DEUTERIUM INFLUX PROFILE ALONG THE APEX OF THE UPPER X-POINT TILES IN JET

ＤＥＵＴＥＲＩＵＭＩＮＦＬＵＸＰＲＯＦＩＬＥＡＬＯＮＧＴＨＥＡＰＥＸＯＦＴＨＥＵＰＰＥＲＸ－ＰＯＩＮＴＴＩＬＥＳＩＮＪＥＴ￥Ｙ．Ｋ．Ｚｈｕ（ＳｏｕｔｈｗｅｓｔｅｒｎＩｎｓｔｉｔｕｔｅｏｆＰｈｙｓｉｃｓ，Ｐ．Ｏ．Ｂｏｘ４３２，Ｃｈｅｎｇｄｕ６１００４１...     

KH_2PO_4和（CH_3NHCH_2COOH）_3CaCl_2在高压下的状

在活塞圆筒式Ｐ～Ｖ关系测量装置上，研究了ＫＨ＿２ＰＯ＿４（ＫＤＰ）和（ＣＨ＿３ＮＨＣＨ＿２ＣＯ－ＯＨ）＿３ＣａＣＩ＿２［Ｔｒｉｓ－ｓａｒｃｏｓｉｎｅｃａｌｃｉｕｍｃｈｌｏｒｉｄｅ（ＴＳＣＣ）］在室温下、４．５ＧＰａ内的ｐ～Ｖ关系。实验结果表明：ＫＤＰ在２．１ＧＰａ左右有一个相变；ＴＳＣＣ在０．８ＧＰａ和３．２ＧＰａ左右各有一个相变。本工作还给出了它们在相变前后的状态方程，以及它们的格临爱森参数γＯ、体积模量Ｂ＿ｏ和Ｂ＿ｏ的压力导数Ｂ＿ｏ。     

THERMAL INSTABILITY OF D-T BURNING PLASMAS

ＴＨＥＲＭＡＬＩＮＳＴＡＢＩＬＩＴＹＯＦＤ－ＴＢＵＲＮＩＮＧＰＬＡＳＭＡＳ￥Ｂ．Ｒ．ＳＨＩＹ．Ｘ．ＬＯＮＧＨ．Ｇ．ＺＨＡＯ（ＳｏｕｔｈｗｅｓｔｅｒｎｉｎｓｔｉｔｕｔｅｏｆＰｈｙｓｉｃｓ，Ｐ．Ｏ．Ｂｏｘ４３２，Ｃｈｅｎｇｄｕ６１００４１，Ｓｉｃｈｕａｎ...     

MPEG音频编码的高效算法研究及单片DSP实现

ＩＳＯ标准推荐ＭＰＥＧ－１音频编码算法要实现起来运算复杂,为便于单片ＤＳＰ实现,算法可以优化。本文简要分析了该算法的各个模块并估计了它们的运算复杂性,然后对子带滤波和心理声学模型作了进一步的探讨,提出了基于逆余弦变换（ＩＤＣＴ）的子带滤波的高效实现算法和不用ＦＦＴ或只用部分ＦＦＴ的快速心理声学掩蔽计算模型。最后用单片ＴＭＳ３２０Ｃ３１＠４０ＭＨｚ实现了ＭＰＥＧ－１层２编码算法。     

PUSHING THE ENVELOPE IN RAMAN SPECTROSCOPY

ＰＵＳＨＩＮＧＴＨＥＥＮＶＥＬＯＰＥＩＮＲＡＭＡＮＳＰＥＣＴＲＯＳＣＯＰＹＧ．Ｄ．Ｐｉｔｔ（ＵｎｉｖｅｒｓｉｔｙｏｆＬｅｅｄｓａｎｄＲｅｎｉｓｈａｗｐｌｃＵ．Ｋ）ＮｅｗｔｅｃｈｎｉｑｕｅｓｈａｖｅｂｅｅｎｄｅｖｅｌｏｐｅｄｆｏｒＵＶＲａｍａｎｍｉｃｒ...     

NaS和Na_2S分子解离能和生成热的从头算研究

使用分于轨道从头算方法，在ＭＰ２（ｆｕ）／６—３１Ｇ优化几何基础上，采用ＨＦ，ＭＰｎ（ｎ＝２，３，４）和ＱＣＩＳＤ（Ｔ）理论，使用６—３１Ｇ（ｄ，Ｐ），６—３１１Ｇ（ｄ，Ｐ），６—３１Ｇ（Ｚｄｆ，Ｐ）和６—３１１＋Ｇ（３ｄｆ，２ｐ）基组，计算了ＮａＳ和Ｎａ２Ｓ分子以及Ｎａ和Ｓ原子的总能量和各级电子关联修正能。在此基础上应用Ｇ１和Ｇ２理论估算ＴＮａＳ和Ｎａ２Ｓ分子的解离能和生成热。在Ｇ２理论框架下，理论估算的解离能和生成热（在室温）分别是６７．０７ｋｃａｌｍｏｌ－１和０．３５ｋｃａｌｍｏｌ－１（Ｎａ２Ｓ）以及５０．４８ｋｃａｌｍｏｌ－１和４１．５７ｋｃａｌｍｏｌ－１（ＮａＳ）。     

非晶PET纤维的分子链段取向分布的研究

用对称透射法,收集在室温下不同拉伸比的非晶聚酯（ＰＥＴ）纤维之二有Ｘ射线散射（ＷＡＸＳ）强度函数Ｉ（κ,α）,采用自编ＭＰＳ９６型全谱分峰程序,将不同方位角α测得的ＷＡＸＳ强度曲线上的４个非晶重叠峰进行分解,获得分子链间原子 射引起的２个峰为：ＫＡ＝０．１４１ｎｍ＾－１,ＫＢ＝０．１７２ｎｍ＾－１;分子链内原子 射引起的２个峰为;ＫＣ＝０．３１１ｎｍ＾－１,ＫＤ＝０．５５５ｎｍ＾－１,详细分析了     

碳纳米管和高取向热解石墨的拉曼光谱对比研究

同时测量了碳纳米管（Ｄ－ＣＮＴ）和高取向热解石墨（ＨＯＰＧ）的拉曼光谱，第一次分别在ＨＯＰＧ和Ｄ－ＣＮＴ中观察到了位于４２６５ｃｍ－１和４２４８ｃｍ－１的（Ｇ＋Ｄ＊）三级模的拉曼散射；拉曼谱的研究显示，Ｄ－ＣＮＴ的结构比ＨＯＰＧ的无序，同时也进一步表明此无序程度主要来源于结构的缺陷，而非石墨层卷曲成筒状之故     

激光散射研究聚甲基丙烯酸烷基磺酸钠／氯化钠水溶液的性质

本文用激光散射技术研究了具有不同长度烷基侧链的聚甲基丙烯酸烷基磺酸钠（ＰＳＳＲＭＡ）在０．１ＭＮａＣｌ水溶液中的性质。用静态光散射测定了这一系列聚合物的重均分子量Ｍ＿ｗ，均方半径Ｒ＿ｇ以及第二维利系数Ａ＿２．用动态光散射测定了光强的时间相关函数，运用Ｌａｐｌａｃｅ反演得到线宽分布Ｇ（Г）。当高聚物溶液的浓度大于１０￣（－３）ｇ／ｍＬ时，特性平均线宽〈Г〉迅速增加、表面张力降低；在２５℃－４７℃的范围内，〈Г〉随温度增加；烷基侧链越长，在０．１ＭＮａＣｌ水溶液中ＰＳＳＲＭＡ的分子排列更紧密。静态与动态光散射的结果建立以下方程：Ｄ＝ｋ＿ＤＭ￣ａ＿Ｄ，当α＿Ｄ～０．５６，对应于不同侧链，Ｋ＿Ｄ在１．５４×～１０￣（－４）－２．０７×１０￣（－４）（ｍＬ／ｇ）时，从线宽分布Ｇ（Г）得到ＰＳＳＲＭＡ的分子量分布。     

双波长倍增差示法测定双嘧啶片中磺胺嘧啶和甲氧苄啶的含量

本论述了双波长倍增差示法同时测定了双嘧啶片中磺胺嘧啶（ＳＤ）和甲氧苄啶（ＴＭＰ）的含量，本法是基于测定标准溶液ＳＤ（８～１４）×１０＾－６ｇ／ｍＬ，ＴＭＰ（１～２）×１０＾－６ｇ／ｍＬ，和样品溶液在２４２．５ｎｍ和２２８．０ｎｍ处的吸收度。ＳＤ和ＴＭＰ的平均回收率与ＲＳＤ分别为１００．６０％，０．３３％，和９８．２７％，１．５８（ｎ＝２０）。本法简便快捷，结果满意。     

Spectro-temporal envelope changes caused by temporal fine structure modification

The study of speech from which the temporal fine structure (TFS) has been removed has become an important research area. Common procedures for removing TFS include noise and tone vocoders. In the noise vocoder, bands of noise are modulated by the envelope of the speech within each band, and in the tone vocoder the carrier is a sinusoid at the center of each frequency band. Five different procedures for removing TFS are evaluated in this paper: the noise vocoder, a low-noise noise approach in which the noise envelope is replaced by the speech envelope in each frequency band, phase randomization within each band, the tone vocoder, and sinusoidal modeling with random phase. The effects of TFS modification on the speech envelope are evaluated using an index based on the envelope time-frequency modulation. The results show that for all of the TFS techniques implemented in this study, there is a substantial loss in the accuracy of reproduction of the envelope time-frequency modulation. The tone vocoder gives the best accuracy, followed by the procedure that replaces the noise envelope with the speech envelope in each band.     

Differential contribution of envelope fluctuations across frequency to consonant identification in quiet

Two experiments investigated the effects of critical bandwidth and frequency region on the use of temporal envelope cues for speech. In both experiments, spectral details were reduced using vocoder processing. In experiment 1, consonant identification scores were measured in a condition for which the cutoff frequency of the envelope extractor was half the critical bandwidth (HCB) of the auditory filters centered on each analysis band. Results showed that performance is similar to those obtained in conditions for which the envelope cutoff was set to 160 Hz or above. Experiment 2 evaluated the impact of setting the cutoff frequency of the envelope extractor to values of 4, 8, and 16 Hz or to HCB in one or two contiguous bands for an eight-band vocoder. The cutoff was set to 16 Hz for all the other bands. Overall, consonant identification was not affected by removing envelope fluctuations above 4 Hz in the low- and high-frequency bands. In contrast, speech intelligibility decreased as the cutoff frequency was decreased in the midfrequency region from 16 to 4 Hz. The behavioral results were fairly consistent with a physical analysis of the stimuli, suggesting that clearly measurable envelope fluctuations cannot be attenuated without affecting speech intelligibility.     

个数可变脉冲线性预测编码研究

不同语音帧的激励信号复杂性不同,所以采用相同个数的脉冲作为激励信号并不合理。针对这一点,提出了个数可变脉冲线性预测编码算法。该算法不固定脉冲个数,而是根据激励信号的复杂度而确定。个数可变脉冲线性预测编码的目的是用尽量少的脉冲数来满足误差约束,这可以看作一个稀疏表示问题。进而,给出了具体的脉冲搜索算法以及个数可变脉冲线性预测编码方案。实验结果发现增加脉冲可以减少误差,但是前面搜索出的脉冲对误差的贡献要大于后搜索出的脉冲。与G.723.1和G.729比较发现,个数可变脉冲线性预测编码可以在约4.2 kbps的编码速率下获得优于G.723.1的合成语音,但略差于G.729。本文算法的编码时间较长,是下一步需要解决的问题。     

Evaluation of two multichannel tactile aids for the hearing impaired

Two multichannel tactile devices for the hearing impaired were compared in speech perception tasks of varying levels of complexity. Both devices implemented the "vocoder" principle in their stimulus processing: One device had a 16-element linear vibratory array worn on the forearm and displayed activity in 16 overlapping frequency channels; the other device delivered tactile stimulation to a linear array of 16 electrodes worn on the abdomen. Subjects were tested in several phoneme discrimination tasks, ranging from discrimination of pairs of words differing in only one phoneme under tactile aid alone conditions to identification of stimuli in a larger set under tactile aid alone, lipreading alone, and lipreading plus tactile aid conditions. Results showed both devices to be better transmitters of manner and voicing features of articulation than of place features, when tested in single-item tasks. No systematic differences in performance with the two devices were observed. However, in a connected discourse tracking task, the vibrotactile vocoder in conjunction with lipreading yielded much greater improvements over lipreading alone than did the electrotactile vocoder. One possible explanation for this difference in performance, the inclusion of a noise suppression circuit in the electrotactile aid, was evaluated, but did not appear to account for the differences observed. Results are discussed in terms of additional differences between the two devices that may influence performance.     

Improving the performance of speech waveform synthesis using WaveNet fused with phase information

Compared with phase spectrum, magnitude spectrum can represent most speech information, hence many speech processing tasks pay much attention on manipulating magnitude spectrum and use the imperfect vocoder parameters or mismatched phase spectrum to synthesize the waveform, which leads to an obvious distortion of speech quality. To address this problem, a modified version of Wave Net model fused with phase information is proposed to synthesize the speech with higher quality. In the Wave Net model, the original or processed phase spectrum of speech and the enhanced magnitude spectrum are concatenated as the condition input, and then the predicted speech waveform is generated directly from this input, which is a kind of fusion feature. The proposed method can realize the effective utilization of the phase information and is verified in two tasks including voice conversion(VC) and bone-conducted speech enhancement(BSE). Two kinds of phase spectrum, the modified group delay(MGD)spectrum and the instantaneous frequency deviation spectrum, are compared comprehensively in the simulation experiments, and the influence of the fusion feature on the bandwidth extension Wave Net model and the teacher-student Wave Net model is also explored. In VC experiments, the A/B test shows the generated speech using the teacher-student Wave Net model is much better than using the STRAIGHT vocoder. In BSE experiments, the results show that,using the bandwidth extension Wave Net model via the feature fused with MGD spectrum, the mean opinion score(MOS) of the enhanced speech increases by 54.3% compared with the original bone-conducted speech. All the results demonstrate that the phase-fused condition input can supplement single magnitude spectrum efficiently and help the Wave Net vocoder achieve promising improvement on the quality of the synthesized speech.     

An effective cluster-based model for robust speech detection and speech recognition in noisy environments

This paper shows an accurate speech detection algorithm for improving the performance of speech recognition systems working in noisy environments. The proposed method is based on a hard decision clustering approach where a set of prototypes is used to characterize the noisy channel. Detecting the presence of speech is enabled by a decision rule formulated in terms of an averaged distance between the observation vector and a cluster-based noise model. The algorithm benefits from using contextual information, a strategy that considers not only a single speech frame but also a neighborhood of data in order to smooth the decision function and improve speech detection robustness. The proposed scheme exhibits reduced computational cost making it adequate for real time applications, i.e., automated speech recognition systems. An exhaustive analysis is conducted on the AURORA 2 and AURORA 3 databases in order to assess the performance of the algorithm and to compare it to existing standard voice activity detection (VAD) methods. The results show significant improvements in detection accuracy and speech recognition rate over standard VADs such as ITU-T G.729, ETSI GSM AMR, and ETSI AFE for distributed speech recognition and a representative set of recently reported VAD algorithms.     

Spectral and temporal cues for phoneme recognition in noise

Cochlear implant users receive limited spectral and temporal information. Their speech recognition deteriorates dramatically in noise. The aim of the present study was to determine the relative contributions of spectral and temporal cues to speech recognition in noise. Spectral information was manipulated by varying the number of channels from 2 to 32 in a noise-excited vocoder. Temporal information was manipulated by varying the low-pass cutoff frequency of the envelope extractor from 1 to 512 Hz. Ten normal-hearing, native speakers of English participated in tests of phoneme recognition using vocoder processed consonants and vowels under three conditions (quiet, and +6 and 0 dB signal-to-noise ratios). The number of channels required for vowel-recognition performance to plateau increased from 12 in quiet to 16-24 in the two noise conditions. However, for consonant recognition, no further improvement in performance was evident when the number of channels was > or =12 in any of the three conditions. The contribution of temporal cues for phoneme recognition showed a similar pattern in both quiet and noise conditions. Similar to the quiet conditions, there was a trade-off between temporal and spectral cues for phoneme recognition in noise.     

第十讲数字信号处理与数字信号处理器(DSP)

文章简要介绍了“数字信号处理”与“数字信号处理器（DSP）”的发展历史．在数字信号处理的应用中，实时实现是非常重要的，而DSP在实时处理中，扮演了一个重要的角色．文章中还介绍了DSP在实际应用中的一些关键技术，例如DSP的种类和选型，DSP的开发工具，实时软件的开发过程等．最后，还介绍了一些DSP的应用实例，如语音编码器，视频电话和视频会议系统，用于雷达和声纳的DSP并行处理系统     

Evaluation of companding-based spectral enhancement using simulated cochlear-implant processing

This study tested a time-domain spectral enhancement algorithm that was recently proposed by Turicchia and Sarpeshkar [IEEE Trans. Speech Audio Proc. 13, 243-253 (2005)]. The algorithm uses a filter bank, with each filter channel comprising broadly tuned amplitude compression, followed by more narrowly tuned expansion (companding). Normal-hearing listeners were tested in their ability to recognize sentences processed through a noise-excited envelope vocoder that simulates aspects of cochlear-implant processing. The sentences were presented in a steady background noise at signal-to-noise ratios of 0, 3, and 6 dB and were either passed directly through an envelope vocoder, or were first processed by the companding algorithm. Using an eight-channel envelope vocoder, companding produced small but significant improvements in speech reception. Parametric variations of the companding algorithm showed that the improvement in intelligibility was robust to changes in filter tuning, whereas decreases in the time constants resulted in a decrease in intelligibility. Companding continued to provide a benefit when the number of vocoder frequency channels was increased to sixteen. When integrated within a sixteen-channel cochlear-implant simulator, companding also led to significant improvements in sentence recognition. Thus, companding may represent a readily implementable way to provide some speech recognition benefits to current cochlear-implant users.     

Age-related changes in talker recognition with reduced spectral cues

Temporal information provided by cochlear implants enables successful speech perception in quiet, but limited spectral information precludes comparable success in voice perception. Talker identification and speech decoding by young hearing children (5-7 yr), older hearing children (10-12 yr), and hearing adults were examined by means of vocoder simulations of cochlear implant processing. In Experiment 1, listeners heard vocoder simulations of sentences from a man, woman, and girl and were required to identify the talker from a closed set. Younger children identified talkers more poorly than older listeners, but all age groups showed similar benefit from increased spectral information. In Experiment 2, children and adults provided verbatim repetition of vocoded sentences from the same talkers. The youngest children had more difficulty than older listeners, but all age groups showed comparable benefit from increasing spectral resolution. At comparable levels of spectral degradation, performance on the open-set task of speech decoding was considerably more accurate than on the closed-set task of talker identification. Hearing children's ability to identify talkers and decode speech from spectrally degraded material sheds light on the difficulty of these domains for child implant users.