|Publication number||USRE42737 E1|
|Application number||US 11/972,564|
|Publication date||Sep 27, 2011|
|Filing date||Jan 10, 2008|
|Priority date||Jun 15, 1999|
|Also published as||CA2374879A1, CN1201632C, CN1370386A, EP1190597A1, EP1190597A4, EP1190597B1, US6985594, WO2000078093A1|
|Publication number||11972564, 972564, US RE42737 E1, US RE42737E1, US-E1-RE42737, USRE42737 E1, USRE42737E1|
|Inventors||Michael A. Vaudrey, William R. Saunders|
|Original Assignee||Akiba Electronics Institute Llc|
|Export Citation||BiBTeX, EndNote, RefMan|
|Patent Citations (92), Non-Patent Citations (10), Classifications (21), Legal Events (5)|
|External Links: USPTO, USPTO Assignment, Espacenet|
The present application claims the benefit of U.S. provisional patent application Ser. No. 60/139,243 entitled “Voice-to-Remaining Audio (VRA) Interactive Hearing Aid & Auxiliary Equipment,” filed on Jun. 15, 1999.
This patent application is a reissue application for commonly assigned U.S. Pat. No. 6,985,594, issued from U.S. patent application Ser. No. 09/593,149, filed on Jun. 14, 2000.
Embodiments of the present invention relate generally to processing audio signals, and more particularly, to a method and apparatus for processing audio signals such that hearing impaired listeners can adjust the level of voice-to-remaining audio (VRA) to improve their listening experience.
As one ages and progresses through life, over time due to many factors, such as age, genetics, disease, and environmental effects, one's hearing becomes compromised. Usually, the deterioration is specific to certain frequency ranges.
In addition to permanent hearing impairments, one may experience temporary hearing impairments due to exposure to particular high sound levels. For example, after target shooting or attending a rock concert one may have temporary hearing impairments that improve somewhat, but over time may accumulate to a permanent hearing impairment. Even lower sound levels than these but longer lasting may have temporary impacts on one's hearing, such as working in a factory or teaching in a elementary school.
Typically, one compensates for hearing loss or impairment by increasing the volume of the audio. But, this simply increases the volume of all audible frequencies in the total signal. The resulting increase in total signal volume will provide little or no improvement in speech intelligibility, particularly for those whose hearing impairment is frequency dependent.
While hearing impairment increases generally with age, many hearing impaired individuals refuse to admit that they are hard of hearing, and therefore avoid the use of devices that may improve the quality of their hearing. While many elderly people begin wearing glasses as they age, a significantly smaller number of these individuals wear hearing aids, despite the significant advances in the reduction of the size of hearing aids. This phenomenon is indicative of the apparent societal stigma associated with hearing aids and/or hearing impairments. Consequently, it is desirable to provide a technique for improving the listening experience of a hearing impaired listener in a way that avoids the apparent associated societal stigma.
Most audio programming, be it television audio, movie audio, or music can be divided into two distinct components: the foreground and the background. In general, the foreground sounds are the ones intended to capture the audiences attention and retain their focus, whereas the background sounds are supporting, but not of primary interest to the audience. One example of this can be seen in television programming for a “sitcom,” in which the main character's voices deliver and develop the plot of the story while sound effects, audience laughter, and music fill the gaps.
Currently, the listening audience for all types of audio media are restricted to the mixture decided upon by the audio engineer during production. The audio engineer will mix all other background noise components with the foreground sounds at levels that the audio engineer prefers, or at which the audio engineer understands have some historical basis. This mixture is then sent to the end-user as either a single (mono) signal or in some cases as a stereo (left and right) signal, without any means for adjusting the foreground to the background.
The lack of this ability to adjust foreground relative to background sounds is particularly difficult for the hearing impaired. In many cases, programming is difficult to understand (at best) due to background audio masking the foreground signals.
There are many new digital audio formats available. Some of these have attempted to provide capability for the hearing impaired. For example, Dolby Digital, also referred to as AC-3 (or Audio Codec version 3), is a compression technique for digital audio that packs more data into a smaller space. The future of digital audio is in spatial positioning, which is accomplished by providing 5.1 separate audio channels: Center, Left and Right, and Left and Right Surround. The sixth channel, referred to as the 0.1 channel refers to a limited bandwidth low frequency effects (LFE) channel that is mostly non-directional due to its low frequencies. Since there are 5.1 audio channels to transmit, compression is necessary to ensure that both video and audio stay within certain bandwidth constraints. These constraints (imposed by the Federal Communications Commission (FCC)) are more strict for terrestrial transmission than for digital video disk (DVD)s, currently. There is more than enough space on a DVD to provide the end-user with uncompressed audio (much more desirable from a listening standpoint). Video data is compressed most commonly through MPEG (moving pictures experts group) developed techniques, although they also have an audio compression technique very similar to Dolby's.
The DVD industry has adopted Dolby Digital (DD) as its compression technique of choice. Most DVD's are produced using DD. The ATSC (Advanced Television Standards Committee) has also chosen AC-3 as its audio compression scheme for American digital TV. This has spread to many other countries around the world. This means that production studios (movie and television) must encode their audio in DD for broadcast or recording.
There are many features, in addition to the strict encoding and decoding scheme, that are frequently discussed in conjunction with Dolby Digital. Some of these features are part of DD and some are not. Along with the compressed bitstream, DD sends information about the bitstream called metadata, or “data about the data.” It is basically zero's and ones indicating the existence of options available to the end-user. Three of these options are dialnorm (dialog normalization), dynrng (dynamic range), and bsmod (bit stream mode that controls the main and associated audio services). The first two are an integral part of DD already, since many decoders handle these variables, giving end-users the ability to adjust them. The third bit of information, bsmod, is described in detail in ATSC document A/54 (not a Dolby publication) but also exists as part of the DD bitstream. The value of bsmod alerts the decoder about the nature of the incoming audio service, including the presence of any associated audio service. At this time, no known manufacturers are utilizing this parameter. Multiple language DVD performances are currently provided via multiple complete main audio programs on one of the eight available audio tracks on the DVD.
The dialnorm parameter is designed to allow the listener to normalize all audio programs relative to a constant voice level. Between channels and between program and commercial, overall audio levels fluctuate wildly. In the future, producers will be asked to insert the dialnorm parameter which indicates the sound pressure level (SPL)s at which the dialog has been recorded. If this value is set as 80 dB for a program but 90 dB for a commercial, the television will decode that information examine the level the end-user has entered as desirable (say 85 dB) and will adjust the movie up 5 dB and the commercial down 5 dB. This is a total volume level adjustment that is based on what the producer enters as the dialnorm bit value.
A section from the AC-3 description (from document A/52) provides the best description of this technology. “The dynrng values typically indicate gain reduction during the loudest signal passages, and gain increase during the quiet passages. For the listener, it is desirable to bring the loudest sounds down in level towards the dialog level, and the quiet sounds up in level, again towards dialog level. Sounds which are at the same loudness as the normal spoken dialogue will typically not have their gain changed.”
The dynrng variable provides the end-user with an adjustable parameter that will control the amount of compression occurring on the total volume with respect to the dialog level. This essentially limits the dynamic range of the total audio program about the mean dialog level. This does not, however, provide any way to adjust the dialog level independently of the remaining audio level.
One attempt to improve the listening experience of hearing impaired listeners is provided for in The ATSC, Digital Television Standard (Annex B). Section 6 of Annex B of the ATSC standard describes the main audio services and the associated audio services. An AC-3 elementary stream contains the encoded representation of a single audio service. Multiple audio services are provided by multiple elementary streams. Each elementary stream is conveyed by the transport multiplex with a unique PID. There are a number of audio service types which may be individually coded into each elementary stream. One of the audio service types is called the complete main audio service (CM). The CM type of main audio service contains a complete audio program (complete with dialogue, music and effects). The CM service may contain from 1 to 5.1 audio channels. The CM service may be further enhanced by means of the other services. Another audio service type is the hearing impaired service (HI). The HI associated service typically contains only dialogue which is intended to be reproduced simultaneously with the CM service. In this case, the HI service is a single audio channel. As stated therein, this dialogue may be processed for improved intelligibility by hearing impaired listeners. Simultaneous reproduction of both the CM and HI services allows the hearing impaired listener to hear a mix of the CM and HI services in order to emphasize the dialogue while still providing some music and effects. Besides providing the HI service as a single dialogue channel, the HI service may be provided as a complete program mix containing music, effects, and dialogue with enhanced intelligibility. In this case, the service may be coded using any number of channels (up to 5.1). While this service may improve the listening experience for some hearing impaired individuals, it certainly will not for those who do not employ the proscribed receiver for fear of being stigmatized as hearing impaired. Finally, any processing of the dialogue for hearing impaired individuals prevents the use of this channel in creating an audio program for non-hearing individuals. Moreover, the relationship between the HI service and the CM service set forth in Annex B remains undefined with respect to the relative signal levels of each used to create a channel for the hearing impaired.
Other techniques have been employed to attempt to improve the intelligibility of audio. For example, U.S. Pat. No. 4,024,344 discloses a method of creating a “center channel” for dialogue in cinema sound. This technique disclosed therein correlates left and right stereophonic channels and adjusts the gain on either the combined and/or the separate left or right channel depending on the degree of correlation between the left and right channel. The assumption being that the strong correlation between the left and right channels indicates the presence of dialogue. The center channel, which is the filtered summation of the left and right channels, is amplified or attenuated depending on the degree of correlation between the left and right channels. The problem with this approach is that it does not discriminate between meaningful dialogue and simple correlated sound, nor does it address unwanted voice information within the voice band. Therefore, it cannot improve the intelligibility of all audio for all hearing impaired individuals.
In general, the previously cited inventions of Dolby and others have all attempted to modify some content of the audio signal through various signal processing hardware or algorithms, but those methods do not satisfy the individual needs or preferences of different listeners. In sum, all of these techniques provide a less than optimum listening experience for hearing impaired individuals as well as non-hearing impaired individuals.
Finally, miniaturized electronics and high quality digital audio has brought about a revolution in the digital hearing aid technology. In addition, the latest standards of digital audio transmission and recordings including DVD (in all formats), digital television, Internet radio, and digit radio, are incorporating sophisticated compression methods that allow an end-user unprecedented control over audio programming. The combination of these two technologies has presented improved methods for providing hearing impaired end-users with the ability to enjoy digital audio programming. This combination, however, fails to address all of the needs and concerns of different hearing impaired end-users.
The present invention is therefore directed to the problem of developing a system and method for processing audio signals that optimizes the listening experience for hearing impaired listeners, as well as non-hearing impaired listeners, individually or collectively.
An integrated individual listening device and decoder for receiving an audio signal including a decoder for decoding the audio signal by separating the audio signal into a voice signal and a background signal, a first end-user adjustable amplifier coupled to the voice signal and amplifying the voice signal, a second end-user adjustable amplifier coupled to the background signal and amplifying the background signal, a summing amplifier coupled to outputs of said first and second end-user adjustable amplifiers and outputting a total audio signal, said total signal being coupled to an individual listening device.
Embodiments of the present invention are directed to an integrated individual listening device and decoder. An example of one such decoder is a Dolby Digital (DD) decoder. As stated above, Dolby Digital is an audio compression standard that has gained popularity for use in terrestrial broadcast and recording media. Although the discussion herein uses a DD decoder, other types of decoders may be used without departing from the spirit and scope of the present invention. Moreover, other digital audio standards besides Dolby Digital are not precluded. This embodiment allows a hearing impaired end-user in a listening environment with other listeners, to take advantage of the “Hearing Impaired Associated Audio Service” provided by DD without affecting the listening enjoyment of the other listeners. As used herein, the term “end-user” refers to a consumer, listener or listeners of a broadcast or sound recording or a person or persons receiving an audio signal on an audio media that is distributed by recording or broadcast. In addition, the term “individual listening device” refers to hearing aids, headsets, assistive listening devices, cochlear implants or other devices that assist the end-user's listening ability. Further, the term “preferred audio” refers to the preferred signal, voice component, voice information, or primary voice component of an audio signal and the term “remaining audio” refers to the background, musical or non-voice component of an audio signal.
Other embodiments of the present invention relate to a decoder that sends wireless transmissions directly to a individual listening device such as a hearing aid or cochlear implant. Used in conjunction with the “Hearing Impaired Associated Audio Service” provided by DD which provides separate dialog along with a main program, the decoder provides the hearing impaired end-user with adjustment capability for improve intelligibility with other listeners in the same listening environment while the other listeners enjoy the unaffected main program.
Further embodiments of the present invention relate to an interception box which services the communications market when broadcast companies transition from analog transmission to digital transmission. The intercept box allows the end-user to take advantage of the hearing impaired mode (HI) without having a fully functional main/associated audio service decoder. The intercept box decodes transmitted digital information and allows the end-user to adjust hearing impaired parameters with analog style controls This analog signal is also fed directly to an analog play device such as a television. According to the present invention, the intercept box can be used with individual listening devices such as hearing aids or it can allow digital services to be made available to the analog end-user during the transition period.
Significance of Ratio of Preferred Audio to Remaining Audio
The present invention begins with the realization that the listening preferential range of a ratio of a preferred audio signal relative to any remaining audio is rather large, and certainly larger than ever expected. This significant discovery is the result of a test of a small sample of the population regarding their preferences of the ratio of the preferred audio signal level to a signal level of all remaining audio.
Specific Adjustment of Desired Range for Hearing Impaired or Normal Listeners
Very directed research has been conducted in the area of understanding how normal and hearing impaired end-users perceive the ratio between dialog and remaining audio for different types of audio programming. It has been found that the population varies widely in the range of adjustment desired between voice and remaining audio.
Two experiments have been conducted on a random sample of the population including elementary school children, middle school children, middle-aged citizens and senior citizens. A total of 71 people were tested. The test consisted of asking the end-user to adjust the level of voice and the level of remaining audio for a football game (where the remaining audio was the crowd noise) and a popular song (where the remaining audio was the music). A metric called the VRA (voice to remaining audio) ratio was formed by dividing the linear value of the volume of the dialog or voice by the linear value of the volume of the remaining audio for each selection.
Several things were made clear as a result of this testing. First, no two people prefer the identical ratio for voice and remaining audio for both the sports and music media. This is very important since the population has relied upon producers to provide a VRA (which cannot be adjusted by the consumer) that will appeal to everyone. This can clearly not occur, given the results of these tests. Second, while the VRA is typically higher for those with hearing impairments (to improve intelligibility) those people with normal hearing also prefer different ratios than are currently provided by the producers.
It is also important to highlight the fact that any device that provides adjustment of the VRA must provide at least as much adjustment capability as is inferred from these tests in order for it to satisfy a significant segment of the population. Since the video and home theater medium supplies a variety of programming, we should consider that the ratio should extend from at least the lowest measured ratio for any media (music or sports) to the highest ratio from music or sports. This would be 0.1 to 20.17, or a range in decibels of 46 dB. It should also be noted that this is merely a sampling of the population and that the adjustment capability should theoretically be infinite since it is very likely that one person may prefer no crowd noise when viewing a sports broadcast and that another person would prefer no announcement. Note that this type of study and the specific desire for widely varying VRA ratios has not been reported or discussed in the literature or prior art.
In this test, an older group of men was selected and asked to do an adjustment (which test was later performed on a group of students) between a fixed background noise and the voice of an announcer, in which only the latter could be varied and the former was set at 6.00. The results with the older group were as follows:
To further illustrate the fact that people of all ages have different hearing needs and preferences, a group of 21 college students was selected to listen to a mixture of voice and background and to select, by making one adjustment to the voice level, the ratio of the voice to the background. The background noise, in this case crowd noise at a football game, was fixed at a setting of six (6.00) and the students were allowed to adjust the volume of the announcers' play by play voice which had been recorded separately and was pure voice or mostly pure voice. In other words, the students were selected to do the same test the group of older men did. Students were selected so as to minimize hearing infirmities caused by age. The students were all in their late teens or early twenties. The results were as follows:
Setting of Voice
The ages of the older group (as seen in Table I) ranged from 36 to 59 with the preponderance of the individuals being in the 40 or 50 year old group. As is indicated by the test results, the average setting tended to be reasonably high indicating some loss of hearing across the hoard. The range again varied from 3.00 to 7.75, a spread of 4.75 which confirmed the findings of the range of variance in people's preferred listening ratio of voice to background or any preferred signal to remaining audio (PSRA). The overall span for the volume setting for both groups of subjects ranged from 2.0 to 7.75. These levels represent the actual values on the volume adjustment mechanism used to perform this experiment. They provide an indication of the range of signal to noise values (when compared to the “noise” level 6.0) that may be desirable from different end-users.
To gain a better understanding of how this relates to relative loudness variations chosen by different end-users, consider that the non-linear volumen control variation from 2.0 to 7.75 represents an increase of 20 dB or ten (10) times. Thus, for even this small sampling of the population and single type of audio programming it was found that different listeners do prefer quite drastically different levels of “preferred signal” with respect to “remaining audio.” This preference cuts across age groups showing that it is consistent with individual preference and basic hearing abilities, which was heretofore totally unexpected.
As the test results show, the range that students (as seen in Table II) without hearing infirmities caused by age selected varied considerably from a low setting of 2.00 to a high of 6.70, a spread of 4.70 or almost one half of the total range of from 1 to 10. The test is illustrative of how the “one size fits all” mentality of most recorded and broadcast audio signals falls far short of giving the individual listener the ability to adjust the mix to suit his or her own preferences and hearing needs. Again, the students had a wide spread in their settings as did the older group demonstrating the individual differences in preferences and hearing needs. One result of this test is that hearing preferences is widely disparate.
Further testing has confirmed this result over a larger sample group. Moreover, the results vary depending upon the type of audio. For example, when the audio source was music, the ratio of voice to remaining audio varied from approximately zero to about 10, whereas when the audio source was sports programming, the same ratio varied between approximately zero and about 20. In addition, the standard deviation increased by a factor of almost three, while the mean increased by more than twice that of music.
The end result of the above testing is that if one selects a preferred audio to remaining audio ratio and fixes that forever, one has most likely created an audio program that is less than desirable for a significant fraction of the population. And, as stated above, the optimum ratio may be both a short-term and long-term time varying function. Consequently, complete control over this preferred audio to remaining audio ratio is desirable to satisfy the listening needs of “normal” or non-hearing impaired listeners. Moreover, providing the end-user with the ultimate control over this ratio allows the end-user to optimize his or her listening experience.
The end-user's independent adjustment of the preferred audio signal and the remaining audio signal will be the apparent manifestation of one aspect of the present invention. To illustrate the details of the present invention, consider the application where the preferred audio signal is the relevant voice information.
Creation of the Preferred Audio Signal and the Remaining Audio Signal
Once the relevant speakers are identified, their voices will be picked up by the voice microphone 301. The voice microphone 1 will need to be either a close talking microphone (in the case of commentators) or a highly directional shot gun microphone used in sound recording. In addition to being highly directional, these microphones 301 will need to be voice-band limited, preferably from 200-5000 Hz. The combination of directionality and band pass filtering minimize the background noise acoustically coupled to the relevant voice information upon recording. In the case of certain types of programming, the need to prevent acoustic coupling can be avoided by recording relevant voice of dialogue off-line and dubbing the dialogue where appropriate with the video portion of the program. The background microphones 302 should be fairly broadband to provide the full audio quality of background information, such as music.
A camera 303 will be used to provide the video portion of the program. The audio signals (voice and relevant voice) will be encoded with the video signal at the encoder 304. In general, the audio signal is usually separated from the video signal by simply modulating it with a different carrier frequency. Since most broadcasts are now in stereo, one way to encode the relevant voice information with the background is to multiplex the relevant voice information on the separate stereo channels in much the same way left front and right front channels are added to two channel stereo to produce a quadraphonic disc recording. Although this would create the need for additional broadcast bandwidth, for recorded media this would not present a problem, as long as the audio circuitry in the video disc or tape player is designed to demodulate the relevant voice information.
Once the signals are encoded, by whatever means deemed appropriate, the encoded signals are sent out for broadcast by broadcast system 305 over antenna 313, or recorded on to tape or disc by recording system 306. In case of recorded audio video information, the background and voice information could be simply placed on separate recording tracks.
Receiving and Demodulating the Preferred Audio Signal and the Remaining Audio
In either case, these signals would be sent to a decoding system 309. The decoder 309 would separate the signals into video, voice audio, and background audio using standard decoding techniques such as envelope detection in combination with frequency or time division demodulation. The background audio signal is sent to a separate variable gain amplifier 310, that the listener can adjust to his or her preference. The voice signal is sent to a variable gain amplifier 311, that can be adjusted by the listener to his or her particular needs, as discussed above.
The two adjusted signals are summed by a unity gain summing amplifier 132 to produce the final audio output. Alternatively, the two adjusted signals are summed by unity gain summing amplifier 312 and further adjusted by variable gain amplifier 315 to produce the final audio output. In this manner the listener can adjust relevant voice to background levels to optimize the audio program to his or her unique listening requirements at the time of playing the audio program. As each time the same listener plays the same audio, the ratio setting may need to change due to changes in the listener's hearing, the setting remains infinitely adjustable to accommodate this flexibility.
Configuration of a Typical Individual Listening Device
Although the components of a hearing aid have been illustrated above, other individual listening devices as discussed above, can be used with the present invention.
Individual Listening Device and Decoder
In a room listening environment, there may be a combination of listeners with varying degrees of hearing impairments as well as listeners with normal listening. A hearing aid or other listening device as described above, can be equipped with a decoder that receives a digital signal from a programming source and separately decodes the signal, providing the end-user access to the voice, for example, the hearing impaired associated service, without affecting the listening environment of other listeners.
As stated above, preferred ratio of voice to remaining audio differs significantly for different people, especially hearing impaired people, and differs for different types of programming (sports versus music, etc.).
According to one embodiment of the present invention, the bitstream from bitstream source 220 is also supplied to repeater 222. Repeater 222 retransmits the bitstream to a plurality of personal VRA decoders 223. Each personal VRA decoder 223 includes a demodulator 266 and a decoder 267 for decoding the bitstream and variable amplifiers 225 and 226 for adjusting the voice component signal and the remaining audio signal component, respectively. The adjusted signal components are downmixed by summer 227 and may be further adjusted by variable amplifier 281. The adjusted signal is then sent to individual listening devices 224. According to one embodiment of the present invention, the personal VRA decoder is interfaced with the individual listening device and forms one unit which is denoted as 250. Alternatively, personal VRA decoder 223 and individual listening device 224 may be separate devices and communicate in a wired or wireless manner. Individual listening device 224 may be a hearing aid having the components shown in
For 5.1 channel programming, voice is primarily placed on the center channel while the remaining audio resides on left, right, left surround, and right surround. For end-users with individual listening devices, spatial positioning of the sound is of little concern since most have severe difficulty with speech intelligibility. By allowing the end-user to adjust the level of the center channel with respect to the other 4.1 channels, an improvement in speech intelligibility can be provided. These 5.1 channels are then downmixed to 2 channels, with the volume adjustment of the center channel allowing the improvement in speech intelligibility without relying on the hearing impaired mode mentioned above. This aspect of the present invention has an advantage over the fully functional AC3-type, in that an end-user can obtain limited VRA adjustment without the need of a separate dialog channel such as the hearing impaired mode.
One solution contemplated by the present invention is to provide the end-user with the ability to block the ambient sound while delivering the signal from the VRA personal decoder. This is accomplished by using an earplug as shown in
While this method will work up to the limits of the earplug ambient noise rejection capability, it has a notable drawback. For someone to enjoy a program with another person, it will likely be necessary to easily communicate while the program is ongoing. The earplug will not only block the primary audio source (which interferes with the decoded audio entering the hearing aid), but also blocks any other ambient noise indiscriminately. In order to selectively block the ambient noise generated from the primary audio reproduction system without affecting the other (desirable) ambient sounds, more sophisticated methods are required. Note that similar comments can be made concerning the acceptability of using headset decoders. The headset earcups provide some level of attenuation of ambient noise but interfere with communication. If this is not important to a hearing impaired end-user, this approach may be acceptable.
What is needed is a way to avoid the latency problems associated with airborne transmission of digital audio programming while allowing the hearing impaired listener to interact with other viewers in the same room.
One other possibility is available that combines adaptive feedforward control with fixed gain feedforward control. This option, illustrated in
Ideally, the hearing aid (H) will invert the hearing impairment, G3. Therefore the last three terms where both G3 and H appear, will have, those coefficients to be approximately one. The resulting equation is then
This does not provide the sound quality needed. While the desired and decoder signals do have level adjustment capability, the last three terms will deliver significant levels of distortion and latency both through the electrical and physical signal paths. The desired result is a combination of the pure decoder signal and the desired ambient audio signal where the end-user can control the relative mix between the two with no other signals in the output. The variables “S” and “d+G1S” are available for direct measurement and the values of H, w1, and w2 are controllable by the end-user. This combination of variable permits the adjustment capability desired. If the adaptive filter and the plant estimate (G2 hat) are now included in the equation for the output to the end end-user's nerve, it becomes:
Now, if the adaptive filter converges to the optimal solution, it will be identical to G1 so that the third and fourth terms in the above equation cancel. And if the estimate of G2 approaches G2 due to a good system identification, the last two terms in the previous equation will also cancel. This leaves only the decoder signal “S” end-user modified by w2 and the desired ambient sound “d” end-user modified by w1, the desired result. The limits of the performance of this method depend on the performance of the adaptive filter and on the accuracy of the system identification from the outside of the hearing aid to the inside of the hearing aid while the end-user has it comfortably in position. The system identification procedure itself can be carried out in a number of ways, including a least mean squares fit.
VRA set top terminal 60 includes a decoder 61 for decoding a digital bitstream supplied by a digital source such as a digital TV, DVD, etc. Decoder 61 decodes the digital bitstream and outputs digital signals which have a preferred audio component (PA) and a remaining audio portion (RA). The digital signals are feed into a digital-to-analog (D/A) converters 62 and 69 which converts the digital signals into analog signals. The analog signals from D/A converter 62 are feed to transmitter 63 to be transmitted to receivers such as receivers 270 shown in
While many changes and modifications can be made to the invention within the scope of the appended claims, such changes and modifications are within the scope of the claims and covered thereby.
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|1||"Digital Audio Compression Standard (AC-3)," ATSC, Annex C AC-3 Karaoke Mode, pp. 127- 130.|
|2||"Guide to the Use of ATSC Digital Television Standard," ATSC, Oct. 4, 1995, pp. 54-59. Available on-line at www.atsc.org/Standards/A54/.|
|3||*||ATSC Digital Television Standard, ATSC, Sep. 16, 1995, Annex B. www.atsc.org/Standards/A53/.|
|4||Chen Yingying "Transitional Product for Digital TV-Development of Set-Top-Box," Mar. 1999.|
|5||Chen Yingying "Transitional Product for Digital TV—Development of Set-Top-Box," Mar. 1999.|
|6||Digidesign's web page listing of their Aphex Aural Exciter. Available on-line at www.digidesign.com/products/all-prods.php3?location=main&product-id=8. The Examiner is encouraged to review the entire website for any relevant subject matter.|
|7||Digidesign's web page listing of their Aphex Aural Exciter. Available on-line at www.digidesign.com/products/all—prods.php3?location=main&product—id=8. The Examiner is encouraged to review the entire website for any relevant subject matter.|
|8||Digital Audio Compression Standard (AC-3), ATSC, Annex C AC-3 Karaoke Mode pp. 127-133, available on-line at www.atsc.org/Standards/A52/.|
|9||Japan Patent Office, Decision of Rejection, Japanese Patent Application 2001-502617, 2 pages, Jan. 27, 2011.|
|10||Share Incorporated homepage, available on-line at www.shure.com. The Examiner is encouraged to review the entire website for any relevant subject matter.|
|U.S. Classification||341/96, 381/307, 381/104, 381/18|
|International Classification||H03G3/00, H04R1/10, H04R5/02, H04S3/00, G10L21/00, G10L19/00, G10L21/02, H04R3/00, H04R25/00, H04R|
|Cooperative Classification||G10L21/0272, H04R25/407, H04R3/005, G10L2021/065|
|European Classification||H04R3/00B, H04R25/40F, G10L21/0272|
|Jan 20, 2011||AS||Assignment|
Owner name: AKIBA ELECTRONICS INSTITUTE LLC, DELAWARE
Free format text: ASSIGNMENT OF ASSIGNORS INTEREST;ASSIGNOR:HEARING ENHANCEMENT COMPANY LLC;REEL/FRAME:025671/0902
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Owner name: HEARING ENHANCEMENT COMPANY LLC, VIRGINIA
Free format text: ASSIGNMENT OF ASSIGNORS INTEREST;ASSIGNORS:VAUDREY, MICHAEL A.;SAUNDERS, WILLIAM R.;REEL/FRAME:025671/0852
Effective date: 20000609
|Jun 19, 2012||CC||Certificate of correction|
|Mar 18, 2013||FPAY||Fee payment|
Year of fee payment: 8
|Nov 13, 2015||AS||Assignment|
Owner name: BENHOV GMBH, LLC, DELAWARE
Free format text: MERGER;ASSIGNOR:AKIBA ELECTRONICS INSTITUTE, LLC;REEL/FRAME:037039/0739
Effective date: 20150811
|Jun 23, 2017||FPAY||Fee payment|
Year of fee payment: 12