要环节,针对低信噪比环境下的河南方言语音基音检测准确率低的问题,提出了一种语音信号增强和基音检测相结合的算法.通过多窗谱估计的改进谱减法对语音信号进行降噪处理,对增强后的语音信号用中心削波法消除偏离基音轨迹的野点,再通过自相关法实现基音检测.仿真结果表明,对于低信噪比环境下河南方言语音信号的基音估值检测结果准确,估算出的基音频率和实际基音频率能很好的重合. 相似文献
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本文提出了一种新的语音信号的基音周期检测方法,该方法根据语音信号的三阶累积量去确定语音信号的基音周期,能有效地排除白色或有色的高斯加性噪声所带来的干扰.与传统的基音周期估计的自相关函数法或平均幅度差函数法(AMDF)相比,该方法更精确、有效,具有更强的鲁棒性. 相似文献
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为了实现由男声变换到女声,在语音信号参数分析过程采用短时自相关法提取语音信号的基音周期,同时用LPC倒谱分析法分析共振峰的范围,通过matlab编写程序修改语音参数并接近于女声的范围,构置GUI界面。在实验中,输入一段语音信号,输出时即实现了由男声到女声的变换效果。因此对于语音信号参数的修改能够实现男女声音之间的变换。 相似文献
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基音周期是语音信号最重要的参数之一。MBE模型的基音周期搜索算法存在着运算量大,抗噪性能一般等缺点。本文基于小波变换,对算法进行了改进。阐述了小波变换基音周期估计原理,基音估计算法实现和实验。分析和实验表明改进算法更具实用性。 相似文献
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《Solid-State Circuits, IEEE Journal of》1983,18(1):10-24
Time domain harmonic scaling (TDHS) has been realized in real time on the Bell Laboratories digital signal processing (DSP) integrated circuit. It is an algorithm that can expand or compress the bandwidth and sampling rate of speech by taking advantage of the pitch structure in the speech signal. As such it is useful in a variety of speech applications including speech coding, speech enhancement, and rate modification. A single DSP can perform compression and a second DSP can perform expansion. Both operations require pitch information to be supplied with the input speech. Included in the system is a real-time pitch/periodicity detector which has also been implemented on a single DSP. Its design is based on a novel modification of the autocorrelation function type pitch detector. This paper presents details of both the TDHS and pitch detector implementation and discusses their performances. In particular in this paper we discuss a 2:1 compression and expansion system that has been used as part of a 9.6 kbit/s speech coder. TDHS was previously thought to require a much larger buffer than the RAM memory available in the DSP. We show that for all the compression/expansion ratios of interest the buffer size needed is twice the maximum pitch period. 相似文献
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An integrated system for multichannel neuronal recording with spike/LFP separation, integrated A/D conversion and threshold detection 总被引:1,自引:0,他引:1
A mixed-signal front-end processor for multichannel neuronal recording is described. It receives 12 differential-input channels of implanted recording electrodes. A programmable cutoff High Pass Filter (HPF) blocks dc and low-frequency input drift at about 1 Hz. The signals are band-split at about 200 Hz to low-frequency Local Field Potential (LFP) and high-frequency spike data (SPK), which is band limited by a programmable-cutoff LPF, in a range of 8-13 kHz. Amplifier offsets are compensated by 5-bit calibration digital-to-analog converters (DACs). The SPK and LFP channels provide variable amplification rates of up to 5000 and 500, respectively. The analog signals are converted into 10-bit digital form, and streamed out over a serial digital bus at up to 8 Mbps. A threshold filter suppresses inactive portions of the signal and emits only spike segments of programmable length. A prototype has been fabricated on a 0.35-microm CMOS process and tested successfully, demonstrating a 3-microV noise level. Special interface system incorporating an embedded CPU core in a programmable logic device accompanied by real-time software has been developed to allow connectivity to a computer host. 相似文献
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The authors describe an integrated speech feature extraction method consisting of: (1) a pitch detector; (2) a voicing decision to correctly partition speech into voiced and unvoiced intervals; (3) a confidence measure which reflects the probabilistic accuracy of the voicing decision; (4) a confidence measure which reflects the expected deviation of the pitch estimate from the true pitch and the probabilistic accuracy of this deviation; and (5) smoothing techniques for the pitch detector, the voicing decision, and the two confidence measures. The focus of their research is on voiced and unvoiced speech corrupted by high levels of white noise. The voicing decision and the confidence measures are developed by observing the behavior of three features derived from the autocorrelation function and experimentally fitting curves to the data. This integrated set of algorithms is statistically analyzed for speech at seven signal-to-noise ratios 相似文献
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In speech processing an estimation of the speech pitch period is important. Real time pitch detection is only possible by the selection of an efficient algorithm suitable for implementation on a programmable processor or in special-purpose hardware. The use of the periodogram algorithm (p.a.) is proposed to detect the pitch period of voiced speech. This algorithm is attractive for the following reasons: (a) it has no multiply operation; (b) when implemented on a 16-bit computer (e.g. microprocessor) the computation can be done in integer arithmetic without exceeding the microprocessor's dynamic range; (c) it is a simple technique for estimating the pitch period with reasonable accuracy. Results of the analysis of speech signals and sinusoids using the periodogram algorithm are presented and comparisons are made with the average magnitude difference function (a.m.d.f.) which is an alternative method of estimating the pitch period of the voiced speech. 相似文献
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本文研究了时频表示的实时实现算法及其算法优化问题,从广义局域自相关函数的物理意义出发,论证了广义时频表示的实时计算采用短时处理技术的可行性,并从两处着眼优化了算法,一得利用核函数共轭对称性,使时频表示的计算复杂度降低一半多;二是在计算解析信号过程中,给出了以任意帧移间隔的时域递推算法,进一步减少了实时算法的计算量。 相似文献
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《Solid-State Circuits, IEEE Journal of》1983,18(1):4-9
A very small, flexible, high-quality, full-duplex 2.4-kbit/s linear predictive vocoder has been implemented with commercially available integrated circuits. This fully digital realization is based on a distributed signal processing architecture employing three Nippon Electric Company (NEC) µPD7720 signal processing interface (SPI) single-chip microcomputers. One SPI implements the LPC analyzer, a second implements the Gold pitch and voicing decision algorithm, white the third µPD7720 implements the excitation generator and synthesizer. An Intel 8085-based 8-bit microcomputer is used for data transfer, control and multiplexing functions, and communications with the host terminal. The LPC chip set achieves high flexibility by accepting run time initialization options from the Intel 8085. These parameters include choice of linear predictive model (<= 15), analysis and synthesis frame size, and speech sampling frequency, as well as choice of speech input and output coding formats (linear or µ-255 law) and choice of analog or digjtal pre- and deemphasis. A total of 16 integrated circuits is used in the LPC vocoder with a power disipation of 5.5 W and occupying 18 in/sup 2/ of circuit area. 相似文献
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本文介绍的语音检测器以DSP芯片TMS320VC5402为核心,对短波电台接收到的信号进行分析和处理。数字语音信号采用串行输入/输出方式,语音检测算法则采用对语音信号进行降噪处理后,再进行短时平均幅度差和短时能量计算的方法。该语音检测器的电路简洁小巧,语音检测准确度高。 相似文献