Abstract: A method and apparatus are provided for generating a noise reduced output signal from sound received by a first microphone. The method includes transforming the sound received by the first microphone into a first input signal and transforming sound received by a second microphone into a second input signal. The method includes calculating, for each of a plurality of frequency components, an energy transfer function value as a real-valued quotient by dividing a temporally averaged product of an amplitude of the first input signal and the second input signal by a temporally averaged absolute square of the second input signal, calculating a gain value as a function of the calculated energy transfer function value, and generating the noise reduced output signal based on the product of the first input signal and the calculated gain value at each of the plurality of frequency components.

Abstract: An earpiece designed to fit a substantial majority (over 90%) of people without customization to the outer ears, i.e., without the need to make customized measurements or a mold of the actual ear of an individual. The device is generally C-shaped and contacts the outer ear in four locations.

Abstract: A computer-implemented method comprising: determining one or more features of a subject signal; revising one or more control signals on the basis of the one or more features; modifying a level of the subject signals based on the control signals. At least one of the features is determined by: comparing a given one of the subject signals against a boundary signal to produce a corresponding given boundary comparison signal; and summarizing the behavior of the given boundary comparison signal over a time interval.

Abstract: A method of processing audio signals for an auditory prosthesis is provided. The method includes a noise reduction step and a later compression step. A gain control step is provided prior to the compression step. The gain control step operates so as to minimize the occurrence of signal compression in the compression step. An auditory prosthesis arranged to provide the method is also provided.

Abstract: Improving decoding of a set of k data symbols received from several receivers, the data symbols being encoded by a systematic block error correcting code of dimension k and size n. The set of data symbols is received along with a corresponding subset of parity symbols, forming a partial data block comprising m symbols. A partial data block transmitted by one emitter, comprising a set of k data symbols and a subset of (m?k) parity symbols, is received from each receiver. For each received partial data block, a subset of parity symbols is generated and an item of reliability information is computed as a function of the received parity symbols and parity symbols generated from a received set of data symbols. The items of computed reliability information are compared with each other to select one received set of data bits.

Abstract: A system and method is provided for converting Dynamic Range Control/Compression (DRC) gain values into a spline representation that is compatible with the current standards. The system and method may: 1) minimize the bitrate for encoding and/or 2) minimize the approximation error between reference gain and interpolation values. A strategy for bitrate minimization may be the reduction of the number of spline nodes since gain and slope information must be transmitted for each node. Accordingly, an efficient heuristics based approach is provided that reduces the number of spline nodes needed to represent a series of DRC gain values using interpolation while accounting for overshoots and other inaccuracies.

Abstract: There is provided a bias arrangement for an amplifier adapted to amplify a varying input signal, the arrangement comprising a control circuit arranged to adaptively vary a bias current to the amplifier in dependence on an envelope of the varying input signal.

Abstract: A method of processing a sound segment is used in a hearing aid. If the sound segment is a high-frequency type, the high-frequency portion of the sound segment will be processed with a frequency lowering process. If the sound segment is a mixed-frequency type (between high-frequency and low-frequency), the energy of at least some portion of the high-frequency portion of the sound segment will be decreased and then processed with a frequency lowering process.

Abstract: Filters 106-1-106-N divide an input signal 100 into N band-limited signals 107-1-107-N, and multipliers 108-1-108-N carry out dynamic range control of the N band-limited signals 107-1-107-N, respectively. After that, filters 111-1-111-N eliminate odd harmonics caused by the dynamic range control, and a signal synthesis unit 113 combines the signals passing through the filters 111-1-111-N into a single output signal 114.

Abstract: An audio signal processing circuit includes a first amplifier to generate a first analog audio signal; a second amplifier to generate a second analog audio signal; an analog/digital converter to generate a first digital audio signal; a digital signal processing unit to output a second digital audio signal; a digital/analog converter to generate differential third analog audio signals; a third amplifier for inverting a positive signal of the third analog audio signals and adding it to a negative signal to generate a single-ended fourth analog audio signal; and a fourth amplifier to generate an output audio signal.

Abstract: An automatic audio signal level adjustment circuit is capable of automatically adjusting a level of an input audio signal within a specific range. The automatic audio signal level adjustment circuit includes an amplitude adjustment determining unit and an amplitude adjusting unit. The amplitude adjustment determining unit is configured to generate an amplitude reduction instruction when the level is greater than a first reference value, and an amplitude augmentation instruction when the level is small than a second reference value. The amplitude adjusting unit is configured to output an output audio signal having an amplitude reduced from that of the input audio signal upon the amplitude reduction instruction, and an output audio signal having an amplitude augmented from that of the input audio signal upon the amplitude augmentation instruction. Further, the amplitude adjusting unit is configured to output an output audio signal equal to the input audio signal upon no instruction.

Abstract: A constant current generation circuit of the invention includes: a temperature variable voltage generation unit that generates a first variation voltage whose voltage value fluctuates with temperature; a variation gradient adjustment unit that generates a second variation voltage based on a reference voltage smaller in the amount of variation with temperature than the first variation voltage and the first variation voltage; and a current generation unit that includes a current setting resistor whose resistance value fluctuates with temperature and generates an output current based on the second variation voltage and the current setting resistor. The variation gradient adjustment unit sets the coefficient of variation with temperature of the second variation voltage so that the difference between it and the coefficient of variation with temperature of the resistance value of the current setting resistor is within a preset first stipulated range.

Abstract: A novel audio ducking method that is aware of the loudness levels of the audio content is provided. The method specifies a minimum loudness separation between audio tracks that are designated as masters and audio tracks that are designated as slaves. The method attenuates the volume of the slave tracks in order to provide at least the minimum loudness separation between the slave tracks and the master tracks. The amount of attenuation for a slave is determined based on the loudness levels of the slave and of a master.

Abstract: The performance of an echo canceller is assessed using a) a test signal launched from originating test equipment and b) a simulated echo of the test signal launched from terminating test equipment. The launch of the simulated echo signal is timed in such a way that it arrives at the tandem echo canceller(s) at a particular point in time relative to the arrival of the test signal, at the tandem echo canceller(s), when the tandem echo canceller(s) is (are) not able to cancel the simulated echo signal. The latter thus arrives uncanceled at the target echo canceller. The launch of the simulated echo signal is further timed in such a way that it arrives at the target echo canceller at a point in time relative to the arrival of the test signal, at the target echo canceller, when the target echo canceller is able to cancel the simulated echo signal.

Abstract: Systems and methods for amplitude compressing a digital signal. An input signal is divided into frames having a first and second sets of samples. The samples in the second set are also in a subsequent frame. Peak values are determined for the first and second sets. One or more slopes are calculated based on the peak values. The slopes are used to define a scale factor which is applied to the first set to produce the output signal. For example, if the first peak value exceeds an amplitude threshold, first and last samples in the first set to exceed the amplitude threshold are found. Slopes are calculated for each of three regions of the first set demarcated by the first and last samples. In each region a slope is selected. These slopes along with an initial scale factor are used to calculate the scale factor.

Abstract: A first differential value is acquired between first current data and first previous data in an i number (i being a natural number) of sampling periods before the current data. A second differential value is acquired between second current data and second previous data in a j number (j being a natural number) of sampling periods before the current data. Both first data and both second data are of a first and a second digital audio signal, respectively, having a sound level of a digital stereo audio signal in the left and right channels, respectively. A first and a second correction coefficient are acquired by adding the first and second differential values at a first and a second ratio, respectively. The first signal is corrected by multiplying the first signal by the first correction coefficient. The second signal is corrected by multiplying the second signal by the second correction coefficient.

Abstract: Embodiments are directed toward user profile based audio adjustment techniques. The techniques are used to render various audio and/or audio/video content having different audio output parameter values in accordance with a user profile that characterizes a user's desired value and/or range of one or more of the output parameter levels.

Abstract: Peak reduction and power limitations are used to prevent distortion and protect components. In a cellular telephone, peak reduction can be based on battery power level to prevent electrical distortion from saturation. In addition peak reduction can be used to prevent mechanical distortion such as rub and buzz. Dynamic range compression can be used for peak reduction. In another application dynamic range compression can be used to control the power output to protect a speaker from damage. One example of a dynamic range compressor/peak limiter comprises a look-ahead buffer and an analysis engine. For example, the look-ahead buffer holds a window of samples of a signal. The analysis engine selects a gain envelope function on the basis of the samples, for example, by selecting the Pth sample in the buffer whenever that sample exceeds a given threshold.

Abstract: For a media clip that includes audio content, a novel method for performing dynamic range compression of the audio content is presented. The method performs an analysis of the audio content. Based on the analysis of the audio content, the method generates a setting for an audio compressor that compresses the dynamic range of the audio content. The generated setting includes a set of audio compression parameters that include a noise gating threshold parameter (“noise gate”), a dynamic range compression threshold parameter (“threshold”), and a dynamic range compression ratio parameter (“ratio”).

Abstract: The present invention features systems for dynamically adjusting audio signals by applying a gain to the signal in a spectrally varying manner to compensate for ambient noise, such that the sound is perceived to be unchanged in volume and spectral composition by the listener. The system obtains a threshold elevation for each frequency component by analyzing the spectral composition of the ambient noise. This threshold elevation is then used by a psychoacoustic model of hearing to determine an appropriate gain adjustment for the corresponding frequency component of the source signal which will make that source signal perceived by the human ear to be just as loud as if the noise were not present.

Abstract: Systems and methods to automatically equalize coloration in speech recordings is provided. In example embodiments, a reference spectral shape based on a reference signal is determined. An estimated spectral shape for an input signal is derived. Using the estimated spectral shape and the reference spectral shape a comparison is performed to determine gain settings. The gain settings comprise a gain value for each filter of a filter system. Using gain values associated with the gain setting, automatic equalization is performed on the input 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: An apparatus for generating a drive signal for a sound transducer (109) comprises a sound generator (101) which provides an input audio signal. A divider (101) divides the input audio signal into at least a low frequency signal and a high frequency signal and an expander (105) generates an expanded signal by applying a dynamic range expansion to the low frequency signal. A combiner (107) then generates the drive signal by combining the expanded signal and the higher frequency signal. The threshold for applying the dynamic range extension may be adjusted depending on the amplitude of the low frequency signal. The low frequency signal may furthermore be compressed into a narrow frequency band around a resonance frequency. The approach may allow improved audio quality especially from high Q low frequency sound transducers by attenuating decay parts of bass signals thereby reducing sustain or ringing for bass notes.

Abstract: In a multiband compressor 100, a level calculation unit 121 calculates a signal level inputted for each of bands, a gain calculation unit 122 calculates a gain value from the calculated signal level, and a gain limitation unit 130 limits a gain value by comparison with a gain value of the other band in a compressor for each band. With this configuration, provided is a multiband compressor capable of achieving a balance between the quality of sound and the effect of enhancing the sound level at a high level.

Abstract: An audio signal in which an audio signal is received as a stream of digital samples, each being a numerical value representing a sampled signal level. A first zero crossing point is identified and the received audio samples are stored until a second zero crossing point is identified, thereby storing a first half-wave of samples. The highest intensity sample is identified from the stored samples and this is compared against a predetermined threshold. All stored samples are scaled by an initial scaling factor so that the intensity of the highest intensity sample is not above this threshold. A second half-wave of samples is stored in which all samples of the second half-wave are below the threshold. All stored samples of the second half-wave are also scaled but by a modified scaling factor derived from a combination of the initial scaling factor and a decay factor.

Abstract: The present invention features systems for adjusting audio signals by applying a gain to the signal in a spectrally varying manner to compensate for ambient noise in the environment of the listener. The system allows a listener to hear what ought to be heard, over the ambient noise, by applying a gain to the source that varies according to the spectral composition of the noise, rather than cancelling or filtering the noise. The spectral composition of the source is thus preserved in the listener's awareness without the removal of the noise signal. After application of these corrective gains to the source, the listener's perception of the source sound is as if the noise was not present. Systems may be incorporated into apparatuses including but not limited to mobile phones and music players.

Abstract: The disclosed BTSC encoder includes a left high pass filter means for receiving a digital left channel audio signal and for digitally high pass filtering the digital left channel audio signal and thereby generating a digital left filtered signal; a right high pass filter means for receiving a digital right channel audio signal and for digitally high pass filtering the digital right channel audio signal and thereby generating a digital right filtered signal; a matrix means for receiving the digital left and digital right filtered signals, and including means for summing the digital left and digital right filtered signals and thereby generating a digital sum signal, and including means for subtracting one of the digital left and digital right filtered signals from the other of the digital left and digital right filtered signals and thereby generating a digital difference signal; a difference channel processing means for digitally processing the digital difference signal; and a sum channel processing means for d

Abstract: A sound signal processing device that is capable of suitably extracting main sound from mixed sound in which unnecessary sound (for example, leakage sound and reverberant sound) is mixed with the main sound. More specifically, a mixed sound signal in the time domain including first sound and second sound, and a target sound signal in the time domain including sound corresponding to at least the second sound, which have temporal relation in their entirety or in part, are each divided into a plurality of frequency bands. A level ratio between the two signals is calculated at each frequency. Based on the level ratio, a signal of the first sound that is included in the mixed sound signal is extracted.

Abstract: The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values (4) based on received audio metadata are sufficient for protection against clipping of the audio signal. The audio metadata is embedded in a first audio stream (1). In case a first gain value (4) is not sufficient for protection, the respective first gain value (4) is replaced with a gain value sufficient for protection against clipping of the audio signal. Preferably, in case no metadata related to dynamic range control is present in the first audio stream (1), the method may add gain values sufficient for protection against signal clipping.

Abstract: Integrated processing of multimedia signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated multimedia system includes a main player that captures and processes signals digitally. The main player may adjust the audio signal to provide audio output of equal loudness across all frequencies by accounting for sensitivity of the human ear for sounds of varying frequencies. The main player can also account for perceived differences in loudness based on the angle of a listener to a speaker by detecting the position of a user and making an adjustment accordingly. The invention further provides a speaker that has embedded performance characteristics or an identifier that allows the system to provide an optimal speaker driving current for a particular system or determine how that speaker would be best implemented in the integrated system.

Abstract: There are provided methods and an apparatus for conditioning an audio signal. According to one aspect of the present invention there is included a method for conditioning an audio signal having the steps of: receiving at least one audio signal, each audio signal having at least one channel, each channel being segmented into a plurality of frames over a series of time; calculating at least one measure of dynamic excursion of the audio signal for a plurality of successive segments of time; filtering the audio signal into a plurality of subbands, each frame being represented by at least one subband; deriving a dynamic gain factor from the successive segments of time; analyzing at least one subband of the frame to determine if a transient exists in the frame; and applying the dynamic gain factor to each frame having a transient.

Abstract: This invention discloses a method and a plurality of compressive amplitude distortion units for enhancing the perceptibility of the low-frequency portion of a sound by introducing residue harmonics of low-frequency signal components into the sound, where the residue harmonics are generated by a nonlinear function emulating the middle-ear response of a human being. The low-frequency portion in the resultant sound is perceivable to a human listener even if this portion is removed from this sound.

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 processing device is designed to enhance the low-pitch register of an audio signal by generating harmonics causing a missing fundamental effect with a light load of processing but without damaging an audio waveform. The audio signal processing device includes a filtering part (e.g. a band-pass filter configured of a high-pass filter and a low-pass filter) that extracts a low-pitch signal from an audio signal input thereto; a dynamic range compression part that compresses a dynamic range of the low-pitch signal by use of a time-variant gain relative to a peak of the low-pitch signal, which is detected via a peak hold operation using a predetermined time constant, thus producing a compressed signal; and an adder that adds the compressed signal to the audio signal so as to produce a processed audio signal including harmonics.

Abstract: An apparatus for upmixing a downmix audio signal describing one or more downmix audio channels into an upmixed audio signal describing a plurality of upmixed audio channels includes an upmixer configured to apply temporally variable upmixing parameters to upmix the downmix audio signal in order to obtain the upmixed audio signal. The apparatus also includes a parameter interpolator, wherein the parameter interpolator is configured to obtain one or more temporally interpolated upmix parameters to be used by the upmixer on the basis of a first complex-valued upmix parameter and a subsequent second complex-valued upmix parameter.

Abstract: The application relates to a compressor circuit for a listening device, the compressor circuit comprising a pair of input terminals and a pair of output terminals, the compressor circuit being adapted for receiving an electric input signal representing an audio signal with an input voltage swing Vipp at the input terminals and for providing a possibly compressed version of the input signal as an output signal with an output voltage swing Vopp at the output terminals, the compressor circuit comprising. The application further relates to a listening device and to the use of the compressor circuit. The object of the present application is to provide a compressor circuit with a proper functioning also at relatively low input levels.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

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: An input audio signal is detected to provide a fast time constant control signal which is clamped for immediate response when the input signal drops below a threshold level and filtered to provide a modified control signal with a slow time constant. As the modified control signal level decreases toward the control signal level, a time constant differential control signal decreases and slows the rate of change of the decreasing modified control signal which, as it nears the control signal level, reverts to a slow response. The resulting control signal has an exponential release response which can be applied to an audio dynamics processor. The time constant differential control signal may also be detected to provide an inverted output when it is significantly less than the modified control signal so as to result in the control signal also having an exponential attack response to be applied to the processor.

Abstract: A method for modifying a current sound-volume value assigned by an on-board system to an acoustic alert signal emitted on board an aircraft is provided. The method includes furnishing, on board the aircraft, the on-board system with at least one information item defining a modification of the current sound-volume value to a new sound-volume value in accordance with a predetermined rule.

Abstract: A method of extending battery life of a wireless microphone unit includes muting the wireless microphone unit responsive to a mute signal from a base station unit, transmitting, by the wireless microphone unit, compressed muted audio data, wherein the compressed muted audio data is compressed via a first compression scheme, determining, by the wireless microphone unit, whether an unmute signal has been received from the base station unit, and responsive to a determination that the unmute signal has been received, unmuting the wireless microphone unit. The method further includes discontinuing transmission of the compressed muted audio data and transmitting compressed audio data via a second compression scheme, wherein the first transmitting step causes the wireless microphone unit to consume less power per unit of transmission time than the second transmitting step.

Abstract: A microphone can include a capacitor capsule, an output buffer amplifier connected to the output of the capacitor capsule, and an audio limiter connected to the output of the output buffer amplifier, wherein the audio limiter limits the output level of the microphone at a threshold level. The microphone can include an adjustable output level. The microphone can include an integrated high-pass filter. The microphone can have an omnidirectional polar pattern.

Abstract: An audio signal processing device includes: a microphone configured to collect noise; an analyzing unit configured to analyze an audio signal collected by the microphone to detect the level and frequency property of the collected audio signal; and a signal processing unit configured to subject an audio signal to be reproduced to signal processing based on the analysis results of the analyzing unit.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: A method for multi-channel processing in a multi-channel sound system, in which a channel or a channel mixture is first split into individual channels, the individual channels are limited by setting the values of the parameters channel fader, threshold, release, and output level and then encoding the individual channels. At least two channels are compressed and/or limited with a uniform output level value in method step, one channel is provided with a deviating output level value, which is set, depending on the audio material to be processed, and every further channel is compressed and/or limited in such a manner that it has an output level value that is at least one decibel less than the uniform output level value. The individual channels are combined into an encoded (coded) channel by setting a value of at least one of the parameters channel fader, threshold, release, and output level.

Abstract: A communications device is configured to control the intelligibility of speech in a downlink voice signal during a call. The device determines a current noise level based on sampling ambient acoustic noise and based on a previously determined noise level. The device then determines an overall output gain and a frequency response based on the current noise level and based on a user-selected volume setting of the device. The device modifies the downlink voice signal during the call in accordance with the determined overall output gain and the determined frequency response. Other embodiments are also described and claimed.

Abstract: A reproducing apparatus includes: a digital signal processing block configured to execute first boost processing for setting a volume level for an entered digital audio signal and boosting an amplitude level of the signal; a digital-to-analog conversion block configured to convert the digital audio signal into an analog one; an analog signal processing block configured to execute second boost processing for boosting an amplitude level of the analog audio signal; an analog volume adjusting block configured to set a volume level for the analog audio signal from the analog signal processing block; a loudspeaker configured to output the analog audio signal from the analog volume adjusting block; an operating block configured to indicate a volume level of an audio signal from the loudspeaker and turn on/off the boost processing for the audio signals; and a control block configured to control components in accordance with an operation by the operating block.

Abstract: In accordance with an embodiment, a method includes determining an amplitude of an input signal provided by a capacitive signal source, compressing the input signal in an analog domain to form a compressed analog signal based on the determined amplitude, converting the compressed analog signal to a compressed digital signal, and decompressing the digital signal in a digital domain to form a decompressed digital signal. In an embodiment, compressing the analog signal includes adjusting a first gain of an amplifier coupled to the capacitive signal source, and decompressing the digital signal comprises adjusting a second gain of a digital processing block.

Abstract: An audio signal output level control method used in an audio device includes detecting input levels of an audio signal, determining control modes to be applied to the audio signal based on the input levels of the audio signal, controlling the input levels of the audio signal according to the control modes, and determining output levels corresponding to the controlled input levels.

Abstract: A method for controlling a maximum signal level output to headphones of a wireless device is provided. The method includes: determining an impedance of the headphones; determining a carrier specific maximum signal level for headphones having the impedance; and, adjusting an audio amplifier of the wireless device coupled to the headphones to restrict the maximum signal level output to the headphones to the carrier specific maximum signal level.