Abstract: A dynamic range manipulation system, for use for example in an audio playback system such as a two-way communication system, mobile telephone, MP3 player, and the like, obtains a measure of ambient noise at the playback location and adjusts the gain of the drive signal provided to the loudspeaker based on this measure and based on the audio signal to be played back.

Abstract: A method and/or apparatus for encoding and/or decoding an audio signal is disclosed, in which a downmix gain is applied to a downmix signal in an encoding apparatus which, in turn, transmits, to a decoding apparatus, a bit stream containing information as to the applied downmix gain. The decoding apparatus recovers the downmix signal, using the downmix gain information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the encoding apparatus can apply an arbitrary downmix gain (ADG) to the downmix signal, and can transmit a bit stream containing information as to the applied ADG to the decoding apparatus. The decoding apparatus recovers the downmix signal, using the ADG information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the method and/or apparatus can also vary the energy level of a specific channel, and can recover the varied energy level.

Abstract: A variable equalizer apparatus for forward and/or reverse equalizers in an amplifier. The system can include a structure to allow continuous contact of the signal flow as an equalizer is removed; jumpers with fixed resistors and associated capacitors and inductors to produce a variable range over many different values; and/or variable resistance potentiometers with fixed resistors and associated capacitors and inductors to produce a variable range over separate value equalizers.

Abstract: A method and/or apparatus for encoding and/or decoding an audio signal is disclosed, in which a downmix gain is applied to a downmix signal in an encoding apparatus which, in turn, transmits, to a decoding apparatus, a bitstream containing information as to the applied downmix gain. The decoding apparatus recovers the downmix signal, using the downmix gain information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the encoding apparatus can apply an arbitrary downmix gain (ADG) to the downmix signal, and can transmit a bitstream containing information as to the applied ADG to the decoding apparatus. The decoding apparatus recovers the downmix signal, using the ADG information. A method and/or apparatus for encoding and/or decoding an audio signal is also disclosed, in which the method and/or apparatus can also vary the energy level of a specific channel, and can recover the varied energy level.

Abstract: When an inputted audio signal is outputted from a speaker device having a predetermined input-output characteristic, the predetermined input-output characteristic being selected such that the linearity of an output level with respect to an input signal is approximately assured at a level equal to a predetermined level or more and the output level with respect to the input signal is lowered at a level equal to the predetermined level or less, a correction process for compensating a lowered output level is carried out with respect to a signal component approximately of a level equal to the predetermined level or less in the inputted audio signal. Owing to the correction process, the reproduction characteristic of a small volume signal from a speaker is improved.

Abstract: A request is received to play an audio file. A determination is made regarding whether volume normalization parameters associated with the audio file are stored in a media library. If the volume normalization parameters associated with the audio file are stored in the media library, the volume normalization parameters are retrieved from the media library. If the volume normalization parameters associated with the audio file are not stored in the media library, retrieving the volume normalization parameters from the audio file. The volume normalization parameters are applied while playing the audio file. The volume normalization process can be applied across multiple audio files during playback.

Abstract: A demodulator (10) converts television signals to video baseband signals and audio baseband signals including stereo signals representing a right channel signal value and a left channel signal value. A DSP (60) recursively finds a preferred coefficient value for a scaling that reduces stereo separation due to amplitude variation of the right and left channel signal values. The preferred coefficient value is thereafter used for scaling the right and left channel signal values.

Abstract: In a method and hearing aid (200) for processing sound signals for hearing impaired persons by providing multi-band compression processing an input sound signal is filtered by a band split filter (202) into a number of frequency bands to obtain band split signals. A signal level for each of the band split signals is determined and the frequency bands are arranged into a number of groups. Based on the signal levels in each of the groups, a compressor input level for a number of band split compressors each associated to one of the frequency bands is calculated. A compressor gain for each band split compressor is determined based on the corresponding compressor input signal and the band split signals are amplified with the corresponding compressor gain and summed in a summing unit (208) to produce an output sound signal.

Abstract: A distortion compensation system minimizes distortion in an audio system by monitoring a supply voltage and adjusting a clipping threshold and/or compression knee. An adjustable gain circuit controls the gain of the audio signal according whether the audio signal exceeds a variable threshold. The variable threshold is adjusted within a threshold range based on the supply voltage. Distortion due to clipping of the audio signal is minimized while available power at any given time is maximized.

Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values.

Abstract: A method for automatically tuning a frequency modulator in a mobile device is described. A frequency band is automatically scanned using a frequency modulation (FM) receiver. The FM receiver is integrated as a part of the mobile device. Quality associated with channels of the frequency band is analyzed to identify at least one available channel at a first frequency. The first frequency is assigned to an FM modulator. The FM modulator is integrated as a part of the mobile device. A determination is made whether a command to scan for a second frequency is received. If the command to scan for the second frequency is not received, a signal on the first frequency is transmitted by the FM modulator.

Abstract: Scaling, by a desired amount sm, the overall perceived loudness Lm of a multichannel audio signal, wherein perceived loudness is a nonlinear function of signal power P, by scaling the perceived loudness of each individual channel Lc by an amount substantially equal to the desired amount of scaling of the overall perceived loudness of all channels sm, subject to accuracy in calculations and the desired accuracy of the overall perceived loudness scaling sm. The perceived loudness of each individual channel may be scaled by changing the gain of each individual channel, wherein gain is a scaling of a channel's power. Optionally, in addition, the loudness scaling applied to each channel may be modified so as to reduce the difference between the actual overall loudness scaling and the desired amount of overall loudness scaling.

Abstract: A hearing aid includes a microphone for conversion of sound into an input audiosignal, a signal processor for processing the input audiosignal, the signal processor including a compressor, and a receiver for conversion of the processed signal into sound, wherein the compressor is configured to adapt attack and release time constants in response to input signal fluctuations.

Abstract: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: A system and method for graphic equalization of audio signals is disclosed. Traditional graphic equalizers provide control over the gains in each of a set of frequency bands. However, the actual band gains vary from the desired gains due to crosstalk between bands. Prior art methods for addressing this difficulty include applying a correction filter to the equalizer, and adjusting the shape of the individual band filters, both of which increase the computational cost. In an embodiment of the present invention, the input gains are processed to produce a set of adjusted gains which take into account the crosstalk, and result in an equalization interpolating the input band gains.

Abstract: The invention provides a softscaled frequency compensation function that allows the evaluation of a first quality measure indicating a global impact of all distortions in an audio transmission system, including linear frequency response distortions and second quality measure that only lakes into account the impact of linear frequency response distortions. The softscaled frequency compensation function is derived from a softscaled ratio between a time integrated output and a time integrated input power density functions. The first quality measure is derived from the difference loudness density function as function of time and frequency, using the frequency compensated input loudness density function and the gain compensated output loudness density function both as a function of time and frequency, in the same manner as carried out in ITU standard P.862.

Abstract: A method and an apparatus for switching speech or audio signals, wherein the method for switching speech or audio signals includes when switching of a speech or audio, weighting a first high frequency band signal of a current frame of speech or audio signal and a second high frequency band signal of the previous M frame of speech or audio signals to obtain a processed first high frequency band signal, where M is greater than or equal to 1, and synthesizing the processed first high frequency band signal and a first low frequency band signal of the current frame of speech or audio signal into a wide frequency band signal. In this way, speech or audio signals with different bandwidths can be smoothly switched, thus improving the quality of audio signals received by a user.

Abstract: A control system comprising a computing device, a memory device comprising a database, and a plurality of audio sources. The memory device is coupled to the computing device. The database comprises input data. The computing device is adapted to control an audio level for each of said audio sources in response to said input data to resolve audio conflicts between the audio sources.

Abstract: An audio output apparatus has a measuring circuit which measures the levels of at least two sound signals, a sound level adjusting module (a sound level adjusting circuit and a gain control circuit) which adjusts a sound level so as to equal the levels of the sound signals based on the levels measured at the measuring circuit, and an array speaker unit (a delay circuit, a multiplier, an adder, an amplifier and a speaker unit) which emits sounds in accordance with the sound signals outputted from the sound level adjusting module in different directivities respectively.

Abstract: Systems for an expander circuit are disclosed. An expander circuit comprising a transmit amplifier for amplifying electrical signals received from a microphone. A rectifier circuit is responsive to the electrical signals and generates a D.C. voltage. A dynamic resistor circuit is coupled to the transmit amplifier for regulating the gain of the transmit amplifier. The dynamic resistor circuit includes a bipolar junction transistor for receiving the D.C. voltage. The bipolar junction transistor is operated such that the collector current is zero and the bipolar junction transistor has a variable impedance seen by the transmit amplifier dependent upon the D.C. voltage.

Abstract: A noise reduction system is used in a BTSC system to reduce noise of an audio signal. The noise reduction system has an audio spectral compressing unit that has a filter and a memory in the approach of the digital processing. The filter is arranged to filter an input signal according to a transfer function, a variable d, and several parameters b0/a0, a0/b0, b1/b0 and a1/a0. The memory is arranged to store the parameters.

Abstract: An integrated audio transducer with associated signal processing electronics is disclosed. A silicon audio transducer, such as a MEMS microphone or speaker, can be integrated with audio processing electronics in a single package. The audio processing electronics can be configured using control signals. The audio processing electronics can provide a single line serial data interface and a single line control interface. The audio transducers can be integrated with associated processing electronics. A silicon microphone can be integrated with an Analog to Digital Converter (ADC). The ADC output can be a single line serial interface. The ADC can be configured using a single line serial control interface. A speaker may be integrated with a Digital to Analog Converter (DAC). Audio transducers can also be integrated with more complex processing electronics. Audio processing parameters such as gain, dynamic range, and filter characteristics may be configured using the serial interface.

Abstract: Automatic measurements are made of audio presence and level in an audio signal by direct processing of an MPEG data stream representing the audio signal, without reconstructing the audio signal. Sub-band data is extracted from the data stream, and the extracted sub-band data is dequantized and denormalized. An audio level for the dequantized and denormalized sub-band data is measured without reconstructing the audio signal. Channel characteristics are used in measuring the audio level of the sub-band data, wherein the channel characteristics are used to weight the measured levels. The measured levels are compared against at least one threshold to determine whether an alarm should be triggered.

Abstract: A personal audio playback device having gain control responsive to environmental sounds provides for improved enjoyment of program material played back through headphones, while further providing features for personal safety and communications with others. A microphone is incorporated on the surface of the playback device, which includes an audio output connection for headphones and internal storage for audio program material. The entire device may be incorporated within the headphones, or the headphones may connect through a connector on the housing of the device. The gain, type or position of the program material is controlled in conformity with a detected characteristic of ambient sounds received by the microphone, which may be the amplitude of the signals in one or more frequency bands, or a particular type of sound, such as speech or vehicular sounds. Multiple modes are selectable for processing the audio, selecting program material type and/or re-positioning the program material.

Abstract: A system and method for achieving extended low-frequency response and increased low-frequency sound pressure output capability in a loudspeaker system is provided. The system and method comprise mounting a low-frequency driver in a ported box, tuning the ported box to a sufficiently low frequency so as to result in a frequency response that can be modeled substantially as a second-order response, and equalizing the response of said driver-box combination with a second-order biquadratic filter function to achieve the desired frequency response characteristic.

Abstract: The present invention relates to a method for preventing an output device from being damaged, which comprises the steps of: dividing the output value of an output device into a plurality of areas, defining a ratio value to every area, receiving a desired output value, determining the current area where the desired output value device is located, and calculating the desired output value with the ratio value to obtain an actual output volume and output the actual output volume.

Abstract: An audio processing chip includes a connecting port, an audio amplifier module and a pulse width modulation (PWM) control circuit. The connecting port receives a pulse width modulation (PWM) signal; the audio amplifier module amplifies an audio signal according to a control signal to thereby output an audio output signal; and the pulse width modulation (PWM) control circuit is coupled between the connecting port and the audio amplifier module, and outputs the control signal to the audio amplifier module according to the PWM signal to thereby control an operation of the audio amplifier module.

Abstract: A signal processing device includes: a frequency conversion processing unit that sets, as a processing target signal, a section in which a peak signal level exceeds a first threshold in an input sound signal and applies frequency conversion processing to the processing target signal to acquire power levels in respective plural bands; and an amplitude compressing unit that executes, when a power level exceeding a second threshold is present among the power levels in the respective plural bands acquired by the frequency conversion processing unit, amplitude compression processing for compressing a signal level of the processing target signal at a compression ratio at which the peak signal level of the processing target signal falls within the first threshold and, otherwise, prohibits the execution of the amplitude compression processing.

Abstract: An audio signal processing apparatus includes: a first filtering unit which outputs an audio signal while attenuating frequency components except for preset frequency components; a detecting unit which detects a sound volume level; an amplitude limiting unit which calculates an amplitude limiting level corresponding to the sound volume level, and which limits a part of a waveform of the audio signal output from the first filtering unit; a second filtering unit which outputs the audio signal output from the amplitude limiting unit while attenuating frequency components except for preset frequency components including a part of the frequency band of the audio signal output from the first filtering unit, and a part of the frequency band of the harmonics; a compressing unit which compresses a dynamic range of the audio signal; and an adding unit which adds the audio signal output from the compressing unit to the input audio signal.

Abstract: An audio data processing device including: a first processor; and a second processor which is connected to the first processor wherein the first processor includes: an audio data acquisition which acquires audio data of digital data; an omitting section which omits a bit corresponding to low volume which is hard to be heard by human ears from the audio data; and a transmitter which transmits the audio data in which the bit corresponding to the low volume is omitted by the omitting section from the first processor to the second processor; wherein the second processor includes: a receiver which receives the audio data transmitted from the first processor; and a reproduction data generator which generates audio reproduction data necessary to reproduce the audio data based on the received audio data.

Abstract: A spectral enhancement system is disclosed that includes an input node for receiving an input signal, at least one broad band pass filter coupled to the input node and having a first band pass range, at least one non-linear circuit coupled to the filter for non-linearly mapping a broad band pass filtered signal by a first non-linear factor n, at least one narrow band pass filter coupled to the non-linear circuit and having a second band pass range that is narrower than the first band pass range, and an output node coupled to the narrow band pass filter for providing an output signal that is spectrally enhanced.

Abstract: A system for amplifying a digital audio signal comprises a receiver 12 for receiving a digital audio signal, a level estimator 14 arranged to calculate the audio level of the digital audio signal, a gain control 16 arranged to receive a gain level, the gain level defining the desired amplification of the digital audio signal, a logic circuit 18 arranged to calculate the headspace in the digital audio signal and to divide the gain level into a scaling gain and an amplifier gain, the scaling gain not exceeding the calculated headspace, a digital signal processor 20 arranged to amplify the digital audio signal with the scaling gain, a digital-to-analogue converter 22 arranged to convert the amplified digital audio signal into an analogue signal, and an amplifier 24 arranged to amplify the analogue audio signal with the amplifier gain.

Abstract: Disclosed is an automatic audio distortion control method and apparatus, wherein the automatic audio distortion control apparatus comprises an amplifier (6) and a feedback loop which has its both ends coupled to an input and an output of the amplifier (6), wherein said feedback loop has clipping distortion of signals outputted from the amplifier (6) as a control parameter for automatic control on the distortion of the amplifier (6). Once the outputted level gets close to a limit-value of speakers, the method and the apparatus will regulate power amplifier gain automatically, in order to control distortion, prevent damage to the speakers, and provide compatibility with high or low levels inputted from various audio sources.

Abstract: For synthesizing at least three output channels using two stereo input channels, the stereo input channels are analyzed to detect signal components occurring in both input channels. A signal generator is operative to introduce at least a part of the detected signal components into the second channel associated with a second speaker in an intended speaker scheme, which is positioned between a first and a third speaker in the speaker scheme. When, however, feeding of the complete detected signal components would result in a clipping situation, then only a part of the detected signal components is fed into the second channel as a real center channel and the remainder is located in the first and third channels as a phantom center channel.

Abstract: A method for signal processing for a hearing aid aims to better match signal processing for a hearing aid and in particular a hearing device to a situation and includes processing an input signal in accordance with a first processing algorithm to form a first intermediate signal and processing the input signal in accordance with a second processing algorithm to form a second intermediate signal in parallel with the processing of the input signal in accordance with the first processing algorithm. The input signal is classified by a classifier. Finally, an output signal with a constant mixture ratio is formed both from the first and from the second intermediate signals, taking into account the result of the classification. This allows the advantages of a plurality of algorithms to be used at the same time. A corresponding hearing aid is also provided.

Abstract: At least one exemplary embodiment is directed to a method of generating a Personalized Audio Content (PAC) comprising: selecting Audio Content (AC) to personalize; selecting an Earprint; and generating a PAC using the Earprint to modify the AC.

Abstract: A clipping prevention device, includes: a compression section adapted to compress an input digital audio signal level; a digital-analog conversion section adapted to operate at a predetermined first operating voltage and convert the digital audio signal into an analog audio signal; an electronic volume control adapted to operate at a second operating voltage and amplify or attenuate the analog audio signal with a user-changeable amplification factor; and a control section adapted to calculate a clipping level based on the maximum amplification factor of the electronic volume control and the user-changeable amplification factor, the maximum amplification factor being determined when the analog audio signal at the maximum signal level is amplified to the maximum signal level, and also adapted to control the compression section so that the signal is compressed so as to prevent clipping of the analog audio signal amplified or attenuated by the electronic volume control.

Abstract: Psychophysical tests are administered to cochlear implant (CI) users to determine a spectral modulation transfer function (SMTF), the smallest detectable spectral contrast as a function of spectral modulation frequency, for each individual CI user. The determined SMTF for individual CI user is compared against a SMTF of a normal hearing person to determine the specific enhancements needed. A spectral contrast enhancement that best fits the needed enhancements for the individual CI user is selected, and a sound processing strategy is adjusted to provide customized spectral contrast enhancement for the individual CI user. The sound processing strategy implemented includes an outer hair cell model.

Abstract: An apparatus (12) which overcomes the foregoing inconveniences of manually adjusting the audio device (10) volume related to a vessels speed and participants distance behind the vessel in a towable activity. This apparatus (12) controls audio volume levels (14) set at V2 and (16) set at V1 where V2 is greater than V1 and audio volume level (14) is achieved when vessels engine or propulsion shaft RPM is above set point (22) for a time greater than designated by (24), thereby controlling audio volume as a relation to vessel speed.

Abstract: Bandwidth expansion for audio signals by frequency band translations plus adaptive gains to create higher frequencies; use of a common channel for both stereo channels limits computational complexity. Adaptive cut-off frequency determination by power spectrum curve analysis, and bass expansion by both fundamental frequency illusion and equalization.

Abstract: A satisfactory sound volume sensation from a hearing point of view can be obtained, and an effective sound volume correction operation can be obtained with an algorithm as simple as possible. Correction step 0 (correction function off, level correction value 0) to correction step 16 (level correction value—16 dB) are defined. When a first condition that is defined in such a manner as to correspond to a state in which distortion occurs is satisfied, the correction step is made to proceed by one step from the current correction step (?1 dB is added to a level correction value). When a second condition that is defined in such a manner as to correspond to a state in which distortion does not occur is satisfied, the correction step is made to return by one step from the current correction step (?1 dB is subtracted from the level correction value).

Abstract: Systems and methods for providing protection from failure events in a digital audio amplification system. One embodiment of the invention comprises a system having a digital amplifier controller, an amplifier output stage coupled to the controller and configured to receive audio signals from the controller, one or more sensors coupled to the output stage and one or more low-pass filters coupled to receive sensor signals from the one or more sensors. The low-pass filter is configured to filter the sensor signals and to provide the filtered sensor signals to the controller, which provides a programmable response to the filtered sensor signals. The response may range from not taking any action, to limiting the amplification of audio signals, to shutting down the system.

Abstract: The invention describes a graphical method for detecting and adjusting audio overload conditions. The graphical user interface provides a user complete playback control of several audio tracks, detection of overload conditions such as audio clipping, and graphical methods to correct the overload conditions. The graphical interface provides drag handles which the user can use to adjust the various characteristics of an audio file. The characteristics, such as amplitude and tempo, may be adjusted as a function of time.

Abstract: An acoustic shock protection method and device are provided. A pattern analysis-based approach is taken to an input signal to perform feature extraction. A parameter space is identified, which is corresponding to the signal space of the input signal. A rule-based decision approach is taken to the parameter space to detect an acoustic shock event. The device may be advantageously implemented using a weighted overlap-add approach to provide low group delay, high-fidelity and a high degree of protection from acoustic shock events.

Abstract: A level adjustment method is applicable to an audio processing apparatus having a plurality of amplifiers corresponding to three or more of input channels of an audio signal for amplifying signal levels of the respective input channels. The level adjustment method is carried out by a group arrangement process of arranging the plurality of the input channels into one or more group, and a group control process of controlling each group such as to decrease an amplification rate of all the amplifiers corresponding to the input channels belonging to the same group as a maximal one of the signal levels of the input channels belonging to the same group increases. Typically, the group arrangement process is applied to an audio signal of a surround system having at least six input channels including a left channel, a right channel, a left surround channel, a right surround channel, a center channel and an LFE channel.

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: 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: A method and apparatus for detecting speech segments of a speech signal processing device is provided. A critical band is divided into a certain number of regions according to noise frequency characteristics, a signal threshold and a noise threshold are set for each of the regions, and it is determined whether each frame is a speech segment or noise segment by comparing the log energy calculated for each region to the corresponding signal threshold and noise threshold. Therefore, a speech segment can be detected rapidly and accurately by using a small number of operations even in a noise environment.

Abstract: A coded signal conveys encoded audio information and metadata that may be used to control the loudness and dynamic range of the audio information during its playback. If the values for these metadata parameters are set incorrectly, annoying fluctuations in loudness during playback can result. The present invention overcomes this problem by detecting incorrect metadata parameter values in the signal and replacing the incorrect values with corrected values.

Abstract: An expander circuit for decoding audio signals which were encoded in accordance with the BTSC multichannel sound system standard including DBX commanding is presented. A wideband expander circuit is utilized and a DBX expander can be accommodated. The wideband expander circuit is provided with a signal path having an input LPF (low pass filter), a stereo difference signal (L?R) demodulator, a second LPF, and a voltage controlled amplifier (VCA), the gain of which is controlled by a control signal derived from the demodulated difference signal which has been operated on by a bandpass filter (BPF) and an integrating peak detector. The output of the VCA is provided to a deemphasis network before being fed to a decoder matrix for combining with the sum stereo signal (L+R) for reconstructing the original L and R signals.