Patent Abstract:
A method and apparatus for processing an audio signal to enhance the perceived lower frequency content of the audio signal when played through an audio output device, includes an input configured to receive an audio input signal, a processor configured to filter the audio input signal to produce a high frequency signal and a low frequency signal, generate an enhancement signal by producing higher frequency harmonics from the low frequency signal, including a process of self convolution, and combine the high frequency signal with the enhancement signal to produce an output signal; and an output configured to receive the output signal and produce an audio output.

Full Description:
CROSS REFERENCE TO RELATED APPLICATIONS 
     This application claims priority from United Kingdom Patent Application No. 09 06 594.7, filed 17 Apr. 2009, the whole contents of which are incorporated herein by reference in their entirety. 
     BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The present invention relates to processing an audio signal in order to enhance the sound output. 
     2. Description of the Related Art 
     In loudspeaker designs, the ability of the woofer to produce low frequencies is dictated by its size and power. With the increasing drive towards Small speaker designs, the bass cut-off frequency of such loudspeaker systems becomes higher. The missing fundamental effect is known. The brain perceives the pitch of a tone by the ratio of higher harmonics related to a fundamental, and not just by the fundamental itself. Thus, if the ear detects a series of harmonic frequencies not containing the fundamental frequency itself, the brain will still perceive the fundamental frequency to be present. Generating higher harmonics above the bass cut-off frequency of a loudspeaker when polyphonic sound is being played is difficult. A technique is therefore required to pitch shift the audio by one octave, thereby producing a first harmonic, and then produce additional higher-order harmonics. 
     BRIEF SUMMARY OF THE INVENTION 
     According to an aspect of the present invention, there is provided a method of processing an audio signal to enhance the perceived low frequency content of the audio signal when played through an audio output s device. The method comprises the steps of: receiving an audio input signal; 
     filtering the audio input signal to produce a high frequency signal and a low frequency signal; generating an enhancement signal by producing higher frequency harmonics from the low frequency signal; and combining the high frequency signal with the enhancement signal to produce an output signal. 
    
    
     
       BRIEF DESCRIPTION OF THE SEVERAL VIEWS OF THE DRAWINGS 
         FIG. 1  shows an example of components contained within an audio processing system in accordance with an embodiment of the present invention; 
         FIG. 2  shows an overview of processes according to an embodiment of the present invention; 
         FIG. 3  shows an expansion of the processing at step  105 ; 
         FIG. 4  shows the F 1  and F 2  buffers; 
         FIG. 5  shows examples of weighting values; 
         FIG. 6  illustrates the method of self convolution; 
         FIG. 7  shows a worked example of self convolution of F 1 ; 
         FIG. 8  shows self convolution of F 2 ; 
         FIG. 9  shows ring modulation; and 
         FIG. 10  shows production of the enhancement signal. 
     
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     
       FIG. 1 
     
     An example of components contained within an audio processing system in accordance with an embodiment of the present invention is shown in  FIG. 1 . A central processing unit  101  is provided, as well as a random access memory  102 , the latter being provided for the storage of programs and operation data executed by the central processing unit  101 . 
     Storage for programs and operational data is also provided by a hard disk drive  103 , although alternative forms of storage are possible, such as solid-state flash memory. An input/output interface  104  is provided for receiving input commands from, for example a mouse, keyboard or other input device, and for providing output to output devices, which may be audio output devices such as loudspeaker  105 , headphones or other types of output device. A network card  106  provides a facility to communicate over a network and new programs and data may be loaded across such a network, or indeed from portable storage devices, such as disc  107 , by a DVD drive  108 . The components communicate via a system bus  109 . 
     
       FIG. 2 
     
     An overview of processes according to an embodiment of the present invention is shown in  FIG. 2 . In order to enhance the sound output, a series of processes are carried out on the input signal. 
     An audio input signal is received at  201 , and a filter is applied at  202 . In an embodiment of the invention, the audio input signal is represented as digital samples, and thus the filtering step is performed in the digital domain. 
     In a preferred embodiment a single filter may be used, such as a high pass filter. When being deployed for enhancing the characteristics of a loudspeaker, the response of the high pass filter is preferably matched to the low frequency performance of the loudspeaker. The filtered signal may be subtracted from a copy of the original signal to produce a second signal that has only the low frequencies present. In an alternative embodiment, two separate filters may be used, one being a low pass filter and the second being a high pass filter. This alternative embodiment could be implemented using notch filters and/or a band pass filter. The filtering process at  202  separates the low frequencies shown at  203  from the high frequencies shown at  204 . The low frequencies are then processed at  205  as is further described with reference to  FIG. 3 . 
     A result of the above described processing is the production of an enhancement signal shown at  206 . This enhancement signal has been produced from the low frequencies, but is itself at a higher frequency. The enhancement signal  206  is then combined with the high frequencies  204  at  207 . Thus, the resulting output signal at  208  is produced with relatively high frequency content. However, due to processing that took place at  205 , the output signal sounds similar to the input signal due to psychoacoustic effects. 
     In particular, low frequencies contained in the input signal appear to the ear to still be present in the output signal. 
     The processing undertaken is performed by windowing (by using a function such as a Hann function) of an incoming audio sample, and convolving the windowed sample with the original audio sample. This can be seen as self-convolution. This process is further described with reference to  FIG. 3 . Thus, in the time/frequency-domain, the audio progresses sample by sample in one direction, whilst the impulse response is travelling in the opposite direction sample by sample. This results in a polyphonic linear pitch shift of a perfect octave. 
     Digital techniques generally produce odd order harmonics with relative ease, these being the type of harmonics that generally sound distorted and undesirable. The types of distortions that are considered desirable are generally even order harmonics, which are harder to produce digitally. The present invention provides a facility to produce the entire even order harmonic series by taking the second harmonic which has been generated by the above processing, and performing the processing again to produce a fourth, and so on. Further even order harmonics may be created in this way. 
     To achieve the missing fundamental effect, the method involves adding in the produced even order harmonic series with around 60% total harmonic distortion of a pure sine wave at certain prescribed amounts. The result is that without actually playing the fundamental lowest note, the ear will hear the total harmonic distortion and imagine the low note. This results in the perception of tones lower than are actually produced by an output device. Indeed, the ear will hear tones produced from a speaker that the speaker is in fact incapable of producing. 
     
       FIG. 3 
     
     An expansion of the processing at step  205  is shown in  FIG. 3 . The input of lower frequencies is as shown at  203 . A series of buffers are provided with samples of windowed signal. At step  301  a first buffer, the F 1  buffer (shown in  FIG. 4 ) is updated. 
     The most recent sample is added to the buffer and the oldest sample previously stored in the buffer is discarded. 
     At step  302  the F 1  buffer is convolved with itself. This is further described with reference to  FIG. 7 . As a result of this convolution a value F 2  is produced. A further buffer (shown in  FIG. 4 ) stores F 2  values and this buffer is updated with the new value at step  303 . The F 2  buffer is self convolved at step  304  as described with reference to  FIG. 8 . The result of this convolution is the value F 4 . 
     F 1  is the first harmonic, F 2  is the second harmonic and F 4  is the fourth harmonic. The self convolution process imposes a latency which is different for the F 1 , F 2  and F 4  values. The F 1  and F 2  samples are thus delayed at step  305  so that the F 1 , F 2  and F 4  values are realigned in time. 
     A process of ring modulation is then carried out at step  306 , as further described with reference to  FIG. 9 . This creates further harmonics. 
     The harmonics which have been produced are then summed with weighting factors at step  307 . This is further described with reference to  FIG. 10 . The result of this sum is an enhancement signal  308 . 
     
       FIG. 4 
     
     The F 1  buffer and the F 2  buffer are shown in  FIG. 4 . The F 1  buffer stores a series of samples of the incoming lower frequencies (F 1 ). The buffers are, in this example, of fixed length. In this case there are spaces for eight samples in the F 1  buffer. A value N is used to represent the number of spaces in a buffer so in this example N=8 and the eight spaces in the buffer are represented by A, B, C, D, E, F, G and H. 
     The F 2  buffer is also shown in  FIG. 4 . In this example the F 2  buffer stores the previous N/2 (N divided by 2) samples of the F 2  signal. So in this example as the F 1  buffer stores 8 values the F 2  buffer stores 4 values. 
     
       FIG. 5 
     
     Examples of arrays of weighting values are shown in  FIG. 5 . At  501  a first array is shown which relates to the F 1  harmonic and provides a series of eight weighting values which correspond with the eight samples which are stored in the H buffer. 
     At  502  a second weighting array is shown which corresponds with the F 2  harmonic and are used in order to self convolve the F 2  harmonic to produce the F 4  harmonic. 
     
       FIG. 6 
     
     The method of self convolution is illustrated in a general form in  FIG. 6 . For a block of size N contained in an array A[N], a set of window weight values contained in an array W[N] and an array index i with values from zero to N−1, the self convolution is as shown in  FIG. 6 . 
     
       FIG. 7 
     
     A worked example of self convolution of F 1  (used to produce an F 2  value) is shown in  FIG. 7 . The first value A from the F 1  buffer is convolved with the last value H from the F 1  buffer which is convolved with the first value W 1  from the F 1  weighting array. This is added to the result of the convolution of B with G and the second value W 2  from the F 1  weighting array, etc in accordance with the formula shown in  FIG. 7 . Thus the buffer is convolved with a windowed version of itself to produce a single sample of the second harmonic signal F 2 . This F 2  value produced is used to update another buffer as shown in  FIG. 8 . 
     
       FIG. 8 
     
     The F 2  buffer is convolved with a windowed version of itself to produce a fourth harmonic F 4  as illustrated in  FIG. 8 . 
     The self convolution process imposes a latency which is different for the F 1 , F 2  and F 4  values. Therefore the F 1  and F 2  samples are delayed so that the F 1  , F 2  and F 4  values are realigned in time. Afterwards ring modulation is used to create further harmonics. This is further illustrated in  FIG. 9 . 
     
       FIG. 9 
     
       FIG. 9  shows ring modulation in order to create harmonics F 13 , F 35  and F 26 . Each of these is created by convolution of previously created harmonics. Ring modulation of two signals containing frequencies A and B produces a signal with frequencies A plus B and A minus B. 
     
       FIG. 10 
     
     To produce the enhancement signal containing the harmonic series the separate harmonics are summed with weighting factors (represented here as W2, W4, W13, W35 and W26). The weighting values are used to control the relative contribution of each harmonic to the series. The enhancement signal is produced by the convolution of each weighting factor with its harmonic value as illustrated in  FIG. 10 . 
     The result of this processing is a signal having a realistic sounding and stable pitch shift of one octave. The self convolution technique is polyphonic, and so the pitch shift can be achieved completely in phase at all frequencies. 
     The enhancement signal produced as previously described is combined with the higher frequencies from the input signal in order to produce the final output signal. In an embodiment of the invention, the output signal is converted to an analog signal and thereafter amplified and supplied to an audio output device such as a loudspeaker. 
     The result of this processing is that the resulting output signal is perceived to include harmonics which are not actually part of the signal. This means that sounds are perceived which may not be within the production capability of the audio output device. For example, a small speaker which is incapable of reproducing low frequencies will apparently generate lower frequencies than it is physically capable of producing because the ear perceives fundamentals which are not present.

Technology Classification (CPC): 7