WO2009045649A1

WO2009045649A1 - Phase decorrelation for audio processing

Info

Publication number: WO2009045649A1
Application number: PCT/US2008/073307
Authority: WO
Inventors: Robert Reams; James D. Johnston
Original assignee: Neural Audio Corporation
Priority date: 2007-08-20
Filing date: 2008-08-15
Publication date: 2009-04-09

Abstract

A system for decorrelating audio data is provided. The system includes a noise generator generating a random noise signal. A phase shift system receives an input channel of audio data and the random noise signal and generates a phase shifted channel of audio data having a phase shift based on the random noise signal.

Description

IN THE UNITED STATES PATENT AND TRADEMARK OFFICE

SPECIFICATION accompanying

Application for Grant of U.S. Letters Patent

TITLE: PHASE DECORRELATION FOR AUDIO PROCESSING

FIELD OF THE INVENTION

[0001] The invention relates to systems for processing audio data, and more particularly to phase decorrelation for audio processing that reduces or eliminates audio artifacts that can be caused by phase interference.

BACKGROUND OF THE INVENTION

[0002] Audio signals from sound sources are often processed using microphones or amplified and transmitted to a listener over speakers. When the audio signals are received from multiple microphones, such microphones are often used to receive audio signals from a single source but may also receive audio signals from other sources that also have independent microphones that are used to receive sound signals from those sources. Likewise, when the audio signals are amplified and transmitted over speakers, a predetermined phase relationship for the audio signals may be required. For example, a system of speakers that each have a flat frequency response over a predetermined range may be used, where the audio signal is provided to each signal and a cross-over filter is used to provide only those signals to each speaker that correspond to the frequency range for that speaker.

[0003] In both cases, phase distortion can be introduced that creates audio artifacts that adversely affect the sound quality. For systems utilizing inputs from multiple microphones that are subsequently mixed to produce a stereo signal, the signals received at each microphone from sound sources other than the intended sound source are typically out of phase with the signals received at the microphones for those sound sources, due to the additional time it takes for those signals to be received. For multiple speaker systems, the cross-over filter or spatial orientation of the speakers can result in unintended phase distortion. In either situation, the phase distortion creates audio artifacts that reduce the quality of the sound or that require additional processing or design features to reduce or eliminate. SUMMARY OF THE INVENTION

[0004] In accordance with the present invention, a system and method are provided for reduction of phase distortion in audio processing systems.

[0005] In particular, a system and method for decorrelatmg sound signals are provided that reduce or eliminate phase distortion in audio processing systems.

[0006] In accordance with an exemplary embodiment of the present invention, a system for decorrelatmg audio data is provided. The system includes a noise generator generating a random noise signal. A phase shift system receives an input channel of audio data and the random noise signal and generates a phase shifted channel of audio data having a phase shift based on the random noise signal.

[0007] The present invention provides many important technical advantages. One important technical advantage of the present invention is a system and method for decorrelatmg channels of audio data so as to reduce or eliminate phase interference between the channels of audio data.

[0008] Those skilled in the art will further appreciate the advantages and superior features of the invention together with other important aspects thereof on reading the detailed description that follows m conjunction with the drawings.

BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWINGS [0027] FIGURE 1 is a diagram of a system for providing spatial separation for a stereo output signal in accordance with an exemplary embodiment of a present invention;

[0028] FIGURE 2 is a diagram of a system for decorrelating the phase of an input signal in accordance with exemplary embodiment of the present invention;

[0029] FIGURE 3 is a flowchart of a method for decorrelating the phase of an input signal in accordance with an exemplary embodiment of the present invention;

[0030] FIGURE 4 is a diagram of a system for decorrelating microphonic inputs into a mixer in accordance with an exemplary embodiment of the present invention;

[0031] FIGURE 5 is a diagram of a system for decorrelating speaker outputs in accordance with an exemplary embodiment of the present invention;

[0009] FIGURE 6 is a diagram of a method for decorrelating sound signals in accordance with an exemplary embodiment of the present invention;

[0001] FIGURE 7 is a diagram of a system for sound field widening and phase decorrelation in accordance with an exemplary embodiment of the present invention;

[0002] FIGURE 8 is a diagram of a system for multi-channel phase decorrelation in accordance with an exemplary embodiment of the present invention;

[0003] FIGURE 9 is a diagram of a system for performing sound field widening and phase decorrelation in accordance with an exemplary embodiment of the present invention;

[0004] FIGURE 10 is a diagram of a method for processing a plurality of channels of audio data to decorrelate phase in accordance with an exemplary embodiment of the present invention;

[0005] FIGURE 11 is a diagram of a system for performing phase modification of an audio channel in accordance with an exemplary embodiment of the present invention;

[0006] FIGURE 12 is a diagram of a system for performing phase decorrelation and sound field widening in accordance with an exemplary embodiment of the present invention; and

[0010] FIGURE 13 is a diagram of a system for sound field widening and phase decorrelation with phase modulation in accordance with an exemplary embodiment of the present invention .

DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS

[0032] In the description that follows, like parts are marked throughout the specification and drawings with the same reference numerals, respectively. The drawing figures might not be to scale, and certain components can be shown in generalized or schematic form and identified by commercial designations in the interest of clarity and conciseness.

[0033] FIGURE 1 is a diagram of a system 100 for providing spatial separation for a stereo output signal in accordance with an exemplary embodiment of a present invention. System 100 can be used to provide an apparent spatial location for a stereophonic output channel from a monaural input, so as to allow multiple voice channels to be provided to a stereophonic headset. In this manner, an apparent spatial location for an input channel is provided to a listener, so as to allow the listener to distinguish different input signals based on the apparent spatial location.

[0034] System 100 includes decorrelators 102A through 102N and 104A through 104N, each of which receives left and right channels, respectively. In one exemplary embodiment, the left and right channels can be a monaural signal, such that the left and right channels are the same signal and have the same phase. Decorrelators 102A through 102N and 104A through 104N can be implemented in hardware, software, or a suitable combination of hardware and software, and can be one or more software systems operating on a digital signal processing platform, a general purpose processing platform, or other suitable platforms. As used herein, "hardware" can include a combination of discrete components, an integrated circuit, an application-specific integrated circuit, a field programmable gate array, or other suitable hardware. As used herein, "software" can include one or more objects, agents, threads, lines of code, subroutines, separate software applications, two or more lines of code or other suitable software structures operating in two or more software applications or on two or more processors, or other suitable software structures. In one exemplary embodiment, software can include one or more lines of code or other suitable software structures operating in a general purpose software application, such as an operating system, and one or more lines of code or other suitable software structures operating in a specific purpose software application.

[0035] Decorrelators 102A through 102N and 104A through 104N decorrelate the phase of the monaural signal received at the left and right inputs. In one exemplary embodiment, the left and right inputs can be transformed from a time domain to a frequency domain such that decorrelators 102A through 102N and 104A through 104N decorrelate the phase of the signal in the frequency domain. In this exemplary embodiment, a time to frequency domain transform system (not shown) is used to perform the time to frequency domain transformation of the input signal. [0036] Pinnae model filters 106A through 106N and 108A through 108N receive the decorrelated left and right channel inputs and apply a pinnae model filter to the input. In one exemplary embodiment, the pinnae model can be a frequency filter based on the generalized response of human hearing to frequency inputs .

[0037] Variable delays IIOA through HON are coupled to pinnae model filters 106A through 106N and variable delays 112A through 112N are coupled to pinnae model filters 108A through 108N. Variable delays HOA through HON and 112A through 112N provide an adjustable delay to the decorrelated and filtered signals, so as to generate an apparent spatial separation in the stereophonic output. In one exemplary embodiment, a listener that receives left and right channel inputs through a stereophonic listening device such as headphones may perceive a spatial separation based on the variable delay between the left and right channels. In a real-world environment, a listener determines the location of a point sound source based on the delay between when the sound signals are received at the listener's left and right ears. For example, a sound signal generated from a point sound source that is closer to a listener' s left ear will be received at the left ear sooner than sound is received at the listener's right ear. This time delay allows the location of the point sound source to be determined. In this exemplary embodiment, the apparent location of an input channel can be moved relative to the listener based on the amount of variable delay settings of variable delays IIOA through HON and 112A through 112N. In one exemplary embodiment, the amount of variable delays IIOA through HON and 112A through 112N can vary from 230 to 600 microseconds, so as to represent the amount of delay that is typically observed in three-dimensional listening environments. Likewise, a single delay can be used for each pair of channels, the delays can be fixed so as to provide a predetermined spatial location for each pair of channels, or other suitable embodiments can be provided. [0038] Variable pass filters 114A through 114N and variable pass filters 116A through 116N can be implemented in hardware, software, or a suitable combination of hardware and software, and can be one or more software systems operating on a general purpose processing platform. Variable pass filters 114A through 114N and 116A through 116N provide a variable band pass filter having a break point that can be related to the spatial separation of the left and right channels to the listener, and can be first order low pass filters, such as having a break point frequency of 2 kHz or other suitable frequencies. By adjusting the delay of variable delays IIOA through HON and 112A through 112N and the break point frequency of variable pass filters 114A through 114N and 116A through 116N, the apparent location of the signal received at left and right channel inputs can be altered for a listener using stereophonic headphones or suitable listening devices. Likewise, a single filter can be used for each pair of channels, the filters can be fixed so as to provide a predetermined spatial location for each pair of channels, or other suitable embodiments can be provided. [0039] Summation systems 118 and 120 can be implemented in hardware, software, or a suitable combination of hardware and software, and can be used when one or more software systems are operating on a general purpose processing platform. Summation systems 118 and 120 receive the output from variable pass filters 114A through 114N and 116A through 116N and add the signals to output the left shifted channel signal and right shifted channel signal, respectively. In one exemplary embodiment, the left shifted channel outputs and right shifted channel outputs are frequency domain signal, and can be transformed back to the time domain by suitable frequency-to- time transform system (not explicitly shown) .

[0040] In operation, system 100 allows left and right channel input signals to be processed so as to create an apparent spatial location when the signal is provided to stereophonic headphones or other suitable listening devices. System 100 utilizes variable time delay and frequency filters to create an apparent spatial separation to the listener. In addition, the left and right channel signals are decorrelated so as to eliminate any potential phase interference. Pinnae model filtering can be used to further optimize the apparent spatial location of the signal perceived by a listener through a stereophonic headphone device or other listening device so as to allow the left and right channel data to have a specific apparent location to the listener. In one exemplary embodiment, a plurality of left shifted and right shifted audio channels can be combined, such as to allow two or more audio inputs to be generated having different apparent spatial locations. In this manner, the user can distinguish various inputs based on their apparent spatial location. The variable delays and filters can also or alternatively be fixed, so as to provide a predetermined apparent spatial location for each of a plurality of input channels, such as to associate a predetermined source with a predetermined apparent location.

[0041] FIGURE 2 is a diagram of a system 200 for decorrelating the phase of an input signal in accordance with exemplary embodiment of the present invention. System 200 includes noise generator 204, quadrature phase shift 202, first order filter 206 and amplifier 210, each of which can be implemented in hardware, software, or a suitable combination of hardware and software, and which can be one or more software systems operating on a general purpose processing platform, a digital signal processor, or other suitable platforms. [0042] System 200 receives an input signal which is provided to quadrature phase shift 202. Quadrature phase shift 202 provides a 90 degree phase shift to the input signal. Potentiometer 208 provides an adjustable phase shift to the input signal ranging from 0 degrees to 90 degrees based on the setting of potentiometer 208. The setting of potentiometer 208 is randomly varied based on output from noise generator 204, which is filtered through a first order filter 206. In order to avoid generation of audio artifacts, noise generator 204 is controlled to generate random noise in a frequency range corresponding to the input frequency of the input signal. In one exemplary embodiment, the following relationships can be used to determine the frequency of noise to be generated based on the frequency range of the input signal:

0-50 Hz N(f) - 10 hZ

50-200 Hz N(2f) ~ 20 hZ

200-800 Hz N(4f) ~ 40 hZ

80-3.2 kHz N(8f) ~ 80 hZ

3.2-12.8 kHz N(16f) ~ 160 hZ

12.8 kHz - ∞ N(32f) ~ 320 hZ

[0043] In one exemplary embodiment, noise generator 204 can be varied based upon the measured frequency of the input signal, different noise generators can be used based on different frequency bands for the input signal or other suitable embodiments can be used. The output signal from variable potentiometer 208 is provided to amplifier 210, which amplifies the signal .

[0044] In operation, system 200 provides a decorrelator for use in decorrelating the phase of an input signal. In one exemplary embodiment, decorrelator system 200 can be used to provide the correlation to adjust the apparent spatial relationship of a stereophonic input woofer for the suitable purposes as described herein.

[0045] FIGURE 3 is a flowchart of a method 300 for decorrelating the phase of an input signal in accordance with an exemplary embodiment of the present invention. Method 300 begins at 302 where channels of sound are decorrelated. In one exemplary embodiment, the channels can be monaural signals that are decorrelated so as to decorrelate the in-phase monaural signal. Likewise, other suitable channels can be decorrelated . The method then proceeds to 304.

[0046] At 304, each channel of decorrelated audio data is filtered using a pinnae model or other suitable filters. The method then proceeds to 306.

[0047] At 306, it is determined whether a change in the apparent location to the listener of the input signal should be created. For example, two or more input channels can be used and an apparent location for each input channel can be created so as to allow a listener to perceive the apparent location of each input channel separately so as to facilitate the separation of the input channels by the listener. If it is determined at 306 that a change in location is not required, the method proceeds to 312. Otherwise the method proceeds to 308 where a variable delay is adjusted. In one exemplary embodiment, the amount of delay can be adjusted based on a range of 230 to 600 microseconds, where the amount of delay changes the apparent location of the audio channel. For example, if the amount of delay of the left channel relative to the right channel is 230 microseconds, then the apparent location of the sound to the listener will be closer to the center of the listener than the right side. Likewise, if the delay between the left and right channel is 600 microseconds, the apparent location of the sound will be closer to the left side of the listener. Other suitable delays can also or alternatively be used. In another exemplary embodiment, a predetermined location can be assigned based on the source of a sound channel. In this exemplary embodiment, if the listener is a pilot, then the location of communication channel received from a central control location can be assigned to a first location, such as the listener' s left side, the location of a communications channel received from a co-pilot can be assigned to a second location, such as left of center of the listener, the location of a communications channel received from a squadron leader can be assigned to a third location, such as right of center of the listener, and the location of a communications channel received from voice commands or instructions from guidance or weapons systems can be assigned to a fourth location, such as the listener's right side. [0048] After the delay is adjusted at 308, the method proceeds to 310 where a band pass filter is adjusted. In one exemplary embodiment, the band pass filter can be a first order band pass filter having a break point at approximately 2 khZ, where the frequency of the band pass is adjusted based on the frequency of the input data or other suitable factors. The method then proceeds to 312.

[0049] At 312, it is determined whether additional parties or channels should be added. In one exemplary embodiment, method 300 can be used to provide spatial separation between input channels for two or more inputs to a person using stereophonic headphones or other suitable equipment for listening to the output, such as a pilot or other suitable personnel who are receiving voice channel data from various parties such as ground control, co-pilots, or other suitable parties. If it is determined at 312 that additional parties are to be added, the method returns to 302. Otherwise the method proceeds to 314 and terminates .

[0050] In operation, method 300 allows changes to be made to provide apparent spatial separation to a listener for two or more input channels. Method 300 allows different voice channels to be processed so as to create an apparent location for each voice channel, where the apparent locations can be changed or modified based upon the number of voice channels, parties, or other suitable sound inputs.

[0051] FIGURE 4 is a diagram of a system 400 for decorrelating microphonic inputs into a mixer in accordance with an exemplary embodiment of the present invention. System 400 allows microphonic input to be decorrelated so as to avoid phase distortion caused by overlap of signals received at various microphones .

[0052] System 400 includes microphones 402 through 408, each of which is coupled to decorrelators 410 through 416, respectively. As used herein, the "couple" and its cognate terms such as "coupled" or "couples" can include a physical connection (such as through a copper conductor), a virtual connection (such as through randomly assigned data memory locations), a logical connection (such as through one or more logical devices) , other suitable connections, or a suitable combination of such connections.

[0053] Decorrelators 410 through 416 are coupled to mixer 418, which receives the inputs from decorrelators 410 through 416 and generates a stereo output 420. Mixer 418 can be a standard mixer that is used to mix a plurality of signal channel inputs so as to generate a stereo output.

[0054] In operation, system 400 applies random phase decorrelation to inputs received at microphones 402 through 408, so as to avoid phase distortion that may be caused by the delayed reception of sound signals at each microphone. In one exemplary embodiment, microphones 402 and 404 can be placed in proximity to each other, such as to record sound signals from a snare drum and a cymbal of a drum set, respectively. Because the sound signals received at microphone 404 will include some sound signals generated by the snare drum that is slightly out of phase with the sound signals received from the snare drum at microphone 402 (because of the time delay) , and the sound signals received at microphone 402 will include some sound signals generated by the cymbal that is slightly out of phase with the sound signals received from the cymbal at microphone 404 (because of the time delay), audio artifacts will be created when the sound signals are mixed because of the phase differences from the different sound sources. In an environment where multiple microphones are used for multiple different sound sources, the creation of audio artifacts can be a significant impediment to creating a sound mix that does not have an unacceptable level of such audio artifacts.

[0055] By decorrelating the signals received by microphones 402 and 404 using phase decorrelators 410 and 412, the effect of picking up out-of-phase cymbal sound signals at microphone 402 and out of phase snare drum sound signals at microphone 404 can be reduced or eliminated, so as to allow the operator of mixer 418 to more readily mix the sound signals received from the decorrelators 410 and 412 without having to compensate for phase distortion and creation of audio artifacts. As such, system 400 can be used in environments where a large number of microphones are provided that receive sound signals from multiple sources but which are oriented for receiving sound from primarily a single source. In this manner, the decorrelated signal inputs can help prevent the creation of phase distortion that can generate audio artifacts. The generation of such audio artifacts renders the job of mixing such stereo signals more difficult, such that decorrelating the phase of the inputs reduces the complexity of mixing and provides improved stereo outputs 420. [0056] FIGURE 5 is a diagram of a system 500 for decorrelating speaker outputs in accordance with an exemplary embodiment of the present invention. System 500 allows decorrelation of audio signals provided to multiple speakers so as to avoid phase distortion and interference from each speaker. [0057] System 500 includes decorrelators 502 and 504, which receive audio input and perform phase decorrelation on the audio input. In one exemplary embodiment, the audio input can include an audio signal that has been amplified and that is to be provided to speakers 506 and 508, such as a "tweeter" and "woofer" speaker pair that has been optimized for providing improved performance over a frequency range that is wider than can be properly handled by a single speaker. While a crossover filter is typically used to provide the high frequency signals from audio input to the tweeter and the low frequency signals to the woofer, both speakers may receive the output signal within the crossover frequency band. In this exemplary embodiment, decorrelators 502 and 504 provide phase decorrelation so as to avoid phase interference in the crossover region for the signals provided to speakers 506 and 508. In this manner, audio artifacts are not created by phase distortions created by the crossover filter or the signals provided to speakers 506 and 508.

[0058] Likewise, in other exemplary embodiments, speakers 506 and 508 can be speakers in different locations that are providing the same audio output over the same frequency range, where decorrelators 502 and 504 are used to decorrelate phase data. In this exemplary embodiment, speakers 506 and 508 may be providing parametric stereo signal, such as where the phase information has been removed, such that decorrelators 502 and 504 can be used to ensure that phase information is not inadvertently created between speakers 506 and 508 so as to create audio artifacts.

[0059] FIGURE 6 is a diagram of a method 600 for decorrelating sound signals in accordance with an exemplary embodiment of the present invention. In one exemplary embodiment, method 600 can be used to decorrelate inputs from microphones, outputs to speakers, or other suitable sound signals .

[0060] Method 600 begins at 602 where an input is received. In one exemplary embodiment, the input can be from a microphone, an input for provision to a speaker, or the suitable inputs. The method then proceeds to 604.

[0061] At 604, a frequency range is determined. In one exemplary embodiment, the frequency range can be optimized for a specific input, such as where a microphone is used for receiving sound from a specific sound source having a predetermined frequency range, for audio output signals that are to be amplified over a speaker that has been optimized for a certain frequency range, or other suitable frequency ranges. Likewise, method 600 can be performed on a signal that can vary over a wide frequency range, such that the frequency range determined at 604 is a selected for a specific frequency band to be decorrelated. Other suitable embodiments can also or alternatively be used. The method then proceeds to 606. [0062] At 606, it is determined whether a change in the frequency range is required. In one exemplary embodiment, when the input has a frequency variation such that a range adjustment is required, the method can proceed to 608. Otherwise, where a frequency range is set and is not varied, the method proceeds to 612. [0063] At 608, the noxse frequency for the decorrelator is adjusted. In one exemplary embodiment, noise frequencies can be set so as to prevent generation of audio artifacts from noise variations that are greater than a predetermined range, such as a noise frequency that is related to the frequency of the input signal. The method then proceeds to 610 where a first order filter is adjusted. In one exemplary embodiment, the first order filter and noise frequency can be related so as to provide a controllable level of decorrelation so as to prevent generation of audio artifacts. The method then proceeds to 612. [0064] At 612, it is determined whether a variable input is being received. In one exemplary embodiment, the input being processed can be received from a microphone such that the decorrelation is based on the frequency range of the signal being received. Likewise, the frequency can be variable based on a user control for a multiple speaker system or other suitable inputs. If it is determined at 612 that a variable input is not received the method proceeds to 614 and terminates. Otherwise, the method returns to 602.

[0065] In operation, method 600 allows an input signal to be decorrelated so as to change its phase based on a randomly generated noise frequency. Method 600 thus allows input signals from microphones, output signals to speakers, or other suitable signals to be phase decorrelated so as to prevent the generation of audio artifacts that can result from phase distortions between received signals at different microphones, phase distortions resulting from crossover between speakers, or other phase distortions.

[0007] FIGURE 7 is a diagram of a system 700 for sound field widening and phase decorrelation in accordance with an exemplary embodiment of the present invention. System 700 can be used to increase the perceived width of a sound field by phase decorrelation of the audio channel signals being processed. [0008] System 700 includes all pass filter Ll 702, all pass filter L2 704, all pass filter L3 706 and all pass filter L4 708, each of which can be implemented in hardware, software, or a suitable combination of hardware and software, and can be one or more software systems operating on a general purpose processing platform. As used herein, "hardware" can include a combination of discrete components, an integrated circuit, an application-specific integrated circuit, a field programmable gate array, or other suitable hardware. As used herein, "software" can include one or more objects, agents, threads, lines of code, subroutines, separate software applications, two or more lines of code or other suitable software structures operating in two or more software applications or on two or more processors, or other suitable software structures. In one exemplary embodiment, software can include one or more lines of code or other suitable software structures operating in a general purpose software application, such as an operating system, and one or more lines of code or other suitable software structures operating m a specific purpose software application. [0009] All pass filter Ll 702 is coupled to all pass filter L2 704, and all pass filter L3 706 is coupled to all pass filter L4 708. As used herein, the term "coupled" and its cognate terms such as "couples" or "couple, " can include a physical connection (such as a wire, optical fiber, or a telecommunications medium), a virtual connection (such as through assigned memory locations of a data memory device or a hypertext transfer protocol (HTTP) link) , a logical connection (such as through one or more semiconductor devices in an integrated circuit), or other suitable connections. [0010] System 700 receives left channel audio data and right channel audio data, which can be a series of digital time domain samples of audio data or other suitable audio data. The left channel audio data is provided to all pass filter Ll 702, which has a delay length of Ll. In one exemplary embodiment, where the audio data is digital time domain samples of an audio signal, the delay length Ll can be equal to the number of samples in the delay element of all pass filter Ll 702. After the left channel audio data is processed with all pass filter Ll 702, the processed data is then provided to all pass filter L2 704, which has a delay length L2 that is different from delay length Ll. The processed left channel audio data is then output, such as to speakers, for recording, or for other suitable applications.

[0011] The right channel audio data is likewise provided to all pass filter L3 706, which has a delay length L3 that is not equal to either delay length Ll or L2. The processed right channel data is then provided to all pass filter L4 708, which has a delay length L4 that is not equal to any of delay lengths Ll, L2 or L3. The processed right channel data is then output, such as to speakers, for recording, transmission, rendering m headphones or for other suitable applications.

[0012] In one exemplary embodiment, all pass filter Ll 702, all pass filter L2 704, all pass filter L3 706 and all pass filter L4 708 can each be sparse Nth order pole zero filters having an impulse response that is determined by the filter coefficients. The delay lengths of all pass filter Ll 702, all pass filter L2 704, all pass filter L3 706 and all pass filter L4 708 are selected so that the total delay length of a signal processed by all pass filter Ll 702 and all pass filter L2 704 is the same as the total delay length of a signal processed by all pass filter L3 706 and all pass filter L4 708. In one exemplary embodiment, the delay lengths of all pass filter Ll 702 and all pass filter L3 706 can be a first prime pair, and the delay lengths of all pass filter L2 704 and all pass filter L4 708 can be a second prime pair, such that the total length of the all pass filters can be set to an equal value by selection of suitable prime pairs. In one exemplary embodiment, the delay lengths of the first prime pair can be 41 and 43 and the delay lengths of the second prime pair can be 71 and 73, such that all pass filter Ll 702 can have a delay length of 41 and all pass filter L2 704 can have a delay length of 73, and all pass filter L3 706 can have a delay length of 43 and all pass filter L4 708 can have a delay length of 71. In this manner, the overall processing delay for both the left channel audio data (41 + 73) and the right channel audio data (43 + 71) will be equal (114) . [0013] In operation, the all pass filters create a phase shift in the left channel audio data and right channel audio data that depends on the frequency and the length of the all pass filters. Because the phase shift created by each all pass filter is different, any pre-existing phase correlations between the left channel audio data and the right channel audio data will be eliminated, which results in the apparent widening of the sound field to a listener, such as when the signals are amplified and transmitted over loudspeakers. Because correlation between the phase of individual frequency bins can result in the apparent location of that frequency bin as being in the center between the left and right channel speaker, decorrelation of such frequencies will widen the apparent sound field. In other words, correlation m phase results in a collapse of the sound field towards the center of the left and right channel speaker, whereas decorrelation in phase avoids the collapse of the sound field towards the center of the left and right channel speaker. Additional all pass filters can also or alternatively be utilized to create additional phase decorrelation, where suitable.

[0014] Furthermore, it is possible to connect two different pairs of sections to each channel, and to modulate the filter coefficients while keeping the coefficients for one pair in quadrature with those of the other pair, so as to modulate the decorrelation by a third signal, creating a time-varying change in the phase shifts. This third signal can be a sine wave of low frequency, a narrowband noise signal of low frequency or other suitable signals, such that reflection and cancellation patterns within a room or a listening environment are made to move about the room in a fashion preventing the human auditory system from strongly noticing the cancellations or reflections.

[0015] FIGURE 8 is a diagram of a system 800 for multichannel phase decorrelation in accordance with an exemplary embodiment of the present invention. System 800 can be implemented in hardware, software or a suitable combination of hardware and software, and can be one or more software systems operating on a general purpose processing platform. This system can be accordingly made time varying in the same fashion as above .

[0016] System 800 includes a plurality of channels of audio data from first audio channel, second audio channel, to an Nth audio channel. All pass filter Ll 802 having a delay length of Ll and all pass filter L2 804 having a delay length of L2 are used to process the first audio channel, and all pass filter L3 806 having a delay length of L3 and all pass filter L4 808 having a delay length of L4 are used to process the second audio channel. Likewise, each additional audio channel is processed by different all pass filters having different delay lengths, until the last or Nth audio channel is processed by all pass filter L2N 210 having a delay length of L2N and all pass filter L2N+1 212 having a delay length of L2N+1. The individual delay lengths of each of the all pass filters are different, but the total delay length of each channel is equal. In this manner, the phase shift applied to each of the individual frequency bins of each channel of audio data will be decorrelated, which can be used to avoid creation of audio artifacts, to prevent audio artifacts from being generated in a recording studio environment, such as where different microphones are used to record sound from the different parts of a recording sound stage, or in other suitable applications.

[0017] FIGURE 9 is a diagram of a system 900 for performing sound field widening and phase decorrelation in accordance with an exemplary embodiment of the present invention. System 900 includes a first channel signal processing circuit and an Nth channel signal processing circuit, such that system 900 can be used for stereo, four channel audio, 5.1 channel audio, 7.1 channel audio, 9.1 channel audio or any other suitable numbers of channels.

[0018] System 900 includes summation unit 902, gain unit 906, gain unit 910, delay 904, and summation unit 908, which form an all pass filter configuration that may be referred to as a Schroeder section, and which provides an all pass filter of length Ll when the gain factors of gain unit 906 and gain unit 910 are equal and opposite in sign, and which will pass all frequencies but will also add a phase shift depending on the frequency being passed, the delay, and the gain factor of gain unit 906 and gain unit 910. By selecting suitable values for the length of the delay and the gain of gain unit 906 and gain unit 910, the amount of phase shift at each frequency can be controlled. In one exemplary embodiment, by setting the gain of gam unit 906 equal to +0.25 and the gam of gain unit 910 equal to -0.25, the amount of phase shift, such as for audio signals from 0 hertz to 20,000 hertz, will vary in a predetermined manner for a given delay value between -180 degrees to +180 degrees, such that a frequency dependent phase shift will be generated in the processed signal. The delay length Ll can be selected based on a number of sampled signals that are processed and the sample rate. In one exemplary embodiment, digital audio signals that are sampled at a rate of 44.IkHz can be processed, where the length of the delay Ll is equal to the number of samples divided by 44,100, or can alternatively be specified by the integer number of samples processed.

[0019] Likewise, a second all pass filter is provided using summation unit 912, gain units 916 and 920, delay 914 with length L2, and summation unit 918. The phase modified signal received from the first all pass filter is then modified by the second all pass filter, which has different phase shift characteristics based on the delay length L2 and the amplitude settings of gain units 916 and 920.

[0020] Likewise, for the Nth channel processing chain, summation units 922 and 928, gain units 926 and 930, and delay 924 are used to form a first all pass filter having a delay length of 2N, and summation units 932 and 938, gain units 936 and 940, and delay 934 are used to form a second all pass filter having a delay length of 2N+1. The delay length of delay 924 and 934 are different from the delay length of delays 904 and 914, but the total delay for each filter chain is equal, such that the total delay length of delay 904 and 914 equals the total delay length of delay 924 and 934. In this manner, sound being processed from the first channel to the Nth channel will be phase shifted so as to eliminate any inadvertent phase correlation and thus widen the perceived sound field where the sound is being provided to speakers, to decorrelate any inadvertent phase correlations and reduce audio artifacts, such as where the sound is being recorded, and for other suitable purposes .

[0021] Any suitable all pass filter may be substituted, with proper mathematical treatment, for the Schroeder sections discussed above.

[0022] FIGURE 10 is a diagram of a method 1000 for processing a plurality of channels of audio data to decorrelate phase in accordance with an exemplary embodiment of the present invention. Method 1000 can be used to widen the sound field or for other suitable purposes related to phase decorrelation, and can be implemented as software on a general purpose processor. [0023] Method 1000 begins at steps 1002 and 1004 in parallel, where a first channel data stream is received at 1002 and an Nth channel data stream is received at 1004. Likewise, additional channels can also be separately received, so that any suitable number of channels can be processed using method 1000. [0024] The method then proceeds to 1006 and 1008 in parallel, where the first channel data is processed with an all pass filter having a delay length of Ll, and the Nth channel data is processed with an all pass filter having a delay length of 2N. Likewise, additional channels can be provided, each channel having a delay length that is different from the delay length of the all pass filter for any other channel. The method then proceeds to 1010 and 1012 in parallel.

[0025] At 1010, the first channel data that has been processed with all pass filter Ll is then processed with all pass filter L2, and at 412 the Nth channel data that has been processed with all pass filter L2N is processed with all pass filter L2N+1. Likewise, additional channels can each be processed by a corresponding second all pass filter, where the lengths of each of the corresponding second all pass filters are different from the lengths of the other second all pass filters. Nevertheless, the sum of the delay lengths of the first and second all pass filters for any channel of audio data should equal the sum of the delay lengths of the all pass filters for all other channels, so as to avoid loss of synchronization between the channels when real-time processing is performed. When the audio data is being processed for storage, the delay lengths for each filter chain can be different as long as the processed audio data for each channel is subsequently correlated so as to allow the audio data for each of the channels to be synchronized. The method then proceeds to 1014 and 1016 in parallel .

[0026] At 1014 and 1016, the phase modified first channel data is output and the phase modified Nth channel of data is output. Likewise, additional channels can be output. In this manner, the data can be provided to speakers, so as to generate an audio signal that has an apparent sound field that is wider than the unprocessed audio signal, where phase correlations between frequency bends of each channel can cause the sound field to collapse towards the center. Likewise, the decorrelated audio channels can be recorded for subsequent processing, such as mixing, or other suitable purposes. [0027] In operation, method 1000 can be used to decorrelate the phase of channels of data, such as to widen the apparent sound field to a listener, to avoid creation of audio artifacts, or for other suitable purposes. Although two exemplary filter stages are disclosed, additional filter stages can also or alternatively be used where suitable, as long as the delay length of each filter chain is the same for each channel of data .

[0028] FIGURE 11 is a diagram of a system 1100 for performing phase modification of an audio channel in accordance with an exemplary embodiment of the present invention. System 1100 can be implemented m hardware, software, or a suitable combination of hardware and software, and can be one or more software systems operating on a general purpose processing platform. [0029] System 1100 includes summation units 1102 and 1108, gain units 1106 and 1110 and delay 1104, having a length of 2N. Likewise, a second all pass filter unit is comprised of summation units 1112 and 1118, gain units 1116 and 1120, and delay 1114, having a length of 2N+1. Although a single Nth channel filter chain is shown, a suitable number of channels can be selected, as long as each have a total delay length that is equal to the delay length of each other filter chain, but where the length of any individual delay unit 1104 and 1114 is different from each other delay unit.

[0030] In operation, system 1100 provides a different architecture for performing stereo or multi channel phase decorrelation. By selecting suitable gain values and different delay lengths, each all pass filter will create phase variations as a function of frequency in the processed signal that are different from the phase variations of the other all pass filters, so as to eliminate unwanted phase correlation. System 1100 can be used for sound field widening, elimination of audio artifacts, or other suitable purposes. Additional filters can also or alternatively be added to the filter chain, so long as the total delay lengths of all filter chains are equal. [0031] FIGURE 12 xs a diagram of a system 1200 for performing phase decorrelation and sound field widening in accordance with an exemplary embodiment of the present invention. System 1200 utilizes a very long cascade of second order all pass filters, and can be implemented in hardware, software, or a suitable combination of hardware and software, and can be one or more software systems operating on a general purpose processing platform. Other suitable forms of all pass filters can also be substituted, with proper mathematical treatment, for the Schroder sections. The calculation of the many stages of gain elements for the second-order cascade that will provide useful phase-shift can likewise be developed accordingly.

[0032] FIGURE 13 is a diagram of a system 1300 for sound field widening and phase decorrelation with gain modulation in accordance with an exemplary embodiment of the present invention.

[0033] System 1300 includes summation unit 1302, gain unit 1306, gain unit 1310, delay 1304, and summation unit 1308, which form an all pass filter configuration that may be referred to as a Schroeder section, and which provides an all pass filter of length Ll when the gain factors of gam unit 1306 and gam unit 1310 are equal and opposite in sign, and which will pass all frequencies but will also add a phase shift depending on the frequency being passed, the delay, and the gam factor of gain unit 1306 and gam unit 1310. Likewise, a second all pass filter is provided using summation unit 1312, gain units 1316 and 1320, delay 1314 with length L2, and summation unit 1318. The phase modified signal received from the first all pass filter is then modified by the second all pass filter, which has different phase shift characteristics based on the delay lenqth L2 and the amplitude settings of gam units 1316 and 1320. [0066] Gam modulation units 1322 and 1324 are used to change the filter coefficients, such as by ramping them between -.25 and .25 or other suitable values. The coefficients for one pair of gain units are maintained in quadrature with those of the other pair, so as to modulate the decorrelation by a third signal, creating a time-varying change in the phase shifts. This third signal can be a low frequency sine wave, a narrowband noise signal of low frequency, or other suitable signals. In this manner, reflection and cancellation patterns within a room or a listening environment are moved relative to the listener so as to prevent the listener's auditory system from perceiving the cancellations or reflections. While two gain modulation units are shown, a single unit or other suitable numbers of units can also or alternatively be used.

[0067] Although exemplary embodiments of a system and method of the present invention have been described in detail herein, those skilled in the art will also recognize that various substitutions and modifications can be made to the systems and methods without departing from the scope and spirit of the appended claims .

Claims

What is claimed is:

1. A system for decorrelating audio data comprising: a noise generator generating a random noise signal; and a phase shift system receiving an input channel of audio data and the random noise signal and generating a phase shifted channel of audio data having a phase shift based on the random noise signal.

2. The system of claim 1 further comprising a filter coupled between the noise generator and the phase shift system and filtering the noise signal.

3. The system of claim 1 wherein the phase shift system further comprises: a quadrature phase shift system receiving the input channel of audio data and shifting the phase of the channel of audio data by ninety degrees; and a potentiometer having a first input coupled to the quadrature phase shift system, a second input coupled to the input channel of audio data, and an output that varies between the first input and the second input, where the random noise signal controls a setting of the output between the first input and the second input.

4. The system of claim 1 wherein the noise generator generating the random noise signal comprises means for generating a random signal.

5. The system of claim 1 wherein the phase shift system comprises means for generating a phase-shifted channel of audio data having a random phase shift.

6. A method for decorrelating audio data comprising: receiving an input signal of audio data; generating a control signal having a random value; and shifting a phase of the input signal of audio data based on the control signal.

7. The method of claim 6 wherein generating the control signal having the random value comprises generating a random noise signal.

8. The method of claim 7 wherein generating the random noise frequency comprises generating the random noise frequency at a frequency that is less than the frequency range of the input signal of audio data.

9. A system for processing audio data comprising: a first decorrelator receiving a first channel of audio data and randomly changing a phase of the first channel of audio data; and a second decorrelator receiving a second channel of audio data and randomly changing a phase of the second channel of audio data.

10. The system of claim 9 further comprising a microphone generating the first channel of audio data.

11. The system of claim 9 further comprising a mixer receiving the decorrelated first channel of audio data and the decorrelated second channel of audio data and generating a stereophonic signal.

12. The system of claim 9 further comprising a speaker receiving the decorrelated first channel of audio data.

13. The system of claim 9 further comprising: a first speaker having a first predetermined frequency response range receiving the decorrelated first channel of audio data; a second speaker having a second predetermined frequency response range receiving the decorrelated second channel of audio data.

14. A system for decorrelating signals comprising: a first all pass filter having a first delay length for processing a first signal; a second all pass filter having a second delay length coupled to the first all pass filter and for processing the first signal after processing by the first all pass filter; a third all pass filter having a third delay length for processing a second signal; a fourth all pass filter having a fourth delay length coupled to the third all pass filter and for processing the second signal after processing by the third all pass filter; and wherein the first delay length, the second delay length, the third delay length and the fourth delay length each have a unique value, and the sum of the first delay length and the second delay length is equal to the sum of the third delay length and the fourth delay length.

15. The system of claim 14 wherein each of the first all pass filter, the second all pass filter, the third all pass filter and the fourth all pass filter are pole zero filters.

16. The system of claim 14 wherein the first all pass filter comprises: a first summation unit; a first gain unit coupled to an output of the first summation unit; a delay coupled to the output if the first summation unit; a second summation unit coupled to the delay and the first gain unit; and a second gain unit coupled to the delay and the first summation unit.

17. The system of claim 14 wherein the first delay length and the third delay length comprise a first prime pair, and the second delay length and the fourth delay length comprise a second prime pair.

18. The system of claim 14 further comprising one or more additional all pass filters coupled to the second all pass filter, each having a unique delay length, and one or more additional filters coupled to the fourth all pass filter, each having a unique delay length, wherein the total delay of the first filter, the second filter and the one or more additional all pass filters coupled to the second all pass filter is equal to the total delay of the third all pass filter, the fourth all pass filter, and the one or more additional filters coupled to the fourth all pass filter.

19. The system of claim 14 wherein each of the first all pass filter, the second all pass filter, the third all pass filter and the fourth all pass filter are first order filters.

20. The system of claim 14 wherein each of the first all pass filter, the second all pass filter, the third all pass filter and the fourth all pass filter are digital filters.

21. The system of claim 14 further comprising a phase shift modulator modulating a gain of one of the all pass filters in quadrature with a gain of another of the all pass filters.

22. A method for processing a plurality of signals comprising : processing a first wide band audio signal with a first all pass filter chain to generate a phase shift in the first wide band audio signal that has a first variation as a function of frequency; processing a second wide band audio signal with a second all pass filter chain to generate a phase shift in the second wide band audio signal that has a second variation as a function of frequency that is different from the first variation as a function of frequency; and wherein a phase variation generated by the first all pass filter chain and the second all pass filter chain varies from no more than +180 degrees to no less than -180 degrees.

23. The method of claim 22 wherein processing the first wide band audio signal with the first all pass filter chain comprises : processing the first wide band audio signal with a first all pass filter having a first delay length; and processing the first wide band audio signal with a second all pass filter having a second delay length, wherein the first delay length is different from the second delay length.

24. The method of claim 22 wherein processing the second wide band audio signal with the second all pass filter chain comprises : processing the second wide band audio signal with a third all pass filter having a third delay length; and processing the second wide band audio signal with a fourth all pass filter having a fourth delay length, wherein the third delay length is different from the first delay length, the second delay length and the fourth delay length, and the sum of the first delay length and the second delay length is equal to the sum of the third delay length and the fourth delay length.

25. The method of claim 22 wherein the first delay length and the third delay length form a first prime pair, and the second delay length and the fourth delay length form a second prime pair.

26. The method of claim 22 wherein processing the first wide band audio signal with the first all pass filter chain comprises processing the first wide band audio signal with a first all pass filter having a first delay length and a second all pass filter having a second delay length.

27. The method of claim 26 wherein the first all pass filter and the second all pass filter are first order all pass filters .

28. The method of claim 22 wherein processing the first wide band audio signal with the first all pass filter chain comprises processing the first wide band audio signal with a first Schroeder section having a first delay length and a second Schroeder section having a second delay length.

29. The method of claim 22 further comprising modulating a gain of one of the all pass filters in quadrature with a gain of another of the all pass filters.

30. A system for decorrelating signals comprising: means for processing a first signal to generate a first phase shift as a function of frequency; means for processing the first signal to generate a second phase shift as a function of frequency; means for processing a second signal to generate a third phase shift as a function of frequency; means for processing the second signal to generate a fourth phase shift as a function of frequency; and wherein the first phase shift as a function of frequency, the second phase shift as a function of frequency, the third phase shift as a function of frequency and the fourth phase shift as a function of frequency are each unique.

31. The system of claim 30 wherein the means for processing the first signal to generate the first phase shift as a function of frequency comprises a first order all pass filter.

32. The system of claim 30 wherein the means for processing the first signal to generate the second phase shift as a function of frequency comprises a second order all pass filter.

33. The system of claim 30 further comprising a phase shift modulator modulating a gain of a first pass filter in quadrature with a gain of a second all pass filter.