CN1894855A - System and method for detection of amplitude limit and modification of signal - Google Patents
System and method for detection of amplitude limit and modification of signal Download PDFInfo
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Abstract
Systems and methods for detecting clipping conditions in an audio signal and processing the signal to reduce the clipping conditions. In one embodiment, a system comprises a noise shaper, a modulator, an output stage and other components. A detector detects clipping in the noise shaper and a signal processor processes the audio signal input to the noise shaper based on feedback received from the detector. The signal processor may function to modify the input audio signal in different ways in response to different conditions that are detected by the detector. A filter may be included to filter tie output of the detector before being provided to the signal processor. A flag circuit may be coupled between the filter and the signal processor to assert an output signal until the signal processor resets the flag circuit.
Description
Background technology
Pulse width modulation (PWM) or D class signal amplification technique have existed many years, yet, along with the flourish of switched-mode power supply (SMPS) becomes more popular.Because the appearance of this technology, amplify application facet increasing interest has been arranged pulse width modulation being applied to signal.This is at least in part by using D class power output topological structure to replace the result of the significant efficiency improvement that traditional (linear AB class) power output topological structure can realize.
The early stage trial that the exploitation signal amplify to be used has been used in early days used identical processing method in the switched-mode power supply for amplification.More particularly, the modulation scheme of simulation has been used in these trials, and such modulation scheme causes extremely low performance applications.These application implementations are got up extremely complicated and expensive.Therefore, these schemes are not adopted acceptance widely.Therefore, the simulation embodiment of the prior art of D class technology can not replace traditional class ab ammplifier in the main flow amplifier is used.
Recently, digital pwm modulation schemes emerges.These schemes use ∑-Δ modulation technique to produce the pulse width modulating signal that is used in the new digital D class embodiment.Yet these digital pwm modulation schemes are very little for the contribution that the elimination pulse-width modulator is integrated in the major obstacle in total amplifier solution.Therefore, D class technology still can not replace traditional class ab ammplifier in mainstream applications.
One of problem that exists in current pulse-width modulated amplifier system is to control the performance of processing to realize optimizing for audio signal at an easy rate.Use the schematic diagram of Fig. 1 that this problem can be described.Fig. 1 is the calcspar of some basic element of character of the digital pulsewidth modulation amplification channel of explanation prior art.As shown in Figure 1, the parts of amplification channel 100 comprise: noise reshaper 110, modulator 120 and output stage 130.
High accuracy pulse code modulation (PCM) voice data (common 16 bits or its above bit width) is imported into noise reshaper 110, here said data is quantized.In the ordinary course of things, with this data-measuring into about being the 5-10 bit.Quantification with noise reshaper 110 is input to modulator 120 then, and modulator 120 becomes pulse width modulation (PWM) data with the pulse code modulation (pcm) data compilation.In the ordinary course of things, the data that produced by modulator 120 comprise high-side signal and low-side signal.Use these signals to drive the high-end of output stage 130 and low side field-effect transistor (FET), output stage 130 produces the pulse width modulating signal that amplifies.Usually, carry out low-pass filtering treatment to the signal that offers output stage 130 to remove high-frequency noise.
Referring now to accompanying drawing 2, wherein the expression explanation is according to the functional block diagram of the noise reshaper structure of prior art.As can be seen from Figure 2, noise reshaper 110 is made of quantizer 210 and filter 220.The input traffic that comprises the pulse code modulation voice data is combined with the filter data that is produced by filter 220.With the transfer function of filter 220 be designed to can the filtering input traffic and output stream between poor so that the noise that the attenuate quantization device produces in audio band and amplify the noise of upper frequency.It can be that the output valve of discrete number is handled said data splitting by shining upon said data that quantizer 210 is designed to.Thus, quantizer 210 can be realized roundoff function (promptly, the pulse code modulation data that receives is rounded to immediate quantized value) and amplitude limit function (that is the pulse code modulation data of reception that, will be outside quantizing range is mapped to maximum or the minimum value that can express).
When the input of quantizer 210 surpasses the quantizing range value, the said data of quantizer amplitude limit.The final output class of quantizer 210 is similar to the amplitude limit in common analogue amplifier, wherein will remove peak value from signal.So just changed by data represented audio signal and to have produced may be the audio artifacts that can hear.Such amplitude limit of the signal that quantizer 210 is received also may cause quantization error.Quantization error is poor between the output of the input of quantizer 210 and quantizer 210.When quantizer 210 carried out amplitude limit for signal, quantization error increased.Quantization error may produce unsteadiness in noise reshaper, and may produce can audiblely influencing that other does not expect.
Press traditional approach, the method that solves the problem that causes by the said signal of amplitude limit in noise reshaper is to use the amplitude limiter circuit amplitude limit input signal that separates with quantizer then to be entered into noise reshaper again.To be configured amplitude limiter circuit, so that with the level of signal limiter clip level of quantizer in being lower than noise reshaper.Though this has alleviated the quantization error problem really, if not because the amplitude limit of quantizer may produce quantization error, but but there be not basic the elimination because the distortion of the signal that amplitude limit causes, and, also may make problem worse even, this is because the low cause of limitation signal level ratio quantizer of amplitude limiter circuit.Change another kind of mode and see, use amplitude limiter circuit can produce not desired effects: limited the maximum magnitude (modulation index) of output, this is not indispensable.
Summary of the invention
More than one or more problems of Gai Kuoing can solve by various embodiment of the present invention.In a broad sense, the present invention includes and be used for detecting the amplitude limit state of audio signal and handle said signal to relax the system and method for amplitude limit state.
One embodiment of the present of invention comprise: be used to relax a system of some problem of prior art systems, here, be not with input audio signal amplitude limit to a fixing level, and be to use variable mode to handle input audio signal.For example, constantly can the said input audio signal of amplitude limit at some, constantly can compress said input audio signal at other, and not change said input audio signal constantly completely at other.
An embodiment comprises a system, said system comprises a detector and a signal processor, said detector is configured to detect the amplitude limit state in the audio signal, said signal processor can receive the feedback signal of self-detector, signal processor can be configured to respond the feedback signal that receives from detector here, and revise audio signal.In one embodiment, said system comprises noise reshaper, modulator, output stage and several additional parts.These parts comprise and are used for detecting amplitude limit status detection device and being used for basis is input to the audio signal of noise reshaper from the feedback processing of detector reception signal processor at noise reshaper.The effect of signal processor can be that the different state that detector response detects is revised input audio signal in a different manner.
Among the embodiment, said system also comprises a filter, carries out filtering for the output of detector, then this information is offered signal processor as feedback.The filtering of feedback can be used for preventing that signal processor from revising audio signal under the occurrent condition of the amplitude limit of audio signal, allows signal processor to revise audio signal simultaneously under the more recurrent condition of the limit ratio of audio signal.In one embodiment, said system also comprises a marking circuit that is coupling between filter and the signal processor.Said marking circuit is configured to receive filtering feedback signal from filter, and when confirming the filtering feedback signal, can confirms to offer the output signal of signal processor.The output signal state that maintenance always is identified before signal processor reset flag circuit by the marking circuit generation.The more stable even feedback signal of said signal of the simple filtering of a ratio so just can be provided to signal processor.
An interchangeable embodiment comprises a kind of method, and said method comprises the steps: to detect the amplitude limit state in audio signal; Revise audio signal with response for the amplitude limit status detection.With regard to the above embodiments, the amplitude limit state can be different for different embodiment with the response for the amplitude limit status detection.In one embodiment, implement said method at the sampling rate conversion device that is used for digital audio amplifier.
Many additional embodiment also are possible.
Description of drawings
After having read following detailed and the following accompanying drawing of reference, other purpose of the present invention and advantage all will become apparent.
Fig. 1 is the calcspar of some basic element of character of prior art digital type pulse width modulated amplification channel.
Fig. 2 is the functional block diagram of explanation according to the structure of the noise reshaper of prior art.
Fig. 3 is the functional block diagram of explanation according to the part of the digital audio amplification channel of an embodiment.
Fig. 4 is the functional block diagram of explanation according to the operating position of the clip detector of an embodiment and signal processor.
Fig. 5 is that explanation is according to the signal processor of the embodiment curve chart for the response of input audio signal.
Fig. 6 is the functional block diagram of explanation according to the operating position of clip detector, filter and the signal processor of an alternative embodiment.
Fig. 7 is the functional block diagram of explanation according to the operating position of clip detector, limit filtration device and the amplitude limit mark and the signal processor of another alternative embodiment.
Though the present invention can carry out various improvement and replacement form, yet, the specific embodiment of the present invention that will represent by means of the example in the accompanying drawing and additional detailed description also.Yet should be appreciated that, do not wish that these accompanying drawings and detailed description limit the present invention to described specific embodiment.On the contrary, disclosure of the present invention is intended to cover all improvement, equivalent and the alternative that drops in the scope of the present invention that is limited by the accompanying claims.
Embodiment
One or more embodiment of the present invention is described below.Be noted that these and any other embodiment described below are exemplary, be intended to illustrate the present invention rather than restriction the present invention.
As described herein, various embodiment of the present invention comprises and is used for detecting the amplitude limit state of audio signal and handles said signal to alleviate the system and method for amplitude limit state.
One embodiment of the present of invention comprise a kind of system that is used to alleviate some problem of prior art systems, here, be not with the input audio signal amplitude limit to fixing level, but handle input audio signal with a kind of variable mode.For example, constantly can the said input audio signal of amplitude limit at some, constantly can compress said input audio signal at other, and not change said input audio signal completely in other the moment.In one embodiment, said system comprises noise reshaper, modulator, output stage and several additional parts.These parts comprise and are used for detecting amplitude limit status detection device and being used for basis is input to the audio signal of noise reshaper from the feedback processing of detector reception signal processor at noise reshaper.The effect of signal processor can be that the different state that detector response detects is revised input audio signal in a different manner.
In one embodiment, said system also comprises a filter, carries out filtering for the output of detector, then this information is offered signal processor as feedback.Filtering to feedback can be used for preventing that signal processor from revising audio signal under the occurrent condition of the amplitude limit of audio signal, allows signal processor to revise audio signal simultaneously under the more recurrent condition of the limit ratio of audio signal.In one embodiment, said system also comprises a marking circuit that is coupling between filter and the signal processor.Said marking circuit is configured to receive filtering feedback signal from filter, and when confirming the filtering feedback signal, can confirms to offer the output signal of signal processor.By the output signal state that maintenance always is identified before signal processor reset flag circuit that marking circuit produces, so just can provide the more stable even feedback signal of said signal of the simple filtering of a ratio to signal processor.
Various interchangeable embodiment is described below, and other embodiment will be conspicuous for the person of ordinary skill of the art.
Referring now to accompanying drawing 3, wherein expression is the functional block diagram of explanation according to the part of the digital audio amplification channel of an embodiment.In this embodiment, the digital audio and video signals with pulse code modulation is input to amplitude limit prevention unit 310.Amplitude limit stops unit 310 can handle or not handle input audio signal, and this depends on the feedback that derives from noise reshaper 320.By behind the noise reshaper 320 quantization audio signals, audio signal is offered modulator 330, modulator 330 converts said audio signal to a pair of pulse width modulating signal.These signals offer output stage 340 then, and output stage 340 produces a pulse width modulation audio signals.
As shown in Figure 3, audio signal offers system with the form of pulse code modulation signal or the form of PCM signal.Pulse code modulation is a kind of scheme that is used for the coding simulation data.Pulse code modulation signal is binary.That is, how complicated no matter analog waveform has, only there are two possible states, with logical one (high level) and logical zero (low level) expression.Use the pulse code modulation scheme, the analogue data of digitlization form of ownership all is possible, comprising voice data.
In order to obtain the pulse code modulation audio stream of analog waveform, be with the amplitude of the time sampling analog signal of rule.Sampling frequency is several times of peak frequency of analog waveform.The method that quantizes the instantaneous amplitude of each analog signal of taking a sample is that it is rounded to apart from one group of integer that specific predetermined level is nearest.The number of level is 2 power, and the amplitude of sampling is by one group of binary digit (bit) expression corresponding to 2 power.For example, if analog signal is quantified as 64 (2
6) level, then each sampling is by a group expression of 6 bits.If analog signal is quantified as 1024 (2
10) level, then each sampling is by a group expression of 10 bits.
The pulse code modulation audio signal is handled by signal processor 310, and sends it to noise reshaper 320.Noise reshaper 320 has quantizer 321 and filter 322.The pulse code modulation voice data normally is quantized into the high accuracy data of 16 bits or more bits.The purpose of quantizer 321 is the bit numbers that voice data are quantified as smaller amounts.In the ordinary course of things, quantizer 321 produces the voice data of 5 to 10 bits.The purpose of filter 322 is that the noise that is produced by quantizer 321 is shifted out audio band.In other words, the noise in the filter 322 attenuation audio frequency bands is amplified in the noise of audio band outside simultaneously.This is by obtaining in the audio signal that is input to noise reshaper 320 with by poor, the difference signal that filtering is final between the audio signal of quantizer 321 output and it is added to audio signal then is input to quantizer 321 again and realizes.
Quantizer 321 in the process of quantization audio data again for the data processing of rounding off, and also have potential may the said data of amplitude limit.The reason that takes place that rounds off is, the level of the data that input audio data begins the to quantize quantizer 321 that might not match quantizes the level of said data again.The contingent reason of amplitude limit is that input audio data may be included in the value outside the scope that may occur in the output of quantizer 321.Being mapped to quantizer in these values outside this scope exports on the maximum or minimum value that may represent.
When input audio signal has surpassed the scope of quantizer 321, this signal of quantizer 321 amplitude limits.In other words, remove peak value from this signal.This amplitude limit is similar to the amplitude limit that takes place in common analogue amplifier.Remove peak value from signal and will make signal distortion, and if signal distortion is enough obvious, this distortion can be heard so.As previously discussed, the amplitude limit of signal also causes the increase (between the output of the input of quantizer 321 and this quantizer poor) of quantization error.If quantization error is enough big, it is unstable that noise reshaper may become.
Therefore, can utilize algorithm to describe the operation of quantizer 321 according to following mode:
if(in＞max)
out＝max
else?if(in＜min)
out＝min
else
out＝delta*round(in/delta)
As used herein, " in " is input audio signal, " Out " is output audio signal, " max " is the Maximum Output Level by the signal of quantizer 321 generations, " min " is the minimum output level of quantized signal, " delta " is the step sizes that quantizes, and " round " turns back to an immediate integer-valued function with input." max " and " min " is the multiple of " delta ".After quantizing, " Out " can be taken at any one in interior (max-min) delta+1 the possible values of scope [min, max].The maximum that is used for determining the scope (value of " min " and " max ") of quantizer 321 has been determined will be by the minimum and the maximum pulse of modulator 330 generations.Usually, set for these values, they can be mated will be by the driver of output stage 340 in the embodiment of appointment and the minimum and the maximum pulse of field-effect transistor processing.
Export by noise reshaper 320 by the signal that quantizer 321 produces, and offer modulator 330.The pulse code modulation data that modulator 330 compilings quantize again is pulse width modulation (PWM) data.In other words, each the group bit corresponding to audio data sample all converts a pulse with the width that is proportional to sample in pulse code modulation data stream.For modulator 330, produce two output signals.These signals are identical, and just phase place is opposite each other.In this embodiment, output stage 340 comprises two field-effect transistors (FET).Use drives one of field-effect transistor in the output stage 340 by each signal that modulator 330 produces.In one embodiment, all to carry out low-pass filtering for each output signal that modulator 330 produces, to remove high-frequency noise.
Those of ordinary skill in the art should be realized that this is a description of highly simplifying.In the ordinary course of things, on pulse code modulation data, carry out interpolation and correction,, and then enter data into noise reshaper so that for non-linear correction the in the pulse width modulation processing.And then, also to handle in the ordinary course of things, so that support output stage protection and edge setting the output of noise reshaper.Output stage can have any in a series of different configurations, for example configuration of semibridge system N+N field-effect transistor.Replaceable configuration can be used N or P field-effect transistor, put in order the bridge-type configuration, or the like.Other noise reshaper configuration also is possible.After having read disclosure of the present invention, many such variations all are conspicuous for the person of ordinary skill of the art.
In one embodiment, all the corresponding component with the above prior art systems of describing at Fig. 1 100 is identical for noise reshaper 320, modulator 330 and output stage 340.Like this, it may be the pulse code modulation audio signal of 16 bit signals that noise reshaper 320 receives, and quantizes said signal and make it to become bit than peanut.
System shown in Fig. 3 is in the system that is different from Fig. 1 aspect the use of amplitude limit prevention unit 310.Amplitude limit stop unit 310 be not simply with the input audio signal amplitude limit to fixed level, but it is opposite, handle for input audio signal, make its amplitude limit or revise audio signal as required, to alleviate above those problems of describing in conjunction with Fig. 1 according to alternate manner.In other words, amplitude limit stops the mode of operation of unit 310 to depend on audio signal itself.If input audio signal has low comparatively speaking modulation index and does not therefore comprise any peak value of the range of signal that may quantize above noise reshaper 320, amplitude limit stops unit 310 not revise input audio signal completely, provides this signal to noise reshaper 320 then.On the other hand, therefore will be by the peak value of quantizer 321 amplitude limits of noise reshaper 320 if input audio signal has very high comparatively speaking modulation index and has, then amplitude limit stops unit 310 can revise audio signal significantly, provides this signal to noise reshaper 320 then.If input audio signal has medium modulation index, then the function of amplitude limit prevention unit 310 is to revise the some parts of audio signal occasionally, and the part that keeps other is not revised.
With reference to Fig. 4, wherein expression explanation is according to the functional block diagram of the operating position of the clip detector of an embodiment and signal processor.In this embodiment, amplitude limit stops unit 410 to comprise clip detector 411 and signal processor 412.A pulse code modulation audio signal is input to signal processor 412, and signal processor 412 is according to this signal of feedback processing that receives from clip detector 411, and the signal forwarding that will handle is to noise reshaper 420 then.Noise reshaper 420 quantizes this signal and it is transferred to modulator 430, and modulator 430 provides a pair of signal to driving output stage 440.
In this embodiment, clip detector 411 receives the same signal of the quantizer 421 that is input to noise reshaper 420.Clip detector 411 is handled this input signal concurrently with quantizer 421, should revise the state of the input audio data of pulse code modulation to detect expression signal processor 411.In one embodiment, this state can be simply: audio signal has surpassed that quantizer 421 may quantize again and without the scope of the said signal of amplitude limit.In other words, the amplitude limit in the clip detector 411 detection quantizers 421.The operation of clip detector 411 can be described according to the form of following mode with algorithm:
if(in＞maxclip)
clip＝true
else?if(in＜min?clip)
clip＝true
else
clip＝false
As used herein, " in " is the input of quantizer, and " maxclip " and " minclip " is the threshold level that amplitude limit begins to take place, and, as the scope [minclip of input (" in ") in permission; Maxclip] outside the time, be set at very by the signal " clip " of clip detector 411 output, otherwise be vacation.
In the ordinary course of things, " maxclip " and " minclip " can set arbitrarily.In a preferred embodiment, minclip is set to the minimum limiting threshold (" min " of quantizer, as used in the arthmetic statement of above-mentioned quantizer), maxclip is set at the maximum limiting threshold (" max " is as used in the arthmetic statement of above-mentioned quantizer) of quantizer.This just allows to implement effectively the following combination quantizer and the scheme of clip detector:
if(in＞max)
out＝max
clip＝true
else?if(in＜min)
out＝min
clip＝true
else
out＝delta*round(in/delta)
clip＝false
Should be noted that, though the embodiment that is described by aforementioned algorithm has confirmed an output signal (promptly no matter when just confirming an output signal in case amplitude limit has taken place) according to the simple amplitude limit of the signal that is input to quantizer, interchangeable embodiment can implement more complicated detection scheme.For example, these embodiment can be when detecting the particular state group or confirm an output signal when detecting the instable amplitude limit type that may cause quantizer.Any such amplitude limit state can be detected by clip detector 411, and can be used as the basis of affirmation by the corresponding output signal of clip detector 411 generations.
The output signal (" amplitude limit ") that is produced by clip detector 411 offers signal processor 412 as feedback, and signal processor 412 is positioned at the place ahead of noise reshaper 420.The purpose of signal processor 412 is scopes of audio signal that restriction is input to noise reshaper 420, but in doing so will be according to than the mode of the said signal to of a simple amplitude limit fixing level " intelligence " mode more.In one embodiment, signal processor 412 is configured to be used as a compressor reducer-limiter.In this embodiment, signal processor 412 is configured to compress at least a portion audio signal, so that avoid or reduce to occur in amplitude limit in the noise reshaper 420 as far as possible.In many cases, this implements may be easily, because implement amplifier of the present invention a compressor reducer-limiter may have been arranged, and this compressor reducer-limiter may be used to realize this function by signal processor 412.
Explanation may be by the compressor reducer-restrictor function of signal processor 412 realizations in Fig. 5.Fig. 5 is the curve chart of explanation signal processor 412 for the response of an input audio signal.The response of curve 510 expression signal processors 412.Response curve 510 can be regarded as having three segmentations.First segmentation is for from 0 response to the input of the scope of first threshold 505 (corresponding to output level 501).Input audio level in this scope is to export with 1: 1 proportional linearity.In other words, output equals input.If input is in the scope of first threshold 505 to second threshold values 506, then output is compressed.Under another kind of mode, output level is less than incoming level, but is not limited.Output level reduces since output in 1: 1, and this begins a upwardly extending dotted line by first segmentation from response curve 510 and represents in Fig. 5.At last, at Maximum Output Level 502, carry out amplitude limit for the incoming level that is higher than second threshold value 506.
Should be noted that compressor reducer-restrictor function of representing is a kind of possible variation of the response of signal processor 412 in Fig. 5.According to another kind of mode, this response may be that quadrant width of cloth function is simple like that, perhaps may be complicated more, has additional response segmentation, level and smooth response curve or other variation.During the feedback that receives from clip detector 411 (perhaps limit filtration device or amplitude limit marking circuit also will be described below in greater detail this below) in response, response function also may change.
Be noted that, signal processor 412 can be placed on various other functional module of amplifier channel (as above-mentioned interpolation and pulse width modulation linearity correction module,) before, among or afterwards, for example above-mentioned interpolation and pulse width modulation linearity correction module.Can also implement signal processor 412 according to various mode.For example, can in the hardware that uses gate, implement signal processor 412.According to another kind of mode, and may be simple more and solution more flexibly, be used in programmable digital signal processor (DSP) and go up the software of operation and implement signal processor 412, in digital amplifier, have sort processor usually.
With reference to Fig. 6, wherein represent functional block diagram according to the operating position of clip detector, filter and the signal processor of an interchangeable embodiment.In this embodiment, amplitude limit stops unit 610 to comprise: clip detector 611, limit filtration device 613 and signal processor 612.Again, a pulse code modulation audio signal is input to signal processor 612, signal processor 612 is handled said signal according to the feedback that receives.Yet, in this embodiment, receive said feedback through filter 613 from clip detector 611.Signal processor 612 passes on the signal of handling and gives noise reshaper 620, and noise reshaper 620 quantizes said signals and it is transferred to modulator 630.Modulator 630 provides a pair of signal to driving output stage 640 then.
In this embodiment, clip detector 611 receives the signal of the quantizer 621 that is input to noise reshaper 620 and handles this signal concurrently with quantizer 621.Clip detector 621 detects the amplitude limit state, and this amplitude limit state representation signal processor 611 should be revised the pulse code modulation voice data of input, and produces corresponding signal.This signal is offered limit filtration device 613.This signal offers signal processor 612 as feedback after 613 filtering of process limit filtration device, signal processor 612 can feed back represented modification input audio signal according to this then.As previously discussed, audio signal is to change according to the feedback that receives by the modification of signal processor 612.
At high incoming level, the quantizer 621 in the noise reshaper 620 may be an amplitude limit audio signal by accident.This may be for example owing to the noise shaping of signal or because due to the of short duration big input spike.According to the design of noise reshaper 620, noise reshaper 620 may be handled the amplitude limit of several such spikes and can not become unstable.Therefore, compress input signal in these cases and may unnecessarily limit output signal.Therefore, can use the single outbreak or the of short duration burst of limit filtration device 613 filterings " amplitude limit detection ", prevent that their from triggering the modification of the input audio signal that is undertaken by signal processor 612.
In one embodiment, can use a counter that an almost complete embodiment of limit filtration device 613 is provided.Can implement said counter according to following arthmetic statement:
if(clip＝true)
if(clipcnt＝clipmax)
clipcnt＝clipcnt
clipfiltered＝true
else
clipcnt＝clipcnt+1
clipfiltered＝false
else
clipcnt＝0
clipfiltered＝false
As used herein, " clip " here is the output of clip detector 611.The number of effective " amplitude limit " input that internal state variable " clipcnt " counting is continuous.Before handling beginning, " clipcnt " is initialized as 0.When " clipcnt " reached value " clipmax ", counter stopped counting, and was set at " clipfiltered " output true.The input " clip " become invalid before " clipfiltered " remain effectively always.This will be reset to 0 with " clipcnt " simultaneously.
Those of ordinary skill in the art should be realized that, implements the mode of limit filtration device and has nothing in common with each other.Various possible embodiment can be based on hardware, software or their combination.These embodiments may be used various dissimilar algorithms, for example IIR, FIR, majority voting, increase/subtract counting, and so on.Should be noted that when implementing the useful practice is limits value " clipmax " can be programmed, thereby can regulate this value at the several applications of circuit with mounting hardware.
With reference to Fig. 7, wherein represent the functional block diagram of the operating position of clip detector, limit filtration device, amplitude limit mark and signal processor according to another interchangeable embodiment.In this embodiment, amplitude limit stops unit 710 to comprise: clip detector 711, limit filtration device 713, amplitude limit marking circuit 714 and signal processor 712.As in other embodiments, a pulse code modulation audio signal is input to signal processor 712, signal processor 712 is handled said signal according to the feedback that receives.Yet, in this embodiment, receive said feedback through marking circuit 714 from clip detector 711 and filter 713.Signal processor 712 passes on the signal of handling and gives noise reshaper 720, and noise reshaper 720 quantizes said signals and it is transferred to modulator 730.Modulator 730 provides a pair of signal to driving output stage 740 then.
In this embodiment, clip detector 711 receives the signal of the quantizer 721 that is input to noise reshaper 720 and handles this signal concurrently with quantizer 721.Clip detector 721 detects the amplitude limit state, and produces the signal of the correspondence that is provided for limit filtration device 713.This signal offers amplitude limit marking circuit 714 after 713 filtering of process limit filtration device.If confirmed, till then amplitude limit marking circuit 714 is confirmed output signals and kept this signal when signal processor 712 resets this amplitude limit marking circuit 714 from the signal (expression should be revised the audio signal that is input to signal processor 712) through filtering of filter 713 receptions.The output signal that amplitude limit marking circuit 714 is produced offers signal processor 712 as feedback.Signal processor 712 can feed back the indicated mode according to variable according to this and revise input audio signal.
Using the amplitude limit marking circuit may be useful in certain embodiments, especially all the more so based on those embodiment of software.Amplitude limit marking circuit 714 can be inserted between limit filtration device 713 (perhaps clip detector 711) and the signal processor 712, perhaps marking circuit 714 be added the inside of one of these parts.The purpose of amplitude limit marking circuit 714 be the relevant amplitude limit input audio signal of storage once or the sufficiently long time of amplitude limit to trigger the information of limit filtration device.In case signal processor 712 has moved (or the affirmation of response flag circuit 714 is moved) when confirming marking circuit 714, signal processor 712 is just to " removing " signal of marking circuit 714 excitations, so that remove this mark.When implementing in the software on digital signal processor, can inquire the output of amplitude limit marking circuit 714, perhaps use and interrupt handling.
Will be understood by those skilled in the art that, can use in the various technology any one to come expression information and signal.For example, can represent data, instruction, order, information, signal, bit, symbol and the chip quoted in the above description by voltage, electric current, electromagnetic wave, magnetic field or particle, optical field or particle or their combination in any.Use any suitable transmission medium, comprising metal wire, metallic traces, through hole, optical fiber and analog, can between each parts of disclosed system, communicate by letter these information and signal.
Those of ordinary skill in the art will recognize also that further the various illustrative logical block, module, circuit and the algorithm steps that are combined in embodiment description disclosed herein may be embodied as electronic hardware, computer software or the combination of the two.For this interchangeability of hardware and software clearly is described, more than at their function various illustrative parts, logical block, module, circuit and step have been described prevailingly.These functions are implemented as hardware and also are implemented as software, depend on specific application and are applied to design constraint on the whole system.Those of ordinary skill in the art can realize described function according to various mode at each specific application, but such enforcement decision should be interpreted as departing from for scope of the present invention.
Being combined in various illustrative logical block, module and circuit that embodiment disclosed herein describes can utilize with lower member and implement or realize: general processor, digital signal processor (DSP) or other logical device, application-specific integrated circuit (ASIC) (ASIC), field programmable gate array (FPGA), discrete gate circuit or transistor-transistor logic circuit, discrete hardware component and being used to finished their any combination of function described herein.General processor can be any conventional processors, controller, microcontroller, state machine or similar equipment.Processor can also be implemented to become the combination of calculation element, for example combination of a digital signal processor and a microprocessor, a plurality of microprocessor, the one or more microprocessors that combine with the digital signal processor core or any other such configuration.
The step that is combined in method that embodiment disclosed herein describes or algorithm can be in hardware, directly implement in software of being carried out by processor or firmware module or in the combination at them.Any other form storage medium that software module can be put into RAM memory, flash memory, ROM memory, eprom memory, eeprom memory, register, hard disk, removable disk, CD-ROM or be known in the art.A typical storage medium can be coupled on the processor so that processor can be from storage medium sense information and information can being write in the storage medium.In alternative, storage medium can be integrated on the processor.Processor and storage medium can be placed in the application-specific integrated circuit (ASIC).Application-specific integrated circuit (ASIC) can be arranged on the user terminal.In alternative, processor and storage medium can be used as discrete parts and are arranged in the user terminal.
Former description for the disclosed embodiments makes those of ordinary skill in the art can produce or use the present invention.The various improvement of these embodiment all becomes at an easy rate for the person of ordinary skill of the art and shows and easily, and Ding Yi General Principle can be applied among other the embodiment and can not depart from design of the present invention and scope here.Therefore, do not expect to limit the present invention to the embodiment of expression here, but the present invention is met and principle disclosed herein and new Ying's feature the wideest consistent scope.
Abovely benefit provided by the invention and advantage have been described with reference to specific embodiment.These benefits and advantage and the key element or the limits value that their are taken place or become outstanding more are not considered to the feature of key, that require or the essence of any claim or all authority requirement.Just as used herein, term " comprises " or its any other variation can be interpreted as not comprising the key element or the limits value of these term back with getting rid of.Therefore, comprise that a system, method or other embodiment of one group of key element is not only limited to these key elements, the key element of other that can also comprise that clearly do not list or claimed embodiment is intrinsic.
Though described the present invention with reference to specific embodiment, should be appreciated that these embodiment are illustrative, scope of the present invention is not limited to these embodiment.For many variations of embodiment described above, modification, additional and to improve all be possible.Can expect that these change, revise, add and improve and all will drop in the scope of describing in detail in following claims of the present invention.
Claims (24)
1. system comprises:
A detector is configured to detect the amplitude limit state in audio signal;
A signal processor is coupled to receive the feedback signal from described detector to it;
Wherein, described signal processor is configured to respond the feedback signal that receives from described detection device and revises described audio signal.
2. the system of claim 1, wherein: is variable by described signal processor for the modification of described audio signal.
3. the system of claim 1, further comprise a noise reshaper, wherein: described detector is coupled to described noise reshaper, and is configured for described detector, makes it can detect the amplitude limit state of the described audio signal in the described noise reshaper.
4. the system of claim 3, wherein: described system comprises one or more parts of digital audio amplifier.
5. the system of claim 1, wherein: described signal processor is configured to and can revises described audio signal by the described audio signal of amplitude limit.
6. the system of claim 1, wherein: described signal processor is configured to and can revises described audio signal by compressing described audio signal.
7. the system of claim 6, wherein: described signal processor is configured to and can revises described audio signal by the part that compression surpasses the described audio signal of threshold level.
8. the system of claim 1 further comprises a filter, filter coupledly wherein is configured for described filter between described detector and described signal processor, so that carry out filtering for the feedback signal of described detector.
9. the system of claim 8, wherein: described limit filtration device comprises a counter, described counter of each clock cycle for the output signal of confirming described clip detector all will add 1, for the described counter reset of each clock cycle of the output signal of not confirming described clip detector.
10. the system of claim 9, wherein: be configured for described limit filtration device, can confirm described output signal when reaching threshold level through filtering with the described counter of box lunch.
11. the system of claim 8, further comprise a marking circuit, described marking circuit is coupling between described filter and the described signal processor, wherein: the feedback signal that described marking circuit is configured to receive process filtering, and, if described feedback signal through filtering is in the state that is identified, then to keep described feedback signal to be in the state that is identified through filtering, until described marking circuit when resetting by described signal processor till.
12. the system of claim 1, wherein: described amplitude limit state comprises the simple amplitude limit state of described audio signal.
13. a method comprises:
In audio signal, detect the amplitude limit state;
Respond described amplitude limit status detection and revise described audio signal.
14. the method for claim 13, wherein: revise described audio signal and comprise in variable mode and revise described audio signal.
15. the method for claim 13, wherein: in described audio signal, detect described amplitude limit state and comprise: the described amplitude limit state that detects the described audio signal in noise reshaper.
16. the method for claim 15, wherein: in digital audio amplifier, amplify described audio signal by described noise reshaper output.
17. the method for claim 13, wherein: revise described audio signal and comprise the described audio signal of amplitude limit.
18. the method for claim 13, wherein: revise described audio signal and comprise the described audio signal of compression.
19. the method for claim 18, wherein: revise the part that surpasses threshold level that described audio signal comprises the described audio signal of compression.
20. the method for claim 13 further comprises: the feedback signal that filtering is corresponding with the amplitude limit state of detection wherein responds described amplitude limit status detection and revises described audio signal and comprise that the described feedback signal through filtering of response revises described audio signal.
21. the method for claim 20, wherein: the described feedback signal of filtering comprise the affirmation that responds described feedback signal make described counter add 1 and respond described feedback signal do not confirm to make described counter reset.
22. the method for claim 21, wherein: the described feedback signal of filtering further comprises confirms described feedback signal through filtering when described counter reaches threshold level.
23. the method for claim 20 further comprises: if confirm described feedback signal through filtering, then keep the affirmation of described feedback signal through filtering, until the described feedback signal of response through filtering when revising described audio signal till.
24. the method for claim 13, wherein: described amplitude limit state comprises the simple amplitude limit of described audio signal.
Applications Claiming Priority (5)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| US45641403P | 2003-03-21 | 2003-03-21 | |
| US60/456,414 | 2003-03-21 | ||
| US46977003P | 2003-05-12 | 2003-05-12 | |
| US60/469,770 | 2003-05-12 | ||
| PCT/US2004/008534 WO2004086616A2 (en) | 2003-03-21 | 2004-03-19 | Clip detection in pwm amplifier |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN1894855A true CN1894855A (en) | 2007-01-10 |
| CN1894855B CN1894855B (en) | 2011-06-15 |
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| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN200480009689.9A Expired - Fee Related CN1894855B (en) | 2003-03-21 | 2004-03-19 | System and method for detection of amplitude limit and modification of signal |
Country Status (5)
| Country | Link |
|---|---|
| EP (1) | EP1606884A2 (en) |
| JP (1) | JP4617298B2 (en) |
| CN (1) | CN1894855B (en) |
| HK (1) | HK1094626A1 (en) |
| WO (1) | WO2004086616A2 (en) |
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103955784A (en) * | 2014-04-01 | 2014-07-30 | 南京航空航天大学 | Manufacturing workshop individual electronic display board system based on data collection coprocessor |
| CN107221335A (en) * | 2017-05-27 | 2017-09-29 | 大连理工大学 | A kind of digitalizer and method for monitoring audio signal |
| CN109766889A (en) * | 2018-11-19 | 2019-05-17 | 浙江众合科技股份有限公司 | Orbital image based on curve matching identifies post-processing approach |
Families Citing this family (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP4618017B2 (en) * | 2004-07-02 | 2011-01-26 | ヤマハ株式会社 | Clip suppression circuit for pulse width modulation amplifier |
| US7286010B2 (en) * | 2006-01-26 | 2007-10-23 | D2Audio Corporation | Systems and methods for over-current protection |
| JP5266830B2 (en) * | 2008-03-26 | 2013-08-21 | ヤマハ株式会社 | Self-excited class D amplifier |
| WO2016162156A1 (en) * | 2015-04-09 | 2016-10-13 | Bang & Olufsen Icepower A/S | Duty cycle clipper |
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| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| JP3047420B2 (en) * | 1990-03-15 | 2000-05-29 | ソニー株式会社 | Data compression encoder |
| JP3121987B2 (en) * | 1994-05-11 | 2001-01-09 | シャープ株式会社 | Speaker drive |
| JP3214981B2 (en) * | 1994-06-30 | 2001-10-02 | 旭化成マイクロシステム株式会社 | Delta-sigma A / D converter with AGC function |
| JP2000022544A (en) * | 1998-07-01 | 2000-01-21 | Mitsubishi Electric Corp | Da conversion device |
| US6157331A (en) * | 1998-10-01 | 2000-12-05 | Tritech Microelectronics, Ltd. | Sigma delta modulator with automatic saturation detection and recovery |
| US6122331A (en) * | 1999-06-14 | 2000-09-19 | Atmel Corporation | Digital automatic gain control |
| WO2001026222A1 (en) * | 1999-10-01 | 2001-04-12 | Smith Carroll W | Dynamic closed-loop audio amplifier system |
| WO2001039370A2 (en) * | 1999-11-29 | 2001-05-31 | Syfx | Signal processing system and method |
| US6577737B1 (en) * | 2000-02-17 | 2003-06-10 | Visteon Global Technologies, Inc. | Method of detecting a DC offset in an automotive audio system |
| JP4251769B2 (en) * | 2000-11-15 | 2009-04-08 | ヤマハ株式会社 | Digital audio amplifier |
| JP3815607B2 (en) * | 2001-11-16 | 2006-08-30 | 横河電機株式会社 | Temperature transmitter using analog-digital converter |
-
2004
- 2004-03-19 EP EP04757922A patent/EP1606884A2/en not_active Withdrawn
- 2004-03-19 WO PCT/US2004/008534 patent/WO2004086616A2/en active Search and Examination
- 2004-03-19 CN CN200480009689.9A patent/CN1894855B/en not_active Expired - Fee Related
- 2004-03-19 JP JP2006507396A patent/JP4617298B2/en not_active Expired - Fee Related
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2007
- 2007-02-15 HK HK07101872.4A patent/HK1094626A1/en not_active IP Right Cessation
Cited By (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103955784A (en) * | 2014-04-01 | 2014-07-30 | 南京航空航天大学 | Manufacturing workshop individual electronic display board system based on data collection coprocessor |
| CN103955784B (en) * | 2014-04-01 | 2016-09-14 | 南京航空航天大学 | Manufacturing shop based on data acquisition coprocessor personal electric banner system |
| CN107221335A (en) * | 2017-05-27 | 2017-09-29 | 大连理工大学 | A kind of digitalizer and method for monitoring audio signal |
| CN107221335B (en) * | 2017-05-27 | 2020-07-14 | 大连理工大学 | Digitization device and method for monitoring audio signal |
| CN109766889A (en) * | 2018-11-19 | 2019-05-17 | 浙江众合科技股份有限公司 | Orbital image based on curve matching identifies post-processing approach |
| CN109766889B (en) * | 2018-11-19 | 2021-04-09 | 浙江众合科技股份有限公司 | Rail image recognition post-processing method based on curve fitting |
Also Published As
| Publication number | Publication date |
|---|---|
| CN1894855B (en) | 2011-06-15 |
| WO2004086616A3 (en) | 2004-11-25 |
| EP1606884A2 (en) | 2005-12-21 |
| HK1094626A1 (en) | 2007-04-04 |
| JP2006523414A (en) | 2006-10-12 |
| JP4617298B2 (en) | 2011-01-19 |
| WO2004086616A2 (en) | 2004-10-07 |
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