MXPA98005726A - Process that improves an audio concert using a synchronized audifo system - Google Patents

Abstract

An audio enhancement system and method is provided in which a wireless hearing aid system comprises a transmitter (40) and a receiver (30) including a synchronization device (109) which uses electromagnetic location signals to locate the position of the position of the receiver with respect to the transmitter. The transmitter for this system will transmit a frequency modulated (FM) signal on several separate channels in the 900 MHz band interval. Each channel will carry the same audio information, however, each successive channel will tend its audio signal delayed for a period of time pre-established, for example, 50 ms, in relation to the previous channel. The hearing aid receiver (30), which supports the location and associated hardware location signals, will select the appropriate channel based on the distance of the listener from the main loud speakers. These channels are placed so that when they are in a large area, and the appropriate channel is chosen, the sound received electronically on the wireless channel will be approximately in phase with the sound that reaches the listener from the main speakers

Description

* _ _ _ PROCESS THAT IMPROVES AN AUDIO CONCERT USING A AUDIO FONTS SYSTEM J.INTECIDENTS OF THE INVENTION a) Field of the Invention The present invention relates generally to audio systems and more particularly to systems for improving the sound received by individuals in transit and located in defined positions separated from the primary speaker system. The present audio system allows individuals in transit to move within a predetermined area without experiencing nerves in the sound quality provided to these individuals. b) Description of the Related Art The current state of the art for the reproduction of sound or sound support equipment used in concert halls or in other indoor or outdoor spaces involves the use of one or more positions of groups of loudspeakers. These locations are typically located at or near the physical location of the actual sound source or virtual sound source. Unfortunately, the reproduction quality - 2 - of such conventional systems, the acoustic sound is affected in a harmful way by distortion of the frequency and time spectrum that results from the distances through which the sound travels. In addition, distortions of a non-linear type are introduced due to the physical compression of air and branches through which the sound propagates. Further, since the perceived sound intensity and sound pressure level decreases in proportion to the instances that the sound sources travel, in order to obtain the desired sound pressure level, a remote listener 10 places substantially more pressure on the sound. sound that must be developed at the source. However, increasing the sound pressure level in these defined places produces an increase in distortion. People who attend concerts, performances or ; ^, 15 conferences in large halls or in auditoriums (interior as well as exterior) become more demanding in their desire for high quality sound; they want to have a sound quality supplied to their specific position by public addressing systems which mimic the quality of recording studio or so least imitate the sound quality in the mixing cabinet with the main speakers. A common approach taken by sound system designers is to use "delayed speaker systems" in combination with the main speaker system. In particular, additional speakers are provided in positions remote in order to direct the reproduction of sound quality to individuals who are placed in a bad position to receive the sound from the main speaker system. These fixed remote speakers typically have their input signals delayed in time with respect to the signal and proportionate to main loudspeaker systems to synchronize their acoustic output with the sound coming from the main system - ^ k speakers; this approach reduces feedback echo which results in two sources of sound which are deviated in distance. However, these fixed remote speakers fail to adequately serve individuals in transit. In an attempt to provide an improved audio system, U.S. Patent No. 5,432,858 to Clair Jr., et al. , describes an audio system comprising a wireless transmitter and a plurality of reproductive systems - 15 enhanced sound. Each sound subsystem is a unit ~ Portable arranged to be transported by a person located in a remote position with respect to the main speaker. Each sound subsystem includes a receiver to receive a broadcast signal, and a microphone placed in a hearing aid to detect the sound coming from the main speakers. The sound subsystem also includes circuits which increase this broadcast signal in order to synchronize the broadcast signal with the sound coming from the main speakers. In order to increase the broadcast signal In accordance with the description of this patent, the subsystem uses a set of delay circuits provided in the set of subsystem hearing aids which delays the broadcast signal received by the receiver for a predetermined period of time which generally corresponds with time 5 is needed for the sound to come from the main speakers to propagate through the air to the remote position of the hearing instruments. The sound enhancement system described by U.S. Patent No. 5,432,858 takes one of three forms: a "zone" system, a "manually synchronized" system and a "self-synchronized" system. For the "zone" system the audience is divided into defined zones, which cover a known distance from the main source of sound. Each listener is located within a given area and receives an increased sound from a particular receiver / transducer subsystem delayed by a predetermined time. Consequently, the augmented sound and the main sound reach the ears of each listener within that zone with substantial synchrony. More particularly, members of the audience within each zone tune in personally its respective receiver to the appropriate channel for its zone, and in this way listen to the sound reproduced by the associated remote transducer in substantial synchrony with the arriving main sound. However, each person who arrives at a concert where the "zone" system of this The invention must receive instructions on how and why to tune its receiver / amplifier unit in a particular channel adjusted based on that individual's position. It will be understood by anyone familiar with the typical environments in a concert, however, that such a system will be too complicated 5 and impractical for its distribution and use. In addition, this system excessively limits the portable capacity of an audio system ^ ¡^ * because the "zone" system requires the user to manually tune his receiver during the movement in the concert hall. The second "manually synchronized" system of U.S. Patent No. 5,432,858 is even more limiting than the "zone" system described above. The "manually synchronized" system requires the listener to manually adjust their time delay circuits. With this provision, the whole of the audience is covered by a single transmitting zone, where the audio signal is broadcast on a single frequency by a single wireless transmitter common to all the receiver / transducer subsystems located throughout the concert hall. Again it will be understood for anyone familiar with the typical concert environments, however, that such a "manually synchronized" system will be too complicated and impractical for both distribution and use. The third "auto-synchronized" system of US Pat. No. 5,432,858 carries out the synchronization of the broadcast signal and the sound coming from the speakers # by providing a sampling microphone in the portable transducer unit. The circuit of the portable transducer unit automatically adjusts the time delay in response to the sound picked up by the sampling microphone. East The "self-synchronized" system suffers from the drawback that it requires very complex, expensive and bulky circuits. £ ^ Specifically, the receiving / amplifying unit requires a wireless receiver, dynamic signal processors with an activating circuit, a control signal delay circuit programmable, a signal gate, a microphone preamplifier, an adder circuit and a signal correlation circuit. The signal correlation circuit itself comprises a correlation circuit and a controller. Of course, the sampling microphone is inherently susceptible That is to 15 background noise, and therefore requires additional w-means to inactivate the microphone when it is not in the presence of the main sound that arrives. Although the previous approaches to obtain an improvement in sound have some aural benefits, these systems Conventional ones, however, suffer from numerous drawbacks resulting from a diminished quality of the sound supplied to remote listeners. These systems also limit the listener to specific listening areas, and therefore do not meet the listener's needs of a mobile audience. In addition, systems of the prior art result in a relatively complex, unmanageable and inflexible sound reproduction system. Therefore, the resulting size, weight and cost of these prior art receivers are limiting. Accordingly, there is a need for an audio enhancement system which overcomes the disadvantages of the prior art.

BRIEF DESCRIPTION OF THE INVENTION The general objective of this invention is to provide an audio enhancement system which solves the disadvantages in the prior art. A further objective of this invention is to provide an audio enhancement system to provide a signal ¿^ ~ 15 synchronized to people in transit who are located in distances "remote from the main speakers so that the synchronized signal provides studio-quality sound, or at least a mix-room-quality sound, in synchronization with the sound supplied by the speakers main. In accordance with these and other objects of the present invention, there is provided an audio enhancement system and method in which a wireless hearing aid system comprises a transmitter and a receiver which use a frequency band that does not require a license defined by FCC. for home use and short range. The transmitter for this system will broadcast a frequency modulated (FM) signal on several separate channels in the 900 MHz band interval. However, each channel will transmit the same audio information, each succeeding channel will have its audio signal delayed by a preset time period, for example 50 ms, in relation to the previous channel. The receiver of the hearing aids, which supports the positional location signals, and the associated hardware will select the appropriate channel based on the distance of the listener from the main speakers. These channels are set so that when they are in a large area, and the appropriate channel is chosen, the sound received electronically on the wireless channel will be approximately in phase with the sound reaching the listener from the main speakers. The location of the listener is determined, and the appropriate transmission channel is automatically selected in a novel way, so that the dedicated pulse transmitters are strategically located in the area. Each individual hearing aid and associated receiver will calculate its approximate position based on the signals provided by these dedicated pulse transmitters, and will tune to one of the channels that broadcast the Fm signal in the 900 MHz band.

Therefore, this system provides a method and apparatus for accurately receiving a broadcast signal which provides studio-quality sound, and synchronizes this signal with the sound coming from the main system. speakers. The system of the invention is easy to use, does not require manual operation by the user and allows each The individual moves with respect to the main speaker system without suffering from feedback, distortion or out-of-sync sound reproduction. Other advantages and benefits of the present invention will become apparent to those familiar with the art in view of the following drawings, and the detailed description that follows. -15 BRIEF DESCRIPTION OF THE DRAWINGS Figure 1 is a schematic representation of the area served by the audio system of this invention. Figure 2 is a schematic representation of the receiver and transducer unit 20 of this invention. Figure 3 illustrates an example of the set of circuits for channel division and transmission via the transmitter in the hearing aids. Figure 4 illustrates the channel selection circuitry of this invention. -0 - 10 - DESCRIPTION p? TAT.T.? R) A. OF THE PREFERRED MODALITY With reference to Figures 1-4, an audio enhancement system for use with systems will now be described. conventional sound reproduction, with reference to several preferred modalities. It will be understood that the embodiments described herein are not intended to limit the scope of the invention, but only provide examples of the present invention as they are used in various environments. The primary sound reproduction system can be any type of system having at least one primary speaker or at least one main group of loudspeakers 15 located in one position, eg, a stage or podium 12. The speaker system produces sound in response to an electronic input signal provided by any suitable audio source, for example a microphone 18, which is processed by a main sound board or mixer board. Although the invention is primarily designed for use in live broadcast or entertainment, it should be noted that the invention is equally suitable for use in simulated broadcast or recorded broadcast, or any stage (indoor and outdoor) where improvement can be integrated. audio with a primary speaker system. The loudspeaker or main loudspeakers 15 propagate the sound produced by them through the air so that it can be heard by people located in various positions around the arena. The audio enhancement system of this invention serves to increase or improve the sound heard by individuals in transit by providing distortion-free sound, and synchronized via personal transducer devices which are located near or transported by such persons. To ensure that distortion-free sound improves instead of degrading the primary sound coming from the speakers The main system, the system of this invention is designed so that the audio enhancement system provides a synchronized signal, that is, the sound reaches the ear of the listener in synchronization with the sound coming from the main speakers. As will be appreciated by those familiar with the -15 technique, the implementation of audio enhancement in accordance with ^^ The teachings of this invention can take various configurations. However, these modalities are only exemplary. Therefore, other configurations may be constructed in accordance with the teachings of this invention.

Each one of the audio enhancement modalities basically comprises at least one transmitting subsystem and at least one remote receiving subsystem. Such subsystems will be described in detail in the following. In general, each receiving subsystem basically comprises a receiver housed compactly within the portable unit, and an associated portable transducer unit, i.e., a pair of hearing aids. Each receiving subsystem is arranged to be located in any remote location inhabited by the listener of so that it can receive electrical signals transmitted from the transmitting subsystems. Signal diffusion by the transmitting subsystems represents the signals provided by the audio source to the main speakers, and preferably comprises a signal derived from a board central mixer. The receiving unit of the subsystem receives the broadcast signals, then converts them, processes and amplifies signals to propel the associated transducer device, i.e., hearing aids, to produce a sound in sync with the sound coming from the main speakers. T15 In order to facilitate the location of a subsystem w receiver as close as possible to the listener, the electrical signal provided to the receiver is transmitted without wires. Therefore, the system makes use of wireless transmitters in the transmitting subsystem for broadcasting audio signals to a plurality of receiving subsystems and remote transducers in transit. As mentioned previously, the audio enhancement system of that invention basically comprises at least one transmitting subsystem and at least one remote receiving subsystem. In order to synchronize the sound that reaches the With the receiving subsystem with the sound coming from the main speakers, the present invention provides a means of synchronization. The synchronization means includes a pulse transmitting subsystem which locates the receiving subsystem and tunes in its receiving system to an appropriate delay channel 5 which is received by the receiving subsystem. The signal supplied through this delay channel has its & delayed audio in a predetermined time period provided to compensate for the period of time it requires that the primary sound supplied by the main speakers be spread through the air to the remote position of the receiving subsystem. The receiving subsystem of this invention is designed to detect electromagnetic information approaching a radial distance from its main source of sound. More specifically, the synchronization means of this invention supplies RF pulses to the listening area occupied by listeners in transit. These RF pulses are used to approximate the distances of each receiver subsystem from the main speaker. In the preferred embodiment, the receiving subsystem compares the arrival times of various RF pulses to approximate their distance from the main speakers. For example, two RF pulse transmitters may be located in the stadium to be served by this invention; a first RF pulse transmitter is located in the front portion of the stadium next to the main sound source, and the second RF pulse transmitter is located at the rear portion of the sand, distal from the primary sound source. The receiver compares the arrival times of these two RF pulses to approximate the distance of the stage. 5 In an alternative mode, an RF generator generates standing waves by means of the beat frequency of two # RF pulses. The beat frequency, for example, has a wavelength of approximately 4 times the approximate depth of the area. With this information, you can make a position determination by the receiver. For these synchronization means, the receiver uses the position location information to pick up one of a plurality of channels that will be broadcast at approximately 900 MHz by the transmitting subsystem. The plurality of channels is chosen so that each successive channel is delayed by a fixed amount relative to the other. For position location and channel determination of this invention, position X, Y is not needed; instead, only an approximate radial distance from the front of the system is needed main speakers. It should be noted that the human ear can only perceive the difference in the arrival time of two sounds (in the same ear) when the sounds are separated by more than 25 ms. In view of these facts, the radial position of the receiver only needs to be precise with 15-20 meters.

Many different location location methods are possible that include the following preferred method: two dedicated pulse transmitters are placed in a single area, one in the front and one in the back. The pulse transmitter forward can transmit a 900 MHz RF pulse with a width of 10 ns. These pulses will be repeated every 1 ms. He • The transmitter in the back of the area will receive the pulse from the front pulse transmitter and transmit its own pulse. 900 MHz 10 ns; 50 ns after he receives his first pulse. By Therefore, each hearing aid in the area will receive two pulses, every 1 ms. The earphones in the front of the area will receive their pulses 500-1000 ns apart, based on the size of the area, while the units in the back of the area will receive their pulses 50 ns apart. This difference in delay is noticeable - rjJ-5 electronically and can be used to find an approximate position of an individual's hearing aids. Within the hearing aid unit, the variable delay can change the voltage of the VCO in the down converter so that the appropriate channel can be chosen. It should be considered that the system of this invention does not attempt to match electromagnetic waves, but actually attempts to match the phase of sound pressures from the stage and through the hearing aids. When working with sound pressures, the ear is much more error tolerant that an electronic receiver is to phase errors in electromagnetic waves. Therefore, errors in the phase matching of the two sounds combined will not be easily perceived by the user. In fact, laboratory simulations show that if the difference in delay of these two sound signals is made coincide in the next 25 ms, then no difference is perceived between the two waveforms by a listener. fu The receiver works as follows. With reference to Figure 2, the signal is received by the antenna 102 and travels directly to a multi-purpose integrated circuit 104, for example, the Philips SA620 multiple purpose IC. Such an integrated circuit contains a low amplifier 106, noise (LNA), a down converter 108 (double balanced mixer), and a voltage controlled oscillator 110 (VCO or local oscillator, LO). The low noise amplifier 106 first amplifies the radio signal supplied by the antenna 102. Subsequently, the signal is converted downwardly to the mixer 110 using a frequency provided by the local oscillator 108. The IF 112 output of the multi-purpose IC 104 will be in the frequency range of a broadcast FM signal standard (approximately 100 MHz, and much stronger so that local stations will not interfere with the operation). Before being supplied to the data band amplification and detection unit 114, the IF signal 112 is processed by the channel selection circuitry 109 of the described below with reference to Figure 4.

Then, a data band detection and amplification will be performed by a single-chip FM receiver 114, for example, Philips TDA7021T, which "receives" the 100 MHz signal, and converts it to a multiplexed stereo signal in one second. IF 5 116 of 70 kHz. This 70 kHz signal 116 can then be passed to a stereo demultiplexer 118, i.e., a demultiplexer. ^ Philips TDA7040T stereo, and an audio amplifier 120, is "* say, a Philips TDA7050T audio amplifier, for the final output to the user and to the left and right speakers 122a, 122b.10 The final amplifier 120 will connect to a control of volume (not shown) on the outside of the hearing aid unit so that the user can adjust the audio power to a desired level All of the ICs considered by this invention can be contained in assembly packaging with a surface ? J-5 small, and require relatively little energy. * The audio enhancer system of this invention will now be described with reference to Fig. 1. The sound is first picked up by the instrument or voice microphones 18. This sound is directed to the central sound board 10 where all 20 individual sounds are processed and mixed together.Effects and equalization are carried out at this point.Afterwards, the sound is sent to amplifi power amplifiers, and from here to the speaker system 15. The mixed and equalized sound is also sent to the transmitting subsystem, i.e. the headphones 40 (at audio frequencies, electronically on signal cables). In the transmitter or transmitters 40 of hearing aids, the arriving audio signal is divided into 10 channels, and each channel is then delayed by a pre-set amount of time. Each of these delayed copies of the original signal is then modulated in its own 900 MHz carrier for transmission to the receiver 30 in the headphones. Figure 3 illustrates an example of the circuit set for channel division and transmission via the transmitters 40 of hearing aids. Separated the transmitters 40 from hearing aids are two transmitters 50 of RF pulse of hearing aids. The pulse synchronization of these two transmitters is chosen so that a receiver in the area can receive and determine an approximate radial position based on the difference in the arrival time of these pulses. The RF pulses are the lowest frequency of the hearing aids generated at 900 MHz signals so that in the IF section of the receiver, a simple low pass filter can be used to reject the audio information, and allow the pulse information to pass. Based on the time of arrival of the pulses, the channel selection circuits (see Figure 4) in the receiver adjust a control voltage of the single-chip receiver 114, for example Philips TDA7021T. This control voltage receives one of the 900 MHz RF channels that has the audio portion delayed. More specifically, the control voltage changes the chosen IF frequency within the receiver 114. With this arrangement, the chosen channel will have its audio portion delayed by approximately the same amount of time it requires for the sound to travel from the speakers of the receiver. scenario to the position of the receiver. Therefore, the electronic sound and the sound traveling through the air will be approximately in phase, and the listener will not perceive any echo or mismatch between the synchronization between the two sounds. With reference to figure 4, we will now describe the circuit selection set 109 of channels (see figure 2) . The RF pulses received by the antenna must be converted downward to an IF signal by the mixer 110. The diode 109b detects the RF pulses that have been converted downward to IF. Since IF is a low pass filter p 15 to LPF 109a, most of the modulated signals have been rejected. The frequency plan is such that the RF pulses that end at the top in the pass band of this filter 109a, while the information signal is rejected. A ramp generator 109c receives > the pulse signals from the diode 109b. Upon receiving the first pulse, the ramp generator 109c is started. At the reception of the second, the ramp is set at the current voltage. Therefore, varying the arrival times of the pulses will change the control voltage on the channel selection leg of the band detection and amplification unit 114. data, for example, Philips TDA 7021T.

An FM modulation scheme with the same modulation characteristics is preferred for this invention, among other reasons because: (1) FM single-chip integrated circuit FM receivers are currently available at a reasonable cost, 5 small; (2) over short distances (and therefore reasonable energy limits), an FM system will have a relatively high noise signal ratio jfk--, and will be close to the quality of a compact disc; and (3) when using FM analog modulation in the 900 MHz band avoids the use of space and an integrated microcontroller circuit 10 that consumes a prohibitively high amount of energy, as well as its supporting hardware. Although the description of this invention has focused on the use of ten channels, it will be understood by those skilled in the art that the number of channels can be chosen based on the size of the particular area to be served and the range of desired precision. Using ten channels, each delayed successively by 50 ms, a maximum delay of 500 ms This corresponds to a maximum coincident distance of 165 meters, a range of coverage considered adequate for most areas. If the correct channel is chosen on the receiver, the maximum delay error between the electronically transmitted sound and the sound probe traveling from the stage will be 25 ms. As mentioned before, a difference of 25 ms time is not easily perceived by the human ear.

Although the present invention has been shown and described with reference to numerous preferred embodiments, it will be understood by those familiar with the art that various changes in shape and detail can be made without departing from the spirit and scope of the present invention.

Claims (13)

1. An audio enhancement system to provide an improved audio signal from a primary source 5 to a plurality of discrete locations located within an area, the audio enhancement system is characterized J0 because it comprises: an audio source means for generating a first audio signal and for converting the first audio signal into a first electromagnetic signal; a primary signal propagation means for spreading the first audio signal; a first transmitting means for transmitting the first electromagnetic signal via a wireless means; 15 a receiving means for receiving the first electromagnetic signal and converting the first electromagnetic signal into a second audio signal; a second transmitting means for transmitting an electromagnetic localization signal, the location signal The electromagnetic comprises information related to a relative position of the receiving means with respect to the primary signal propagation means; a synchronization means for automatically delaying the first electromagnetic signal based on the signal of In the electromagnetic localization, the receiving means supplies the second audio signal by substantially synchronizing the first audio signal with the second audio signal by the synchronizing means.

2. The audio enhancing system, according to claim 1, characterized in that the first transmitting means transmits the first electromagnetic signal in a plurality of channels.

3. The audio enhancing system, according to claim 2, characterized in that at least two of the plurality of channels are deviated in time.

4. The audio enhancing system, according to claim 2, characterized in that each of the plurality of channels are deviated in time by a predetermined amount.

5. The audio enhancing system, according to claim 1, characterized in that the synchronization means comprises an electromagnetic localization means for determining a position of the receiving means based on the electromagnetic localization signal.

6. The audio enhancing system, according to claim 2, characterized in that the second transmitting means comprises at least one electromagnetic pulse transmitter that transmits the electromagnetic localization signal in the form of at least one electromagnetic pulse.

7. The audio enhancing system, according to claim 6, characterized in that the synchronizing means comprises a position determining means for determining a position of the receiving means with respect to the primary signal propagation means, the position determining means calculates the position of the receiving means based on at least one electromagnetic pulse.

8. The audio enhancing system, according to claim 7, characterized in that the synchronizing means selects a channel of the plurality of channels based on the position of the receiving means.

9. The audio enhancing system, according to claim 1, characterized in that the synchronization means automatically delays the first electromagnetic signal based on a radial distance of the receiving means from the transmitting means.

10. The audio enhancing system, according to claim 6, characterized in that the second transmitting means comprises two electromagnetic pulse transmitters, each transmitting the electromagnetic localization signal and 5 form of electromagnetic pulses at regular intervals, where a radial distance of the receiving medium is calculated from the medium j ^, transmitter based on the electromagnetic pulses.

11. The audio enhancing system, according to claim 1, characterized in that both the receiving means and the synchronizing means are placed in portable hearing aids used by the listener in transit.

12. The audio enhancing system, in accordance with ^. =. 15 claim 1, characterized in that the synchronization means automatically synchronizes the first and second audio signals.

13. The audio enhancing system, according to claim 1, characterized in that the second transmitting means comprises pulse transmitters placed in discrete positions on the stadium, and the synchronization means comprises a set of circuits for position determination and channel selection. within the receiving means, the pulse transmitters 25 transmit the electromagnetic localization signal in the form of a plurality of electromagnetic pulses and the set of position determination and channel selection circuits calculate a position of the receiving means based on the electromagnetic pulses and they select the first electromagnetic signal based on the position.