Abstract: A method for encoding a digitized audio signal which includes the step of selecting one of two or more psycho acoustic model provided for generating masking thresholds used in a data reduction process. The selecting criterion is the available data rate for the encoded bit stream. Each one of the two or more psycho acoustic models is adapted to a specific data rate of the encoded bit stream. In a second embodiment, the method includes the step of combining two or more masking thresholds resulting from different psycho acoustic models, thereby leading to a more accurate calculation of a masking threshold for the data reduction process. Further, there are appropriate apparatuses for the encoding of digitized audio signals.

Abstract: A method of transmitting speech signals with reduced bandwith requirements. With this invention an original speech signal is first converted to a textual representation, and a facsimile of the original speech is determined from the textual representation. Then a minimum error turn is derived from the difference between the original speech signal and the facsimile of the original speech signal. The minimum error turn is then compressed, and it is this compressed minimum error turn, along with the textual representation, that is transmitted on the communications medium. At the receiving end, the textual representation and the difference representation are split through a demultiplexer. The textual representation is then passed through a synthesizer while the difference representation is passed through a mapper.

Abstract: A first vector quantizer generates output codevectors corresponding in number to a number determined by a predetermined number of bits through linear coupling of integer coefficients of a predetermined number of base vectors stored in a base vector memory. A second vector quantizer determines coefficients of the base vectors according to at least one of output indexes of the output codevectors.

Abstract: A data processing device is programmed to decode and transform a stream of data representing a plurality of subband encoded channels of audio data into one or more channels of PCM encoded data for reproduction by a speaker subsystem. An improved method for decoding and transforming utilizes downmix matrices (1021 and 1022) to form downmixed frequency domain channels in buffers (1031-1034). Only two long DCT transform operations (1041 and 1042) and two short DCT transform operations (1043 and 1044) are needed to transform the downmixed frequency domain channels into a left PCM output (1071) and a right PCM output (1072).

Abstract: A speech encoding system for encoding a digitized speech signal into a standard digital format, such as MIDI. The MIDI speech encoding system includes a memory storing a dictionary comprising a digitized pattern and a corresponding segment ID for each of a plurality of speech segments (i.e., phonemes). A speech analyzer identifies each of the segments in the digitized speech signal based on the dictionary. One or more prosodic parameter detectors measure values of the prosodic parameters of each received digitized speech segment. A MIDI speech encoder converts the segment IDs and the corresponding measured prosodic parameter values into a MIDI speech signal. A MIDI speech decoding system includes a MIDI data decoder and a speech synthesizer for converting the MIDI speech signal to a digitized speech signal.

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: A split-band coding system combines multiple channels of input signals into various forms of composite signals and generates spatial-characteristic signals representing soundfield spatial characteristics in a plurality of frequency subbands. The spatial-characteristics signals may be generated in either or both of two forms. In a first form, the signal represents measures of signal levels for subband signals from the input channels. In a second form, the signal represents one or more apparent directions for the soundfield. The type of the spatial-characteristics signal may be adapted dynamically in response to a variety of criteria including input signal characteristics. Temporal smoothing and spectral smoothing of the spatial-characteristics signals may be applied in an encoder. Temporal smoothing and spectral smoothing may be applied to gain factors derived from the spatial-characteristics signals in a decoder.

Abstract: An MPEG audio/video decoder has memories, a signal processing unit (SPU) including a multiplier and a butterfly unit, a main CPU, and a memory controller which are time division multiplexed between decoding video and audio data. For audio decoding, the butterfly unit determines combinations of components of a frequency-domain vector to reduce the number of multiplies required to transform to the time domain (matrixing). Matrixing is interwoven with MPEG filtering to increase throughput of the decoder by increasing parallel use of the multiplier, the butterfly unit, and a memory controller. A widowing process for the MPEG standard uses only independent components of the audio vectors. This reduces the required number of components to be stored, thereby reducing the size of required memory, the time to write the components after matrixing, and the time to retrieve the components for windowing.

Abstract: An audio signal compression and decompression method and apparatus that provide lossless, realtime performance. The compression/decompression method and apparatus are based on an entropy encoding technique using multiple Huffman code tables. Uncompressed audio data samples are first processed by a prediction filter which generates prediction error samples. An optimum coding table is then selected from a number of different preselected tables which have been tailored to different probability density functions of the prediction error. For each frame of prediction error samples, an entropy encoder selects the one Huffman code table which will yield the shortest encoded representation of the frame of prediction error samples. The frame of prediction error samples is then encoded using the selected Huffman code table. A block structure for the compressed data and a decoder for reconstructing the original audio signal from the compressed data are also disclosed.

Abstract: An audio decoder which can reduce a memory circuit capacity necessary for performing a series of decoding processes and can perform a down mixing. The audio decoder decodes audio data of a plurality of channels encoded in a frequency domain by using a time base to frequency base conversion. After a down mixing process was performed to the audio data of the frequency domain by frequency domain down mixing circuit, it is converted into audio data of a time domain by frequency base to time base converting circuit, thereby reducing memories by the number corresponding to the reduced number of channels. Further, by executing an inverse quantizing process of each channel and a frequency base to time base converting process of each channel by pipeline processes, a work buffer can be shared in both of the processes.

Abstract: A method and apparatus for speech transmission in a telecommunications system in which a speech signal is compressed to a small number of speech coding bits by a speech coding method, and the speech coding bits are subjected to channel coding. Several different speech coding methods, which may all operate at different transmission rates, are involved in the speech transmission. The method is based on the use of two-stage channel coding. The first channel coding is dependent on the speech coding method, and it is performed in connection with the speech coding in such a manner that the total transmission rate provided by the speech coding and the first channel coding is always constant irrespective of the speech coding method. The second channel coding performed thereafter is always exactly the same regardless of the speech coding method and the first channel coding method, and it is used with all speech coding methods.

Abstract: An encoding system and an encoding method for encoding a digital signal having at least a first and a second digital signal component. The encoding system includes a splitter unit for dividing the bandwidth of the digital signal components into M successive frequency bands, and generating in response to the digital signal components M sub signals (SB.sub.m1,SB.sub.

Abstract: A method, system and product are provided for selectively modifying an encoded audio signal. The method includes receiving the encoded audio signal, the encoded audio signal having a first frequency bandwidth, and identifying a delivery point for the encoded audio signal, the delivery point having a second frequency bandwidth. The method also includes selecting a plurality of subbands from the first frequency bandwidth based on the second frequency bandwidth, and modifying the encoded audio signal based on the plurality of subbands selected. The system includes control logic for performing the method. The product includes a storage medium having computer readable programmed instructions for performing the method.

Abstract: A method and apparatus which allows the transmission of the perceptually important features of a speech-coding parameter at a low bit rate. The speech coding parameter may, for example, comprise the signal power of the speech. The parameter is processed on a block by block basis. The parameter value at the block boundaries is transmitted by conventional methods such as, for example, by means of differential quantization. The shape of the reconstructed parameter contour within block boundaries is based on a classification. The classification determines perceptually important features of the parameter contour within a block. The classification can be performed either at the transmitting end of the coder (using, for example, the original parameter contour with high time resolution and possibly other speech parameters as well) or at the receiving end of the coder (using, for example, the transmitted parameter values, and possibly other transmitted speech parameters as well).

Abstract: A method and apparatus for the decoding of digital audio data encoded in accordance with layer 1 or 2 of the MPEG format. The inverse quantization of the quantizised data samples and the resealing of the inverse quantizised data samples takes place contemporaneously with windowing of the data samples transformed into the time domain and not contemporaneously with transformation of data samples from the frequency domain into the time domain using a matrix operation. The apparatus has fixed-wire frame unpacking and filter bank units. The frame unpacking unit is used for frame synchronization of a data stream, header decoding, reading of page information, inverse quantization of quantizised subband data samples and resealing of the inverse quantizised data samples. The filter bank unit is used for transformation of rescaled data samples from the frequency domain into the time domain using a matrix operation and for windowing of the data transformed into the time domain.

Abstract: Digital data divided into blocks each having a predetermined number of samples is transformed into data on a frequency axis for each block to generate coefficient data for each frequency. Coefficient data of a predetermined number of blocks are stored in a buffer. From the buffer, coefficient data are inputted to a floating-point transforming circuit for each one block. The coefficient data are divided into a plurality of sub-bands, each sub-band including one or a plurality of coefficient data. The coefficient data are floating-point transformed for each sub-band and transformed into one sub-band common characteristic data which is common to the coefficient data included in each sub-band and mantissa data of the number equal to the number of coefficient data included in each sub-band. The sub-band common characteristic data and the mantissa data are stored in a memory. Sub-band common characteristic data of a predetermined number of blocks are read from the memory for each sub-band.

Abstract: A synthesis filter for an MPEG-2 audio decoder comprising a first coefficient ROM for storing cosine matrix coefficients therein, a first queue memory for storing subband data therein, a first MAC unit for performing a matrixing operation by multiplying the subband data stored in the first queue memory by the cosine matrix coefficients stored in the first coefficient ROM, a second queue memory for storing output data from the first MAC unit therein, a third queue memory being copied with the contents of the second queue memory, a second coefficient ROM for storing window coefficients therein, a second MAC unit for synthesizing the contents copied to the third queue memory with the window coefficients stored in the second coefficient ROM to produce audio data, an FIFO memory for storing the audio data from the second MAC unit therein, a D/A converter for converting the audio data stored in the FIFO memory into an analog audio signal and outputting the converted analog audio signal to a speaker, and a controlle

Abstract: A speech signal processing circuit includes an input buffer for receiving inverse quantization samples and for temporarily storing those samples. The circuit also includes a band synthesis filter for reading the inverse quantization samples stored in the input buffer one by one, and for conducting quadrature conversion processing and sum-of-product operation processing to decode the samples into speech signals. The circuit further includes a control circuit for controlling operation of the band synthesis filter. When the inverse quantization samples are recognized as being stored in the input buffer, the control circuit controls the band synthesis filter to execute, as an initial operation, the quadrature conversion processing of the inverse quantization samples as many times as a number corresponding to an operation delay time of the band synthesis filter.

Abstract: A method and apparatus for encoding an input signal in which the input signal is transformed into frequency components which are encoded by quantization from one encoding unit to another. The information specifying zero encoding units among the encoding units in which encoding is done on the assumption that all frequency components contained in them are deemed to be zero are encoded, is encoded, while the quantization step information of the zero encoding units is outputted without encoding. With the encoding method and apparatus, the number of the encoding bits may be decreased while deterioration in the input signal is prohibited for improving the encoding efficiency.

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.