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WO1999008380A1 - Systeme et procede perfectionnes pour ameliorer l'ecoute - Google Patents

Systeme et procede perfectionnes pour ameliorer l'ecoute Download PDF

Info

Publication number
WO1999008380A1
WO1999008380A1 PCT/US1998/010693 US9810693W WO9908380A1 WO 1999008380 A1 WO1999008380 A1 WO 1999008380A1 US 9810693 W US9810693 W US 9810693W WO 9908380 A1 WO9908380 A1 WO 9908380A1
Authority
WO
WIPO (PCT)
Prior art keywords
signal
voice
audio
filter
end user
Prior art date
Application number
PCT/US1998/010693
Other languages
English (en)
Inventor
Ronald D. Blum
William E. Kokonaski
William R. Saunders
Michael A. Vaudrey
Original Assignee
Hearing Enhancement Company, L.L.C.
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Hearing Enhancement Company, L.L.C. filed Critical Hearing Enhancement Company, L.L.C.
Priority to AU76956/98A priority Critical patent/AU7695698A/en
Publication of WO1999008380A1 publication Critical patent/WO1999008380A1/fr

Links

Classifications

    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G9/00—Combinations of two or more types of control, e.g. gain control and tone control
    • H03G9/005—Combinations of two or more types of control, e.g. gain control and tone control of digital or coded signals
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G5/00—Tone control or bandwidth control in amplifiers
    • H03G5/16—Automatic control
    • H03G5/18—Automatic control in untuned amplifiers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B1/00—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission
    • H04B1/06—Receivers
    • H04B1/10—Means associated with receiver for limiting or suppressing noise or interference

Definitions

  • the present invention covers the separation and or blending of predominantly voice[ band ] bandwidth audio from[ background ] remaining audio in an audio program in such a way that the end user can adjust the volume of the mostly voice bandwidth audio and the[ background ] remaining audio separately to his hearing needs or taste
  • the present invention includes a variety of means for separating the majority of voice band from the background audio such as analog or digital bandpass filtering, [ or ]adaptive speech filtering using various digital signal processing techniques or obtaining a previously recorded, mostly voice, signal
  • the applications for this invention include all audio playback or received modalities including but not limited to, video recording, television and radio broadcasting, as well as, tape[ and ]. DVD.
  • the present invention pertains to the application of mostly separate audio signal controls to adjust the background and voice band components in a[ program] signal to optimize the mixture of these two signals to the listeners personal taste
  • This invention can be utilized with CDs, television, radio, tape recorded audio programming or any kind of audio playback or received modalities but not limited to hearing aids, in which the listener can personally adjust or blend background and voice band signals to optimize the combination of both for his particular hearing needs BACKGROUND OF THE INVENTION
  • Shiraki U S Patent No 5, 197,100
  • Shiraki The technique employed by Shiraki involves the use of band pass filtering in combination with[ summering ] summing and subtracting circuits to form a " voice channel" that would be differentiated from the rest of the audio programming
  • the problem with the approach of Shiraki is that the individual listener cannot adjust the degree to which the center channel is amplified, attenuated or blended to optimize his hearing abilities O 99/08380
  • This invention contemplates the use of "pure voice" as recorded in a studio and its mixing with whatever background is desired, e g, an actual or simulated crowd noise at a football game while the announcer, in a soundproof booth, records the play-by-play of the game
  • the separate tracks are separately provided and mixed by the end user, not by any audio engineer, so as to better provide quality and intelligible listening specifically attuned to the end user's particular aural characteristics
  • consumers have had no input regarding their preferential mix of vocals to remaining audio, whether it be background noise or music Only the audio engineer has had any input by utilizing his or her own preference and bias in mixing the audio inputs for broadcast or recording
  • audio programming for television ( entertainment, educational, informative, etc ) movies and music is created in a manner that is suitable for obtaining the mostly "pure voice" or preferred signal referred to in this disclosure
  • each component of the overall sound is usually recorded separately Vocals, guitar, bass and drum
  • any remaining audio signal component occurring simultaneously with voiced audio can affect speech intelligibility
  • Some examples occurring in general audio background include[s] music, traffic noise, wind, running water, etc
  • a significant portion of these interfering sounds may not reside in the voice frequency band ( roughly 200 to 6000Hz ) and if [filtered] removed from the audio signal by filtering would result in an audio signal which is predominantly voice It is acknowledged, however, that certain background sounds that fall within the voice frequency band will remain along with the voice components
  • a listener is provided a means of adjusting the volume of the frequency limited audio, relative to the unfiltered audio program, an optimal combination of mostly voice and background is achieved by the end user in a ratio of the unfiltered audio to the frequency limited audio specifically customized for the end user
  • This invention provides a means for the end listener to adjust and blend the voice band audio and the background audio independently, to compensate for his hearing loss or obtain a more tailored audio program, specific to
  • the voice band signal and the full audio signal is adjusted, by way of example only and not limiting the scope of this invention, by sending the two signals to separate variable gain amplifiers that the listener adjusts a feature such as a button, [ with a dial or Jknob, dial or other equivalent adjustment on the audio receiver, tape player , VCR, DVD or CD player, or any other such audio playback or receiving modality such as, but not limited to, hearing aids or a headset which can be controlled by remote control such as by infra-red or by radio signals
  • Yet another object of this invention is to provide a novel circuit means for splitting an audio signal into two components, one containing mostly voice band and the other [background] remaining audio, which can be background noise, and providing an adjustment capability whereby the listener can adjust the total signal, later summed, to his own needs including those incident to hearing loss in certain frequency ranges
  • Still another object of this invention is to provide a system and method for the separation and blending of voice band audio from background or remaining audio in an audio program in such a way that the end listener can adjust the volume of the mostly voice band audio and the background separately to satisfy his hearing needs or taste It yet another object of this invention to provide a system for separating the voice band audio from the background audio in an audio program of any nature using a variety of means such as analog or digital bandpass filtering or adaptive speech filtering using digital signal processing techniques
  • a further object of this invention is to provide mostly pure voice to an end user for his or her blending with other audio, whether the other audio be background, music or another blend of audios
  • An added object of this invention is to provide a system and method by which an end user can blend mostly pure voice with other audio to "fine tune" the mix relationship to the particular characteristics of his or her own ear
  • Still a further object of this invention is to provide a method of obtaining mostly pure voice from standard audio formatting techniques prior to it being mixed with other audio by the audio engineer
  • Figure 1 is a schematic of the system using a band pass filter to create a primarily voice band signal that is adjusted together with an unfiltered audio background signal
  • Figure 2 is a schematic of the system of this invention utilizing an adaptive speech filter
  • FIG. 3 is a schematic of the system of this invention using an analog band pass filtering with Butterworth filters
  • Figure 4 is a schematic of the system utilizing a digital components including an anti-aliasing filter, converters, low pass filters, attenuators and processing compressors, and
  • Figure 5 is a schematic of the system showing the use of a band stop filter to create a background audio signal
  • Figure 1 illustrates the use of band pass filtering to create a primarily voice band signal that is adjusted, together with the unfiltered audio signal to provide the listener with the ability to adjust the amplitude of the voice bandwidth limited signal with the full spectrum audio
  • the incoming audio signal 1 is split [with part of the signal going to a] such that the same signal goes to a voice band pass filter 2 and [part going directly] to a variable gain amplifier 3
  • the band pass filtering is most easily accomplished using multistage analog filters built from discrete components (such as resistors, capacitors and op- amps ) Such a filter design is commonplace and readily available to those skilled in the art
  • the band pass filtering can be done using digital signal processing [77z/.y requires converting the analog audio signal to a digital signal and using an FFT algorithm on the incoming audio signal and subsequently performing a convolution with a band pass filter function ]
  • Digital filtering can be performed in either the time domain or frequency domain In the time domain the sampled input can be continuously convolved with the
  • Fig 2 illustrates a technique for creating the voice signal using an adaptive speech filter
  • adaptive speech filters exist in the open literature and have met with varying success in reducing and/or eliminating background noise from the speech portion of an audio signal
  • the incoming audio signal 6 is again split [ A portion of the ]
  • the signal is sent to a band pass filter 7 to remove the information below and above the frequency band information
  • the filtered signal is then sent to an adaptive filter 8 to remove the stationary audio from the voice audio which is in general nonstationary audio
  • the output from the adaptive speech filter is then sent to a variable gain amplifier 9 for adjustment of the amplitude by the listener
  • the unfiltered [portion ] of the signal is sent through a delay 10 so that the signals are in phase when recombined
  • the output from the delay is then sent to a variable gain amplifier 1 1 to adjust its volume relative to voice band signal
  • Both signals are summed with a summing amplifier 12 to produce the final audio output
  • speech extraction algorithms currently on the market and still in development stages This invention does not
  • FIG. 3 illustrates an example of an analog approach to band pass filtering using Butterworth filters [ Butterworth filters are often used as the best compromise between flatness of response in the pass band and minimal phase delay ] Butterworth filters provide a design solution to filtering whose magnitude response is maximally flat in both the passband and stopband
  • an audio signal 13 [ is split, part of the signal ] goes through a five pole high pass Butterworth filter 14 with a corner frequency determined by specific values of the resistors and capacitors in the circuit
  • the component values are as follows
  • This high pass filter produces the lower corner frequency of the band pass filter, rapidly attenuating any frequencies below 200 Hz
  • the signal output from the high pass filter is then sent to a low pass filter 15
  • This filter is also a five pole Butterworth filter For a cut off frequency of approximately 3000 Hz, the following component values are chosen
  • This low pass filter forms the upper [frequency] corner frequency of the band pass filter, in this case 3000 Hz
  • This example is similar to telephony applications where the full range of voice frequencies are limited to just the range required intelligibility The selection of this range should not in anyway limit the scope of the instant invention
  • Each [file] five pole filter produces approximately 30 dB of attenuation per octave, which should provide sufficient isolation of the desired voice band audio signal
  • variable gain amplifier 16 which is used to adjust the volume of the voice band limited audio signal[,] [ a]
  • variable gain amplifier 17 is used to adjust the volume on the unfiltered audio signal
  • the two signals are then added together at a summing junction of another variable gain amplifier 18 which controls the volume of the total combined signalf.
  • the variable gain amplifier can be[ simple ] implemented using potentiometers with isolation amplifiers or digitally controlled variable gain amplifiers It is understood that many methods of analog filtering are available in the literature and these are all equivalents and the instant example should not be considered the only method for providing such filtering nor is it intended to limit the scope of the present invention
  • Figure 4 illustrates a digital approach to the current invention Initially the audio signal 20 is sent to an anti-aliasing filter 21 This filter is designed in a manner similar to the low pass stage of the band pass filter from figure 3 The filtered signal is then digitized with [ Sigma Delta ] an A/D converter 22 that converts the time varying voltage into a string of digital pulses
  • Figure 5 illustrates the use of a band stop filter to create a background audio signal, where the voice band frequencies are attenuated or removed from the signal, and a voice band signal that is formed by using a band pass filter as before
  • the incoming audio signal 43 is split with part of the signal going to a band stop filter 44
  • the output from the band stop filter is sent to a variable gain amplifier 45
  • the other part of the split signal is sent to a band pass filter 46 to form the voice band signal
  • the output from the band pass filter is then sent to a variable gain amplifier 47
  • the listener can adjust the volume of the background signal with a knob that controls the gain of the variable gain amplifier
  • the corner frequencies for the band pass and band stop filters are the same or very similar so that the frequencies that are attenuated by the band stop filter are the same frequencies that are parsed by the band pass filter
  • the signals from the variable gain amplifiers are then sent to a summing amplifier 48 to produce the final audio output 49
  • the summing amplifier may be adjusted by the listener and
  • the pure voice signal can be obtained separately from the remaining audio ( voice and background ) signal
  • These two signals are the two signals which are derived or extracted by the methods of bandpass filtering and speech extraction as previously described
  • the two signals are independent of one another and the speech intelligibility of the mostly pure voice depends only on extraneous noise that is present in the vocal track
  • the announcers microphone will also measure the crowd noise as well as his own voice, and thus speech intelligibility of the "pure voice" track will begin slightly degraded
  • the method of simultaneous delivery of the mostly voice signal and the remaining audio signal is different for different mediums and is not the specific focus of this invention
  • the instant invention seeks to claim that keeping the voice or vocal components of any programming separate from the remaining audio programming ( background or non-voiced components ) when the recording process initiates, can drastically improve the ability of the user adjustment hardware to improve speech intelligibility This is primarily because the mostly voice signal that is being adjusted already has excellent intelligibility as a result of the initial recording.
  • each signal is then sent through a variable gain amplifier and summed as described earlier for the other methods of obtaining mostly voice
  • While the instant invention does not have as a specific object to claim new methods of delivering another audio track to the end user, it does contemplate and claim that providing an audio track with mostly pure voice only to an end user for mixing by him or her with remaining audio is unique
  • the ability of the end user to mix audios requires that two signals be delivered to the end user simultaneously
  • This delivery mechanism will be different for different mediums and may include airwaves, such as the additional bandwidth now existing in total aural carrier for television programming Additional tracks on CDs or tapes can be used DVDs now come equipped with multiple tracks for storing alternative information but have no means of allowing end user adjustment of separate signals to his or her hearing preferences

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)

Abstract

La présente invention vise à séparer le signal sonore de bande vocale du signal sonore de fond dans un programme audio (43). L'utilisateur final peut régler séparément le volume de la bande principalement vocale (47) et du fond (45) en fonction de ses besoins ou de son goût en matière d'audition. Un filtre passe-bande (46) analogique ou numérique ou bien un filtre vocal adaptatif peuvent être utilisés pour extraire le signal sonore de bande vocale.
PCT/US1998/010693 1997-08-08 1998-05-29 Systeme et procede perfectionnes pour ameliorer l'ecoute WO1999008380A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
AU76956/98A AU7695698A (en) 1997-08-08 1998-05-29 Improved listening enhancement system and method

Applications Claiming Priority (4)

Application Number Priority Date Filing Date Title
US90750397A 1997-08-08 1997-08-08
US08/907,503 1997-08-08
US5930398A 1998-04-14 1998-04-14
US09/059,303 1998-04-14

Publications (1)

Publication Number Publication Date
WO1999008380A1 true WO1999008380A1 (fr) 1999-02-18

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PCT/US1998/010693 WO1999008380A1 (fr) 1997-08-08 1998-05-29 Systeme et procede perfectionnes pour ameliorer l'ecoute

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AU (1) AU7695698A (fr)
WO (1) WO1999008380A1 (fr)

Cited By (10)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2000057673A1 (fr) * 1999-03-24 2000-09-28 Koninklijke Philips Electronics N.V. Filtre a elimination de bande
GB2353193A (en) * 1999-06-22 2001-02-14 Yamaha Corp Sound processing
EP1076928A1 (fr) * 1998-04-14 2001-02-21 Hearing Enhancement Company, Llc. Commande de volume reglable par l'utilisateur d'adaptation de la capacite auditive
US6985594B1 (en) * 1999-06-15 2006-01-10 Hearing Enhancement Co., Llc. Voice-to-remaining audio (VRA) interactive hearing aid and auxiliary equipment
US7003120B1 (en) 1998-10-29 2006-02-21 Paul Reed Smith Guitars, Inc. Method of modifying harmonic content of a complex waveform
EP1558061A3 (fr) * 2004-01-16 2007-01-17 Anthony John Andrews Système pour le positionnement d'événements de son
US7415120B1 (en) 1998-04-14 2008-08-19 Akiba Electronics Institute Llc User adjustable volume control that accommodates hearing
CN101123830B (zh) * 2006-08-09 2010-10-06 索尼株式会社 用于处理音频信号的设备及方法
WO2013191953A1 (fr) * 2012-06-18 2013-12-27 Google Inc. Système et procédé pour élimination sélective de contenu audio d'un enregistrement audio mixte
WO2016054763A1 (fr) * 2014-10-06 2016-04-14 Motorola Solutions, Inc. Procédés et systèmes pour réglage intelligent de volume à double canal

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4052559A (en) * 1976-12-20 1977-10-04 Rockwell International Corporation Noise filtering device
US5323467A (en) * 1992-01-21 1994-06-21 U.S. Philips Corporation Method and apparatus for sound enhancement with envelopes of multiband-passed signals feeding comb filters
US5569038A (en) * 1993-11-08 1996-10-29 Tubman; Louis Acoustical prompt recording system and method

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4052559A (en) * 1976-12-20 1977-10-04 Rockwell International Corporation Noise filtering device
US5323467A (en) * 1992-01-21 1994-06-21 U.S. Philips Corporation Method and apparatus for sound enhancement with envelopes of multiband-passed signals feeding comb filters
US5569038A (en) * 1993-11-08 1996-10-29 Tubman; Louis Acoustical prompt recording system and method

Cited By (19)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP1076928A4 (fr) * 1998-04-14 2007-10-17 Hearing Enhancement Co Llc Commande de volume reglable par l'utilisateur d'adaptation de la capacite auditive
US8284960B2 (en) 1998-04-14 2012-10-09 Akiba Electronics Institute, Llc User adjustable volume control that accommodates hearing
EP1076928A1 (fr) * 1998-04-14 2001-02-21 Hearing Enhancement Company, Llc. Commande de volume reglable par l'utilisateur d'adaptation de la capacite auditive
EP2009785A3 (fr) * 1998-04-14 2009-03-25 Hearing Enhancement Company, Llc. Méthode et dispositif permettant un utilisateur final de adjuster des préférences d'un auditeur pour des malentendants et des non-malentendants
US7415120B1 (en) 1998-04-14 2008-08-19 Akiba Electronics Institute Llc User adjustable volume control that accommodates hearing
US7003120B1 (en) 1998-10-29 2006-02-21 Paul Reed Smith Guitars, Inc. Method of modifying harmonic content of a complex waveform
WO2000057673A1 (fr) * 1999-03-24 2000-09-28 Koninklijke Philips Electronics N.V. Filtre a elimination de bande
US6985594B1 (en) * 1999-06-15 2006-01-10 Hearing Enhancement Co., Llc. Voice-to-remaining audio (VRA) interactive hearing aid and auxiliary equipment
USRE42737E1 (en) 1999-06-15 2011-09-27 Akiba Electronics Institute Llc Voice-to-remaining audio (VRA) interactive hearing aid and auxiliary equipment
US7162045B1 (en) 1999-06-22 2007-01-09 Yamaha Corporation Sound processing method and apparatus
GB2353193B (en) * 1999-06-22 2004-08-25 Yamaha Corp Sound processing method and apparatus
GB2353193A (en) * 1999-06-22 2001-02-14 Yamaha Corp Sound processing
EP1558061A3 (fr) * 2004-01-16 2007-01-17 Anthony John Andrews Système pour le positionnement d'événements de son
CN101123830B (zh) * 2006-08-09 2010-10-06 索尼株式会社 用于处理音频信号的设备及方法
WO2013191953A1 (fr) * 2012-06-18 2013-12-27 Google Inc. Système et procédé pour élimination sélective de contenu audio d'un enregistrement audio mixte
US9195431B2 (en) 2012-06-18 2015-11-24 Google Inc. System and method for selective removal of audio content from a mixed audio recording
US11003413B2 (en) 2012-06-18 2021-05-11 Google Llc System and method for selective removal of audio content from a mixed audio recording
WO2016054763A1 (fr) * 2014-10-06 2016-04-14 Motorola Solutions, Inc. Procédés et systèmes pour réglage intelligent de volume à double canal
US10014839B2 (en) 2014-10-06 2018-07-03 Motorola Solutions, Inc. Methods and systems for intelligent dual-channel volume adjustment

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