Abstract: A system and method for improved audio controls on a personal computer is provided. The system and method provide a unified architecture for audio controls across hardware and software interfaces of the personal computer. An intelligent facility may automatically change audio controls for users to simply interact with various communications and media applications. To this end, a configurable audio controller intelligently handles various aspects of the system's audio devices by following various rules that may be based at least in part on user-configurable settings and a current operating state. The present invention also provides audio controls so that a user may easily change audio settings such as the volume of an audio output device. There are many applications that may use the present invention for automatic control of audio devices based upon the user's context.

Abstract: Variable attack and release system and method are disclosed for dynamically modifying the various elements of the system, including modifying compander response time to dynamically adjust for changing parameters such as environmental or input signal changes, to maintain output signals within predetermined limits. The system and method include permitting accurate and quick response to noise sources having a duration which exceeds a predetermined threshold while at the same time ignoring transient or short duration environmental noise.

Abstract: A method of matching input amplitudes in a system wherein one or more of a plurality of inputs may be selected, with each input capable of having different characteristics, involving selecting an input signal and mapping the input signal to a predetermined signal amplitude through the use of level matching logic. The level matching logic may include a gain cell for increasing or decreasing the amplitude of the input signal.

Abstract: An audio converting device including a digital high-pass filter, an expander, a digital low-pass filter, a delta-sigma modulator, a digital-to-analog converter, an analog low-pass filter and a gain control unit is provided. The digital high-pass filter in this invention can filter out a direct-current component of digital audio data such that the production of noise is avoided when the volume is adjusted by users.

Abstract: A sound signal reproducing apparatus for supplying sound signals of a plurality of channels to speakers via sound amplifier circuits includes: a power supply voltage controller for controlling a power supply voltage to the sound amplifier circuits; a plurality of filter circuits for passing predetermined frequencies of the output signal of each of the sound amplifier circuits in the plurality of channels; a plurality of level detection circuits for detecting levels of output signals of the plurality of filter circuits; an adder circuit for adding output signals of the plurality of level detection circuits; and a comparator circuit for comparing an output signal of the adder circuit with a reference level, so that an output signal of the comparator circuit controls the power supply voltage controller.

Abstract: The invention concerns a method (300) and system (100) for controlling audio output. The method includes the steps of inputting (312) an audio signal and a voltage level signal, measuring (314) the audio signal and the voltage level signal, mapping (316) the audio signal against at least one table (134) of predetermined corresponding gain targets (138) and selecting (318) at least one gain target for the audio signal. The mapping step and the selecting step are based at least in part on the measurement of the voltage level signal and the measurement of the audio signal. The method also includes the step of applying (320) the gain target to the audio signal.

Abstract: The present invention relates to a process for adjusting the sound volume of a digital sound recording characterized in that it comprises: a step consisting of determining, in absolute values, for a recording, the maximum amplitude values for sound frequencies audible for the human ear, a step consisting of calculating the possible gain for a specified sound level setting, between the maximum amplitude value determined above and the maximum amplitude value for all frequencies combined, a step consisting of reproducing the recording with a sound card by automatically adjusting the amplification gain level making it possible to obtain a sound level for the recording of a specified value so that it corresponds to the gain calculated for this recording.

Abstract: An audio signal encoding apparatus includes a device for compressing multiple-channel digital audio signals into compression-resultant multiple-channel signals respectively. The multiple-channel digital audio signals relate to a sampling frequency and a quantization bit number. The compression-resultant multiple-channel signals, a signal representative of the sampling frequency, and a signal representative of the quantization bit number are formatted into a formatting-resultant signal. The formatting-resultant signal contains a sub packet and a sync information portion. The sub packet contains at least portions of the compression-resultant multiple-channel signals. The sync information portion contains the signal representative of the sampling frequency and the signal representative of the quantization bit number.

Abstract: Output control system and method includes calculating gain for an input signal, detecting a predetermined condition, and modifying the gain calculation in accordance with one or more kneepoints to provide a varying output signal. A companding ratio is then established in accordance with the kneepoints. System gain is then adjusted, and the companding ratio may be further adjusted by adjusting one of the kneepoints relative to another.

Abstract: Maximal and minimal values represented by samples of a digital audio signal are detected. A number of samples from a sample representing a minimal value to a maximal-value-corresponding sample is detected. A number of samples from a sample representing a maximal value to a minimal-value-corresponding sample is detected. Calculation is given of a first difference between the maximal-value-corresponding sample and the immediately-preceding sample. Calculation is given of a second difference between the minimal-value-corresponding sample and the immediately-preceding sample. First and second coefficients are calculated from the detected sample numbers. The first coefficient and the first difference are multiplied to generate a first multiplication result. The second coefficient and the second difference are multiplied to generate a second multiplication result. The maximal value represented by the maximal-value-corresponding sample is incremented by the first multiplication result.

Abstract: With the goal of presenting a waveform compression and expansion apparatus with which the sound quality of such things as musical tones that are expressed by waveforms is satisfactory following the compression and expansion of the waveforms of the musical tones etc., a method and system for waveform compression and expansion is disclosed in which all of the multiple number of band divided waveforms that comprise the original waveform which has been band divided are apportioned to at least two kinds of compression and expansion formats and form a multiple number of compressed and expanded waveforms by compression or expansion an identical amount only in the direction of the temporal axis.

Abstract: A hearing aid device providing instantaneous gain compression for sound signals and adaptive control of nonlinear waveform distortion, the device comprising: (a) at least one bandpass nonlinearity (BPNL) amplifier comprising a first bandpass filter, a second bandpass filter, and a memoryless nonlinear (MNL) compressive audio amplifier configured to receive a sound signal from the first bandpass filter and provide an MNL compressive audio amplifier output to the second bandpass filter, wherein the MNL compressive audio amplifier is configured to produce the MNL compressive audio amplifier output by providing memoryless gain compression directly on a sound signal that is (1) received from the first bandpass filter and (2) exhibits instantaneous amplitudes greater than a compression threshold, the BPNL amplifier thereby producing a desired gain compression on the received sound signal at an output of the second bandpass filter, and (b) a controller in communication with the BPNL amplifier, the controller being c

Abstract: An earmuff comprising a headpiece and a circuit. The headpiece supports the earmuff on the head of an individual. The circuit has an input device for receiving external sound energy and converting the external sound energy to electrical sound signals.

Abstract: A method for computing the “clippings” of an audio signal in the digital domain to prevent aliasing is disclosed. An offset signal is found by subtracting a threshold from samples of the audio signal. These are multiplied by a pulse after the pulse is lined up in time with the crossing of the audio signal at the threshold. The pulse is the sum of two step functions, both of which are bandwidth-limited.

Abstract: A device for decoding audio signals subjected to a noise-reduction encoding such as a Dolby-B encoding comprises a plurality of processing blocks (612, 616, 620, 624, 628, 632, 636, 640) for generating, starting from an input signal (610) containing the audio signal subjected to encoding superimposed on a noise component, an output signal (626) consisting of a replication of the audio signal with the noise component reduced. The aforesaid processing blocks are implemented in a digital form and comprise a sliding-band filtering structure (612) fed with the input signal (610) and designed to generate a filtered signal (614). The filtered signal (614) is fed, according to a general feedforward scheme, to an overshoot-suppression stage (616) in view of the generation, in an adder node (620), of a difference signal (622) starting from which the output signal is obtained via filtering (624).

Abstract: An apparatus is provided for processing a wide dynamic range analog signal which comprises multiple components such as a signal with an in-phase component and a quadrature-phase component in for example separate I and Q data channels, wherein each channel has a dynamic range compressor stage and an operator stage which processes the compressed signals. Optionally the apparatus has a dynamic range expander stage following the operator stage. A method according to the invention involves processing I and Q information after first independently compressing the dynamic range of the signal according to a logarithmic transfer characteristic over a frequency range of interest. A mathematical operation through a F(i,q) function (corresponding to the operator stage) is performed on the compressed components, thereby producing normalized components. The operating transfer function F(i,q) cross links the data channels to effect normalization based on amplitude of information in each of the channels.

Abstract: A circuit and related method for digital volume control are provided, where the circuit includes a digital filter configured to process samples of an input stream in a manner that processes a previous input sample during a time interval before a subsequent input sample, and outputs a series of exponentially decaying waveforms. The result is an exponential response to a volume change made by a user, where the change feels more pleasant and natural than a conventional linear response.

Abstract: A hearing compensation system for the hearing impaired comprises a plurality of bandpass filters having an input connected to an input transducer and each bandpass filter having an output connected to the input of one of a plurality of multiplicative AGC circuits whose outputs are summed together and connected to the input of an output transducer. The multiplicative AGC circuits attenuate acoustic signals having a constant background level without the loss of speech intelligibility. The identification of the background noise portion of the acoustic signal is made by the constancy of the envelope of the input signal in each of the several frequency bands. The background noise that will be suppressed includes multi-talker speech babble, fan noise, feedback whistle, florescent light hum, and white noise.

Abstract: Computer processor method and apparatus for creating an audio multiplier control signal for controlling the dynamic range of a recorded audio work. The technique includes determining an envelope of the amplitude of amplitude versus time values the audio signal, and then determining, for values of the envelope, respective minimum and maximum multiplication factors (MinMF and MaxMF) that can be multiplied times the values such that the products are above a predetermined minimum amplitude and below a predetermined maximum amplitude of the dynamic range. Then a control signal function of amplitude versus time is created such that all values of the control signal function at particular times are between respective MinMF and MaxMF values for the times, and such that segments of the control signal function have reduced slopes. Also disclosed is a method of creating a reduced-slope series of line segments passing through a pair of Max and Min limiting functions specifying y values with respect to a variable x.

Abstract: Improved approaches are disclosed to filter and compress sound signals so as to achieve not only speech audibility and intelligibility at low levels but also preserves spectrum contrast at high levels. According to one aspect of the invention, gain amounts for different frequency bands are individually constrained based on signal levels for the frequency bands. Hence, the gain amounts for each of the frequency bands may or may not be constrained depending on the corresponding signal levels. As a result, the most critical information for speech intelligibility, speech clarity, and speech quality can be made available to hearing impaired people over wide range of signal level. The invention is particularly useful for hearing aids or other sound systems for the hearing impaired.

Abstract: The invention relates to an apparatus (1) for amplifying bass frequencies of an audio signal (AL, AR). The bass components are applied to a controllable amplifier (17).

Abstract: The present invention involves a digital audio player and a method for processing encoded digital audio data, wherein the digital audio data is encoded using one of a plurality of encoding formats. The audio data player has a hard disk or other data storage medium for storing data files, a microcontroller buffer memory for anti-skip protection, and an audio decoder. The encoded audio data files and associated decoder files are downloaded from a personal computer or similar device to the audio data player hard drive. The player provides a menu-driven user interface for selection, sorting, and playback of stored audio data files. During playback, elapsed playback time is computed and displayed. For variable bit-rate audio data files, the audio data player generates an elapsed playback timekeeping map concurrently with playback and fast forward scan of audio data files.

Abstract: The invention relates to a fully digitized audio system comprising a power supply and a decode and sound field effect process unit, characterized in that a digital audio signal output from the decode and sound field effect process unit is transmitted to a control and encode unit where audio and control signals are encoded, and coupled to a digital sound box through a digital transmission terminal in the control and encode unit. The invention adopts totally digitized audio signal processing according to the concept of mechatronics, solving the problem of distortions rising throughout the procedure from input, process, distribution, transmission and amplification to sounding, to ensure a controllable hi-fi output of audio signal. The audio system according to the present invention provides high fidelity, good controllability, and easiness of assembling, and is suitable for mass production of advance audio systems.

Abstract: A single control determines both volume and the degree of compression in the reproduction of motion picture soundtracks. For some settings of the control, compression or limiting reduces the highest levels on a movie soundtrack, leaving the dialogue level substantially unchanged, thus removing the reason for complaints from the audience without the danger of making the dialogue too quiet for intelligibility.

Abstract: An audio signal processing circuit reduces noise in an incoming audio signal and is particularly useful in telephone communication systems utilizing one or more hands-free microphones. A preferred embodiment of the audio signal processing circuit includes a pre-emphasis circuit receiving the audio signal from a microphone or other transducer, an amplifier circuit receiving the pre-emphasized audio signal from the pre-emphasis circuit and a de-emphasis circuit receiving the amplified signal from the amplifier circuit. An output of the de-emphasis circuit provides the processed audio signal having an improved signal to noise ratio with minimum audible distortion. A preferred embodiment of the amplification circuit includes an amplifier defining a non-linear transfer function therethrough which provides low gain to the lower amplitude noise signals and higher gain to the higher amplitude audio signals.

Abstract: A multi-channel signal processing system adapted to provide binaural compressing of tonal inputs is provided. Such a system can be used, for example, in a binaural hearing aid system to provide the dynamic-range binaural compression of the tonal inputs. The multi-channel signal processing system is essentially a system with two signal channels connected by a control link between the two signal channels, thereby allowing the binaural hearing aid system to model behaviors, such as crossed olivocochlear bundle (COCB) effects, of the human auditory system that includes a neural link between the left and right ears. The multi-channel signal processing system comprises first and second channel compressing units respectively located in first and second signal channels of the multi-channel signal processing system. The first and second channel compressing units receive first and second channel input signals, respectively, to generate first and second channel compressed outputs.

Abstract: Systems and methods for detecting clipping conditions in an audio signal and processing the signal to reduce the clipping conditions. In one embodiment, a system comprises a noise shaper, a modulator, an output stage and other components. A detector detects clipping in the noise shaper and a signal processor processes the audio signal input to the noise shaper based on feedback received from the detector. The signal processor may function to modify the input audio signal in different ways in response to different conditions that are detected by the detector. A filter may be included to filter the output of the detector before being provided to the signal processor. A flag circuit may be coupled between the filter and the signal processor to assert an output signal until the signal processor resets the flag circuit.

Abstract: An amplitude adjustment device such as an amplitude compression device and amplitude expansion device is basically configured by a PWM modulator, a demodulator and an amplitude detector. Herein, the PWM modulator effects pulse-width modulation on an input signal to produce a pulse-width modulated signal, which is demodulated by the demodulator to produce an output signal. In addition, the amplitude detector detects an amplitude of a demodulated signal or an amplitude of the input signal to produce a control signal. A modulation factor of the pulse-width modulation is adjusted based on the control signal. In the case of the amplitude compression device, an input/output gain is changed inversely proportional to the amplitude of the input signal or amplitude of the output signal. Thus, it is possible to compress a dynamic range with respect to input/output characteristics.

Abstract: An audio codec and a method of compressing audio data makes use of a filterbank which automatically adapts itself to changes in the sampling frequency/bit rate to mimic the characteristics of the human auditory system. The algorithm used compares the bandwidth of each sub-band at a given depth with the critical bandwidth. If the critical bandwidth is less than the bandwidth of the sub-band, then the sub-band is split into two at the next level, and the process is repeated until the bandwidth of every sub-band is less than the critical bandwidth at the corresponding frequency. The codec thus automatically adapts itself to changes in sampling frequency/bit rate, which is particularly advantageous when very low bandwidths are in use.

Abstract: Proposed by the invention is a device for the bidirectional transfer of audio and/or video signals, in particular in the context of sound and/or image reports, with: at least one means (6) for providing an audio input signal; a first mixing device (10), which is connected to the means (6) for providing the audio input signal and which is designed to output a mixed audio transmission signal; a transmission and/or reception device coupled to the first mixing device (10) for transmitting the mixed audio transmission signal and/or receiving an audio reception signal; a control device (22) coupled to the first mixing device (10) for controlling the first mixing device (10); a compression and/or decompression device (12, 32) for compression of the mixed audio transmission signal or, as the case may be, for decompression of the audio reception signal, which compression and/or decompression device is connected to the first mixing device (10) for taking up the mixed audio transmission signal or, as the case may

Abstract: A perceptual encoder divides an audio signal into successive time blocks, each time block is divided into frequency bands, and a scale factor is assigned to each of ones of the frequency bands. Bits per block increase with scale factor values and band-to-band variations in scale factor values. A preliminary scale factor for each of ones of the frequency bands is determined, and the scale factors for the each of ones of the frequency bands is optimized, the optimizing including increasing the scale factor to a value greater than the preliminary scale factor value for one or more of the frequency bands such that the increase in bit cost of the increasing is the same or less than the reduction in bit cost resulting from the decrease in band-to-band variations in scale factor values resulting from increasing the scale factor for one or more of the frequency bands.

Abstract: An audio system limits gain in individual audio channels having a shared clip detect signal from a multi-channel power amplifier such that only channels likely to be exceeding the distortion threshold are gain limited. In addition, any arbitrary channels may be grouped together for common gain limiting. The invention monitors the power level of each audio channel, compares the power level to a power threshold that indicates whether a high power condition with the potential to cause excess distortion exists or not, and makes a decision whether to activate each channel's gain limiter based on whether the corresponding clip detect signal is active and the high power condition exists simultaneously for that channel.

Abstract: An audio signal encoding apparatus includes a device for compressing multiple-channel digital audio signals into compression-resultant multiple-channel signals respectively. The multiple-channel digital audio signals relate to a sampling frequency and a quantization bit number. The compression-resultant multiple-channel signals, a signal representative of the sampling frequency, and a signal representative of the quantization bit number are formatted into a formatting-resultant signal. The formatting-resultant signal contains a sub packet and a sync information portion. The sub packet contains at least portions of the compression-resultant multiple-channel signals. The sync information portion contains the signal representative of the sampling frequency and the signal representative of the quantization bit number.

Abstract: A digital audio dynamic range compressor includes a root mean square estimator receiving first and second audio input samples and generating root mean square values of the samples. A gain calculator receives the root mean square values and computes a gain for each input sample in the linear domain, not in the logarithmic or dB domain. A minimum selector receives the computed gain of each input sample and determines a minimum. An attack and release filter receives the minimum gain value and filters the minimum gain value according to attack and release coefficients and generate a gain output. A multiplier receives the gain output and multiplies the first and second audio input samples with the gain output.

Abstract: As a desired digital signal processor (DSP) mode is selected from DSP setting buttons displayed in a DSP setting window, only those slide bars for changing parameters of the selected DSP mode are displayed. As a slider bar corresponding to a parameter to be changed is selected from the slide bars displayed in the DSP setting window, an area of an impulse response diagram to be influenced by a change in the parameter value is indicated in a discriminatory manner by an arrow or the like. A user can visually confirm the acoustic effect to be changed by the value of each DSP parameter and can change the parameter value with ease.

Abstract: An audio system limits gain in individual audio channels having a shared clip detect signal from a multi-channel power amplifier such that only channels likely to be exceeding the distortion threshold are gain limited. In addition, any arbitrary channels may be grouped together for common gain limiting. The invention monitors the power level of each audio channel, compares the power level to a power threshold that indicates whether a high power condition with the potential to cause excess distortion exists or not, and makes a decision whether to activate each channel's gain limiter based on whether the corresponding clip detect signal is active and the high power condition exists simultaneously for that channel.

Abstract: A digital audio decoder decodes or expands compressed data such as bit stream data, which are compressed based on the MPEG/Audio standard. Inverse quantization circuits perform inverse quantization on plural bit stream data, which are supplied thereto in connection with multiple channels respectively, thus producing inversely quantized data with respect to a prescribed number (e.g., thirty two) of sub-band samples respectively. The inversely quantized data are combined together among the multiple channels with respect to the prescribed number of the sub-band samples respectively. Then, a filter bank synthesizes together combined data corresponding to all of the sub-band samples, thus reproducing original digital audio signals. Multipliers are provided for use in gain control on the inversely quantized data with respect to the sub-band samples respectively.

Abstract: A system for sharing wavelet domain components among encoded signals receives a set of signals decomposed and encoded according to a wavelet transform. The decomposed and encoded signals each include a set of wavelet coefficients at each level of the decomposition of the encoded signal. Using a vector quantization technique, the system identifies one or more sets of wavelet coefficients that are sharable among two or more of the decomposed and encoded signals at a particular level of decomposition. The system then stores the sets of wavelet coefficients of the decomposed and encoded signals. Each identified sharable set of wavelet coefficients at a particular level of decomposition is stored only once and shared by two or more of the decomposed and encoded signals.

Abstract: In a dynamics compressor, an analog signal to be compressed is supplied to a first amplifier stage as an input signal, and the output signal thereof is supplied to the next amplifier stage as an input signal, and so on for a number of successive amplifier stages. Each amplifier stage amplifies its input signal with a stage amplification until it reaches a stage limit level. The analog signal and the respective output signals of the amplifier stages are summed in a summation stage to form a sum signal. Above a minimal level, the magnitude of the curve of the sum signalroughly corresponds to an exponent characteristic of the magnitude of the analog signal.

Abstract: A conventional audio/video transmission system including an audio/video transmitter apparatus and an audio/video receiver apparatus suffers from delay of video relative to audio resulting from reproducing audio from the audio signal output from the audio/video transmitter apparatus and reproducing video from the video signal output from the audio/video receiver apparatus. To solve this inconvenience, according to the invention, in an audio/video transmission system including an audio/video transmitter apparatus and an audio/video receiver apparatus, the audio/video transmitter apparatus additionally has an audio delay circuit that outputs the input analog audio signal with a predetermined delay time.

Abstract: This invention generally relates to audio signal processing apparatus and methods for altering, and particularly increasing, the perceived level of bass frequencies in an audio signal. The apparatus comprises an audio input (202) to receive an audio input signal; a compressor (204) coupled to the audio input and having an output, to compress said audio input signal; a high-cut filter coupled to the output of said compressor to provide a filtered compressor output; and a combiner (206) to combine a signal from said compressor output with a signal from said audio input to provide a combined audio output; and wherein said compressor is configured to distort said audio input signal such that said distortion is perceivable as an increase in the level of bass in said combined audio output.

Abstract: A method (200) and apparatus (100) for classifying a homogeneous audio segment are disclosed. The homogeneous audio comprises a sequence of audio samples (x(n)). The method (200) starts by forming a sequence of frames (701-704) along the sequence of audio samples (x(n)), each frame (701-704) comprising a plurality of the audio samples (x(n)). The homogeneous audio segment is next divided (206) into a plurality of audio clips (711-714), with each audio clip being associated with a plurality of the frames (701-704). The method (200) then extracts (208) at least one frame feature for each clip (711-714). A clip feature vector (f) is next extracted from frame features of frames associated with the audio clip (711-714). Finally the segment is classified based on a continuous function defining the distribution of the clip feature vectors (f).

Abstract: The present invention includes methods of preprocessing soundtracks adapted to hearing impairment, environmental noise or other factors and making the preprocessed soundtracks available as alternate soundtracks. Particular aspects of the present invention are described in the claims, specification and drawings.

Abstract: A feedback controller, embedded in an FM audio processor, that controls the “integrated multiplex power” to the requirements of ITU-R 412 (2.51) is disclosed. This regulation specifies the maximum power, which is the same as the power produced by a sinewave modulating the carrier ±19 kHz (where ±75 kHz is 100% peak modulation). The controller applies the square of the multiplex signal to an integrator. A constant is removed from the integrator; representing the maximum power threshold. The output of the integrator is sampled periodically with a sample-and-hold circuit. A second integrator receives the output of the sample-and-hold. The output of the second integrator circuit is added to the threshold setting of a compressor, which determines the average power output of the compressor. The output of the compressor is applied through a peak controller to the input of the multiplex coder, closing a feedback loop, which controls the integrated multiplex power to a preset threshold.

Abstract: A feedback controller, embedded in an FM audio processor, that controls the “integrated multiplex power” to the requirements of ITU-R 412 (2.51) is disclosed. This regulation specifies the maximum power, which is the same as the power produced by a sinewave modulating the carrier ±19 kHz (where ±75 kHz is 100% peak modulation).

Abstract: Apparatus and methods for audio compression and frequency shifting retain the spectral shape of an audio input signal while compressing and shifting its frequency. The fast Fourier transform of the input signal is generated, to allow processing in the frequency domain. The input audio signal is divided into small time segments, and each is subjected to frequency analysis. Frequency processing includes compression and optional frequency shifting. The inverse fast Fourier transform function is performed on the compressed and frequency shifted spectrum, to compose an output audio signal, equal in duration to the original signal. The output signal is then provided to the listener with appropriate amplification to insure audible speech across the usable frequency range.

Abstract: A system for extracting a signal component from an output signal of a dynamic system. The system comprises a state observer and a corrector filter. The state observer is adapted to track a signal component that represents dominant dynamics in the output signal of the dynamic system and provide an estimation signal representing an estimated signal component in the output signal of the dynamic system. The corrector filter is adapted to compensate for a mismatch between the estimation signal and the actual signal component that represents the dominant dynamics in the output signal. A combination of the estimation signal with an output signal of the corrector filter can provide a synthesized signal including the signal component that represents the dominant dynamics in the output signal.

Abstract: An encoding device (100) includes (i) a first encoding unit (132) that encodes spectral data in the lower frequency band represented by a plularity of parameters, out of the spectral data obtained by transforming an audio signal inputted for a fixed time length, (ii) a second quantizing unit (133) that generates sub information representing characteristics of the spectral data in the higher frequency by fewer parameters than those for the lower frequency band, out of the spectral data obtained by the transformation, (iii) a second encoding unit (134) that encodes the generated sub information, and (iv) a stream output unit (140) that outputs the data encoded by the first encoding unit (132) and the data encoded by the second encoding unit (134).

Abstract: An apparatus for discriminating between signals produced by a desired acoustic source and an undesired acoustic source includes: a first microphone disposed at a first distance from the desired acoustic source; at least a second microphone disposed at a second distance from the desired acoustic source; a proximity estimation block configured to utilize signals form the first microphone and second microphone to produce a signal representing an estimate of the proximity of the desired acoustic source; and a variable gain block configured to adjust the gain of an output signal from at least one of the microphones by utilizing the estimate of the proximity of the desired acoustic source.

Abstract: A companding amplifier wherein compression data is provided to an expansion circuit along with audio data. An analog audio input signal is compressed and converted into digital data by a compression circuit. The compression circuit includes a data framer that produces a digital data stream that contains both the digital audio data and digital compression data. The resultant digital data stream can then be recorded on a medium such as a CD or DAT, or transmitted over a medium such as wire, fiber optic or radio waves. The digital data stream can then be read back or otherwise received by an expansion circuit that includes a data separator that separates the digital audio data and digital compression data. The digital audio data is then converted back to an analog audio signal, and the digital compression data is converted to an analog control signal that is used as the control signal for expansion.