Multi-channel audio enhancement system for use in recording and playback and method for providing same

Multi-channel audio enhancement system used in recording and playback and method provided therefor

Technical field

The present invention relates generally to audio enhancement systems and methods. Specifically, the present invention relates to an apparatus and method for enhancing multiple audio signals and mixing these signals into a two-channel format for reproduction in a normal playback system.

Background technique

Audio recording and playback systems can be characterized by a large number of independent tracks or tracks used to input and/or play a set of sounds. In a basic stereo recording system, two fast channels, each connected to a microphone, can be used to record sounds detected from different microphone positions. During playback, the sound recorded by the two channels is typically reproduced through a pair of speakers, and one speaker reproduces a single channel. Providing two separate audio channels during recording allows the two channels to be processed separately to achieve the specified effect during playback. Similarly, providing multiple separate audio channels has more freedom in separating specific sounds so that these sounds can be processed separately.

Professional audio studios use a multi-channel recording system that can separate and process multiple individual sounds. However, since many common audio reproduction devices are transmitted in conventional stereo, the fast track system used to record sounds requires these sounds to be "mixed" into only two separate signals. In the world of professional audio recording, sound studios use this kind of mixing method, because the pronunciation of individual instruments and given audio products can be recorded on separate tracks, but must be used in the usual stereo system. Replay in stereo format. Professional systems can use 48 or more separate audio channels that are processed separately before being recorded to two stereo sound tracks.

In a multi-channel playback system defined here as more than two separate audio channels, each sound recorded from a single fast track can be individually processed and reproduced through a corresponding speaker or speakers. Sounds recorded at multiple locations around the listener or sounds designated to be placed at multiple locations around the listener can be truly reproduced by good speakers placed in appropriate locations. Such systems find special use in theaters and other audio-visual environments where the audio and video representations are perceived by certain audiences. These systems, including Dolby Laboratories' "Dolby Digital" system; Digital Drama System (DTS); and Sony Dynamic Digital Sound (SDDS), are all designed to initially record and then reproduce multi-channel sound to provide a surround listening experience.

In personal computers and home theater events, the recording medium is standardized so that multiple channels other than the two usual stereo channels are stored on such a recording medium. One such standard is a duo that provides 6 separate audio signals. More channel coding standard than AC-3. In this Dolby AC-3 system, two audio channels are designated for playback on the front left and right speakers, two channels are reproduced on the rear left and right speakers, and one channel is used for the front center dialogue On the loudspeaker, one channel is used for low frequency and effect signals. An audio playback system that can reproduce all 6 channels does not require these signals to be mixed into two channel formats. However, many playback systems including today's typical personal computers and future personal computers/televisions only have two channel playback capabilities (excluding the center and subwoofer channels). Accordingly, the information presented in the additional audio signal, like the different normal stereo signals that appear in AC-3 recording, must be electronically discarded or mixed into a two-channel format.

There are various techniques and methods for mixing multi-channel signals into two-channel formats. A simple mixing method can be to simply combine all the signals into a two-channel format, and at the same time only calibrate the relative gain of the mixed signal. Other techniques are to apply frequency shaping, amplitude calibration, time delay or phase shift in the final mixing process or to combine some of these into a single audio signal. The particular technique or techniques used depends on the format and content of the individual audio signals and the designated use of the initial two-channel mix.

For example, US Patent No. 4,393,270 issued to van den Berg discloses a method of processing electrical signals, which is achieved by modulating each individual signal corresponding to a preselected direction that can compensate for the feeling of placing the speaker. A separate multi-channel processing system is disclosed in U.S. Patent No. 5,438,623 issued to Begault. In this patent, the individual audio signals are divided into two signals, which are delayed and filtered according to the head-related transfer function (HRTF) of the left and right ears. The resulting signals are then combined to produce left and right output signals designated for playback through a pair of headphones.

The technologies appearing in the prior art, including those appearing in professional recording locations, do not provide an effective method for mixing multi-channel signals into a two-channel format to achieve true audio reproduction through a limited number of individual channels. As a result, much of the environmental information that provides the listener with the full surround sound of the immersed sound or the immersed feeling of the sound is lost or masked in the final mixed recording. Although there are a number of previous methods for processing multi-channel audio signals to achieve real feelings through normal two-channel playback, there is more room for improvement to achieve the purpose of real listening feelings.

Summary of the invention

Accordingly, the object of the present invention is to provide an improved method of mixing multi-channel audio signals, which can use all aspects of recording and playback to provide an improved true listening experience. The object of the present invention is to provide an improved system and method for specifying the professional audio recording of the main disc to be reproduced in the usual stereo system. The object of the present invention is to provide a system and method for processing to extract a multi-channel audio signal from an audio video recording and provide no listening experience when reproduced through a limited number of audio channels.

For example, personal computers and video players are seeing the ability to record and reproduce digital video discs (DVD) with 6 or more separate audio channels. However, many such computers and video players do not have more than two audio channels. Replay channels (and possibly a subwoofer channel), they cannot use all the individual audio channels designated as the surround environment. In this way, this requires computers and other transmission systems in the prior art to be able to effectively use all the audio information available in such systems and to provide a two-channel listening experience comparable to a multi-channel playback system. The present invention satisfies such a requirement.

The audio enhancement system and method are disclosed here. It is used to process a group of audio signals, the sound represented by the sound is present in a 360-degree sounding place, and the group of audio signals are combined to generate a pair of signals. When the pair of signals pass through a pair of speakers When playing, they can accurately represent the 360-degree sounding place. The audio enhancement system can be used as a professional recording system, or in personal computers and other home audio systems that include a limited amount of audio reproduction channels.

In a preferred embodiment used in a home audio reproduction system with stereo playback capability, multi-channel recording provides a plurality of individual audio signals, which at least consist of a pair of left and right signals, a pair of surround signals, and a center channel signal . This home audio system is equipped with speakers that reproduce two-channel signals from the front sounding location. The left and right signals and surround signals are first processed and then mixed together to provide a pair of output signals that are reproduced through speakers. In particular, the left and right signals from the recording are intensively processed to provide a pair of spatially corrected left and right signals to enhance the sound from the front sounding location felt by the listener.

The surround signal is processed intensively by first separating the environment and monophonic components of the surround signal. The ambient and monophonic components of the surround signal are modified to achieve the desired spatial effect and to individually correct the position of the playback field sounder. When the surround signal is played through forward speakers as part of the synthesized output signal, the surround sound experienced by the listener is emitted across the entire rear sounding location. Finally, the center signal can be processed and mixed with the left, right and surround signals, or it can be sent directly to the center channel speaker of the home reproduction system when a speaker is present.

According to one aspect of the invention, the system has at least 4 separate audio signals, which include the main left and right signals with audio information to be reproduced from the previous sounding location, and the surround left signal with audio information to be reproduced from the rear sounding location. And right signal. The system generates a pair of left and right output signals reproduced from the previous sounding location to produce a three-dimensional audio and image experience without the need for real speakers to be placed in the rear sounding location.

The system includes a first electronic audio enhancer that receives the main left and right signals. The first audio enhancer processes the environmental components of the main left and right signals and when reproducing the left and right output signals using a pair of speakers placed in the front sounding location to produce a widened sound image sensation that crosses the front sounding location.

The second electronic audio enhancer receives the surround left and right signals, and the second audio enhancer processes the environmental components of the surround left and right signals and generates the left and right output signals when using a pair of speakers placed in the front sounding place to reproduce the left and right output signals. The sound and image perception of the sound at the place where the sound is made later.

The third electronic audio enhancer receives the surround left and right signals. The third audio enhancer processes the monophonic components of the surround left and right signals and reproduces the left and right output signals when a pair of speakers placed in the front sounding site reproduces the left and right output signals. The central position of the sounding place produces the audiovisual feeling of the sound.

By combining the processed ambient components from the main left and right signals, the processed ambient components from the surround left and right signals, and the processed mono components from the surround left and right signals, from at least four separate audio A signal mixer that generates left and right output signals in the signal, wherein the mutual output phase relationship of the environmental components of the main and surround signals is included in the left and right output signals.

In another embodiment, the at least four individual audio signals include a center channel signal having audio information reproduced by a center speaker designated for a front sounding location, and the center channel signal is combined by a signal mixer as the left and right output signals. Part. In another embodiment, the at least four individual audio signals include a center channel signal with audio information designated to be reproduced by a center speaker located in the front sounding site, and the center channel signal is made to be the same as the main left and right through a signal mixer. The single tone components of the signal are combined to produce left and right output signals.

In another embodiment, the at least four separate audio signals include a center channel signal having audio information of a center location acoustically reproduced by an excellent center channel speaker. In another embodiment, the first, second, and third electronic audio enhancers apply HRTF-based transformation functions to their respective individual audio signals and when the left and right output signals are audibly reproduced to generate corresponding individual audio signals The obvious audiovisual.

In another embodiment, the first audio enhancer adjusts the environmental components of the main left and right signals by boosting environmental components below approximately 1 KHz and above approximately 2 KHz relative to frequencies between approximately 1 KHz and 2 KHz. In another embodiment, the gain added to the boosted environmental component is approximately 8 dB relative to the gain added to the environmental component of approximately 1 and 2Khz.

In another embodiment, the second and third audio enhancers adjust the environment surrounding the left and right signals by boosting the environment and single components that are approximately below 1KHz and approximately above 2KHz relative to the frequency between approximately 1 and 2KHz. And monophonic components. In yet another example, the gain added to boost the ambient and tone components of the surround left and right signals is approximately 18 dB relative to the gain added to the ambient and tone components between approximately 1 and 2 KHz.

In another embodiment, the first, second, and third electronic audio enhancers are formed on a semiconductor substrate. In yet another embodiment, the first, second, and third electronic audio enhancers are implemented by software.

According to another aspect of the present invention, a multi-channel recording and reproducing apparatus receives a large number of individual audio signals and processes a large number of audio signals to provide first and second enhanced audio output signals, so as to complete the absence of the output signal when reproducing the output signal. Sound feel. The multi-channel recording device has a large number of parallel audio signal processing devices, which are used to modify the signal content of individual audio signals. Each parallel audio signal processing device has: a circuit for receiving two separate audio signals and converting two audio signals The environmental components of the two audio signals are separated from the monophonic components, and a position processing device is used to electronically apply a head-related transformation function to each of the environmental and monophonic components of the two audio signals to generate processed Environment and monophonic components. The head-related transformation function corresponds to the desired spatial position relative to the listener.

The multi-channel circuit mixer combines the processed monophonic components and environmental components generated by a large number of position processing devices to produce an enhanced audio output signal. The processed environmental components are then combined according to the output phase relationship of the first and second output signals.

In another embodiment, each of the large number of position processing devices further includes a circuit capable of individually modifying two audio signals, and wherein the multi-channel mixer further combines the two modified signals from the large number of position processing devices and The respective environment and monophonic components are combined to produce an audio output signal. In another embodiment, a circuit capable of individually modifying the two audio signals electronically provides a head-related transformation function to the two audio signals.

In another embodiment, a circuit capable of individually modifying the two audio signals electronically applies a time delay to one of the two audio signals. In still another embodiment, the two audio signals include audio information corresponding to the front left position and the front right position related to the listener. In another embodiment, the two audio signals include audio information corresponding to the rear left position and the rear right position related to the listener.

In another embodiment, a large number of parallel processing devices have first and second processing devices, and the first processing device applies a head-related transformation function to the first pair of audio signals so that the first pair of audio signals is completed when the output signal is generated. The first perception direction of the signal. The second processing device applies the head-related transformation function to the second pair of audio signals, so that when the output signal is generated, the second perception direction of the second pair of audio signals is completed.

In another embodiment, a large number of parallel audio processing devices and multi-channel circuit mixers are implemented in a digital signal processing device in a multi-channel recording and playback device.

According to another aspect of the present invention, an audio enhancement system processes a large number of audio source signals to generate a pair of stereo output signals, so that a three-dimensional sound field is generated when the pair of stereo output signals are reproduced by a pair of speakers. The audio enhancement system includes a first processing circuit in communication with a first pair of audio source signals. The first processing circuit is configured to separate the first environmental component and the first monophonic component from the first pair of audio signals. The processing circuit is further configured to modify the first environmental component and the first monophonic component so that the first audio image is generated so that the first audio image felt by the listener is emitted from the first position.

A second audio processing circuit that communicates with the second pair of audio source signals. The second processing circuit is so configured to separate the second environmental component and the second monophonic component from the second pair of audio signals. The second processing circuit is further configured to modify the second environment and the second tone so that the second sound image is generated, so that the second sound image felt by the listener is emitted from the second position.

A hybrid circuit that communicates with the first processing circuit and the second processing circuit. The mixing circuit is so configured to combine the first and second modified monophonic components in phase and the first and second modified environmental components out of phase to produce a pair of stereo output signals.

In another embodiment, the first processing circuit is further configured to modify a large number of frequency components in the first environmental component with the first transformation function. In another implementation, the first transformation function is further configured to emphasize the portion of the low-frequency component in the first environmental component relative to other frequency components in the first environmental component. In still another embodiment, the first transformation function is constructed so as to emphasize a portion of the high-frequency component of the first environmental component with respect to other frequency components in the first environmental component.

In another embodiment, the second processing circuit is so configured to modify a large number of frequency components within the second environmental component with the second transformation function. In yet another embodiment, the second transformation function is so constituted to modify the frequency component in the second environmental component in a different manner from the first transformation function modifying the frequency component in the first environmental component.

In another embodiment, the second transformation function is constructed so as not to emphasize the part of frequency components above 11.5KHz relative to other frequency components in the second environmental component. In another embodiment, The second transformation function is constructed so as not to emphasize the part of the frequency component between approximately 125 Hz and approximately 2.5 KHz relative to other frequency components in the second environmental component. In still other embodiments, the second transformation function is constructed so as to increase the portion of the frequency component between approximately 2.5Khz and approximately 11.5Khz relative to other frequency components within the second environmental component.

According to another part of the present invention, the multi-track audio processor receives a large number of individual audio signals as part of the synthesized audio source. A large number of audio signals are composed of at least two different audio signal pairs, and the different audio signal pairs include audio information that is interpreted by the listener as being emitted from different positions in the sound listening environment.

The multi-track audio processor includes a first electrical device that receives a first pair of audio signals. The first electrical device respectively applies a head-related transformation function to the environmental components of the first pair of audio signals to generate a first audio image, wherein the first audio image is perceived by the listener as being emitted from a first position.

A second electrical device that receives the second pair of audio signals. The second electrical device respectively applies a head-related transformation function to the environmental component and the monophonic component of the second pair of audio signals to generate a second audio image, wherein the second audio image is perceived by the listener as being emitted from the second position.

A device that mixes the components of the first and second pair of audio signals received from the first and second electrical devices. The mixing device combines environmental components out of phase to produce a pair of stereo output signals.

According to another aspect of the present invention, the entertainment system has two main audio reproduction channels to reproduce audio-video recordings to users. The audio-video record includes five separate audio signals, which are front left signal FL, front right signal FR, rear left signal RL, rear right signal RR, and center signal C. Among them, the entertainment system is from two main audio channels Allow users to get the feeling of surround sound. The entertainment system includes an audio-video playback device to extract five individual audio signals from the audio-video recording.

The audio processing device receives five independent audio signals and generates two main audio reproduction channels. The audio processing device includes a first processor for adjusting the environmental components of the front signals FL and FR to obtain a spatially corrected environmental component (FL-FR) P. The second processor adjusts the environmental components of the post signals RL and RR to obtain a spatially corrected environmental component (RL-RR) P. The third processor adjusts the directional field components of the signals RL and RR to obtain a spatially corrected directional field component (RL+RR)P.

The left output signal is generated in the mixer, and the left mixer combines the spatially corrected environmental component (FL-Fr) P, and the spatially corrected environmental component (RL-RR) P, and the spatially corrected directional field component (RL+RR) P To generate the left output signal.

The right mixer produces the right output signal. The right mixer combines the opposite spatially corrected environmental component (FR-FL) P, and the opposite spatially corrected environmental component (RR-RL) P, and the spatially corrected directional field component (RL +RR)P to generate the right output signal.

The device combines audio-video recording and playback to reproduce the left and right output signals through two main channels to create a sense of surround sound for the user.

In another embodiment, the center signal is input by the left mixer and combined into a part of the left output signal and the center signal is input by the right mixer and combined into a part of the right output signal. In another embodiment, the directional field components of the center signal and the front signal FL+FR are combined by the left and right mixers to become part of the left and right output signals, respectively. In another embodiment, the center signal is provided as the third output signal to be reproduced through the center channel speaker of the entertainment system.

In another embodiment, the entertainment system is a personal computer and the audio-video playback device is a digital versatile disc (DVD) player. In another embodiment, the entertainment system is a television and the audio-video playback device is an associated digital versatile disc (DVD) player connected to the television system.

In another embodiment, the first, second, and third processors emphasize low and high frequency regions relative to the middle frequency region. In another embodiment, the audio processing device is implemented as an analog circuit formed on a semiconductor substrate. In another embodiment, the audio processing device is implemented in a software format, and the entertainment system's microprocessor executes the software format.

According to another aspect of the present invention, in order to simulate a surround sound environment, a method enhances a set of audio source signals, where the audio source signals are distributed to a set of speakers placed around the listener to generate left and right output signals for passing A pair of speakers reproduces sound. The audio source signal includes a left front signal (LF), a right front signal (RF), a left rear signal (LR) and a right rear signal (RR).

The method includes actions of modifying an audio source signal and generating a processed audio signal based on the audio content of the selected source signal pair. The processed audio signal is defined in accordance with the following equation: P1=F1(Lf-RF), P2=F2(LR-RR), and P3=F3(LR+RR) where F1, F2 and F3 are emphasized The transform function of the spatial content of the audio signal, and complete the listener's sense of depth when the processed audio signal generated by the speaker is reproduced.

The method further includes the act of combining the processed audio signal and the audio source signal to generate left and right output signals. The left and right output signals include the components cited from the following equation:

Lout=K1LF+K2LR+K3P1+K4P2+K5P3, Rout=K6RF+K7RR-K8P1-K9P2+K10P3, where K1-K10 are independent variables that determine the gain of each audio signal.

In another embodiment, the transformation functions F1, F2, and F3 are characterized by amplifying frequencies between approximately 50 and 500 Hz and approximately 4 and 15 KHz relative to frequencies between approximately 500 Hz and 4 KHz. Level. In yet another embodiment, the left and right output signals further include a center channel audio source signal. In another embodiment, the method is implemented by a digital signal processing device.

According to another aspect of the present invention, a method generates an analog surround sound sensation by reproducing first and second output signals in an entertainment system having at least four audio signal sources. The at least four audio signals include a pair of front audio signals indicating that the opposite listener is audio information emitted from a previous sounding place, and a pair of rear audio signals indicating that the opposite listener is audio information emitted from a rear sounding place.

The method includes the function of combining the front audio signal to generate the front environment component signal and the front direction component signal. The method further includes the function of combining the rear audio signal to generate a rear environment component signal and a rear direction component signal. The method further includes the effect of processing the front environment component signal with an HRTF-based transformation function to generate a receptive source in the direction of the front environment component around the front left and right sides of the listener.

The method further includes the effect of processing the post-environment component signal with a second HRTF-based transformation function to generate a sensory source in the direction of the post-environment component around the rear left and right sides of the listener. The method further includes processing the rear direction component signal with a third HRTF-based transformation function to generate a source of the direction of the rear direction component in the rear center of the listener.

The method further includes combining the first one of the front audio signal, the first one of the rear audio signal, the processed front environment component, the processed rear environment component, and the processed rear direction component to produce the first output signal. . The method forward includes combining the second of the front audio signal, the second of the rear audio signal, the processed front environmental component, the processed rear environmental component, and the processed backward component to produce the second output signal. effect. The method further includes the function of respectively reproducing the first and second output signals through a pair of speakers located at a front sounding location relative to the listener.

In another embodiment, the first, second, and third HRTF-based transformation functions are approximately between 50 and 500Hz and approximately between 4 and 15Khz by magnifying the frequency between approximately 500Hz and 4KHz. The frequency of the signal to adjust the respective input.

In another embodiment, the entertainment system is a personal computer and at least four audio source signals are generated by a digital video disc player attached to the computer system. In another embodiment, the entertainment system is a television and an associated digital video disc player connected to the television system generates at least four audio source signals.

In another embodiment, the at least four audio source signals include a center channel audio signal, and the channel signal is electronically added to the first and second output signals. In another embodiment, a digital signal processor is used to process the first, second, and third HRTF-based transformation functions.

According to another aspect of the present invention, an audio enhancement device used by an audio signal decoder provides a plurality of audio signals that are designated to be reproduced by a set of speakers located in a surround sound listening environment. The audio enhancement device generates a pair of output signals for playback by a pair of speakers from a plurality of audio signals.

The audio enhancement device includes an enhancement device for grouping a large number of multiple audio signals from the signal decoder to separate the audio signals. The enhancement device modifies each of the individual audio signal pairs to generate individual component signal pairs. A circuit combines the component signals to produce an enhanced audio output signal, each enhanced audio output signal including a first component signal from the first pair of component signals and a second component signal from the second pair of component signals.

According to another aspect of the present invention, an audio enhancement device used in an audio signal decoder provides a plurality of audio signals designated to be reproduced by a set of speakers located in a surround sound listening environment. The audio enhancement device generates an output signal pair from a plurality of audio signals for reproduction by a pair of speakers.

The audio enhancement device includes a device for grouping at least some of a plurality of audio signals of the signal decoder into individual audio signal pairs. The grouping device further includes a device for modifying each individual audio signal pair to generate a separate component signal pair.

The audio enhancement device further includes a device for combining the component signals to produce an enhanced audio input signal. Each enhanced audio output signal includes a first component signal from the first pair of component signals and a second component signal from the second pair of component signals.

The following description in conjunction with the following drawings will make the above and other aspects, features and advantages of the present invention more obvious.

Description of the drawings

Fig. 1 is a schematic block diagram of a first embodiment of a multi-channel audio enhancement system for generating a pair of enhanced output signals to produce a surround sound effect.

Fig. 2 is a schematic block diagram of a second embodiment of a multi-channel audio enhancement system for generating a pair of enhanced output signals to produce a surround sound effect.

Fig. 3 is a schematic block diagram describing the audio enhancement processing of a pair of selected audio signals.

Figure 4 is a schematic block diagram of an enhancement circuit for processing selected components from a pair of audio signals.

Fig. 5 is a perspective view of a personal computer having an audio enhancement system constructed in accordance with the present invention that generates surround sound effects from two output signals.

Fig. 6 is a schematic block diagram of the personal computer of Fig. 5 describing its main internal components.

FIG. 7 depicts the perceived and real sound source heard by the listener during the operation of the personal computer shown in FIG. 5.

Fig. 8 is a schematic block diagram of a preferred embodiment for processing to mix a set of AC-3 audio signals to complete surround sound experience from a pair of output signals.

Figure 9 is a graphical representation of a first signal adjustment curve used in a preferred embodiment of processing and mixing a set of AC-3 audio signals to complete surround sound experience from a pair of output signals.

Figure 10 is a graphical representation of a second signal adjustment curve used in a preferred embodiment of processing and mixing a set of AC-3 audio signals to complete surround sound experience from a pair of output signals.

FIG. 11 is a schematic block diagram depicting various filter and amplifier stages used to generate the first signal adjustment curve of FIG. 9.

The schematic diagram of FIG. 12 depicts various filter and amplifier poles used to generate the second signal adjustment curve of FIG.

detailed description

Figure 1 depicts a block diagram of a first preferred embodiment of a multi-channel audio enhancement system 10 that processes a set of audio signals and provides a pair of output signals. The audio enhancement system 10 includes a multi-channel audio signal source 16 that outputs a set of independent audio signals 10 to a multi-channel signal mixer 20. The mixer 20 provides a set of processed multi-channel outputs 22 to the audio without a processor 24 that can be directed to the recording device 30 or power amplifier 32 before being reproduced by a pair of speakers 34 and 36. The left channel signal 26 and the processed right channel signal 28. Depending on the signal input 18 received by the processor 20, the signal mixer can also generate a bass signal 40 including low frequency information corresponding to the bass signal B from the signal source 16 and/or the corresponding center output from the signal source 16 The signal C includes a central audio signal 42 of dialogue or central positioning sound. That is, not all signal sources nor center channel C will provide a separate bass effect channel B. Therefore, it should be understood that these channels shown are optional signal channels. After being amplified by amplifier 32, signals 40 and 42 are represented as output signals 44 and 46, respectively.

In operation, the audio enhancement system 10 of FIG. 1 receives audio information from an audio source 16. Audio information can be in the form of separate analog or digital channels or as a digital data bit stream. For example, the audio source 16 may be a signal generated by a set of microphones attached to various musical instruments during a symphony orchestra or other audio performance. Alternatively, the audio source 16 may be a pre-recorded multi-track reproduction of an audio production. In many cases, the specific form of audio data received from the source 16 is not particularly relevant to the operation of the enhancement system 10.

For the purpose of illustration, the source audio signal described in FIG. 1 includes eight main channels A0-A7, a single bass or low frequency channel B, and a single center channel signal C. Those of ordinary skill in the prior art are welcome that the concept of the present invention can also be applied to multi-channel systems with more or less independent audio channels.

As explained in more detail in conjunction with FIGS. 3 and 4, the multi-channel no processor 24 modifies the output signal 22 received from the mixer 20 and produces a three-dimensional effect when a pair of output signals Lout and Rout are acoustically reproduced. The processor 24 shown in FIG. 1 serves as an analog processor that operates the multi-channel mixed output signal 22 in real time. If the processor is an analog device, and if the audio source 16 provides digital data output, the processor 24 must of course include a digital-to-analog converter (not shown) before processing the signal 22.

Referring now to FIG. 2, a second preferred embodiment of a multi-channel audio enhancement system is shown that provides digital immersion processing of audio sources. The illustrated audio enhancement system 50 includes a digital audio source 52 that transmits audio information along a path 54 to a multi-channel digital audio decoder 56. The decoder 56 transmits multi-audio channel signals along a path 58. Furthermore, the decoder 56 can generate optional bass and center signals B and C. The digital data signals 58, B and C are passed to the audio processor 60, which operates digitally to enhance the received signal. The processor 60 generates a pair of enhanced digital signals 62 and 64, which are fed to a digital-to-analog converter 66. Furthermore, the signals B and C are fed to the converter 66. The generated enhanced analog signals 68 and 70 corresponding to the low frequency and center information are fed to the power amplifier 32. Similarly, the enhanced analog left and right signals 72, 74 are sent to the amplifier 32. The left and right enhanced signals 72 and 74 are transmitted to the recording device 30 for direct storage of the processed signals 72 and 74 on a recording medium such as a magnetic tape or an optical disc. Once stored on the recording medium, the processed audio signals corresponding to the signals 72 and 74 can be reproduced by a normal stereo system without further enhancement processing to achieve the specified immersion effect described herein.

The amplifier 32 transmits the amplified left output signal 80, Lout to the left speaker 34 and transmits the amplified right output signal 82, Rout to the right speaker 36. Similarly, the amplified bass effect signal 84, Bout is transmitted to the optional subwoofer 86. The amplified center signal 88, Cout can be transmitted to an optional center speaker (not shown). For the near-field reproduction of signals 80 and 82, that is, a position where a listener is near and between speakers 34 and 36, it is not necessary to use a center speaker to achieve proper positioning of the center audio image. However, in the case of a far-field application where the position of the listener is relatively far from the two speakers 34 and 36, the center speaker can be used to fix the center audio image between the two speakers 34 and 36.

The combination of the decoder 56 and the processor 60 is indicated by the dash 90, and the combination can be implemented in many different ways, depending on the specific application, design constraints, or just personal preference. For example, the processing performed in the area 90 can be performed in a digital signal processor (DSP), or by software loaded into the memory of a computer, or as a microprocessor signal such as that of Intels Pentium. Part of processing power.

Referring now to FIG. 3, it is shown that the processor 24 of FIG. 1 is not combined with the signal mixer. The processor 24 includes individual enhancement modules 100, 102, and 104, each of which receives a pair of audio signals from the mixer 20. The enhancement modules 100, 102, and 104 partially process a pair of signals of the corresponding stereo level. This is achieved by separating the environmental and monophonic components from each pair of signals to realize that these components and the original signal are modified to The generated signals 108, 110, and 112 are generated. The bass that has undergone separate processing, the center and other signals are transmitted along the path 118 to the module 116, which can provide level adjustment of the received signal 118, simple filtering or other modifications. The generated signal 120 from the module 116 is output to the mixer 124 in the processor 24 together with the signals 108, 110, and 112.

Figure 4 depicts a schematic internal structure of a preferred embodiment of the module 100. The module 100 consists of inputs 130 and 132 that receive a pair of audio signals. The audio signal is sent to a circuit or other processing device to separate the environmental component from the pointing field, or single tone, and sound component appearing in the input signal. In a preferred embodiment, the circuit generates a directional sound component representing the sum signal M1+M2 in the signal path 136, and the difference signal M1-M2 including the environmental component of the input signal is transmitted along the path 138. The sum signal M1+M2 is modified by a circuit 140 having a transformation function F1. Similarly, the difference signals M1-M2 are modified by a circuit 142 having a transformation function F2. The transformation functions F1 and F2 may be the same and in a preferred embodiment provide spatial enhancement to the input signal by emphasizing specific frequencies and simultaneously attenuating other frequencies. In order to realize the perceived signal position during playback, the transformation functions F1 and F2 can realize HRTF-based processing to the input signal. If desired, the circuits 140 and 142 can be used to insert the time delay and phase shift of the input signals 136 and 138 relative to the initial signals M1 and M2.

Circuits 140 and 142 output modified sum and difference signals (M1+M2)P and (M1-M2)P along paths 144 and 146, respectively. The initial input signals M1 and M2 and the processed signals (M1+M2)P and (M1-M2)P. It is fed to a multiplier that adjusts the gain of the received signal. After processing, the modified signal leaves the enhancement module 100 at outputs 150, 152, 154, and 156, the output 150 transmits the signal K1M1, the output 152 transmits the signal K2F1 (M1+M2), and the output 154 transmits the signal K3F2 (M1+M2) and The output 156 conveys the signal K4M2, where K1-K4 are constants determined by the setting multiplier 148. The types of processing performed by the modules 100, 102, 104, and 116 and in particular the circuits 134, 140, and 142 may be user-adjustable to achieve desired effects and/or desired positions for reproducing sounds. In some cases, it may be desirable to process only the environmental component or single tone component of a pair of input signals. The processing performed by each module may be different, or it may be the same as one or more modules.

In a preferred embodiment, a pair of audio signals are jointly enhanced before mixing, and each module 100, 102, and 104 will generate four processed signals for the mixer 24 shown in FIG. 3 to receive. All the signals 108, 110, 112, and 120 can be selectively combined by the mixer 124 in accordance with principles known to those skilled in the art and depending on the user's preference.

By processing multi-channel signals in stereo level, that is, in pairs, the subtle differences and similar performances in a pair of signals are adjusted to achieve no effect when reproduced through the speakers. The immersion effect can be localized by applying an HRTF-based transformation function to the processed signal to generate a sound field that is fully immersed and localized. Each pair of audio signals is individually processed to produce a multi-channel audio mixing system, which can effectively reproduce the feeling of a real 360-degree sound field. Through the separate HRTF processing of the components of the audio signal, namely the environment and the monophonic components, more conditional control is provided, and when the processed signal is reproduced by sound, more real and no sound experience is produced. in the article by EAB.shaw entitle "Transformation of Sound Pressure Level From the Free Field to the Eardum in the HorigontalPlane", J.Acouet Soc.Am., Vol.56, No, Decemler 1974, and in the article by S.Mchrgardt and V. Mellert entitle "Transpormation characteristic of External HumanEar", J. Acoust. Soc. Am, Vol 61, NO. 6, Jwne 1977, describes an example of an HRTF transformation function that can be used to complete a specific sensory orientation. These two articles are here for reference.

Although the principles of the present invention described above in conjunction with FIGS. 1-4 are applicable to professional studios for high-quality recording, a specific application of the present invention is in audio playback devices, which are capable of processing but not reproducing much. Channel audio signal. For example, today's audio-video recording media are encoded with multi-audio channel signals for reproduction in a home theater surround processing system. Such a surround system typically includes forward or front speakers that reproduce left and right stereo signals, a rear speaker that reproduces left and right surround signals, a center speaker that reproduces center signals, and a subwoofer that reproduces low frequency signals. The recording medium played back by such a surround system can use multi-channel audio signals to be encoded by the Dolby AC-3 audio coding standard. Many of today's playback devices are not equipped with surround or center channel speakers. As a result, the full capabilities of the multi-channel recording medium are not fully utilized, leaving users with a poor listening experience.

Referring to FIG. 5, a personal computer system 200 is shown having a positionless audio processor constructed in accordance with the present invention. The computer system consists of a processing unit 202 connected to a display monitor 204. The front left speaker 206 and the front right speaker 208 and the optional subwoofer 210 are all connected to the unit 202 to reproduce the audio signal generated by the unit 202, and the audience 212 operates the computer system 200 through the keyboard 214. The computer system 200 processes the multi-channel audio signal to provide the listener 212 from the cha-cha speakers 206, 208 and, if any, the speaker 210 without a 360-degree surround sound experience. In a preferred embodiment, the processing system disclosed herein will be described using Dolby AC-3 recording media. It is welcome, however, to apply the same or similar principles to other standardized audio recording techniques that use multiple channels to create a surround sound experience. Furthermore, when the computer system 200 is shown and described in FIG. 5, the audio-video playback device that reproduces the AC-3 recording medium may be a TV, a TV/personal computer component, a digital video disc player connected to the TV, or Any other device capable of playing multi-channel audio recordings.

FIG. 6 is a schematic block diagram of the main internal components of the processing unit 202 in FIG. 5. The unit 202 includes the components of a typical computer system constructed in accordance with principles known to ordinary technicians, namely, a central processing unit (CPU) 220, a large-capacity memory and a temporary random access memory (RAM) system 222, and an input/output control device 224 , All components are connected to each other through the internal bus structure. The unit 202 also includes a power supply 226 and a recording media player/recorder 228, which may be DVD devices or other multi-channel audio sources. The DVD player 228 provides video data to the video decoder for display on the monitor. The audio data from the DVD player 228 is transferred to the audio decoder 232, which provides the multi-channel digital audio data from the player 228 to the processor 250. The audio information from the decoder 232 includes the left front signal, the right signal, the left surround signal, the right surround signal, the center signal, and the low frequency signal, all of which are sent to the no audio processor 250. The processor 250 digitally enhances the audio information from the decoder 232 in a manner suitable for playback in a normal stereo playback system. In particular, the left channel signal 252 and the right channel signal 254 are provided as the output of the processor 250. The low frequency sub-bass signal 256 is also provided to convey the bass response to the stereo playback system. The signals 252, 254, and 256 are first provided to the digital-to-analog converter 258, then to the amplifier 260, and then output to connect the corresponding speakers.

Referring now to FIG. 7, a schematic representation of the position of the speakers of the system of FIG. 5 is shown three-dimensionally from above. The listener 212 is located between and in front of the left front speaker 206 and the right front speaker 208. By processing the surround signal generated from the AC-3 compatible recording in accordance with the preferred embodiment, an analog surround experience is produced for the listener. In particular, the normal reproduction of two-channel signals through the speakers 206 and 208 will produce a phantom center speaker 214 of perception, from which the monophonic components of the left and right signals will be presented and emitted. In this way, when reproduced through the speakers 206 and 208, the left and right signals recorded from the AC-3 six channels will produce the center phantom speaker 214. The left and right surround channels of the AC-3 six-channel recording are processed so that the perceived ambient surround sound is emitted from the rear phantom speakers 215 and 216, while the monophonic surround sound appears to be emitted from the rear phantom center speaker 218. Furthermore, the left and right front signals, and the left and right surround signals are spatially enhanced to provide a sense of no sound to cancel the real speakers 206, 208 and the phantom speakers 215, 216, and 218 as the perceived sound source points. Preferably, the low frequency information is generated by an optional subwoofer 210, which can be placed anywhere around the listener 212.

FIG. 8 is a schematic representation of the immersion processor and mixer that complete the immersion sound effect shown in FIG. 7. The processor 250 corresponds to the processor shown in FIG. 6 and receives six audio channel signals. These signals are the front main left signal ML, the front main right signal MR, the left surround signal CL, the right surround signal SR, the center channel signal C, and Low frequency effect signal B. The signals ML and MR are fed to corresponding gain adjustment multipliers 252 and 254 controlled by the volume adjustment signal Mvolume. The gain of the center signal C is adjusted by the first multiplier 256 and the second multiplication controlled by the signal Muolume and the center adjustment signal Cvolume is controlled by 258. Similarly, the surround signals SL and SR are first fed to the respective multipliers 260 and 262 controlled by the volume adjustment signal Svolume.

The main front left and right signals ML and MR are each fed to the summing nodes 264 and 266. The summing node has a reverse input for receiving MR and a forward input for receiving ML and combined to generate ML-MR on the output path 268. The signal ML-MR is fed to an enhancement circuit 270 characterized by a transformation function P1. The processed difference signal (ML-MR) P is transmitted from the output of the circuit 270 to the gain adjustment multiplier 272, and the output of the multiplier 272 is directly fed to the left mixer 280 and the inverter 282. The inverted difference signal (MR-ML) P is transmitted from the inverter 282 to the right mixer 284. The sum signal ML+MR output at the node 266 is fed to the gain adjustment multiplier 286. The output of the multiplier 286 is fed to the summing node, which makes the center channel signal C and the signal ML+MR. The combined signal ML+MR+C leaving the node 290 is directly sent to the left mixer 280 and the right mixer 284. Finally, before being transmitted to the mixers 280 and 284, the original signals ML and MR first pass through fixed gain adjustment circuits, namely amplifiers 290 and 292, respectively.

The surrounding left and right signals SL and SR leaving the multipliers 260 and 262, respectively, are fed to the summing nodes 300 and 302, respectively. The summing node 300 has a reverse input that receives SR and a forward input that receives SL and is combined to generate SL-SR on the output path 304. All the summing nodes 264, 266, 300, and 302 are so configured as inverting amplifiers or non-inverting amplifiers, depending on whether a sum or difference signal is generated. Both the inverting and forward amplifiers are composed of ordinary operational amplifiers in accordance with the principles known to those skilled in the art. The signal SL-SR is fed to an enhancement circuit 306 characterized by a transformation function P2. The processed difference signal (SL-SR) P is transmitted from the output of the circuit 306 to the gain adjustment multiplier 308. The output of the multiplier 308 is directly fed to the left mixer 280 and the inverter 310. The inverted difference signal (SR-SL) P is transmitted from the inverter 310 to the right mixer 284. The sum signal SL+SR leaving the node 302 is fed to a separate enhancement circuit 320 characterized by the transformation function P2. The processed sum signal (SL+SR) P is transmitted from the output of the circuit 320 to the gain adjustment multiplier 332. When making reference to the sum and difference signals, it should be noted that the actual use of the sum and difference signals is only a representation. The same processing can be done regardless of how the environment and tone components of the pair of signals are separated, and the output of the multiplier 332 is directly fed to the left mixer 280 and the right mixer 284. Likewise, before being passed to the mixers 280 and 284, the original signals SL and SR are first fed through fixed gain amplifiers 330 and 334, respectively. Finally, the low-frequency effect channel B is fed through the amplifier 336 to generate the output low-frequency effect signal Bout. Optionally, if there is no subwoofer, the low frequency channel B can be mixed as part of the output signals Lout and Rout.

The enhanced circuit 250 of FIG. 8 can be in the form of an analog discrete form, in the form of a semiconductor substrate, through software running on a main or dedicated microprocessor, in a digital signal processing (DSP) chip or firmware or in some other digital format. achieve. It is also possible to use a hybrid circuit structure composed of analog and digital components, because in many cases the source signal will be digital. Accordingly, individual amplifiers, regulators or other components can be implemented by software or firmware. Furthermore, the enhancement circuit 270 and the enhancement circuits 306 and 320 of FIG. 8 may use various audio enhancement techniques. For example, the circuit devices 270, 306, and 320 may use time delay technology, phase shift technology, signal adjustment, or a combination of all these technologies to achieve the desired audio effect. The basic principles of these audio enhancement technologies are well known to those skilled in the art.

In a preferred embodiment, the submerged processor circuit 250 only reproduces two output signals Lout and Rout on the condition of a group of AC-3 multi-channel signals to provide a surround sound experience. In particular, the ML and MR signals are processed together by separating the environmental information presented in these signals. The environmental signal component represents the difference between a pair of audio signals. The environmental signal components derived from a pair of audio signals are therefore often referred to as "difference" signal components. While the circuits 270, 306, and 320 are shown and described as generating sum and difference signals, other embodiments of the audio enhancement circuits 270, 306, and 320 cannot obviously generate sum and difference signals at all. This can be done in many ways using common circuit design principles. For example, the separation difference signal information and its subsequent adjustment can be implemented digitally or instantaneously at the input stage of the amplifier circuit. In addition to processing AC-3 audio signal sources, the circuit 250 of FIG. 8 will automatically process signal sources with fewer individual audio channels. For example, if a Dolbypro logic signal is input from the processor 250, that is, SL=SR here, only the enhancement circuit 320 will operate to modify the post channel signal, because no environmental component will be generated at the node 300. Similarly, if only two channel stereo signals ML and MR are presented, then the processor 250 operates to generate a spatially enhanced listening experience from only two channels through the operation of the enhancement circuit 270.

In a preferred embodiment, the environmental information of the pre-channel signal that can be expressed as the difference signal ML-MR is adjusted by the circuit 270 in accordance with the frequency response curve 350 of FIG. 9. The curve 350 can be designated as a spatial calibration curve, or a "projection" curve. By selectively enhancing the sound information that provides a spacious feel, such adjustment of the environmental signal information broadens and mixes the sounding location of the sensation generated from a pair of audio signals.

The enhancement circuits 306 and 320 modify the surround signals SL and SR and the environment and tone components, respectively. According to a preferred embodiment, the transformation functions P2 and P3 are the same and both apply the same projection adjustment level to the corresponding input signal. In particular, the circuit 306 adjusts the environmental component of the surround signal represented by the signal SL-SR, and the circuit 320 adjusts the tone component of the surround signal represented by the signal SL+SR. The adjusted level is represented by the frequency-corresponding curve 352 of FIG. 10.

The projection adjustment curves 350 and 352 shown in FIGS. 9 and 10, respectively, are gain functions measured in decibels relative to audio displayed in logarithmic format. The gain level in decibels at each individual frequency is only related to their relative reference signal. This is because the amplification of the entire output signal occurs in the final mixing process. Referring initially to FIG. 9, according to a preferred embodiment, the projection curve 350 has a peak gain at point A at approximately 125 Hz. The gain of the projection curve 350 decays at a rate of approximately 6 dB per octave above and below 125 Hz. The projection curve 350 reaches the minimum gain at point B in the approximate 1.5-2.5KHz region. At frequencies above point B, the gain increases at a rate of approximately 6dB per octave up to point C, which is approximately 7Khz, and then continuously increases to approximately 20KHz, which is approximately the highest frequency that the human ear can hear.

Referring now to FIG. 10, and in accordance with a preferred embodiment, the projection curve 352 has a peak gain at point A located at approximately 125 Hz. The gain of the projection curve 350 decreases at a rate of approximately 6 dB per octave below 125 Hz and at a rate of approximately 6 dB per octave above 125 Hz. The projection curve 352 reaches the minimum gain at point B in the region of approximately 1.5-2.6 KKHz. The gain rises at a rate of approximately 6dB per octave at frequencies above point B, and reaches the maximum gain at point C of approximately 10.5-11.5KHz. The frequency response of curve 352 drops at frequencies above approximately 11.5 KHz.

The apparatus and method applicable to completing the adjustment curves 350 and 352 of FIGS. 9 and 10 are similar to the apparatus and method disclosed in the pending application serial number 08/430751 filed on April 24, 1995, which is hereby incorporated by reference in its entirety. . U.S. Patent Nos. 4,738,669 and 4,866,7440 issued to Amold I. Klayman disclose related audio enhancement techniques for enhancing environmental information, and these two patents are hereby incorporated by reference in their entirety.

In operation, the special function of the circuit 250 of FIG. 8 is that the five main channel signals ML, MR, C, SR, SL positioned around a listener when reproduced by only two speakers are applied to Figure 9 of the signal ML-MR is broadened by the line 350 and spatially enhanced the ambient sound from the signals ML and MR. This produces the feeling of the wide front sounding place emitted from the speakers 206 and 208 shown in FIG. 7. This is done by selectively adjusting the environmental signal information to emphasize low and high frequency components. Similarly, the adjustment curve 352 of FIG. 10 is applied to the signal SL-SR to broaden and spatially enhance the ambient sound from the signals SL and SR. Furthermore, however, the adjustment curve 352 modifies the signal SL-SR to account for the positioning of the HRTF to obtain the perception of the rear speakers 215 and 216 of FIG. 7. As a result, the curve 352 includes higher levels that emphasize the low and high frequency components of the signal SL-SR relative to the one to which the ML-MR is applied. This is needed because the normal frequency response of the human ear to a sound directed at a listener from an azimuth of zero degrees will emphasize a sound centered at approximately 2.75KHz. Emphasizing these sounds creates an inherent transfer function of the average human ear and resonance from the ear canal. The projection curve 352 of FIG. 10 cancels the inherent transformation function of the human ear to produce the feeling of the rear speaker of the signals SL-SR and SL+SR. The resulting processed difference signal (SL-SR) P is driven out of phase to the mixers 280 and 284 to maintain the feeling of a wide post-sounding location, as if reproduced by phantom speakers 215 and 216.

Through separate environmental signal processing into sum and difference components, by allowing the gain of each signal SL-SR and SL+SR to be adjusted individually, greater control is provided. The present invention also recognizes the generation of the center rear phantom speaker 218 as shown in FIG. 7 and requires similar processing to the signal SL+SR because the sound is actually emitted from the front speakers 206 and 208. Therefore, the signal SL+SR is also adjusted by the circuit 320 according to the curve 352 of FIG. 10. The generated processed signal (SL+SR) P is driven in phase to realize the phantom speaker 218 of the feeling, as if two phantom rear speakers 215 and 216 actually exist. For a dedicated center channel speaker and audio reproduction system, the circuit 250 of FIG. 8 can be modified so that the center signal C is directly fed to the center speaker instead of being mixed in the mixers 280 and 284.

The approximate relative gain values of the various signals in the circuit 250 are measured with reference to 0 decibels away from the multipliers 272 and 308. Using this reference, the gains of the amplifiers 290, 292, 330, and 334 according to the preferred embodiment of the present invention are approximately -18dB, the gain of the sum signal leaving the amplifier 332 is approximately -20dB, and the gain of the sum signal leaving the amplifier 286 is approximately At -20dB, the gain of the center channel signal leaving the amplifier 258 is approximately -7dB. These relative gain values are purely design choices according to user preferences and can be changed without departing from the spirit of the present invention. Adjusting the amplifiers 272, 286, 308, and 332 allows the processed signal to be suitable for the type of sound reproduced and to suit the user's personal preferences. Increasing the level of the sum signal emphasizes the audio signal presented at the center point between a pair of speakers. On the contrary, increasing the difference signal level emphasizes the environmental sound information that produces a wider sound image perception. In some audio arrangements where music genre parameters and system configurations are well known, or where adjustments are impractical, multipliers 272, 286, 308, and 332 can be preset and fixed at a desired level. In fact, if the desired level adjustment of the multipliers 308 and 332 is performed using the post signal input level, it is possible to directly connect the boost circuit to the input signals SL and SR. As appreciated by those of ordinary skill in the technical field, the final individual signal strength of the various signals of FIG. 8 will be affected by the volume adjustment and the mixing level applied by the mixers 280 and 284.

According to this, the audio output signals Lout and Rout produce much improved audio effects because the ambient sound is selectively emphasized to fully surround a listener at the reproduced sound location. Ignoring the relative gain of the individual components, the audio output signal Lout and Rout can be expressed by the following digital formula: Lout=ML+SL+(ML-MR)P+(SL-SR)P+(ML+MR+C)+(SL+ SR)P(1)Rout=MR+SR+(MR-ML)P+(SR-SL)P+(ML+MR+C)+(SL+SR)P(2) The above-mentioned enhanced output signal can be magnetically or Electronically stored on various recording media such as vinyl records, compact discs, digital or analog audio tapes, or computer data storage media. The enhanced audio output signal that has been stored can be reproduced by a normal stereo reproduction system to complete the enhancement of the stereo image of the same level.

Referring to FIG. 11, a schematic block diagram shows a circuit for completing the adjustment curve 350 of FIG. 9 according to a preferred embodiment. The circuit 270 inputs the environmental signal ML-MR, which corresponds to the path 268 in FIG. 8. The signal ML-MR is first limited by a high-pass filter 360 having a cut-off frequency of approximately 50 Hz or a -3dB frequency. The use of the high-pass filter 360 is designed to avoid excessive amplification of the bass components presented in the signal ML-MR.

To shape the signal ML-MR in the spectrum, the output of the filter 360 is divided into three separate signal paths 362, 364, and 366. In particular, the ML-MR is transmitted along the path 362 to the amplifier 368 and then to the summing node 378. The signal ML-MR is also passed along the path 364 to the low pass filter 370, then to the amplifier 372, and finally to the summing node 378. Finally, the signal ML-MR is transmitted along the path 366 to the high pass filter 374, then to the amplifier 376, and then to the summing node 378. Each individual limit difference signal ML-MR is combined at the summing node to generate a processed difference signal (ML-MR) P. In a preferred embodiment, the low-pass filter 370 has a cut-off frequency of approximately 200 Hz, and the high-pass filter 374 has a cut-off frequency of approximately 7 KHz. The strict cut-off frequency is not critical, as long as the environmental components of the low and high frequency regions relative to the frequency of the intermediate frequency region of approximately 1 to 3 KHz can be amplified. The filters 360, 370, and 374 are all primary filters to reduce complexity and cost, but the processing levels shown in FIGS. 9 and 10 can also be high-order filters if the processing level is not significantly changed. Likewise, in a preferred embodiment, amplifier 368 will have a gain of approximately 0.5, amplifier 372 will have a gain of approximately 1.4, and amplifier 376 will have a gain of approximately 1.

The signals leaving the amplifiers 368, 372, and 376 constitute the signal (ML-MR) O component. When the summing node 378 combines these signals. The entire spectrum shaping, namely the standardization of the environmental signal ML-MR, took place. This is the processed signal (ML-MR) P mixed by the left mixer 280 as part of the output signal Lout. Similarly, the inverted signal (MR-ML) P mixed by the right mixer 284 (shown in FIG. 8) is used as a part of the output signal Rout.

Referring now to FIG. 9, in a preferred embodiment, the gain difference between point A and point B of the projection curve 350 is ideally designed to be 9 dB, and the gain difference between points B and C should be approximately 6 dB. These numbers are design constraints and the actual numbers will more seem to depend on the true values of the components used by the circuit 270. If the gains of amplifiers 368, 372, and 376 in Figure 11 are fixed, then the projection curve 350 will remain constant, and the adjustment amplifier 368 will tend to adjust the amplitude level of point B, thus changing points A and B and point B. The gain difference between and C. In a surround sound environment, a gain difference of very greater than 9dB may tend to reduce the perception of the middle area of the listener.

The projection curve completed by the digital signal processor in many cases will more accurately reflect the above design constraints. For the simulation implementation. If the frequency and gain difference of corresponding points A, B and C change by plus or minus 20%, it is still acceptable. This deviation from the ideal specification will still produce the desired enhancement effect, albeit a bit worse than the optimal result.

Referring now to FIG. 12, a schematic block diagram shows a circuit for completing the adjustment curve 352 of FIG. 10 according to a preferred embodiment. Although the same curve 352 is used to shape the signals SL-SR and SL+SR, for the convenience of discussion, only the circuit enhancement device 306 is referred to in FIG. 12. In a preferred embodiment, the characteristics of the device 306 and the characteristics of the device 320 are the same. The circuit 306 inputs the environmental signal SL-SR, which corresponds to the location of the path 304 in FIG. 8. The signal SL-SR is first limited by a high-pass filter 380 with a cut-off frequency of 50 Hz. As shown in the circuit 270 of FIG. 11, the output of the filter 380 will be divided into three separate signal paths 382, 384, and 386. The shaping signal SL-SR. In particular, the signal SL-SR is transmitted along the path 382 to the amplifier 388 and then to the summing node 396. The signal SL-SR also travels along the path 384 to the high pass filter 390 and then to the low pass filter 392. The output of the filter 392 is sent to the amplifier 394 and finally to the summing node 396. Finally, the signal SL-SR is transmitted along the path 386 to the low-pass filter 398, then to the amplifier 400, and then to the summing node 396. Each individually limited signal SL-SR is combined at the summing node 396 to produce a processed difference signal (SL-SR)P. In a preferred embodiment, the high-pass filter 390 has a cut-off frequency of approximately 21 KHz, and the low-pass filter has a cut-off frequency of approximately 8 KHz. The filter 392 is used to generate the maximum gain point C of FIG. 10 and can be removed if desired. Furthermore, the low-pass filter has a cutoff frequency of approximately 225 Hz. As appreciated by those of ordinary skill in the art, there are multiple additional filter combinations that can complete the frequency response curve 352 shown in FIG. 10 without departing from the spirit of the present invention. For example, the strict number of filters and cutoff frequency are not critical, as long as the signal SL-SR is adjusted according to Figure 10. In the preferred embodiment, all filters 380, 390, 392, and 398 are primary filters. Likewise, according to a preferred embodiment, the amplifier 388 has an approximate gain of 0.1, the amplifier 394 has an approximate gain of 1.8, and the amplifier 400 will have an approximate gain of 0. 8. The processed signal (SL-SR) P mixed by the left mixer 280 (shown in FIG. 8) is used as a part of the output signal Lout. Similarly, the inverted signal (SR-SL) mixed by the right mixer 284 (shown in FIG. 8) is used as a part of the output signal Rout.

10 again, in a preferred embodiment, the gain difference between points A and B of the projection line 352 is ideally designed to be 18 dB, and the gain difference between points B and C should be approximately 10 dB. These numbers are design limitations and the actual numbers will very much depend on the actual values of the components used in the circuits 306 and 320. If the gains of amplifiers 388, 394, and 400 of Figure 12 are fixed, then the projection curve 352 will remain constant. The adjustment of amplifier 388 will tend to adjust the amplitude level of point B of curve 352, thus changing the gain difference between points A and B, and points B and C.

Through the foregoing and the accompanying drawings, the present invention has shown important advantages over conventional audio reproduction and enhancement systems. The above description has described and indicated the basic novel features of the invention. It should be understood that those of ordinary skill in the art can omit and substitute various omissions, substitutions and changes for the devices, details, and types shown. Depart from the spirit of the present invention. Therefore, the following claims limit the scope of the invention.