Abstract: Nasalized sound effects during reproduction of low-pitch sounds are suppressed to produce playback sounds of high clarity. Amplitude data is processed with high range formant emphasis of crests and valleys of the envelope of the frequency spectrum on the high frequency range and with deepening of the valley of the frequency spectrum over the entire frequency range, above all, over the low to mid frequency range. Next, the amplitude data is processed for emphasizing the peak values of the formant of the voiced frame in the portion of the speech signal which is rising in magnitude and for unconditionally emphasizing the spectral envelope on the high frequency range. The voiced speech spectrum is generated by synthesizing the cosine wave based upon the emphasized amplitude data.

Abstract: An apparatus and method for high efficient signal coding by separating frequency components obtained by converting an input signal into tone property component signals and the other component signals by using a mask level obtained based on the psychoacoustic model and coding these signals respectively to increase signal coding quality and efficiency.

Abstract: A method and apparatus for implementing a vocoder in a application specific integrated circuit (ASIC) is described. The apparatus contains a DSP core that performs computations in accordance with a reduced instruction set (RISC) architecture. The circuit further includes a specifically designed slave processor to the DSP core referred to as the minimization processor. The apparatus further includes a specifically designed block normalization circuitry.

Abstract: Several audio signal processing techniques may be used in various combinations to improve the quality of audio represented by an information stream formed by splice editing two or more other information streams. The techniques are particularly useful in applications that bundle audio information with video information. In one technique, gain-control words conveyed with the audio information stream are used to interpolate playback sound levels across a splice. In another technique, special filterbanks or forms of TDAC transforms are used to suppress aliasing artifacts on either side of a splice. In yet another technique, special filterbanks or crossfade window functions are used to optimize the attenuation of spectral splatter created at a splice. In a further technique, audio sample rates are converted according to frame lengths and rates to allow audio information to be bundled with, for example, video information.

Abstract: Estimating the time-varying spectrum envelope of a time-varying signal facilitates pitch modification and other shifting of signal content in the frequency domain. Local maxima of a spectrum of the signal are identified by applying a masking curve. The masking curve has a peak at the particular maximum and descends away therefrom the local maximum. Local maxima falling below the local maximum are eliminated. The slope of the masking curve is varied in accordance with measured parameters of the spectrum to decrease or eliminate spurious peaks. Thereafter, a smoothing procedure may be applied to smooth the spectrum in frequency.

Abstract: Frequency components are broken into a first signal made up of a plurality of tonal components and a second signal made up of other components. The number of the frequency components making up the tonal components is variable. Tonal signals may be encoded efficiently depending on the manner of distribution of their spectral energy to assure more efficient encoding on the whole. If the signals compression coded in this manner are recorded on a recording medium, the recording capacity may be employed effectively. Also, high-quality acoustic signals may be obtained on decoding signals reproduced from the recording medium.

Abstract: An apparatus for improving the link margin of a communication link includes a variable rate vocoder which decreases the output bit stream rate it produces so as to reduce the amount of information having to be transmit in the communication link. In one embodiment, the variable rate vocoder includes a plurality of vocoder portions, each of which produces a different bit stream rate. The selector is used for selecting among the output bit streams produced by each vocoder. In another embodiment, a logic device is coupled to the output of the vocoder. The logic device, upon receipt of a control signal, truncates the less important bits.The method for improving link margin includes reducing the vocoder output rate thereby reducing the amount of data being transmit in an communication link. The method also includes using increased error correction coding and transmitting at increased per bit power levels to increase link margin.

Abstract: A loudness balancing process includes three operations. In a first operation, the user specifies a plurality of digitally sampled audio time domain waveforms and an adjusted maximum loudness for each waveform is generated and stored. This operation includes a retrieve and filter process that identifies a portion of each waveform with a maximum loudness, and an adjust and store process that generates an adjusted maximum loudness that is a maximum loudness for the waveform which is free of audible distortion due to clipping. In a second operation, each stored adjusted maximum loudness is retrieved and filtered. The filtering selects a minimum adjusted maximum loudness that is selected as a global maximum loudness. In a third operation, each waveform in the plurality of waveforms is loudness-balanced based on the global maximum loudness. This three step process assures a consistent maximum loudness across the plurality of waveforms and assures that no audible noise is introduced by loudness balancing process.

Abstract: A method and a device for discriminating a voiced sound from an unvoiced sound or background noise in speech signals are disclosed. Each block or frame of input speech signals is divided into plural sub-blocks and the standard deviation, effective value or the peak value is detected in a detection unit for detecting statistical characteristics from one sub-block to another. A bias detection unit detects a bias on the time scale of the standard deviation, effective value or the peak value to decide whether the speech signals are voiced or unvoiced from one block to another.

Abstract: A speech coder (100) computes scalar statistics (180), ensemble statistics (190), spectral parameters (150), and a normalized excitation waveform (270) which describe a frame of speech samples. The coder (100) encodes the statistics (220, 230), spectral parameters (155), and the normalized waveform (290) for later decoding and synthesis. A speech synthesizer (900) decodes the encoded scalar statistics (570), encoded ensemble statistics (560), encoded spectral parameters (490), and encoded normalized excitation waveform (550). The synthesizer (900) then denormalizes (670) the normalized excitation waveform using the scalar statistics and the ensemble statistics, resulting in a decoded excitation waveform. Speech is synthesized (710) from the decoded excitation waveform and the decoded spectral parameters.

Abstract: In a speech coding method of the present invention, initially, a plurality of samples of speech data are analyzed by a linear prediction analysis and thereby prediction coefficients are calculated. Then, the prediction coefficients are quantized, and the quantized prediction coefficients are set in a synthesis filter. Moreover, a pitch period vector is selected from an adaptive codebook in which a plurality of pitch period vectors are stored, and the selected pitch period vector is multiplied by a first gain which is obtained, at the same time, with a second gain. In addition, a noise waveform vector is selected from a random codebook in which a plurality of the noise waveform vectors are stored, and is multiplied by a predicted gain and the second gain. Then, the speech vector is synthesized by exciting the synthesis filter with the pitch period vector multiplied by the first gain, and with the noise waveform vector multiplied by the predicted gain and the second gain.

Abstract: In this invention, an approach is employed to carry out blocking of an input signal to transform the blocked signals into spectrum signals to divide the spectrum signals into a plurality of units to normalize them thereafter to implement variable length encoding to all or a portion of the spectrum signals to output the variable-length encoded signals along with normalization coefficient and the number of re-quantization bits of each unit, wherein an upper limit is provided with respect to the number of bits per each block of a signal to be encoded and outputted to compulsorily change, in a block for which the number of bits above the upper limit is required, normalization coefficient of at least one unit thereafter to re-quantize and entropy-encode a corresponding signal to output the encoded spectrum signal, thereby permitting hardware scale to be smaller as compared to the conventional apparatus without depending upon unevenness of the number of bits by variable length encoding.

Abstract: An apparatus and method for performing speech signal compression, by variable rate coding of frames of digitized speech samples. The level of speech activity for each frame of digitized speech samples is determined and an output data packet rate is selected from a set of rates based upon the determined level of frame speech activity. A lowest rate of the set of rates corresponds to a detected minimum level of speech activity, such as background noise or pauses in speech, while a highest rate corresponds to a detected maximum level of speech activity, such as active vocalization. Each frame is then coded according to a predetermined coding format for the selected rate wherein each rate has a corresponding number of bits representative of the coded frame. A data packet is provided for each coded frame with each output data packet of a bit rate corresponding to the selected rate.

Abstract: An information encoding method and apparatus, an information decoding method and apparatus and an information transmission method in which encoding and decoding with higher efficiency and higher sound quality may be achieved by gain control in meeting with the degree of amplitude changes in the attack portion and the pre-echo may be prevented from occurring. Gain control and gain control compensation operations are performed by applying a gain control function with a smaller gain control quantity and by applying a gain control function with a larger gain control quantity to a signal waveform portion having a level just ahead of an attack portion higher than a pre-set level and to a signal waveform portion having an extremely low level just ahead of the attack portion, respectively.

Abstract: A method and apparatus for enhancing the intelligibility of a telephonic speech signal within the available bandwidth and intensity limits of a telephone communication network. The method combines enhancement of both the formant ratio and the consonant/vowel energy ratio to realize a speech signal more intelligible to a hearing impaired user. The invention uses an auditory model of the human ear. A speech signal is put through a filter bank designed to simulate the cochlear filter shapes and filter spacing of a healthy cochlea. The energy output from each of a plurality of filters is computed and used to form an auditory spectrum. The peaks associated with strong first and second formants are identified, and the second formant is enhanced relative to the first formant by attenuating the first formant. Also, consonants in the speech signal are identified as having an energy level below a threshold associated with vowels, but above the threshold associated with silent regions. Consonant regions are amplified.