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System and method for adjusting an audio signal

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20130003991 patent thumbnailZoom

System and method for adjusting an audio signal


Disclosed is a system and method of adjusting a volume level for an audio signal for a communication device to comply with a quality threshold. The method comprises: monitoring for an increase in the volume level; and upon determining that implementing the increase in the volume level would produce an output that would exceed the audio frequency pass mask, processing a digitized signal value of the audio signal to produce a first output signal value for the audio signal implementing the increase in the volume level utilizing a digital signal processing (DSP) device in the communication device, a processing filter defined in the digital signal processing device, a first set of adjustment parameters and the digitized signal value.
Related Terms: Audio Digital Signal Processing Digitize Signal Processing Processing Device
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USPTO Applicaton #: #20130003991 - Class: 381107 (USPTO) - 01/03/13 - Class 381 
Electrical Audio Signal Processing Systems And Devices > Including Amplitude Or Volume Control >Automatic



Inventors:

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The Patent Description & Claims data below is from USPTO Patent Application 20130003991, System and method for adjusting an audio signal.

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RELATED APPLICATIONS

This application is a continuation application of U.S. patent application Ser. No. 12/500,934 filed on Jul. 10, 2009, which itself is a continuation application of U.S. patent application Ser. No. 10/855,407 filed on May 28, 2004 (now U.S. Pat. No. 7,574,010).

FIELD OF DISCLOSURE

The disclosure relates to a system and method for adjusting an audio signal, in particular an audio signal generated by a speaker in a communication device.

BACKGROUND

In electronics, an audible audio signal is generated by generating an electrical signal representative of the audible signal and feeding that signal to a speaker. Various adjustments can be applied to the audio signal, including changing its volume, pitch and frequency, using known analog and digital signal processing techniques.

For a communication device, such as a cellular phone, a current feature provides a volume setting which selectively increases and decreases volume of an audio signal generated by the device's speaker. For a user of a cellular phone who has hearing difficulties, when he talks on the phone, the phone's volume setting may be placed at a “high” volume level, to increase the audio signal of the person calling the user. Alternatively, the volume setting may be set at a sufficiently high level to enable the user to listen to the audio signal generated by the cellular phone without requiring that the cellular phone be placed immediately near the user's ear. This setting is useful when the user checks his voice mail system. Therein, with a high volume level, he can hold the cellular phone in front of him and listen to his messages while still being able to see the keypad of the cellular phone. This enables him to see the keypad and quickly access the appropriate keys as he navigates through the commands of the voice mail system.

Increasing the volume on a cellular phone past a threshold level introduces distortion to the generated audio signal, thereby making it difficult to understand.

Known signal processing techniques provide dynamic monitoring of audio signals to dynamically recognize when an audio signal has excessive distortion and then provide a corrective shaping signal to it to reduce the distortion. Such techniques work very well, but use sophisticated evaluation techniques which require a significant amount of signal processing capabilities to implement them in real time. Smaller devices, such as cellular phones, have limited signal processing capabilities, and may not have sufficient capabilities to implement these techniques.

There is a need for a system and method of adjusting an audio signal in an efficient manner.

BRIEF DESCRIPTION OF THE DRAWINGS

The foregoing and other aspects of the disclosure will become more apparent from the following description of specific embodiments thereof and the accompanying drawings which illustrate, by way of example only, the principles of the disclosure. In the drawings, where like elements feature like reference numerals (and wherein individual elements bear unique alphabetical suffixes):

FIG. 1 is a block diagram of a communication device associated with an embodiment;

FIG. 2 is a schematic diagram of operational elements associated with the communication device of FIG. 1;

FIG. 3 is a frequency response curve illustrating responses for selected output levels of audio signals produced by the communication device of FIG. 1;

FIG. 4 is a harmonic distortion curve illustrating distortion levels for selected output levels of audio signals produced by the communication device of FIG. 1; and

FIG. 5 is a flow chart of an algorithm used for adjusting signals used by the operational elements of FIG. 2.

DETAILED DESCRIPTION

OF AN EMBODIMENT

The description which follows, and the embodiments described therein, are provided by way of illustration of an example, or examples, of particular embodiments of the principles of the present disclosure. These examples are provided for the purposes of explanation, and not limitation, of those principles and of the disclosure. In the description, which follows, like parts are marked throughout the specification and the drawings with the same respective reference numerals.

In a first aspect, a method of adjusting a volume level for an audio signal for a communication device to comply with an audio frequency pass mask is provided. The method comprises: monitoring for an increase in the volume level; and upon determining that implementing the increase in the volume level would produce an output that would exceed the audio frequency pass mask, processing a digitized signal value of the audio signal to produce a first output signal value for the audio signal implementing the increase in the volume level utilizing a digital signal processing (DSP) device in the communication device, a processing filter defined in the digital signal processing device, a first set of adjustment parameters and the digitized signal value.

In the method, the first set of adjustment parameters may provide a set of coefficients to adjust the audio signal to roll off lower frequency components in the audio signal.

In the method, the first set of adjustment parameters may provide a set of coefficients to further adjust the audio signal to boost higher frequency components in the audio signal.

In the method, the audio frequency pass mask may be a mask compliant with GSM standards.

The method may further comprise upon determining that implementing the increase in the volume level would not exceed the quality threshold, processing the digitized signal value to produce a second output signal value for the audio signal that also implements the increase in the volume level utilizing the DSP, the processing filter, a second set of adjustment parameters and the digitized signal value.

In the method, the processing filter may be a finite impulse response (FIR) filter; the first set of adjustment parameters may be a first set of coefficients for the FIR filter; and the second set of adjustment parameters may be a second set of coefficients for the FIR filter.

In the method, the audio frequency pass mask may define a boundary associated with one of the following attributes relating to the audio signal: a signal boost for selected signals; gain adjustment; side tone frequency adjustment; switching adjustments; signal suppression; and adjustments for a microphone in the communication device.

In a second aspect, a system for adjusting a volume level for an audio signal for a communication device in compliance with a quality threshold is provided. The system comprises: a module to accept a request for change of volume for the audio signal; a module to receive the audio signal and convert the audio signal to a digitized audio signal; a microprocessor; a DSP for processing the digitized audio signal and producing a digital output signal utilizing a filter, the digitized audio signal and a set of parameters; and a first module providing a set of instructions operating on the microprocessor. The instructions are to: determine when the request for the change of volume occurs; identify a new volume level for the audio signal for the request; and upon determining that implementing the new volume level would produce an output that would exceed the quality threshold process a digitized signal value of the audio signal to produce a first output signal value to implement the new volume level for the audio signal utilizing the DSP, a processing filter defined in the DSP, a first set of adjustment parameters and the digitized signal value.

In the system, the first set of adjustment parameters may provide a set of coefficients to adjust the audio signal to roll off lower frequency components in the audio signal.

In the system, the first set of adjustment parameters may provide a set of coefficients to further adjust the audio signal to boost higher frequency components in the audio signal.

In the system, the audio frequency pass mask may be a mask compliant with GSM standards.

In the system, the first module may provide a further set of instructions operating on the microprocessor to upon determining that implementing the new volume level would not exceed the quality threshold, process the digitized signal value to produce a second output signal value for the audio signal to implement the new volume level utilizing the DSP, the processing filter, a second set of adjustment parameters and the digitized signal value.

In the system, the filter may be a FIR filter; the first set of adjustment parameters may be a first set of coefficients for the FIR filter; and the second set of adjustment parameters may be a second set of coefficients for the FIR filter.

In a third aspect, a circuit for adjusting an output audio signal of a communication device in compliance with a quality threshold is provided. The circuit comprises: a module for receiving a signal encoding an audio signal and converting the signal to a digitized signal; a module to accept a request for an increase in a current volume setting to an increased volume level for the output audio signal; a microprocessor; and a module to convert the adjusted version of the digitized signal to an analog audio signal and to provide the analog audio signal to a speaker. The microprocessor is provided with instructions to monitor for an increase in the volume level; determine when the request for the change of volume occurs; identify a new volume level for the audio signal for the request; and upon determining that implementing the new volume level would produce an output that would exceed the quality threshold process a digitized signal value of the audio signal to produce a first output signal value to implement the new volume level for the audio signal utilizing a DSP, a processing filter defined in the DSP, a first set of adjustment parameters and the digitized signal value.

In the circuit, the first set of adjustment parameters may provide a set of coefficients to adjust the audio signal to roll off lower frequency components in the audio signal.

In the circuit, the first set of adjustment parameters may provide a set of coefficients to further adjust the audio signal to boost higher frequency components in the audio signal.

In the circuit, the audio frequency pass mask may be a mask compliant with GSM standards.

In the circuit, the microprocessor may be provided with further instructions to upon determining that implementing the new volume level would not exceed the quality threshold, process the digitized signal value to produce a second output signal value for the audio signal to implement the new volume level utilizing the DSP, the processing filter, a second set of adjustment parameters and the digitized signal value.

In the circuit, the processing filter may be a FIR filter; the first set of adjustment parameters may be a first set of coefficients for the FIR filter; and the second set of adjustment parameters may be a second set of coefficients for the FIR filter.

In the circuit, the quality threshold may define a boundary associated with one of the following attributes relating to the audio signal: a signal boost for selected signals; gain adjustment; side tone frequency adjustment; switching adjustments; signal suppression; and adjustments for a microphone in the communication device.

In another aspect, a method of adjusting a volume level for an audio signal for a communication device to comply with a quality threshold is provided. The method comprises: obtaining a digitized signal value of the audio signal and monitoring for an increase in the volume level. In the method, upon determining that implementing the increase in the volume level would produce an output that would exceed the quality threshold, processing the signal value to produce a first output signal value for the audio signal utilizing a DSP device in the communication device, a processing filter defined in the digital signal processing device, a first set of adjustment parameters and the signal value. In the method, upon determining that implementing the increase in the volume level would not exceed the quality threshold processing the signal value to produce a second output signal value for the audio signal utilizing the DSP, the processing filter, a second set of adjustment parameters and the signal value. In the method, the first output signal and the second output signal both implement the increase in the volume level.

In the method, the quality threshold may be an audio frequency pass mask.

In the method, the audio frequency pass mask is a mask compliant with GSM standards.

In the method, the processing filter may be a FIR filter; the first set of adjustment parameters may be a first set of coefficients for the FIR filter; and the second set of adjustment parameters may be a second set of coefficients for the FIR filter.

In the method, the second set of adjustment parameters may provide a set of coefficients to adjust the audio signal to roll off lower frequency components in the audio signal.

In the method, the second set of adjustment parameters may provide a set of coefficients to further adjust the audio signal to boost higher frequency components in the audio signal.

In yet another aspect, a system for adjusting a volume level for an audio signal for a communication device in compliance with a quality threshold is provided. The system comprises: a module to accept a request for change of volume for the audio signal; a module to receive the audio signal and convert the audio signal to a digitized audio signal; a microprocessor; a digital signal processor for processing the digitized audio signal and producing a digital output signal utilizing a filter, the digitized audio signal and a set of parameters; a first module providing a set of instructions operating on the microprocessor; and a second module to convert the digitized audio signal to an analog audio signal and to provide the analog audio signal to a speaker. The set of instructions: determine when the request for change of volume occurs; identify a new volume level for the audio signal for the request; upon determining that implementing the new volume level would produce an output that would exceed the quality threshold, the instructions produce a first output volume signal through the digital signal processor using a first set of adjustment parameters as the set of parameters and the new volume level; and upon determining that implementing the new volume level would produce an output that would not exceed the quality threshold, the instructions produce a second output volume signal through the digital signal processor using a second set of adjustment parameters as the set of parameters and the new volume level. In the system, the first output signal and the second output volume signal both implement the change in the volume level.

In the system, the quality threshold may be an audio frequency pass mask.

In the system, the audio frequency pass mask may be a mask compliant with GSM standards.

In the system, the filter may be a FIR filter; the first set of adjustment parameters may be a first set of coefficients for the FIR filter; and the second set of adjustment parameters may be a second set of coefficients for the FIR filter.

In the system, the second set of adjustment parameters may provide a set of coefficients to adjust the audio signal to roll off lower frequency components in the audio signal.

In the system, the second set of adjustment parameters may provide a set of coefficients to further adjust the audio signal to boost higher frequency components in the audio signal.

In still another aspect, a circuit for adjusting an output audio signal of a communication device in compliance with a quality threshold is provided. The circuit comprises: a module for receiving a signal encoding an audio signal and converting the signal to a digitized signal; a module to accept a request for an increase in a current volume setting to an increased volume level for the output audio signal; a microprocessor provided with instructions to detect when the request for the increase occurs and to generate an adjusted version of digitized signal to implement the increased volume level utilizing a DSP; a filter; and the digitized signal; and a module to convert the adjusted version of the digitized signal to an analog audio signal and to provide the analog audio signal to a speaker. The circuit determines if the request for the increase at the increased volume level would generate an audio output signal which exceeds the quality threshold; upon determining that the increased volume is within the quality threshold, the circuit selects a first set of adjustment parameters to be used by the digital signal processor to generate an adjusted version of the digitized signal to implement the increased volume having a received loudness rating (RLR) level; and upon determining that the increased volume would cause the output audio signal to exceed the quality threshold, the circuit selects a second set of adjustment parameters to be used by the digital signal processor to generate another adjusted version of the digitized signal that provides different output levels for the digitized signal at different frequencies as compared to the adjusted version to have an acceptable characteristic for the quality threshold while implementing the increased volume at about the RLR level.

In the circuit, the quality threshold may be an audio frequency pass mask.

In the circuit, the acceptable characteristic may be a frequency response within an acceptable boundary of the audio frequency pass mask; the filter may be a FIR filter; the first set of adjustment parameters may be a first set of coefficients for the FIR filter; and the second set of adjustment parameters may be a second set of coefficients for the FIR filter.

In the circuit the audio frequency pass mask may be compliant with GSM standards.

In the circuit, the second set of adjustment parameters may comprise a set of coefficients to roll off lower frequency components in the audio signal.

In the circuit, the second set of adjustment parameters may further comprise a set of coefficients to boost higher frequency components in the audio signal.

In the circuit, the audio frequency pass mask may define a boundary associated with one of the following attributes relating to the audio signal: a signal boost for selected signals; gain adjustment; side tone frequency adjustment; switching adjustments; signal suppression; and adjustments for a microphone in the communication device. Further, the filter may be a FIR filter. Further, the first set of adjustment parameters may be a first set of coefficients for the FIR filter; and the second set of adjustment parameters may be a second set of coefficients for the FIR filter.

In another aspect, a method of adjusting an audio signal for a communication device is provided. The method comprises: identifying a characteristic associated with the signal and identifying a quality threshold for the characteristic; identifying a first set of adjustment parameters for the signal for use when the quality threshold is exceeded; and identifying a second set of adjustment parameters for the signal for use when the quality threshold is not exceeded. Further, in the communication device, the method further comprises: monitoring for a current setting value for the characteristic; and detecting whether the current setting value is above or below the quality threshold. If the value is above, then the method processes the signal value to produce a first output signal value utilizing a digital signal processing device in the communication device, a processing filter defined in the digital signal processing device, values of the first set of adjustment parameters and the signal value. If the value is below, then the method processes the signal value to produce a second output signal value utilizing the digital signal processing device, the processing filter, values of the second set of adjustment parameters and the signal value.

In the method, the characteristic may be a volume level; the quality threshold may be a set volume level within operating parameters of the communication device; the filter may be a FIR filter; the first set of adjustment parameters may be a first set of coefficients for the FIR filter; and the second set may be a second set of coefficients for the FIR filter.

In the method, the first set may provide a set of coefficients to adjust the audio signal to comply with an audio frequency pass mask.

In the method, the audio frequency pass mask may be a mask compliant with GSM standards.

In the method, the second set may provide a set of coefficients to adjust the audio signal to roll off lower frequency components in the audio signal.

In the method, the second set may further provide a set of coefficients to further adjust the audio signal to boost higher frequency components in the audio signal.

In yet another aspect, a system for adjusting an audio signal for a communication device is provided. The system comprises: a module to accept a request for change of volume for the audio signal; a module to receive the audio signal and convert the audio signal to a digitized audio signal; a microprocessor; a digital signal processor for processing the digitized audio signal and producing a digital output signal utilizing a filter, the digitized audio signal and a current set of parameters; an algorithm operating on the microprocessor; a module to convert the digitized audio signal to an analog audio signal; and to provide the analog audio signal to a speaker. The algorithm is adapted to: detect when the request for change of volume occurs; identify a new volume level being requested from the request; select a first set of adjustment parameters for the audio signal as a selected set if a volume threshold is exceeded; select a second set of adjustment parameters for the audio signal as the selected set if the volume threshold is not exceeded; and provide the selected set to the digital signal processor if the current set of parameters in the digital signal processor is different than the selected set.

In the system, the filter may be a FIR filter; the first set of adjustment parameters may be a first set of coefficients for the FIR filter; and the second set of adjustment parameters may be a second set of coefficients for the FIR filter.

In the system, the first set of adjustment parameters may provide a set of coefficients to adjust the audio signal to comply with an audio frequency pass mask.

In the system, the audio frequency pass mask may be a mask compliant with GSM standards.

In the system, the second set of adjustment parameters may provide a set of coefficients to adjust the audio signal to roll off lower frequency components in the audio signal.

In the system, the second set of adjustment parameters may further provide a set of coefficients to further adjust the audio signal to boost higher frequency components in the audio signal.

In other aspects, various combinations of sets and subsets of the above aspects are provided.

Referring to FIG. 1, communication device 100 is shown. User of communication device 100 can establish a call with another person using another device which can communicate with device 100. Device 100 may be a telephone, cordless telephone, cellular phone, voice-enabled personal digital assistant or any device providing electronic voice communications. The other person may be using a device connected to a PSTN network (not shown). Therein, communications may be established through a cellular network (not shown) associated with device 100 and a PSTN network associated with the device of the other person. Alternatively, the other device may be a voice mail system.

The main interface elements of communication device 100 for its user include: keypad 102, display 104, speaker 106 and microphone 108. Speaker 106 generates all audible signals received from the other device. Microphone 108 receives all audio signals from the user as he speaks to the other person. Speaker 106 can be any type of speaker having appropriate dimensions and performance characteristics to produce an audio signal for device 100. For a wireless communication interface, antenna 110 is provided to receive and transmit wireless signals for device 100.

Referring to FIG. 2, further detail on functional aspects of elements in device 100 are provided. Therein, circuit 200 provides facilities to receive and transmit audio signals to and from device 100. Circuit 200 comprises microprocessor 202, non-volatile memory 204, digital signal processing (DSP) module 206, radio module 208, antenna 110, coder/decoder (CODEC) 210 and speaker 106.

Microprocessor 202 is the main control element for device 100. Algorithm 212 is a program operating on microprocessor 202, effectively providing control for many operations of device 100, including call control, display control and power management. Microprocessor 202 has access to memory 204, which is used to store routines, variables and data used by algorithm 212. In the embodiment, microprocessor 202 is a commercially available microcontroller, such as a microcontroller available from ARM, Motorola and Intel. Memory 204 is a flash memory device. Other technologies of non-volatile storage devices known in the art can be used. A representative DSP 206 is commercially available from various manufacturers, including Texas Instruments and Analog Devices. A representative CODEC 210 is commercially available from Texas Instruments and Analog Devices. A representative speaker 106 is commercially available from Philips and Foster.

For device 100, antenna 110 provides a wireless interface to receive and transmit voice signals encoded in radio frequencies to other devices. Antenna 110 is connected to radio module 208, which converts received wireless signals 214 received by antenna 110 into electrical signals, which can then be used by the other elements in circuit 200. In particular, module 208 converts received wireless signals 214 into digital data stream 216, which is a sufficient representation of the audio signal encoded in wireless signal 214. Similarly, data representing digitized audio signals spoken by the user and received by microphone 104 is received by radio module 208 then converted into an electrical signal for antenna 110, which then converts the electrical signal to a radio signal and transmits it to the air. Radio module 208 comprises internal circuits and routines known in the art of radio signal processing.

Digital data stream 216 is provided from radio module 208 to DSP 206, which is programmed to selectively shape the digitized voice signals to effect a required acoustic response properties for sounds meant for reproduction on speaker 106. At the heart of DSP 206 is filter 218, which provides an algorithm to process signal inputs (such as data 216) and apply set adjustment coefficients to aspects of the inputs, to produce output data stream 222. The adjustment coefficients are preferably provided by data 220 from microprocessor 202 as another set of inputs to DSP 206. With all of the input and coefficient information, DSP 206 uses its internal specialized DSP circuits to efficiently generate output data stream 222.

Thereafter output data stream 222 is provided to CODEC 210, which converts stream 216 to analog electrical signal 224, and then provides the electrical signal to speaker 106. Speaker 106 converts electrical signal 224 into audio signal 226, which can be heard by user 228.

It will be appreciated that other embodiments and processes also will provide an equivalent result of converting and processing a received radio frequency-based signal into an audible signal.

Further detail on specific adjustment coefficients provided by the embodiment is now provided. In the current regulatory environment, cellular phones and other wireless communication devices, such as device 100, generally must meet minimal operating performance specifications relating to the quality of audio signals produced by its speaker. Such audio specifications include signal frequency response and loudness rating standards.

In FIG. 3, graph 300 is a receive frequency response graph defined under Globale Systems Mobile (GSM) standards. The receive frequency response standard mandates that the measured frequency response of a received signal (which is reproduced on speaker 106) falls within a specific mask (measured in dB) for a given frequency range. Specifically, the GSM standard dictates that for a communication device operating at the nominal volume setting, the frequency response of audio signals produced by its speaker must fall between a floating template defined by upper boundary 302 and lower boundary 304. Upper boundary 302 is a shaped frequency response curve defining acceptable and unacceptable response levels for given frequencies. For upper boundary 302, at a given frequency if the measured response level is above upper boundary 302, then the measured response fails the specification. Similarly, lower boundary 304 is a shaped frequency response curve defining acceptable and unacceptable response levels for given frequencies. For lower boundary 304, at a given frequency if the measured response level is below lower boundary 304, then the measured response fails the specification. The loudness standard is known as the Receive Loudness Rating (RLR). Specifically, the RLR dictates that for a communication device operating at a nominal volume setting, the RLR must be within a certain range. The calculation procedures for RLR are known in the art.

Frequency response curves 306, 308 and 310 are exemplary responses generated by speaker 106. Each curve represents speaker 106 being driven at one level but having different filter parameters being applied thereto by filter 218 and algorithm 212.

Frequency response curve 310 is the acoustic frequency response of speaker 106 with no filtering (i.e., with a flat filter response). Since the region of curve 310 about 650 Hz is not compliant with the frequency response mask specification, its audio drive signal requires equalization or shaping to make it compliant.

Frequency response curve 306 is the acoustic frequency response of speaker 106 with filter parameters applied for the nominal volume setting to provide compliance with the frequency response mask specification. As seen, curve 306 is compliant with the frequency response mask specification throughout its entire frequency range.

Frequency response curve 308 is the acoustic frequency response of speaker 106 for a different set of filter parameters. The region of curve 308 about 300 Hz is not compliant with the frequency response mask specification. Curve 308 shows that speaker 106 has lower output levels at the lower frequencies compared to curve 306. This is due to different shaping of the audio signal characteristic of the different set of filter parameters. Curve 308 has higher output levels than curve 306 in the frequency region above about 520 Hz. In that region, output values for that speaker have been boosted to compensate for the loss in loudness associated with the low frequency roll-off of the output level of curve 308 relative to curve 306. This compensation ensures that the user experiences the expected change in loudness, as determined by the Receive Loudness Rating (RLR), when the volume setting is switch from one volume level to its adjacent volume level and when filter parameters are switched.

Referring to FIG. 4, for a given speaker producing a signal at a given volume level, a related issue to frequency response is the amount of distortion present in the signal. Generally, due to a speaker\'s inherent operational characteristics, as the volume of a signal reproduced on the speaker is increased, distortion in the audio signal increases. Limitations in the speaker include its operating frequency range and power handling capabilities. Often, for a given speaker having a given frequency operating range, signals in the lower frequencies are distorted first, due to physics relating to transducers. However, a speaker may also have sensitive elements for creating signals having higher frequencies. Accordingly, if the sensitive elements are damaged by high voltage signals, then the signals in the higher frequencies become distorted. Graph 400 shows total harmonic distortion (as a percentage) on the y-axis measured against a frequency range (in Hz) on the x-axis.

Curve 402 shows a representative curve of total harmonic distortion of a speaker driven at its nominal operating level. At its nominal operating level, it may have a frequency response as represented by curve 306 (FIG. 3).

When the speaker is driven at a higher loudness level, which is outside its nominal operating range, it may begin to distort the reproduced signals. Representative distortion levels are shown in curve 404. As seen in curve 404, distortion levels are higher in the lower frequency range than the higher frequency range. Accordingly, in order to more effectively reduce the distortion, it is advantageous to roll-off the lower frequencies in the signal.

Using frequency domain analysis techniques, in order to control the distortion, the embodiment effectively examines the current volume level of the audio signal being reproduced on speaker 106 (FIG. 2), and if the volume level exceeds a certain threshold, then certain aspects of the audio signal are attenuated to reduce the distortion. If the volume level is reduced to below the threshold, then the attenuation is removed. More specifically, when the volume level exceeds a predetermined threshold, device 100 effectively imposes a high-pass filter on the audio signal, thereby attenuating lower frequencies past a cut off band and not attenuating signals in frequencies past the cut off band. The shape and cut-off point of the high pass filter may be modified per different performance requirements. In other embodiments, low-pass filters, notch filters, band-pass filters, other types of filters or any combination of filters may be used.

For a given set of performance requirements, parameters must be established to set when a signal is attenuated and by how much. One method of identifying the characteristics is to measure the performance of the given speaker 106 under differing signal levels, by measuring its frequency response and distortion characteristics. The measurements are analyzed to identify volume levels which may produce excessive distortion and the associated frequency responses. Thereafter a cut-off volume level is selected. When the volume level of speaker 106 exceeds the cut-off volume, then signals are attenuated at certain frequencies. The identification of which frequencies require attenuation and what the attenuation amount should be is also determined by analyzing the measurements.



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stats Patent Info
Application #
US 20130003991 A1
Publish Date
01/03/2013
Document #
13608135
File Date
09/10/2012
USPTO Class
381107
Other USPTO Classes
International Class
03G3/20
Drawings
6


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Electrical Audio Signal Processing Systems And Devices   Including Amplitude Or Volume Control   Automatic