Abstract: A pseudo bass generating apparatus includes a first 4th-order LPF, an absolute value circuit, a clip circuit, a multiplier, a first adder for subtracting an output signal of the multiplier from an output signal of the clip circuit, a second adder for adding an output signal of the first adder and an output signal of the absolute value circuit, a second 4th-order LPF, and a third adder for adding the input signal and an output signal of the second 4th-order LPF.

Abstract: An apparatus for processing a signal and method thereof are disclosed. The present invention includes receiving a downmix signal generated from a plural-channel signal, mix information and phase shift information on the plural-channel signal, upmixing the downmix signal into the plural-channel signal by applying the mix information to the downmix signal, and generating an original plural-channel signal by shifting a phase of at least one channel of the plural-channel signal based on the phase shift information. According to the present invention compensate a reconstructed original plural-channel signal using compensation information (phase shift information), thereby compensating a phase or a gain lost in the plural-channel signal reconstructed by upmixing using mix information.

Abstract: Provided is an encoding/decoding apparatus and method of multi-channel signals. The encoding apparatus and method of multi-channel signals may encode phase information of the multi-channel signals using a quantization scheme and a lossless encoding scheme, and the decoding apparatus and method of multi-channel signals may decode the phase information using an inverse-quantization scheme and a lossless decoding scheme.

Abstract: A frequency detection circuit of a clock regeneration circuit measures time which an input clock takes to change a predetermined number of times, and outputs a count value proportional to the time. A division ratio generation circuit truncates bits of the output of the frequency detection circuit by using a quantizer, and outputs the obtained value as a division ratio. A variable frequency divider divides a master clock by the division ratio output from the division ratio generation circuit, and outputs the obtained clock as a new clock. A high-quality clock having reduced jitter is regenerated, so that audio reproduction with high-quality sound is possible.

Abstract: A sound processing apparatus includes an information acquisition unit which acquires control information including at least one of mode information for designating a reproduction mode of the sound processing apparatus and attribute information for designating an attribute of an audio content represented by a sound signal, a frequency band expansion unit which performs frequency band expansion processing for adding an expanded component generated from the sound signal to the sound signal, and a control unit which changes parameters to be applied to the frequency band expansion processing in accordance with the control information acquired by the information acquisition unit.

Abstract: In accordance with the invention, audio signals are specially processed for sound presentation in a high noise environment. The electrical signal representative of the sound is first subjected to equalization to preferentially reduce the magnitude of bass signals while increasing the magnitude of treble signals. The equalized signal is then compressed, and the compressed signal is subjected to “mirror image” equalization which increases the magnitude of bass signals while reducing the magnitude of treble signals. The resulting signal fed to the speakers provides a sound presentation of compressed volume range and a bass-rich sound spectrum. It is particularly useful for providing quality sound presentation in a high noise environment.

Abstract: Systems and methods for providing bass compensation to correct for uneven bass response are disclosed. An example bass compensation system includes a low pass filter configured to receive an audio signal from an audio source and provide a filtered audio signal, the low pass filter having a roll off of at least 18 dB per octave. The bass compensation system further includes a summing amplifier coupled to the low pass filter and configured to sum the audio signal from said audio source and the filtered audio signal to provide a summed audio signal, wherein the summed audio signal provided by the summing amplifier provides a bass boost at a first frequency and mid bass cut at a second frequency greater than the first frequency.

Abstract: Audio circuits for a mobile terminal include a microphone and an audio signal processing circuit having an input coupled to the microphone. The audio signal processing circuit has a switchable portion and is configured to switch between a voice mode utilizing the switchable portion and a full fidelity mode not utilizing the switchable portion responsive to a mode signal coupled thereto.

Abstract: A sound source model storage section stores a sound source model that represents an audio signal emitted from a sound source in the form of a probability density function. An observation signal, which is obtained by collecting the audio signal, is converted into a plurality of frequency-specific observation signals each corresponding to one of a plurality of frequency bands. Then, a dereverberation filter corresponding to each frequency band is estimated by using the frequency-specific observation signal for the frequency band on the basis of the sound source model and a reverberation model that represents a relationship for each frequency band among the audio signal, the observation signal and the dereverberation filter. A frequency-specific target signal corresponding to each frequency band is determined by applying the dereverberation filter for the frequency band to the frequency-specific observation signal for the frequency band, and the resulting frequency-specific target signals are integrated.

Abstract: The present invention provides methods and systems for digitally processing audio signals in broadcasting and/or transmission applications. In particular, the present invention includes a pre-transmission processing module which is structured and configured to generate a partially processed signal. A transmitter is then structured to transmit or broadcast the partially processed signal to a receiver, where the signal is then fed to a post-transmission processing module. The post-transmission processing module is structured and configured to further processes the signal based upon, for example, the listening environment, profile(s), etc. and generate a final output signal.

Abstract: Disclosed herein is a recording apparatus including: an audio signal correction block configured to execute correction for flattening the frequency characteristic of an audio signal supplied from a microphone and/or correction of the level of the audio signal; a correction control block configured to make the audio signal correction block adjust the level of the audio signal attenuate the reference value with time, and, if an audio signal with a level thereof exceeding the reference value is entered, use the absolute value of the level of the audio signal exceeding the reference value as a new reference value; and a recording block configured to record the audio signal to a recording media.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

Abstract: A signal processing apparatus includes: a control section; a signal processing section connected with a plurality of signal processing elements and configured to perform signal processing for enhancing or attenuating an input signal in a specific frequency band; and a crossfade signal section including a crossfade signal processing element capable of replacing at least one of the signal processing elements, wherein the control section is configured to control any one of the signal processing elements among the plurality of signal processing elements, and the crossfade signal processing element, to crossfade to the crossfade signal processing element having the signal processing element as a new characteristic, to perform processing for replacing any one of the signal processing elements by the crossfade signal processing element, and to perform the processing on remaining signal processing elements of the plurality of signal processing elements in the signal processing section.

Abstract: In some embodiments, a method and system for controlling distortion of the output of a miniature speaker by attenuating critical frequency band of the input signal to be reproduced, using tuning parameters that have been predetermined where the critical frequency band is a frequency range of the speaker's frequency response in which Total Harmonic Distortion (THD) peaks. The distortion control is performed in a manner which allows an increase in the average loudness of the speaker's output without significantly increasing distortion. The tuning parameters include a center frequency and a bandwidth of the critical frequency band, and a power threshold value. In some embodiments, the system is a loudness maximizer configured to limit distortion of a speaker's output by limiting distortion in a critical frequency band using predetermined control parameters, and limit the dynamic range of the output signal and increase its perceived overall average loudness level.

Abstract: An apparatus for extending the bandwidth of an audio signal, the apparatus being configured to: generate an excitation signal from an audio signal, wherein in the audio signal comprises a plurality of frequency components; extract a feature vector from the audio signal, wherein the feature vector comprises at least one frequency domain component feature and at least one time domain component feature; determine at least one spectral shape parameter from the feature vector, wherein the at least one spectral shape parameter corresponds to a sub band signal comprising frequency components which belong to a further plurality of frequency components; and generate the sub band signal by filtering the excitation signal through a filter bank and weighting the filtered excitation signal with the at least one spectral shape parameter.

Abstract: Actuator apparatus for generating a physical effect, at least one attribute of which corresponds to at least one characteristic of a digital input signal sampled periodically in accordance with a clock, the apparatus comprising at least one array of moving elements each constrained to travel alternately back and forth along a respective axis in response to an alternating electromagnetic force applied to the array of moving elements, at least one latch operative to selectively latch at least one subset of said moving elements in at least one latching position thereby to prevent the individual moving elements from responding to the electromagnetic force, an electromagnetic field control system operative to receive the clock and, accordingly, to control application of the electromagnetic force to the array of moving elements, and a latch controller operative to receive the digital input signal and to control the latch accordingly.

Abstract: A clock generator receives first and second clock signals, and input representing a desired frequency ratio. A comparison is made between frequencies of an output clock signal and the first clock signal, and a first error signal represents the difference between the desired frequency ratio and this comparison result. The first error signal is filtered. A comparison is made between frequencies of the output clock signal and the second clock signal, and a second error signal represents the difference between the filtered first error signal and this comparison result. The second error signal is filtered. A numerically controlled oscillator receives the filtered second error signal and generates an output clock signal. As a result, the output clock signal has the jitter characteristics of the first input clock signal over a useful range of jitter frequencies and the frequency accuracy of the second input clock signal.

Abstract: A noise suppression system reduces low-frequency noise in a speech signal using linear predictive coefficients in an adaptive filter. A digital filter may update or adapt a limited set of linear predictive coefficients on a sample-by-sample basis. The linear predictive coefficients may be used to provide an error signal based on a difference between the speech signal and a delayed speech signal. The error signal represents an enhanced speech signal having attenuated and normalized low-frequency noise components.

Abstract: The present invention provides methods and systems for digitally processing audio signals. Some embodiments receive an audio signal and converting it to a digital signal. The gain of the digital signal may be adjusted a first time, using a digital processing device located between a receiver and a driver circuit. The adjusted signal can be filtered with a first low shelf filter. The systems and methods may compress the filtered signal with a first compressor, process the signal with a graphic equalizer, and compress the processed signal with a second compressor. The gain of the compressed signal can be adjusted a second time. These may be done using the digital processing device. The signal may then be output through an amplifier and driver circuit to drive a personal audio listening device. In some embodiments, the systems and methods described herein may be part of the personal audio listening device.

Abstract: A method and device for modifying a synthesis of a time-domain excitation decoded by a time-domain decoder, wherein the synthesis of the decoded time-domain excitation is classified into one of a number of categories. The decoded time-domain excitation is converted into a frequency-domain excitation, and the frequency-domain excitation is modified as a function of the category in which the synthesis of the decoded time-domain excitation is classified. The modified frequency-domain excitation is converted into a modified time-domain excitation, and a synthesis filter is supplied with the modified time-domain excitation to produce a modified synthesis of the decoded time-domain excitation.

Abstract: A sound processing apparatus includes a power spectrum operation unit obtaining a power spectrum of an audio signal, an envelope component removal unit removing an envelope component of the power spectrum and generating a signal characteristic that represents a peakness of the power spectrum, a filter characteristic calculation unit calculating a filter characteristic suppressing the signal characteristic by using the signal characteristic, and a suppress filter filtering the audio signal by using the filter characteristic.

Abstract: An audio signal processing device is provided whereby, from two systems of audio signals in which audio signals of multiple audio sources are included, the audio signals of the multiple audio sources can be suitably separated. The audio signal processing device divides each of two systems of audio signals into a plurality of frequency bands, calculates a level ratio or a level difference of the two systems of audio signals, at each of the divided plurality of frequency bands, and extracts and outputs frequency band components of and nearby values regarding which the level ratio or the level difference calculated at the level comparison means have been determined beforehand. The frequency band components have a level ratio or level difference at and nearby the values determined beforehand which are different one from another.

Abstract: Nested inductor arrays magnetically modulate an analog audio input signal recursively, so that the overall amplitude envelope of the output signal replicates the wave pattern of the input signal. The nested inductor arrays produce multiple levels of recursive modulation, so that the output signal incorporates multiple integrated self-similar harmonic layers, such that the phasing of the various layers are locked in by the analog waveform of the output signal itself. As a result, the spatial “depth” and temporal “immediacy” of the original analog recording is restored and can be encoded in digital format.

Abstract: A computer-implemented method of determining a configuration for an audio processing operation, wherein the audio processing operation comprises a predetermined set of one or more audio processing sub-operations, each audio processing sub-operation being configurable with one or more respective configuration parameters, the method comprising: specifying the predetermined set of one or more audio processing sub-operations; specifying a target frequency response; and performing a convergent optimization process to determine a configuration for the audio processing operation that reduces a difference between the frequency response of the audio processing operation and the target frequency response, wherein the configuration comprises a respective value for each configuration parameter of each audio processing sub-operation.

Abstract: Encoding and decoding methods and apparatus as described. An example method of obtaining auxiliary information in an audio signal using a plurality of frequency components residing in a plurality of code bands comprises transforming an audio signal into a frequency domain representation; determining characteristic of frequencies of the frequency domain representation that may contain the auxiliary information; normalizing across the code bands the characteristics of frequencies of the frequency domain representation in a respective one of the code bands that may contain the auxiliary information, wherein the normalization is carried out against a characteristic of a frequency in that code band; summing the normalized characteristics of the frequencies representative of auxiliary information to determine a sum for a frequency representative of auxiliary information; and determining that the sum is representative of the auxiliary information.

Abstract: Users with headsets may share an electronic device such as a portable computer or handheld device. The electronic device may have a connector such as an audio jack for receiving mating audio plugs on headsets. During normal operation with a single user, audio signals may be conveyed through the audio jack to the headset of the single user. When more than one user wishes to share the electronic device, an adapter accessory may be inserted into the audio jack of the electronic device. The headset of each user may be plugged into mating audio jacks in the adapter accessory. Circuitry in the adapter accessory may receive and process user input from each of the users. User input may be used to make local audio adjustments in the adapter accessory. User input may also be provided from the adapter accessory to the electronic device for processing.

Abstract: A videoconferencing system has a videoconferencing unit that use portable devices as peripherals for the system. The portable devices obtain near-end audio and send the audio to the videoconferencing unit via a wireless connection. In turn, the videoconferencing unit sends the near-end audio from the loudest portable device along with near-end video to the far-end. The portable devices can control the videoconferencing unit and can initially establish the videoconference by connecting with the far-end and then transferring operations to the videoconferencing unit. To deal with acoustic coupling between the unit's loudspeaker and the portable device's microphone, the unit uses an echo canceller that is compensated for differences in the clocks used in the A/D and D/A converters of the loudspeaker and microphone.

Abstract: In order to process a subband signal of a plurality of real subband signals which are a representation of a real discrete-time signal generated by an analysis filter bank, a weighter for weighting a subband signal by a weighting factor determined for the subband signal is provided to obtain a weighted subband signal. In addition, a correction term is calculated by a correction term determiner, the correction term determiner being implemented to calculate the correction term using at least one other subband signal and using another weighting factor provided for the other subband signal, the two weighting factors differing. The correction term is then combined with the weighted subband signal to obtain a corrected subband signal, resulting in reduced aliasing, even if subband signals are weighted to a different extent.

Abstract: In one disclosed aspect, dynamic gain modifications are applied to an audio signal at least partly in response to auditory events. In another aspect, an input channel is divided into auditory events by detecting changes in a measurable characteristic of the input signal with respect to time.

Abstract: In certain embodiments, an improved audio equalization filter can be generated by frequency warping one or more digital filters having a plurality of frequency bands. Frequency warping can include, for example, transforming at least some of the frequency bands of the one or more digital filters into lower frequency bands. As a result, in various implementations the audio equalization filter may be more accurate than certain currently-available IIR equalization filters. The audio equalization filter may also be more computing-resource efficient than certain currently-available FIR equalization filters.

Abstract: A device and method for controlling reproduction of an audio signal is provided, wherein the device is operated by means of an energy storage device. The method comprises the steps of deactivating a normal mode and activating an energy saving mode. Power consumption from the energy storage device for reproduction of the audio signal is reduced in the energy saving mode when compared to the normal mode. The method comprises reducing in the energy saving mode, a bass frequency component of a frequency spectrum of the audio signal and outputting the audio signal with reduced bass frequency component. The method further comprises ascertaining a charge state of the energy storage device and controlling the reduction in the bass frequency component based on a decrease in the charge state of the energy storage device.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: An audio decoding method and apparatus and an audio encoding method and apparatus which can efficiently process object-based audio signals are provided. The audio decoding method includes receiving a downmix signal and object-based side information, the downmix signal comprising at least two downmix channel signals; extracting gain information from the object-based side information and generating modification information for modifying the downmix channel signals on a channel-by-channel basis based on the gain information; and modifying the downmix channel signals by applying the modification information to the downmix channel signals.

Abstract: A circuit includes a microphone circuit, an audio processing module, a digital audio processing module, and an active noise reduction (ANR) circuit. The microphone circuit receives acoustic vibrations and generates an audio signal therefrom. The audio processing module generates a representation of the audio signal. The digital audio processing module compensates the representation of the audio signal based on hearing compensation data to produce a hearing compensated audio signal. The ANR circuit receives the hearing compensated audio signal and an ANR signal. The ANR circuit further functions to adjust the hearing compensated audio signal based on the ANR signal to produce an output audio signal, wherein the ANR signal is generated based on the output audio signal.

Abstract: A harmonic generating apparatus and the method are provided, which are used to enhance the quality of the bass audio signals. The method includes the steps of: providing a frequency signal having a present level and a preceding level; comparing the present level with the preceding level to generate a compared result; and generating the plurality of harmonics based on the compared result.

Abstract: A new and more efficient filtering system (e.g., digital microphone decimation filter architecture system) is described. A key to this architecture is the use of two parallel filter paths. Each path operates at the output sample rate, and comprises a shorter FIR filter followed by a series of allpass stages (e.g., implementing IIR filters). The FIR filter is designed to remove all but the last octave of out-of-band noise.

Abstract: An integrated audio codec includes a high-pass filter to prevent damage to personal computer speakers and other components. The audio codec may be compliant with HD audio standards and can operate with generic software drivers. Tuning of the high-pass filter is provided through an external pin-out where either an external capacitor or external resistors provide an ability to tune the high-pass filter. In one implementation, a tuning voltage is digitized into a tuning code used by a digital high-pass filter. In addition, multiplexers can be used to insure only the audio path leading to the speakers is filtered.

Abstract: An audio reproduction device and system are capable of realizing sounds having natural sound quality with no peculiarities, and provide an excellent audio-image ascending effect which is not varied due to differences among individual persons, thereby making audio images and images coincident with each other. A frequency characteristic correction portion forms a frequency characteristic such that a sound-pressure frequency characteristic of reproduced sound radiated to a viewing position from speakers has a first peak and a second peak, wherein the center frequency of the first peak falls within the range of 6 kHz±15%, and the center frequency of the second peak falls within the range of 13 kHz±20%.

Abstract: A sound processing apparatus according to the present invention acquires a test signal for measuring a standing wave state emitted in a listening room, and determines a peak position or a dip position due to a standing wave based on frequency characteristics of the test signal. Next, the sound processing apparatus emits a burst signal corresponding to the frequency of the peak position or the dip position, and acquires this signal. The sound processing apparatus calculates an increment ?P of the acquired signal, which indicates an amount of increase of a peak in the trailing edge portion corresponding to the end position of the burst signal relative to a peak in the portion corresponding to the stationary portion of the burst signal, and attenuates the frequency of the above peak position or dip position of a sound signal to be output by an attenuation depending on ?P.

Abstract: The method relates to audio recording reproduction. The method involves testing the features of a three-dimensional sound field using a dual or multi-channel system of spatially distributed channels, transmitting audio signals, and recording the responses in order to determine the differences between the influence of the natural acoustical properties of a room and the influence of the mutual spatial arrangement of audio sources and audio receivers on the characteristics of the sound field. Determining the differences between the features of the three-dimensional sound fields at different recording and reproduction conditions in accordance with this method makes it possible to adjust the parametric data of the tonal characteristics when reproducing an audio recording in order to produce a sound field identical to the sound field of the recording.

Abstract: A feedback-controlled microphone includes a microphone body and a membrane operatively connected to the body. The membrane is configured to be initially deflected by acoustic pressure such that the initial deflection is characterized by a frequency response. The microphone also includes a sensor configured to detect the frequency response of the initial deflection and generate an output voltage indicative thereof. The microphone additionally includes a compensator in electric communication with the sensor and configured to establish a regulated voltage in response to the output voltage. Furthermore, the microphone includes an actuator in electric communication with the compensator, wherein the actuator is configured to secondarily deflect the membrane in opposition to the initial deflection such that the frequency response is adjusted. An acoustic beam forming microphone array including a plurality of the above feedback-controlled microphones is also disclosed.

Abstract: A player apparatus for playing an input signal after band-extending the input signal includes: an extension controller to determine an extension start band for the input signal in accordance with information relating to the input signal; and a band divider to divide the input signal into a plurality of sub-band signals. The frequency band is extended on the basis of a plurality of the sub-band signals on a side lower than the extension start band, among the plurality of sub-band signals into which the input signal is band-divided by the band divider.

Abstract: An information processing apparatus includes: a sound source configured to generate sound waves; and a housing configured to incorporate the sound source and include a first main surface and a second main surface opposite to each other, the first main surface including a groove-like concave portion configured to communicate in one axial direction along the second main surface, and an opening for transmitting sound waves of the sound source, the opening being provided in an area of the concave portion and at a position offset from the middle in the one axial direction of the first main surface.

Abstract: A controlling unit includes: a memory unit that stores information in which the multiple clock frequencies that satisfy conditions for predetermined reception interferences to a part of multiple carrier frequencies included in a band frequency of the broadcast wave are previously set so that not all clock frequencies satisfy the conditions for the reception interferences to the same carrier frequencies, the information indicating a relation between each clock frequency and the carrier frequency that includes the reception interference; a reception frequency setting unit that sets the carrier frequency to be received to the broadcast wave receiving unit; a clock frequency determining unit that refers to the information stored in the memory unit and determines a clock frequency that does not include a reception interference to the carrier frequency that is set by the reception frequency setting unit, the determined clock frequency being out of the multiple clock frequencies; and a clock frequency setting unit t

Abstract: A semiconductor device is disclosed. The semiconductor device includes a digital audio circuit which converts an input digital signal into an analog audio signal, a DC-DC converter having a switching power source circuit, and an audible frequency determining circuit. In order that a difference between a frequency of a first clock signal for digital to analog conversion which is used in the digital audio circuit and a frequency of a second clock signal for switching control which is used in a DC-DC converter exceeds a maximum audible frequency, a frequency comparing circuit in the audible frequency determining circuit outputs a signal to a frequency changing circuit in the DC-DC converter. The frequency changing circuit causes a second oscillating circuit to change the second frequency.

Abstract: An audio processor (202) receives a non-priority audio signal (302) and a priority audio signal (304). The priority audio signal occupies a frequency band (408). The audio processor filters (320) the non-priority audio signal by suppressing frequency content in the same frequency region occupied by the priority signal, creating a filtered non-priority signal (412). The filtered non-priority signal and the priority signal are combined (328) and played over an audio transducer (110).

Abstract: In lossy data compression of a signal using ADPCM, an adaptive decorrelation or “prediction” filter is used to reduce the amplitude of the signal, the spectral dynamic range of the signal also being reduced. This latter reduction is effected in a nonuniform manner, if known techniques are used, with regions of high spectral density being compressed more than regions of low spectral density. The present invention recognises that using a uniform compression ratio results in a better tradeoff between compression and robustness to transmission channel errors. A method is described for obtaining a uniform compression ratio by adjusting coefficients of the decorrelation filter in dependence on coefficients of an adaptive training filter that is fed from the output of the decorrelation filter. A reverse method is also provided along with encoder, decoder and codec implementing the techniques.

Abstract: A receiver apparatus includes a first receiver portion and an acoustic filter network. The first receiver portion has a housing and is configured to convert at least one electrical signal into first sound energy having a first frequency range. The acoustic filter network communicates with the first receiver portion and is configured to receive the first sound energy. The acoustic filter network includes at least one sound channel and at least one chamber that communicates with the at least one sound channel. The least one sound channel includes a main branch and a first side branch and the at least one chamber comprises a first chamber. The first side branch communicates with the main branch and the first chamber, and the main branch is configured to receive the first sound energy.

Abstract: An audio signal processing circuit includes: a first low-pass filter configured to pass a component whose frequency is in a band lower than a lowest reproducible frequency of a speaker out of an audio signal inputted for reproduction by the speaker; a first high-pass filter substantially similar in phase characteristics to the first low-pass filter configured to pass a component whose frequency is in a band higher than the lowest reproducible frequency of the speaker out of the audio signal inputted for reproduction by the speaker; a harmonic generation unit configured to generate a harmonic from the audio signal having passed through the first low-pass filter; and a first addition unit configured to add the audio signal according to an output of the harmonic generation unit to the audio signal according to an output of the first high-pass filter.

Abstract: A portable electronic device that provides audio sound output from multiple internal speakers to a common output audio opening in a housing of the portable electronic device is disclosed. In one embodiment, the multiple internal speakers are provided in close proximity to one another, such as adjacent to one another, and serve to produce audio sound pertaining to different audio channels. The sound (i.e., pressure waves) produced by each of the internal speakers is directed into a respective acoustic chamber and output via the output audio opening in the housing. Accordingly, the acoustic chambers for the multiple internal speakers can each direct their audio sound output to the same output audio opening in the housing. The respective acoustic chambers can be formed adjacent to one another with a structural barrier serving to separate the distinct acoustic chambers.