Abstract:
A method of converting single channel audio (mono) signals to two channel audio (stereo) signals using simple filters and an Intra-aural Time Difference (ITD) is presented. This method does not distort the spectral content of the original signal very much, and has low computation requirements. A variation is proposed which also uses Intra-aural Intensity Difference (IID).

Description:
CROSS REFERENCE TO RELATED APPLICATIONS 
     This application is related to contemporaneously filed U.S. patent application Ser. No. 11/560,397 BAND-SELECTABLE STEREO SYNTHESIZER USING STRICTLY COMPLEMENTARY FILTER PAIR and U.S. patent application Ser. No. 11/560,390 STEREO SYNTHESIZER USING COMB FILTERS AND INTRA-AURAL DIFFERENCES. 
     TECHNICAL FIELD OF THE INVENTION 
     The technical field of this invention is stereo synthesis from monaural inputs. 
     BACKGROUND OF THE INVENTION 
     Converting mono audio signals to stereo is a common need in current audio electronics. Two channel stereo sound is now standard. Two channel stereo generally has a much more natural and pleasant quality than mono. People naturally hear everyday sounds in stereo. There are still situations where mono sound signals exist such as telephone conversations, old recordings, low-end toys and radios etc. Converting such signals to stereo can greatly enhance their naturalness. 
     A mono signal carries no directional clues to the original location of the recorded sources. Additionally the original sound should be modified as little as possible to avoid coloration. Since mono signals are more common in low-end equipment, the computational cost of the mono to stereo conversion should be at a minimum because the low-end equipment typically has limited computational capacity. 
     SUMMARY OF THE INVENTION 
     This invention decomposes the original mono signal with filters, adds intra-aural time differences (ITD) using delays and optionally attenuates or filters representing intra-aural intensity differences (IID) and mixes to stereo. These intra-aural time differences and the optional intra-aural intensity differences provide directional clues in a mono to stereo conversion with low computational cost and low distortion. 
     Low computation is achieved depending on the filters used. Very good stereo quality can be achieved by centering the vocal range, moving the lower frequencies to the right side and moving the higher frequencies to the left side. This is similar to many musical performance situations. If only ITD is used, there is very little distortion compared to the mono signal while still producing a realistic stereo sensation. A great deal of flexibility is available choice of the cut-off frequencies and the ITDs and optional IIDs. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       These and other aspects of this invention are illustrated in the drawings, in which: 
         FIG. 1  illustrates a first embodiment of this invention in block diagram form; 
         FIG. 2  illustrates the high-pass separation filter response, the low-pass intra-aural intensity difference (IID) and the combined response of the right channel of the embodiment of  FIG. 1 ; 
         FIG. 3  illustrates a second embodiment of this invention in block diagram form; and 
         FIG. 4  illustrates a portable music system such as might use this invention. 
     
    
    
     DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS 
     The basic technique of this invention splits the mono signal into two or more different signals using filters. These different signals are sent to respective left and right channels of the stereo signal output with different delays. This produces different left and right channel signals. Different left and right channel gains may optionally be applied. Using simple complementary filters without gain reduces or eliminates coloration of the stereo signal. 
     A mono signal has few clues about source locations. However, many people are accustomed to hearing speaking or singing the center and high and low frequencies to the sides. For many live orchestras and some rock bands the low instruments tend to be toward the right and the high instruments tend to be on the left. This invention uses three filters corresponding to a mid-range band-pass, a hi-pass and a low-pass. These filters were designed to be complementary. Often in movies and in many recordings, the vocal sounds, whether singing or speaking, tend to be centered. Additionally overall balance between signals appearing to come from the left and right channels is important. For these reasons, the mid-range was chosen to be between approximately 200 Hz and 1500 Hz. The low range is thus 0 to 200 Hz and the high range was everything from 1500 Hz to the Nyquist frequency. The filters are complementary to minimize distortion of the spectral content of the mono signal. 
       FIG. 1  illustrates a basic embodiment  100  of this invention in block diagram form. The input mono signal  110  is sampled at 44.1 KHz. Thus the Nyquist frequency was 22.05 KHz. For the experiment described below, input mono signal  110  was a produced by mixing the left and right channels of a stereo recording of a rock tune. 
     Input mono signal  110  is supplied to high-pass filter  121 , mid-range band pass filter  123  and low-pass filter  125 . For this experiment filters  121 ,  123  and  125  were embodied by 1025 tap linear phase finite impulse response (FIR) filters. Shorter, simpler infinite impulse response (IIR) filters could be used to minimize the computational cost. 
     Left channel  130  and right channel  135  result from summation of various delayed and undelayed signals from filters  121 ,  123  and  125 . Left channel  130  receives an undelayed signal from high-pass filter  121 . Right channel  135  receives the signal from high-pass filter  121  delayed by 60 samples, or 0.00136 seconds at the 44.1 KHz sampling frequency. Similarly, right channel  135  receives an undelayed signal from low-pass filter  125  and left channel  130  receives the signal from low-pass filter  125  delayed by 60 samples. This 60 sample delay corresponds approximately to the intra-aural time difference for a sound coming from the right or left. The embodiment of  FIG. 1  applies no other direction clues such as gain difference to minimize the difference between the synthesized stereo signal and the original mono signal. Equal delays were applied to the signal from mid-range band pass filter  123  to left channel  130  and right channel  135 . Thus the mid-range signal arrives at both ears at the same time to correspond to a frontal location. This tends to center both speaking and singing voices. A 30 sample delay was chosen for the mid-range in order to split the difference between the 0 sample and 60 sample delays used elsewhere to minimize the amount of delay the high frequency and low frequency signals have relative to the mid-range signal. These pure delays are summarized in Table 1 below. 
     
       
         
               
               
               
               
             
           
               
                   
                 TABLE 1 
               
               
                   
                   
               
               
                   
                   
                 Left Channel 
                 Right Channel 
               
               
                   
                 Source 
                 130 
                 135 
               
               
                   
                   
               
             
             
               
                   
                 high-pass filter 121 
                  0 samples 
                 60 samples 
               
               
                   
                 mid-range band pass 
                 30 samples 
                 30 samples 
               
               
                   
                 filter 123 
               
               
                   
                 low pass filter 124 
                 60 samples 
                  0 samples 
               
               
                   
                   
               
             
          
         
       
     
     The resulting synthesized stereo signal had a very reasonable stereo effect. The mid-range, including vocals, seemed to come from the front, while the bass seemed to come more from the right and the high frequencies more from the left. The overall quality of the synthesized stereo signal was similar to the original mono signal. The synthesized stereo signal had nothing close to a complete recovery of the stereo input source. For example, all panning effects were lost for voices. 
     If producing a realistic stereo effect is more important than approximating the original mono signal, then another technique can be used. This second embodiment adds an attenuation term the high-pass signal to the right ear to approximate the intra-aural intensity difference (IID) due to the head&#39;s attenuation of sounds from the opposite side. Likewise an attenuation term can be applied to the low-pass signal to the left ear. This attenuation is not as important since the head tends to attenuate higher frequencies more than lower ones. A simple attenuation term is the least computationally expensive, however a low-pass filter could be included to further enhance the simulated attenuation due to the head. This takes advantage of the fact that the head attenuates lower frequencies less than higher frequencies. Such a low-pass filter could be very gentle and thus could be computationally very simple. 
       FIG. 2  illustrates the magnitude response of the right channel according to this second embodiment. Curve  201  is the response of the high-pass filter such as high-pass filter  121 . Curve  202  is the response of the combined IID attenuation low-pass filter. Curve  203  illustrates the combined response for the right channel. 
       FIG. 3  is a block diagram of this second embodiment. Input mono signal  110  is supplied to high-pass filter  121 , mid-range band pass filter  123  and low-pass filter  125  as previously described in conjunction with  FIG. 1 . There are four delay blocks: 30 sample delay  331  receiving the output of mid-range band pass filter  123  and supplying adder  350 ; 60 sample delay  333  receiving the output of high-pass filter  121  and supplying attenuation unit  340 ; 60 sample delay  335  receiving the output of low-pass filter  125  and supplying attenuation unit  345 ; and 30 sample delay  337  receiving the output of mid-range band pass filter  123  and supplying adder  355 . These delay blocks provide the ITD as previously described. Attenuation units  340  and  345  represent attenuations or combined attenuation units and low pass filters used to represent the IID. Attenuation unit  340  provides a larger attenuation than attenuation unit  345 . This difference is related to the difference in high frequency and low frequency attenuation by the head. In addition attenuation unit  345  may be considered optional. 
     Summer  350  sums the direct output of high-pass filter  121 , the output of delay unit  331  and the output of attenuation unit  345 . Summer  355  sums the direct output of low-pass filter  123 , the output of delay unit  337  and the output of attenuation unit  340 . Attenuation units  360  and  365  are optional. These attenuation units if provided balance the resulting left channel output  370  and right channel  375 . 
       FIG. 4  illustrates a block diagram of an example consumer product that might use this invention.  FIG. 4  illustrates a portable compressed digital music system. This portable compressed digital music system includes system-on-chip integrated circuit  400  and external components hard disk drive  421 , keypad  422 , headphones  423 , display  425  and external memory  430 . 
     The compressed digital music system illustrated in  FIG. 4  stores compressed digital music files on hard disk drive  421 . These are recalled in proper order, decompressed and presented to the user via headphones  423 . System-on-chip  400  includes core components: central processing unit (CPU)  402 ; read only memory/erasable programmable read only memory (ROM/EPROM)  403 ; direct memory access (DMA) unit  404 ; analog to digital converter  405 ; system bus  410 ; and digital input  420 . System-on-chip  400  includes peripherals components: hard disk controller  411 ; keypad interface  412 ; dual channel (stereo) digital to analog converter and analog output  413 ; digital signal processor  414 ; and display controller  415 . Central processing unit (CPU)  402  acts as the controller of the system giving the system its character. CPU  402  operates according to programs stored in ROM/EPROM  403 . Read only memory (ROM) is fixed upon manufacture. Suitable programs in ROM include: the user interaction programs that control how the system responds to inputs from keypad  412  and displays information on display  425 ; the manner of fetching and controlling files on hard disk drive  421  and the like. Erasable programmable read only memory (EPROM) may be changed following manufacture even in the hand of the consumer in the field. Suitable programs for storage in EPROM include the compressed data decoding routines. As an example, following purchase the consumer may desire to enable the system to be capable of employing compressed digital data formats different from or in addition to the initially enabled formats. The suitable control program is loaded into EPROM from digital input  420  via system bus  410 . Thereafter it may be used to decode/decompress the additional data format. A typical system may include both ROM and EPROM. 
     Direct memory access (DMA) unit  404  controls data movement throughout the whole system. This primarily includes movement of compressed digital music data from hard disk drive  421  to external system memory  430  and to digital signal processor  414 . Data movement by DMA  404  is controlled by commands from CPU  402 . However, once the commands are transmitted, DMA  404  operates autonomously without intervention by CPU  402 . 
     System bus  410  serves as the backbone of system-on-chip  400 . Major data movement within system-on-chip  400  occurs via system bus  410 . 
     Hard drive controller  411  controls data movement to and from hard drive  421 . Hard drive controller  411  moves data from hard disk drive  421  to system bus  410  under control of DMA  404 . This data movement would enable recall of digital music data from hard drive  421  for decompression and presentation to the user. Hard drive controller  411  moves data from digital input  420  and system bus  410  to hard disk drive  421 . This enables loading digital music data from an external source to hard disk drive  421 . 
     Keypad interface  412  mediates user input from keypad  422 . Keypad  422  typically includes a plurality of momentary contact key switches for user input. Keypad interface  412  senses the condition of these key switches of keypad  422  and signals CPU  402  of the user input. Keypad interface  412  typically encodes the input key in a code that can be read by CPU  402 . Keypad interface  412  may signal a user input by transmitting an interrupt to CPU  402  via an interrupt line (not shown). CPU  402  can then read the input key code and take appropriate action. 
     Dual digital to analog (D/A) converter and analog output  413  receives the decompressed digital music data from digital signal processor  414 . This provides a stereo analog signal to headphones  423  for listening by the user. Digital signal processor  414  receives the compressed digital music data and decompresses this data. There are several known digital music compression techniques. These typically employ similar algorithms. It is therefore possible that digital signal processor  414  can be programmed to decompress music data according to a selected one of plural compression techniques. 
     Display controller  415  controls the display shown to the user via display  425 . Display controller  415  receives data from CPU  402  via system bus  410  to control the display. Display  425  is typically a multiline liquid crystal display (LCD). This display typically shows the title of the currently playing song. It may also be used to aid in the user specifying playlists and the like. 
     External system memory  430  provides the major volatile data storage for the system. This may include the machine state as controlled by CPU  402 . Typically data is recalled from hard disk drive  421  and buffered in external system memory  430  before decompression by digital signal processor  414 . External system memory  430  may also be used to store intermediate results of the decompression. External system memory  430  is typically commodity DRAM or synchronous DRAM. 
     The portable music system illustrated in  FIG. 4  includes components to employ this invention. An analog mono input  401  supplies a signal to analog to digital (A/D) converter  405 . A/D converter  405  supplies this digital data to system bus  410 . DMA  404  controls movement of this data to hard disk  421  via hard disk controller  411 , external system memory  430  or digital signal processor  414 . Digital signal processor is preferably programmed via ROM/EPROM  403  to apply the stereo synthesis of this invention to this digitized mono input. Digital signal processor  414  is particularly adapted to implement the filter functions of this invention for stereo synthesis. Those skilled in the art of digital signal processor system design would know how to program digital signal processor  414  to perform the stereo synthesis process described in conjunction with  FIGS. 1 to 3 . The synthesized stereo signal is supplied to dual D/A converter and analog output  413  for the use of the listener via headphones  423 . Note further that a mono digital signal may be delivered to the portable music player via digital input for storage in hard disk drive  421  or external memory  430  or direct stereo synthesis via digital signal processor  414 . 
     This invention is a method for creating synthetic stereo from a mono signal using intra-aural time differences. This application describes a particular implementation of the general method which produced good results in the sense of having a realistic stereo image. This application also described an alternative embodiment which includes an approximation of intra-aural intensity differences.