Abstract: An audio decoder makes use of various component sharing techniques and operates to efficiently prevent deadlock without introducing decoding errors or adding significant complexity to the audio decoder. In one embodiment, the audio decoder comprises a bitstreamer, a synchronization controller, a decode controller, a memory module, a data path, and an output buffer. The bitstreamer retrieves compressed data and provides token-aligned data to the synchronization controller and decode controller. The synchronization controller initially controls the bitstreamer to locate and parse audio frame headers. After each frame header is parsed, the decode controller controls the bitstreamer to parse the variable length code compressed transform coefficients. The coefficients are passed to the memory module and data path which operate under the control of the decode controller to inverse transform the coefficients and produce digital output audio data.

Abstract: An audio signal processing apparatus performs digital-to-analog conversion to convert at least a first audio data signal having a first sampling frequency and a second audio data signal having a second sampling frequency which is different from the first sampling frequency into corresponding analog audio signals, respectively. The audio signal processing apparatus is provided with: a frequency converting device for converting the second sampling frequency of the second audio data signal into the same sampling frequency as the first sampling frequency; a digital-to-analog converting device for performing digital-to-analog conversion to convert the first audio data signal having the first sampling frequency and the second audio data signal having the converted second sampling frequency which is the same as the first sampling frequency into the analog audio signals, respectively.

Abstract: An audio recorder obtains a PCM audio signal from input analog or digital audio data, encodes the PCM audio signal, and stores the encoded signal in a semiconductor memory. The encoding method is selectable from at least two different methods, providing different levels of audio quality. The semiconductor memory may be internal to the audio recorder, or may be an external memory device such as a flash memory card. An audio player decodes the signal stored in the semiconductor memory, and converts the decoded signal to a selected one of at least two output formats.

Abstract: A short-delay postfilter (13) for postfiltering an encoded or decoded audio signal, specifically a speech signal, the short-delay postfilter having a transfer function F(z) of the form F(z)=D(z)/E(z), where E(z) and (D(z) are polynomials dependent on the variable z, where z is the inverse of the unit delay operator z.sup.-1 used in the z transform representation of transfer functions and wherein the denominator E(z) of the transfer functions H(z) of the audio signal's corresponding production filter (12) is also derived and the numerator D(z) differs from the denominator E(z) and is derived by using a longer period than used for the denominator E(z).

Abstract: A video and voice signal processing apparatus is provided. The apparatus includes a signal receiving circuit for receiving an input signal containing a plurality of frames, each frame having an encoded voice signal block and an encoded video signal block. The signal receiving circuit separates the encoded voice signal block from the encoded video signal block in each frame. A voice signal processor converts the encoded voice signal block into a voice signal. Also included is a video extracting circuit which decimates a plurality of encoded video signal blocks and extracts one of the encoded video signal blocks as a representative video signal. A video signal processor converts the representative video signal into a video signal.

Abstract: There is provided a decoder and a data recovery apparatus which has enabled power saving by setting the band to which series of processes are not carried out among the N bands to vary the processing time required for the decoding operation within the predetermined time interval and extend non-operable period, on the occasion of independently executing the predetermined decoding operation in every band for the unit data obtained by dividing the compressed data into N (integer N.gtoreq.2) bands for every predetermined time interval.

Abstract: An acoustic signal processing device used in, for example, karaoke systems, includes a signal processor, a digital/analog converter and a mixer. The signal processor converts the input analog acoustic signal into a digital signal and processes the converted digital signal by signal processing for generating a digital signal added to with pre-set acoustic characteristics. The digital/analog converter converts the digital signal generated by the signal processor into an analog signal. The mixer mixes the input analog signal with the analog signal outputted by the digital/analog converter.

Abstract: A frequency range of a digital audio signal is expanded without increasing a sampling frequency f.sub.5. The digital audio signal has a frequency range with an upper frequency defined by a frequency f.sub.5 /2, and also has noise components induced when being digitized. Firstly, the digital audio signal is split into first and second audio signals. The noise components of the first audio signal are shifted up to a predetermined frequency higher than f.sub.5 so as to leave a first frequency range between the first audio signal and the noise components. Following this, the first audio signal is delayed by a predetermined time duration. On the other hand, the second audio signal is band-pass filtered such as to have a second frequency range defined by frequencies f.sub.1 and f.sub.2 wherein 0<f.sub.1 <f.sub.2 <f.sub.5. Subsequently, the second audio signal is Fourier transformed, after which the spectrum of the second frequency range is processed so as to be expanded over new spectrum.

Abstract: A method of hiding information in a host audio signal introduces one or more echoes into the signal. The separation in time between the host signal and an echo is associated with the value of a datum embedded in the signal. The identity of the embedded datum is determined by observing the delay between the host signal and the echo.

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: Voice compression is performed in multiple stages to increase the overall compression between the incoming analog voice signal and the resulting digitized voice signal over that which would be obtained if only a single stage of compression were to be used. A first type of compression is performed on a voice signal to produce an intermediate signal that is compressed with respect to the voice signal, and a second, different type of compression is performed on the intermediate signal to produce an output signal that is compressed still further. As a result, compression better than 1920 bits per second (and approaching 960 bits per second) are obtained without sacrificing the intelligibility of the subsequently reconstructed analog voice signal. Voice compression is also performed by recognizing redundant portions of said voice signal, such as silence, and replacing such redundant portions with a special code in said compressed signal.