CN1452850A - Dynamic power sharing in multi-channel sound system - Google Patents

Abstract

A signal processing system for multi-channel audio systems. The signal processing system includes a first channel (14) having a first sound signal. A second channel (40) having at least a second sound signal is also included. Also comprising a processor (80) responsive to a signal level threshold (76) in the first channel such that a portion of the first channel sound signal at and above the threshold is mixed to at least a second audio frequency channel.

Description

Dynamic power sharing for multi-channel sound systems

technical field

The present invention generally relates to multi-channel sound systems. More specifically, the present invention relates to methods of power distribution in multi-channel sound systems.

Background technique

In today's home entertainment industry, high-fidelity, spatially accurate sound is important, and the surround sound system is a superior sound delivery system. Typically, a surround sound system has 5 or more channels, and at least one woofer or subwoofer channel. Surround sound systems typically use a front-facing center channel for the human voice and main sound of a program source, or flat sound with focused imaging. Additional channels are used for special effects or other sounds that have a non-centre frontal pan position, ie a sense of movement in space. Channels behind the audience or listeners are used to simulate sound from behind the audience, or to provide ambient, spatial, or enveloping sound. This type of speaker arrangement can allow the viewer or listener to hear a virtual jet or spaceship flying from their left to their right or even from behind them .

Surround sound systems also use volume cues to provide the illusion of motion. In the example of recording a jet, the sound he heard became quieter as the jet moved away from the listener. So, as the jet does, the speaker's output can be increased until it reaches its maximum volume, and then that sound is reduced as the jet goes away. Directional cues are often dominated by loudspeakers with loud output. Most program sources send a greater signal level to a particular channel at a given point in time to give the sound a perceived sense of direction or movement.

A disadvantage of such a system is that one or more channels may go into overload due to a high intensity signal or high level directional signal being generated in one channel and moving between channels. When the signal exceeds the maximum signal level threshold of the speaker or amplifier, the sound may be distorted and level limited. Traditional systems offer no solution to this problem other than increasing the size and power capacity of the system to increase the output without overloading. This can be very expensive and may require a system that is larger than it actually is in a home environment.

Brief description of the drawings

Fig. 1 is a schematic diagram showing a preferred embodiment of a dynamic power sharing circuit used in a multi-channel audio system according to the present invention;

FIG. 2 is a schematic diagram of channels 1-3 included in FIG. 1 .

Figure 2a is a schematic diagram of a channel circuit that detects other threshold parameters besides amplifier power clipping.

Figure 3 is a schematic diagram of a multi-channel system using digital power sharing steering logic.

Figure 4 illustrates the power sharing on the center channel.

Figure 5 illustrates power sharing with respect to side channels.

Figure 6 illustrates a general approach for power sharing;

Figure 7 illustrates a more specific method for power sharing.

Summary of the invention

A signal processing system for multi-channel audio systems. The signal processing system includes a first channel having a first sound signal. A second channel having at least a second sound signal is also included. Also included is a processor responsive to a signal level threshold in the first channel to cause a portion of the first channel sound signal at and above the threshold to be mixed into at least a second audio channel.

According to a more detailed aspect of the invention, the system includes a signal processing system for a multi-channel audio system. The system includes N channels, where N>1, and a sound signal corresponding to each channel. A signal level threshold is associated with each channel. a signal processor responsive to the signal level threshold such that once any channel reaches the signal threshold, the signal processor routes at least a portion of the sound signal of the channel that reaches the signal threshold to the multi-channel audio system in at least one other channel.

Another aspect of the present invention provides a method for increasing the characteristic acoustic output of a multi-channel sound system comprising a plurality of channels each having a sound signal. The first step is to select at least one signal of at least one channel of the multi-channel sound system. Another step is to select a predetermined parameter threshold corresponding to the signal level. A further step is sending a portion of the sound signal associated with one channel of the multi-channel sound system to at least one other channel of the multi-channel sound system when the signal reaches the predetermined parameter threshold.

The features and advantages of the present invention will become clearer through the following detailed description of examples and functions of the present invention with reference to the accompanying drawings. A detailed description

For a better understanding of the principles of the present invention, reference will now be made in detail to the exemplary embodiments shown in the accompanying drawings. However, it should be understood that the scope of the present invention is not limited thereto. Variations and modifications to the features of the invention shown herein, as well as other applications of the principles of the invention shown herein, will be apparent to those of ordinary skill in the art and having knowledge of the disclosure herein Yes, they all should belong to the scope of the present invention.

FIG. 1 illustrates an embodiment of a dynamic power sharing circuit in a multi-channel sound system according to the present invention. A multi-channel sound system consists of 3 or more channels, so for any one channel there are two corresponding channels on each side of the channel with direction vector and sound output.

In FIG. 1 , a channel signal 10 enters a summing amplifier 12 . If a signal overload occurs, it is accepted at the corresponding channel input 14 . The original channel signal will be summed with any overdrive signal and sent to channel 1 amplifier 16 . The original or combined signal may overload the channel at some point. Once a signal threshold is specified, such as when the first channel's amplifier is overloaded, the output of the first channel is limited, and any signal boost from that channel is routed to the two corresponding channels on each side of the first channel. This is in contrast to conventional systems where an amplifier going into overload can clip or distort the signal before it is passed to the load 18 or acoustic transducer.

A differential amplifier 20 is used in the system to receive the first input from the channel 1 output and the second input from the summing amplifier. The output of the differential amplifier is the difference signal between the signal entering the amplifier and the signal leaving the amplifier, which is the overload signal amount of the channel. The gain of the differential amplifier is preferably 1, but may not be 1. The differential amplifier has gain only when it needs to transmit an amplified signal to the corresponding channel. Gain can be used, for example, if the corresponding overflow channel is farther from the listener than the original speaker.

The signal from differential amplifier 20 is routed to at least one other corresponding channel. Figure 1 illustrates the case where the difference signal is provided to channels 2 and 3 (40 and 42). The summing amplifiers 32, 36 of channels 2 and 3 sum their channel inputs 30, 34 with the outputs from the differential amplifiers 22a, 22b. The summed outputs are sent to channels 2 and 3 (32 and 36). In this way, the system is not limited by the overload of any given channel, while maintaining substantially the same directivity of sound. Channels 2 and 3 can also divert their overloads to other channels via their own differential amplifiers 44,46. This circuit is shown as an analog circuit, but it could also be a digital signal processor (DSP), or implemented in software with the same digital functionality.

Each channel has a threshold limit beyond which signals above or near that threshold are passed on to other channels. The threshold limit may be determined based on clipping of the amplifier, amplitude limit of the transducer, frequency-dependent limits, thermal limits, and the like.

The source channel is generated to include a phase advance compared to the corresponding auxiliary channel so as to further support psychoacoustic directional cues. When the listener hears the sound of the source channel before the auxiliary channel, the sound quality of the source channel that the user hears as the source of the sound direction is further enhanced. Auxiliary channels affect volume, but the user mentally filters directionality out of those channels because they are heard a split second later. Delay circuits can be added between channels, or included as part of a differential amplifier, to provide the required phase lead.

If the second or third channel receiving the detoured signal also reaches their signal threshold, the overload can be split and routed to one or more additional channels. When the invention is applied to a five-channel system, and channel 1 is overloaded, a portion of the signal at or above the overload can be bypassed to channels 2 and 3. A further advantage is that limiting, compressing or reducing channel gain up to the overload threshold is done in such a way as to limit the channel's sonic distortion. If channels 2 or 3 are also overloaded, a portion of this signal can be bypassed to channels 4 and/or 5. Even if some directivity is lost through multiple detours, it can in fact be compensated, since rerouting only happens when the sound is loud, so some amount of directivity loss is not very significant. In general, tonal distortions are more noticeable or objectionable to the ear than directional distortions in acoustics. Therefore, it is even more important to eliminate tonal distortion, even though some directional distortion may occur. Thus, an embodiment of the present invention can substantially eliminate tonal distortion due to channel overloading while preserving precisely perceived directional cues.

An additional threshold detector could also be included, as soon as channel 1 starts to be limited, more of channel 1's signal is shared to channel 2 than to channel 3 at that point. This way, when that signal is distributed to the other two channels, more of the signal is sent to channel 2 than to channel 3. In some cases, this can maintain a more correct spatial image position, for example, if channel 1 is the right front channel, channel 2 is the center channel and channel 3 is the right surround channel. This asymmetric mix can also be beneficial if channel 2 is a more aggressive channel than channel 3, so it can accommodate more signal before it reaches overload. The source channel may also have a phase lead with respect to the support channel, or, alternatively, the other two support channels may include a time delay with respect to the main source channel, or other known psychoacoustic properties may be imposed, to Directional cues are maintained in the channels. The differential amplifier circuit may include a coefficient separator. In this way, a large proportion of the signal can be sent to the front speakers and a small proportion to the rear speakers, and vice versa.

Using a dynamic power sharing structure can also reduce the cost of the loudspeaker system. Instead of requiring each speaker or amplifier channel to be large enough to support the highest output power, each channel or speaker can be reduced in capacity and need only have a smaller capacity. When the signal exceeds the signal threshold of the smaller loudspeaker, the additional signal is bypassed to other associated channels. This approach can provide the same amount of apparent sound output as a larger system, while using a smaller overall system, including lower output power speakers and/or reduced amplifier power.

FIG. 2 is a schematic illustration of the components of channels 1-3 contained in FIG. 1 . Audio signal 60 enters channel 16 and passes through gain controlled amplifier 62 . Output amplifier 64 then amplifies the signal. The differential amplifier 66 compares the difference between the input signal 71 and the output signal 72 of the output amplifier. When the output amplifier starts clipping or overloading, then the output signal will be smaller than the input signal. The differential amplifier then sends a difference signal to the gain controlled amplifier based on the difference between the input and output of the output amplifier. The gain controlled amplifier has a variable element (such as a variable resistor) that is adjusted to keep the signal at a certain level, as per the input from the difference amplifier, and keeps the signal clipped further. For example, a gain controlled amplifier can reduce the amplifier gain when the output amplifier starts to distort by 1%. This limits clipping in the output amplifier. A rectification circuit 68 is used to generate the absolute value of the differential signal provided by the differential amplifier. This way the positive and negative parts of the signal will have a positive gain control to reduce distortion and/or clipping. A filter 70 is used to remove noise from the feedback circuit before the differential signal reaches the gain controlled amplifier.

The threshold limit at which the first channel begins to transfer power to the other channels can vary depending on signal frequency, thermal characteristics, amplitude limitations of the transducer, amplifier clipping, natural transducer characteristics, hot transducer characteristics, thermal impact on the amplifier , signal effects on amplifiers, power effects on amplifiers, and other similar phenomena that can affect signals or components of the system. Figure 2a illustrates a circuit that can sense threshold parameters other than amplifier clipping. The gain controlled amplifier 62 receives an input signal and passes it to an output amplifier 64 which then passes an output signal 72 to a load. In this case, the gain controlled amplifier is not controlled by amplifier feedback, but it is controlled by gain control circuit 74 . The signal or voltage generated by the gain control circuit is determined by a threshold limit sensor 76 . The threshold limit sensor can be a natural environment sensor, a strain gauge sensor, a heat sensor, a signal sensor, or a voltage sensor.

For example, if the amplitude limit of a transducer is defined as the highest threshold limit, then a sensor can be used at the transducer (e.g. a speaker cone) to determine when the transducer is close to the highest physical displacement. The maximum displacement can also be measured according to the transducer's highest safe voltage threshold. When the voltage is close to the maximum voltage, which can damage the transducer, then the gain control circuit reduces the gain in the gain controlled amplifier. This threshold limit sensor operates in the same way as a temperature sensor or a maximum frequency sensor. Depending on the temperature of the operating component, the signal can also be limited.

Figure 3 is a schematic diagram of a multi-channel system using power sharing steering logic. The analog circuits shown in FIGS. 1 and 2 can be implemented with a digital signal processing chip (DSP) 80 . In summing circuit 84, first input 82 may be summed together with overload signals 88 from other channels. This input signal is then passed to channel 1 (86) and into the power share steering logic. If channel 1 begins to overload, the overloaded signal can be distributed to channels 2 or 3 via their summing circuits 84a, 84b. It is also possible that parts of this overload signal can be distributed to channels 3 and 4 and combined by their summing circuits 84c, 84d.

An overload signal from one channel can be distributed among the other channels in several ways. One way is to select two or more channels corresponding to the main channel, and divide the signal equally between them. Another approach is to distribute the signal between two or more channels based on the physical location of those channels. For example, the rear speakers may have a smaller output than the front speakers. It is also possible that a given channel will have one, two, three or more channels as its associated channels. Channel 1 can route its signal to channel 5 or channels 3, 4 and 5. Overdrive structure is based on the number of channels available, the amount of overdrive present at a given point in time, and the sound image the system should present. Of course, the preferred embodiment of the device bypasses the overloaded portion of the signal to two other channels.

Dynamic power sharing can be used in a two-speaker stereo system. When the first channel reaches the overload signal threshold, then the signal power above the threshold is allocated to the second channel. Similarly, even a multi-channel system can divert power to only one channel instead of splitting it across two channels. While improving tonal distortion due to overloading, it is still possible to mix well above-threshold signal levels to at least two additional channels, preferably a speaker straddling the main channel, which can be physically Placed between two additional channels.

In other words, this power can be routed again to three or more other channels, depending on the desired directivity. For example, several channels and transducers may be physically stacked on top of each other. When the first channel starts to overload, the signal can be rerouted to a second speaker located above the first speaker. This preserves directionality and provides the much stronger undistorted signal that is needed. Since it is only driven to its maximum level for a fraction of the time, using two smaller speakers instead of one large speaker saves space and cost.

Figure 4 illustrates the power sharing on the center channel. When the signal delivered to the center channel 410 reaches threshold, overloads, or reaches a point of clipping, it can be divided symmetrically and diverted to the front channels at 460 degrees counterclockwise and 420 degrees clockwise. In other words, traffic above the threshold is routed to the left 460 degree and right 420 lanes. The signal is distributed symmetrically to substantially avoid sound image movement away from the center channel or transducer. This is possible because it is common practice to align the two front side channels symmetrically against the central channel. When the three channels regenerate the decomposed overdrive signal, a virtual source 412 is created that is larger than the output capacity of the original center channel. Then, if the left and right front channels are overloaded, the signals from these channels can be rerouted to the right 430° and left 450° surround channels and their transducers. Some surround sound systems may optionally include a sixth rear speaker 440 and this sixth channel may be used to receive a rerouted portion of the overloaded signal from the surround sound channel. Conversely, if the sixth channel is overloaded, the overloaded signal can be routed to the adjacent surround channels. If this surround channel is overloaded, other channels can be selected to increase the overall sound output. Additionally, the system may send an overloaded portion of the signal to one or more subwoofers in the system. The solid arrow 470 in Figure 4 represents the primary output direction of the loudspeaker that has reached the threshold, while the dashed vectors 480, 490 represent the directional output and cues provided by the auxiliary horns. The resulting dashed vector creates a virtual direction vector which is summed with the solid direction so that the original direction vector is audibly not shifted.

Figure 5 illustrates power sharing with respect to side channels. Overdriven side channels may be treated differently to maintain the spatial direction of the sound image. When the right front transducer is overloaded, the signal can be broken down unevenly. Most of the overloaded signal can be sent to the center channel 510 , while the remainder of the overloaded signal can be sent to the right rear surround channel 530 . Sending most of the signal to the front right channel helps reduce image drift. If the overload signal is distributed symmetrically, it may cause the sound image to shift to the back of the listener. This is because the surround transducers are usually weaker and placed farther away than the front speakers. As in the previous embodiment, when the loudspeakers left and right of the loudspeaker of interest are overloaded, the signal may be rerouted to adjacent loudspeakers that are not overloaded. For example, in FIG. 5 , if the rear surround channel 530 is overloaded, the overloaded signal may be rerouted to one or more of the other channels 540 , 550 . Also, a virtual sound source 512 is established, but it could actually be moved further back to the surround speakers than shown in the figure. In the present invention, even if the image is shifted slightly, this is much better than clipping the signal, which would produce audible distortion. For the sides of the head, people tend to reduce the psycho-auditory level of sounds that move compared to sounds that move in the front.

The threshold limit at which the first channel begins to transfer power to the other channels can be determined based on various parameters such as: signal frequency, thermal characteristics of the components, amplitude or displacement of the speaker diaphragm, clipping of the amplifier, and possible effects on the original signal Other similar phenomena that can damage system components, alter performance, or even cause localized sound pressure levels to be greater than desired for individual channels. Alternatively, the triggering threshold could be some combination of these parameters, or even an arbitrary value, in order to create the desired sound effect.

Referring now to FIG. 6 , a general method for increasing the characteristic acoustic output of a multi-channel sound system, each of which has an acoustic signal, is described. One step is to select a signal from one channel of the multi-channel sound system 610 . Another step is to select a predetermined parameter threshold corresponding to the signal level 620 . A further step is: sending a portion of the sound signal associated with one channel of the multi-channel sound system to at least one other channel of the multi-channel sound system when the signal reaches the predetermined parameter threshold.

7 illustrates the application of the present invention to systems in which audio program material is encoded with software or hardware control codes before or during recording on audio source medium 710. During playback, the power sharing function 720 is activated when a given channel or channels reaches a parameter threshold, such as an amplitude threshold. Power sharing can be accomplished by limiting the signal level of a given channel and rerouting portions of that signal to one or more other channels 730 . The method can be generalized to any system operation so as to minimize the requirements on any channel of the system.

Specifically, the software method of encoding may be optimized for a particular audio system, or may have adaptive settings to re-adapt the threshold parameters for a variety of different systems, each with different characteristics . For example, pre-programming of power sharing using encoded software or hardware can be achieved with a number of specific applications, including: (i) setting thresholds or pre-programming thresholds during recording or re-recording of audio material for listening; programming; (ii) subjectively given preset levels as estimated thresholds based on the specific type of audio system used for playback; and, (iii) incorporates a simple diagnostic routine as hardware or software preprogramming of recorded material Appropriate thresholds are derived by running this sequence of diagnostic tests as part of the test, thus ensuring an automatic evaluation of the audio system used. In the latter case, the CD, flash memory, hard drive or other recording medium may include an embedded diagnostic sequence for testing the system hardware and speakers to identify the specific thresholds required. Other methods for defining and/or pre-analyzing thresholds will be apparent to those skilled in the art based on the preceding exemplary descriptions.

It should be understood that: the above-mentioned arrangement is only used to illustrate the application program of the principle of the present invention, without departing from the spirit or scope of the present invention, various modifications and changes can be made, and the appended claims will cover these modifications and changes. retrofit. Therefore, while the invention has been shown in the drawings and has been described in full detail with respect to what is presently believed to be the most practical and best embodiment of the invention, it will be clear to a person of ordinary skill in the art that Various modifications and changes may be made without departing from the principles and concepts of the present invention as set forth in the claims, including but not limited to: changes in structure, implementation, shape, operation of function.

Claims (43)

1. a signal processing system is used for multi-channel audio system, comprising: the first passage that (a) has first voice signal; (b) at least one has the second channel of second sound signal; (c) processor is in response to the signal level thresholds in the first passage, so that the part of the first passage voice signal on this threshold value place and threshold value is mixed at least one of second audio channel.

2. according to the described signal processing system of claim 1, further comprise: at least one has the third channel of the 3rd voice signal; This signal processing system is at least in response to the signal level thresholds in this first passage, so that the part of the first passage voice signal on this threshold value place and threshold value is mixed at least one of the second and the 3rd audio channel.

3. according to the described signal processing system of claim 2, it is characterized in that: from LisPos, each first, second and the 3rd audio channel that all has corresponding first, second and the 3rd loudspeaker is placed in first, second and the corresponding acoustic surrounding of third direction vector with separately, along the first direction vector first loudspeaker is placed between the second and the 3rd loudspeaker.

4. according to the described signal processing system of claim 2, it is characterized in that: the three or more audio channels of being represented by three or more corresponding loudspeaker are placed in the acoustic surrounding, so that first audio channel and corresponding loudspeaker are represented unique direction vector of beginning from LisPos, it has presented a real acoustic image to the audience, this signal processor is in response to the signal level thresholds in the first passage at least, so that the part of the first passage voice signal on this threshold value place and threshold value is mixed in two auxiliary audio channels at least.

5. according to the described signal processing system of claim 4, it is characterized in that: at least two auxiliary audio channels have been strengthened should the reality acoustic image, along the first direction vector have one with the corresponding false reputation picture of this reality acoustic image.

6. according to the described signal processing system of claim 1, further comprise be applied to first passage, with the corresponding signal limitations function of signal threshold value, be used for limiting the signal level of first passage at a predetermined quantity.

7. according to the described signal processing system of claim 2, further comprise be applied to first passage, with the corresponding signal limitations function of signal threshold value, be used for limiting the signal level of first passage at a predetermined quantity.

8. according to the described signal processing system of claim 3, further comprise be applied to first passage, with the corresponding signal limitations function of signal threshold value, be used for limiting the signal level of first passage at a predetermined quantity.

9. according to the described signal processing system of claim 4, further comprise be applied to first passage, with the corresponding signal limitations function of signal threshold value, be used for limiting the signal level of first passage at a predetermined quantity.

10. a signal processing system is used for multi-channel audio system, comprising:

N passage, wherein n＞1; With the corresponding voice signal of each passage; The signal level thresholds relevant with each passage; A signal processor, in response to this signal level thresholds, in case so that any passage is when reaching this signal threshold value, this signal processor reaches this passage of signal threshold value to the major general the part of voice signal is routed at least one other passage of this multi-channel audio system.

11., it is characterized in that: when any a plurality of in N the passage reach this signal threshold value, be routed to any N other passage in this multi-channel audio system to the part of their voice signal of major general according to the described signal processing system of claim 10.

12. according to the described signal processing system of claim 10, further comprise the signal limitations function that is applied to the passage that reaches this signal threshold value, be used for limiting the signal level of first passage at a predetermined quantity.

13. according to the described signal processing system of claim 11, further comprise the signal limitations function that is applied to the passage that reaches this signal threshold value, be used for limiting the signal level of first passage at a predetermined quantity.

14. according to the described signal processing system of claim 1, it is characterized in that: this signal threshold value is relevant with the clipping level of amplifier, and this amplifier is related with the passage that reaches this signal threshold value.

15. according to the described signal processing system of claim 10, it is characterized in that: this signal threshold value is relevant with the clipping level of amplifier, and this amplifier is related with the passage that reaches this signal threshold value.

16. according to the described signal processing system of claim 1, it is characterized in that: this signal threshold value is relevant with the amplitude of loudspeaker, and these loudspeaker are related with the passage that reaches this signal threshold value.

17. according to the described signal processing system of claim 10, it is characterized in that: this signal threshold value is relevant with the amplitude of loudspeaker, and these loudspeaker are related with the passage that reaches this signal threshold value.

18. according to the described signal processing system of claim 1, it is characterized in that: described signal threshold value is relevant with frequency.

19. according to the described signal processing system of claim 10, it is characterized in that: described signal threshold value is relevant with frequency.

20. according to the described signal processing system of claim 1, it is characterized in that: this signal threshold value is relevant with the hot state of loudspeaker, and these loudspeaker are related with the passage that reaches this signal threshold value.

21. according to the described signal processing system of claim 10, it is characterized in that: this signal threshold value is relevant with the hot state of loudspeaker, and these loudspeaker are related with the passage that reaches this signal threshold value.

22. according to the described signal processing system of claim 1, it is characterized in that: this signal threshold value is corresponding to the overload related with the passage that reaches this signal threshold value.

23. according to the described signal processing system of claim 10, it is characterized in that: this signal threshold value is corresponding to the overload related with the passage that reaches this signal threshold value.

24. according to the described signal processing system of claim 1, it is characterized in that: this signal threshold value is relevant with predetermined distortion level, and this distortion level is related with the passage that reaches this signal threshold value.

25. according to the described signal processing system of claim 10, it is characterized in that: this signal threshold value is relevant with predetermined distortion level, and this distortion level is related with the passage that reaches this signal threshold value.

26. according to the described signal processing system of claim 1, it is characterized in that: at least one passage that reaches signal threshold value comprises the characteristic leading with respect to the phase place of at least one other passage.

27. according to the described signal processing system of claim 10, it is characterized in that: at least one passage that reaches signal threshold value comprises the characteristic leading with respect to the phase place of at least one other passage.

28. according to the described signal processing system of claim 1, it is characterized in that: at least one other passage that is different from least one passage that reaches signal threshold value has the time delay that reaches the passage of signal threshold value with respect at least one.

29. according to the described signal processing system of claim 10, it is characterized in that: at least one other passage that is different from least one passage that reaches signal threshold value has the time delay that reaches the passage of signal threshold value with respect at least one.

30. according to claim 1 or 10 described signal processing systems, further comprise the recording medium that is used on signal processing system, resetting, they comprise the code that presets that is used for the definition signal level threshold, and this signal level thresholds applies with respect to the program material that is included on this recording medium.

31. according to claim 1 or 10 described signal processing systems, further comprise the recording medium that is used on signal processing system, resetting, they comprise the code that presets that is used for the definition signal level threshold, this signal level thresholds applies with respect to the program material that is included on this recording medium, comprise based on the particular type of the audion system that is used to reset and subjective given preset level as the threshold value of estimating.

32. according to claim 1 or 10 described signal processing systems, further comprise the recording medium that is used on signal processing system, resetting, they comprise the code that presets that is used for the definition signal level threshold, this signal level thresholds applies with respect to the program material that is included on this recording medium, comprise that further a diagnostic program is as the part of programming in advance that writes down material, derive appropriate threshold by moving this diagnostic test sequence, therefore guarantee automatic assessment employed audion system.

33. according to the described signal processing system of claim 10, it is characterized in that: this signal processor is used to sequentially will be routed in other passage of this multi-channel audio system from least one part that reaches the multichannel voice signal of this signal threshold value in proper order.

34. a method that is used to increase the sign sound output of multichannel sound system, this multichannel sound system comprises a plurality of passages, and each passage has voice signal, and described method comprises the steps:

(a) select at least one signal of at least one passage of this multichannel sound system;

(b) select and the corresponding predetermined parameters threshold value of signal level; And

(c) reach in this predetermined parameters threshold value when this signal, will with one of this multichannel sound system

The part of the voice signal that individual passage is relevant sends at least one of this multichannel sound system

Individual other passage.

35. according to the described method of claim 34, further may further comprise the steps: offer this voice signal limitation function is provided, this voice signal is related with at least one passage with the signal that reaches predetermined predetermined parameter threshold.

36., further may further comprise the steps according to the described method of claim 34:

A) select at least three passages of the multichannel sound system represented by three corresponding loudspeaker

At least three signals are so that first audio channel and corresponding loudspeaker are represented from LisPos

Unique direction vector of beginning, it has presented a real acoustic image to the audience, corresponding loudspeaker

Be placed in the acoustic surrounding; And,

B) will mix with at least two residue audio channels from the part of the signal of first audio channel

Close, described signal has surpassed the predetermined parameters threshold value.

37. according to the described method of claim 34, further may further comprise the steps: before the record material of playing on this recording medium, preparation has the recording medium of embedded code, and this embedded code can be analyzed the parameter threshold that is used for given multichannel sound system in advance.

38. according to the described method of claim 37, further may further comprise the steps: before the record material of playing on this recording medium, preparation has the recording medium of embedded code, and this embedded code can apply a diagnostic procedure to given multichannel sound system, with analytical parameters threshold value in advance.

39. a signal processing system is used for multi-channel audio system, comprising:

(a) has the first passage of first voice signal;

(b) at least one has the second channel of second sound signal;

(c) processor is in response to the signal level thresholds that is applicable to first passage, so that at this letter

The part of the first passage voice signal on number level threshold place and this signal level thresholds

Be mixed at least the second audio channel.

40. according to the described signal processing system of claim 39, further comprise: at least one has the third channel of the 3rd voice signal; Described signal processing system is given in response to the signal level thresholds in described at least one first passage, so that the part of the first passage voice signal on described threshold value place and threshold value is mixed in the second and the 3rd audio channel at least.

41. according to the described signal processing system of claim 40, further comprise: first, second and the 3rd audio channel at least, from LisPos, each all have be placed on separately first, second and the corresponding acoustic surrounding of third direction vector in, first, second and the 3rd loudspeaker accordingly, these at least the first loudspeaker are with corresponding at least the first audio channel that is placed on the direction vector between the second and the 3rd loudspeaker.

42. according to the described signal processing system of claim 41, wherein: three or more audio channels comprise the three or more corresponding loudspeaker that are placed in the acoustic surrounding, so that first audio channel and corresponding loudspeaker are represented unique direction vector of beginning from LisPos, any one of this audio channel first audio channel and corresponding loudspeaker have at least two other auxiliary audio channels, they belong to the audio channel with corresponding loudspeaker, and described loudspeaker have and begin from LisPos, the direction vector that departs from the direction vector of first audio channel in the clockwise direction with counter-clockwise direction; This signal processor is given in response to the signal level thresholds in any one first passage, so that the part of any first passage voice signal on this threshold value place and threshold value is mixed in two auxiliary audio channels at least.

43. according to the described multi channel signals treatment system of claim 39, further comprise the signal limitations function that is applied to first passage, this first passage is corresponding with this signal threshold value.

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