Patent Application
20110194699
Not Applicable
Certain embodiments of the invention relate to wireless communications. More specifically, certain embodiments of the invention relate to a method and system for enhanced sound quality for stereo audio.
With the increasing popularity of various wireless standards and technologies, there is a growing demand to provide a simple and complete solution for wireless communications applications. In this regard, electronics manufacturers are increasingly attempting to incorporate multiple wireless technologies into portable electronic devices. For example, wireless technologies that are seeing widespread deployment include FM radio, Bluetooth (BT), GPS, Wi-Fi, and RFID.
Although desirable to users, incorporating multiple wireless communication technologies into devices such as wireless handsets may pose problems in terms of cost and complexity. In this regard, combining a plurality of wireless technologies into a portable electronic device may require separate processing hardware and/or separate processing software. Moreover, coordinating the reception and/or transmission of data to and/or from the portable electronic device may require significant processing overhead that may impose certain operation restrictions and/or design challenges.
Further limitations and disadvantages of conventional and traditional approaches will become apparent to one of skill in the art, through comparison of such systems with some aspects of the present invention as set forth in the remainder of the present application with reference to the drawings.
A system and/or method is provided for enhanced sound quality for stereo audio, substantially as illustrated by and/or described in connection with at least one of the figures, as set forth more completely in the claims.
These and other advantages, aspects and novel features of the present invention, as well as details of an illustrated embodiment thereof, will be more fully understood from the following description and drawings.
Certain embodiments of the invention may be found in a method and system for enhanced sound quality for stereo audio. In various embodiments of the invention, levels of audio signals received via a RF channel may be controlled based on a signal-to-noise ratio (SNR) of the RF channel. The SNR of the RF channel may be determined based on signal strength in an unused portion of the RF channel. The recovered audio signals may comprise an FM multiplex signal, and an unused portion of the FM radio channel may be a guard band adjacent to a carrier of the FM multiplex signal. A level of one or both of a mono component and a stereo component of an FM multiplex signal may be controlled based on the SNR.
A stereo component of an FM multiplex signal may be scaled prior to combining the stereo component and a mono component to generate left and right audio signals. The stereo component may be scaled by a blending factor. The blending factor may, for example, be equal to a first constant when the SNR is less than a first threshold, grow exponentially when the SNR is between the first threshold and a second threshold, and may be equal to a second constant when the SNR is greater than the second threshold. The first threshold and the second threshold may be programmable. The recovered audio signals may be left and right audio signals. The recovered audio signals may be scaled by a muting factor. The muting factor may, for example, be equal to 1 when the SNR is above a threshold, and may decrease based on one or more slope values when the SNR is between 0 and the threshold. The threshold and/or the one or more slope values may be programmable.
The FM MPX 120 comprises a mono component 102, a carrier 104, a stereo component 106, and a radio data services (RDS) component 108. The mono component 102 may comprise a sum of left and right baseband audio information.
The carrier 104 may comprise, for example, a 19 kHz sine wave. There may be unused frequencies on either side of the carrier signal 104 to prevent distortion or interference with the carrier signal. The unused frequencies may be referred to as guard bands 110.
The stereo component 106 may be generated by subtracting a right baseband audio signal from a left baseband audio signal to generate a difference signal. The difference signal may then be modulated onto a double-sideband suppressed carrier (DSBSC) signal which may, for example, be centered at 38 kHz and occupy the baseband range from 23 to 53 kHz.
The RDS component 108 may comprise Manchester encoded data modulated onto a 57 kHz carrier. The RDS component 108 may be utilized to carry data such as text. In this regard, the RDS component 108 is frequently used to identify the call letters of the radio station transmitting on the channel and/or to identify the audio currently being transmitted on the channel.
Also shown in
Although various aspects of the present invention are described with respect to the FM MPX 120, the invention is not so limited. Accordingly, various aspects of the invention may apply to reception of any frequency modulated audio information.
The antenna 382 may be suitable for transmitting and/or receiving electromagnetic signals. Although a single antenna is illustrated, the invention is not so limited. In this regard, the RF front-end 384 may utilize a common antenna for transmission and reception of signals adhering to one or more wireless standards, may utilize different antennas for each supported wireless standard, and/or may utilize a plurality of antennas for each supported wireless standard.
The RF front-end 384 may comprise suitable logic circuitry and/or code that may be operable to perform amplification, down-conversion, filtering, demodulation, and/or analog-to-digital conversion of received signals. For example, the RF front-end may be operable to tune to the channel 122X, process the signal received on the channel 120X to recover the baseband FM MPX 120, convert the FM MPX 120 to a digital representation, and output the digital baseband signal to the DSP 392. In some embodiments of the invention, the RF front-end may also be operable to process the FM MPX 120 to recover mono audio signals, stereo audio signals, and/or RDS signals, convert the mono, stereo, and/or RDS signals to a digital representation, and output the digital audio and/or RDS signals to the DSP 392. Additionally, the RF front-end 384 may comprise suitable logic, circuitry, interfaces, and/or code that may be operable to perform amplification, up-conversion, filtering, modulation, and/or digital-to-analog conversion of to-be-transmitted signals. For example, the RF front-end 384 may be operable to receive digital baseband signals from the DSP 392, generate corresponding RF signals, and transmit the RF signals via the antenna(s) 382.
In addition to FM MPX signals, the RF front-end 384 may be operable to transmit and/or receive signals in accordance with, for example, cellular, WiMAX, Wi-Fi, Bluetooth, Zigbee, T1/E1, Ethernet, USB, IEEE 1394, analog audio standards, analog video standards, digital audio standards, and/or digital video standards. In various embodiments of the invention, the RF front-end 384 may be tuned and/or otherwise controlled via one or more signals from, for example, the processor 390 and/or the DSP 392. In this regard, in various embodiments of the invention, a gain of the RF front-end 384 may be controlled by one or more signals from the processor 390 and/or the DSP 392. In an exemplary embodiment of the invention, the RF front-end 384 may comprise a received signal strength indicator (RSSI) that may be utilized to determine received signal strength over at least a portion of the bandwidth of the received signal.
The DSP 392 may comprise suitable logic, circuitry, and/or code that may be operable to perform computationally intensive processing of digital signals. In various embodiments of the invention, the DSP 392 may encode, decode, transcode, modulate, demodulate, encrypt, decrypt, scramble, descramble, filter, equalize, compress, decompress, convert, format, packetize, rate convert, and/or otherwise process digital signals. In this manner, the DSP 392 may be operable to recover mono audio signals, stereo audio signals, and/or RDS signals from the FM MPX 120, and may be operable to process the recovered signals to, for example, improve the sound quality of the audio. In this regard, the DSP 392 may be operable to process the FM MPX 120 to generate left and right output audio signals 395L and 395R. In addition to FM MPX signals, the DSP 392 may be operable to process and/or generate signals in accordance with, for example, cellular, WiMAX, Wi-Fi, Bluetooth, Zigbee, T1/E1, Ethernet, USB, IEEE 1394, analog audio standards, analog video standards, digital audio standards, and/or digital video standards.
The processor 388 may comprise suitable logic, circuitry, and/or code that may enable processing data and/or controlling operations of the communication device 380. In this regard, the processor 388 may be enabled to provide control signals to the various other components of the communication device 380. The processor 388 may also control transfers of data between various portions of the communication device 380. Additionally, the processor 388 may enable execution of applications, programs, and/or code for processing data and/or effectuating operation of the communication device 380. Exemplary control signals generated by the processor 388 may comprise one or more signals to tune the RF front-end 384, and to control audio signal levels in the communication device 380.
The memory 390 may comprise suitable logic, circuitry, interfaces, and/or code that may be operable to store information such as parameters and/or code that may effectuate the operation of the communication device 380. Stored information may comprise received data and/or data to be presented, transmitted, and/or otherwise processed. The parameters may comprise configuration data and the code may comprise operational code such as software and/or firmware, but the information need not be limited in this regard. In an exemplary embodiment of the invention, the memory 390 may store instructions and/or parameters that may be utilized to control audio levels in the communication device 380. For example, the memory 390 may store tables, instructions, and/or parameters for determining values of the muting factor, M, and/or the blending factor, B, described below with respect to
The display 383 may be operable to provide visual information to, and/or enable interaction by, a user of the communication device 380. In various embodiments of the invention, a graphical user interface may be presented via the display 383. In various embodiments of the invention, a visual media content such as video, images, and text may be presented via the display 383.
The user controls 385 may be operable to enable user interaction with the communication device 380 to control services and/or content handled by the communication device 380. The user controls 385 may comprise, for example, a keypad, a keyboard, a roller ball, a multidirectional button, a scroll wheels, and/or a touch screen. In various embodiments of the invention, the user controls 385 may enable a user to control a desired volume of audio signals, such as 395L and 395R, output by the communication device 380.
The speakers 387 may be operable to present audio information to a user. The speakers 387 may present voice from a phone call and/or music or ringtones received and/or played back by the communication device 380. In this regard, aspects of the invention may enable controlling, based on SNR of received signals, the level of audio signals 395L and 395R driven to the speakers 387.
The headphone jack 393 may be operable to present audio information to a user via headphones and/or a headset plugged into the jack 393. The headphone jack 393 may present voice from a phone call and/or music or ringtones received and/or played back by the communication device 380. In this regard, aspects of the invention may enable controlling, based on SNR of received signals, the level of audio signals 395L and 395R driven to the headphone jack 393.
The microphone 389 may be operable to convert acoustic signals into electronic signals. The microphone may enable a user to participate in a phone call and/or interact with the cellular enabled communication device via oral input.
In operation, the RF front-end 384 may be tuned to a selected FM channel 122X which may lie, for example, in the FM broadcast band of 87 MHz to 108 MHz. The RF front-end 384 may amplify, down-convert, filter, and demodulate the FM MPX 120 received on the selected channel 122X to generate a corresponding digital baseband signal which may comprise the mono component 102, the stereo component 106, and/or the RDS component 108.
In an exemplary embodiment of the invention, the RF front-end 384 may make two signal strength measurements of the received FM MPX 120. The first measurement may be made in the guard band 110. This portion of the FM MPX 120 may ideally contain no signal and thus the first measurement may correspond to noise on the selected channel 122X. The second measurement may be centered at, for example, the frequency of the carrier 104, a frequency within the mono component 102, and/or a frequency within the stereo component 106. The second measurement may thus correspond to strength of a desired signal on the selected channel 122X. The second measurement may be divided by the first measurement to determine a SNR of the selected channel 122X.
In another embodiment of the invention, the RF front-end measure the signal strength in the guard band 110 to determine the noise on the selected channel 122X, and then may estimate the strength of the desired signal on the selected channel 122X based on, for example, standard signal deviations used by the station transmitting the signals.
In another embodiment of the invention, the measurements may be made on the digital baseband signal by the DSP 392 and the DSP 392 may determine the SNR. In this regard, the DSP 392 may be operable to determine signal strength and/or may be operable to utilize other techniques such as a fast Fourier transform (FFT) for determining the SNR.
After the SNR is determined, signal levels may be adjusted based on the SNR. In various embodiments of the invention, the levels of the received RF signal may be adjusted in the RF front-end 384 and/or the levels of the recovered baseband signal may be adjusted in the RF front-end 384 and/or the DSP 392. For example, a level of the mono component 102 and/or a level of the stereo component 106 of the FM MPX 120 may be adjusted by the DSP 392 prior to combining the mono component 102 and the stereo component 106 to generate output audio signals 395L and 395R. Alternatively or additionally, the level of the output audio signals 395L and 395R may be adjusted by the DSP 392 or the processor 388.
The graphs 402 and 403 illustrate an instance where the signal 411 has been recovered from a FM channel having a high SNR, such as the channel 202 described with respect to
The graphs 404 and 405 illustrate an instance where the signal 411 has been recovered from a received FM channel having a moderate SNR, such as the channel 206 described with respect to
The graphs 406 and 407 illustrate an instance where the signal 411 has been received on a FM channel having a low SNR, such as the channel 208 described with respect to
As described with respect to
395R=M*((R+L)−(L−R)) EQ. 3
395
L
=M*((R+L)+(L−R)) EQ. 4
where (R+L) is the mono component 102, (L−R) is the stereo component 106, and M is the muting factor.
In an exemplary embodiment of the invention, M may be described by the characteristic 452. In this regard, the characteristic 452 may be described by a stop point 454, a first slope 456, a second slope 460, and a threshold 458.
The stop point 454 may correspond to a minimum value of the muting factor, M, that is to be applied to the signals 395L and 395R. In this regard, scaling the signal by the start point 454 may, for example, result in the signals 395L and 395R being imperceptible or barely perceptible to a listener.
The threshold 458 may correspond to an SNR below which M has a value of less than 1. That is, below the threshold 458, the audio signal 411 may be reduced below the maximum volume, which may be set by a listener.
The first slope 456 may determine the rate a which M changes over a range 462 of SNR values and the second slop 462 may determine a rate at which M changes over a range 464 of SNR values. In the range 466, may change at a rate somewhere between the rates corresponding to ranges 462 and 464. In this regard, the value of the threshold 458, the slope 456, the slope 460, and/or user preferences may determine how rounded the characteristic 452 is in the region 466
The characteristic 452 is just an exemplary characteristic for M, and the behavior of M may be defined or described by any suitable function(s). In this regard, a characteristic describing M is not limited to being described by a start point a stop point and one or more slopes but may be any suitable function such as, for example, an exponential or quadratic function. Also, N may be determined via, for example, mathematical and/or logical operations performed by the processor 388 or the DSP 392, and/or may be determined via, for example, a look-up table stored in the memory 390. In various embodiments of the invention, the start point 454, the threshold 458, the slope 456, and the slope 460 may be programmable and the slope and/or function defining the behavior of M in the regions 462, 464, and 466 may be programmable.
As described with respect to
395R=(R+L)−B*(L−R) EQ. 3
395L=(R+L)+B*(L−R) EQ. 4
where (R+L) is the mono component 102, (L−R) is the stereo component 106, and B is the blending factor.
In the exemplary embodiment of the invention depicted in
In step 608, a value for the blending factor, B, and/or a value for the muting factor, M, may be determined based on the SNR determined in step 606. In various embodiments of the invention, the value of B may be calculated based on one or more parameters such as the thresholds 504 and 506, based on one or more logic and/or mathematical functions that describe the behavior of B, and/or based on values in a look-up table. In various embodiments of the invention, the value of M may be calculated based on one or more parameters such as the stop point 458, the slope 460, the slope 456, and/or the stop point 454; based on one or more logical and/or mathematical functions that describe the behavior of M; and/or based on values in a look-up table. In this regard, in an exemplary embodiment of the invention, the muting factor M and the blending factor B may both be utilized to adjust audio signal levels and the signals 395L and 395R may be described by the following equations:
395R=M*((R+L)−B*(L−R)) EQ. 5
395L=M*((R+L)+B*(L−R)) EQ. 6
In step 610, the value of M and/or B calculated in step 608 may be applied to signals such as the output audio signals 395L and 395R. In various embodiments of the invention, M and/or B may be applied via a digital multiplier, an analog multiplier, and/or by controlling the gain of an amplifier, buffer, or other analog and/or digital circuit element. Subsequent to step 610, the exemplary steps may return to step 604 or 606.
Various aspects of a method and system for enhanced sound quality for stereo audio are provided. In an exemplary embodiment of the invention, levels of audio signals received via a FM radio channel 122X may be controlled based on a signal-to-noise ratio (SNR) of the FM radio channel 122X. The SNR of the FM radio channel 122X may be determined based on signal strength in an unused portion of the FM radio channel 122X. The recovered audio signals may comprise an FM MPX 120, and an unused portion of the FM radio channel 122X may be a guard band 110 adjacent to a carrier 104 of the FM MPX 120. A level of one or both of a mono component 102 and a stereo component 106 of the FM MPX 120 may be controlled based on the SNR.
A stereo component 106 of the FM MPX 120 may be scaled prior to combining the stereo component 106 and a mono component 102 of the FM MPX 120 to generate left and right audio signals 395L and 395R. The stereo component 106 may be scaled by a blending factor B. The blending factor, B, may, for example, be equal to 0 when the SNR is less than a first threshold 504, grow exponentially when the SNR is between the first threshold 504 and a second threshold 506, and may be equal to 1 when the SNR is greater than the second threshold 506. The first threshold 504 and the second threshold 506 may be programmable. The recovered audio signals may be left and right audio signals 395L and 395R. The recovered audio signals may be scaled by a muting factor, M. The muting factor, M, may, for example, be equal to 1 when the SNR is above a threshold, and may decrease based on slope values 456 and 460 when the SNR is between 0 and the threshold 458 The threshold 458 and/or the slope values 456 and 460 may be programmable.
Another embodiment of the invention may provide a machine and/or computer readable storage and/or medium, having stored thereon, a machine code and/or a computer program having at least one code section executable by a machine and/or a computer, thereby causing the machine and/or computer to perform the steps as described herein for enhanced sound quality for stereo audio.
Accordingly, the present invention may be realized in hardware, software, or a combination of hardware and software. The present invention may be realized in a centralized fashion in at least one computer system, or in a distributed fashion where different elements are spread across several interconnected computer systems. Any kind of computer system or other apparatus adapted for carrying out the methods described herein is suited. A typical combination of hardware and software may be a general-purpose computer system with a computer program that, when being loaded and executed, controls the computer system such that it carries out the methods described herein.
The present invention may also be embedded in a computer program product, which comprises all the features enabling the implementation of the methods described herein, and which when loaded in a computer system is able to carry out these methods. Computer program in the present context means any expression, in any language, code or notation, of a set of instructions intended to cause a system having an information processing capability to perform a particular function either directly or after either or both of the following: a) conversion to another language, code or notation; b) reproduction in a different material form.
While the present invention has been described with reference to certain embodiments, it will be understood by those skilled in the art that various changes may be made and equivalents may be substituted without departing from the scope of the present invention. In addition, many modifications may be made to adapt a particular situation or material to the teachings of the present invention without departing from its scope. Therefore, it is intended that the present invention not be limited to the particular embodiment disclosed, but that the present invention will include all embodiments falling within the scope of the appended claims.
