Abstract:
A method of converting a plurality of input signals on first and second converters, such that the first and second converters are both used when the plurality of signals comprises two signals, characterised in that said method comprises:
       selecting more than two input signals;   determining the type of each selected signal;   combining any signals having the same type to form a combined signal;   converting one type of signal with the first converter;   converting a second type of signal with the second converter wherein the first or second type signals is a combined signal.

Description:
FIELD OF THE INVENTION 
       [0001]    This invention relates to a method and an apparatus to improve the digital-to-analog conversion of multi-channel audio signals with different input sample rates, particularly but not exclusively in a cellular phone. 
       BACKGROUND OF THE INVENTION 
       [0002]    With the constant technology improvements, cellular phones comprise many different functions in addition to the common phone function. Thus a user can use his cellular phone to process different functions. For instance, a user can use his phone to take and store pictures, to make and store a video film, to send a text message to another cellular phone, to download mages or music files from an outside device, to register events in a calendar, to listen to mp3 files already downloaded on the memory of the cellular phone from an external device, to listen to the radio, to play to electronic games, etc. In addition, many of these other functions which are supported on a telephone may themselves have devices which support telephone communication. All of these functions are available because the cell phone or other devices comprise many electronic circuits and components that manage these functions. 
         [0003]    The different functions of the cellular phone relate to different kinds of data content for example video data, text data or audio data. The transfer of these kinds of data from an outside device to the memory of the cell phone or from the memory of the cell phone to the user occurs through signals carrying the data so the data can be visualized, read or heard. For instance an audio signal carries audio data. A digital signal or an analog signal can represent such an audio signal. Similarly other devices support these and other types of signal. 
         [0004]    When a user wants to listen to an mp3 file, the digital audio signal or music signal related to mp3 file data stored in a device such a cellular phone must be transformed or a conversion made before the user can hear the data as an analog audio signal. In fact the data are stored in a digital format and the conversion allows transformation of the said audio signal into an analog signal. Thus the user can hear the signal. 
         [0005]    The same conversion occurs when a user receives a phone call from another person. The conversion will convert the digital audio signal coming from another cell phone as soon as this signal reaches the receiving cell phone. In fact the incoming signal is again a digital signal and the user can only hear an analog signal. So the conversion will transform the said digital signal into an analog one. 
         [0006]    For both situations, mp3 listening and voice call listening, the conversion of corresponding digital audio signal occurs through an electronic component such as a digital to analog converter (DAC). A digital audio signal having an mp3 source is defined by a wide frequency bandwidth as a wide band signal. A digital audio signal having a voice source is defined by a narrow frequency bandwidth as a narrow band signal. Both these digital audio signals are also defined by their input sample rate or their sampling frequency. The input sample rate of a digital audio signal is typically two times its frequency bandwidth as defined by the Shannon Whittaker sampling theorem for example. A narrow band signal such as a voice signal has a relatively low input sample rate (below about 16 kHz). A wide band signal such as a music signal has a relatively high input sample rate (about 44.1 kHz for standard mp3 files). The electronic circuit of a mobile phone comprises different DACs in order to process such conversions for different kind of data. Sometimes the user may be listening to an mp3 file and then receives a phone call. In this situation, three DACs will realize the conversion from digital signal to analog signal. As the wide band signal representing the music signal is generally a stereo signal, the conversion into a corresponding analog signal uses two DACs. The narrow band telephone call signal uses one DAC. Moreover, one type of DAC is required for wide band signal and another type of DAC is required for narrow band signal. 
         [0007]    Therefore this kind of process generates an important current consumption due to the amount of circuitry and the constant battle with expanding battery life. Besides time for developing and manufacturing, the process needs two different kinds of DACs. So the whole system of the cell phone including the different kinds of DACS takes much more time than would otherwise be the case. 
         [0008]      FIG. 1  shows a prior art schematic structure of a circuit  100  for a mobile phone. This circuit  100  processes the conversion of digital audio signals into analog audio signals. This circuit comprises three inputs for three signals. Each signal comprises an amplitude which determines the instantaneous intensity or the average intensity of the signal. Each signal also comprises its own frequency which is different from the sampling frequency. Each signal comprises bits that are serial bits. The input  102  relates to a narrow band digital audio signal  104  such as voice signals with a given, usually low, input sample rate. This voice signal  104  relates to a phone call which the user receives on a mobile phone. A voice signal usually comprises a 13-bits or 14-bits coded structure. This means that all 13-bits or 14-bits belong to one signal. The inputs  106  and  108  relate to wide band audio signals  110  and  112 . The sample rate of these signals  110  and  112  differ from the sample rate of the signal  104 . These wide band signals represent a music signal. The combination of these two signals  110  and  112  provides a stereo music signal. A music signal usually comprises a 16-bits (or more) coded structure. This means that 16 bits belong to one signal. A music signal relates for instance to a signal corresponding to an mp3 file already registered on storage means of the mobile phone for instance. In the prior art situations, connection lines  114 ,  116  and  118  are dedicated for each of the three signals  104 ,  110  and  112 . The connection lines each comprise one serial parallel interface or interface module  120  and one digital analog converter (DAC)  122 . The interface  120  transforms all the serial bits into parallel bits. Concerning the voice signal  104 , the interface module  120  transforms the 13-bits or 14-bits signal  104  into 13 signals or 14 signals with a 1-bit coded structure. In the same way, concerning the music signals  110  and  112 , the interface module  120  transforms the 16-bits coded structure into 16 signals with a 1-bit coded structure. This digital analog converter allows the conversion of a digital signal to a corresponding analog signal. The DAC  122  comprises a digital filter  124 , a sigma delta modulator  126 , a D-to-A filter  128  and a smoothing filter. 
         [0009]    In the prior art, U.S. Pat. No. 6,714,825 describes a multi-channel reproducing method in order to convert multi-channel audio sources having different sample rates. This method employs less DACs than the number of incoming channels. However this method requires a specific sampling rate conversion in order to convert all the different signals to obtain the same bandwidth for all the signals. Also this process increases the digital complexity of the circuit. 
         [0010]    It appears that if a user wants to listen simultaneously to voice call signals and music signals on a device such as a mobile phone, solutions exist but they necessitate a costly hardware implementation as described above. A number of different methods have been proposed to overcome the problem of reducing the number of DAC in a mobile device but these solutions are not very efficient. 
         [0011]    An object of the present invention is to provide a method and an apparatus which overcome at least some of the problems associated with the prior art. 
       SUMMARY OF THE INVENTION 
       [0012]    According to one aspect of the present invention there is provided a method and an apparatus as defined in the appended claims. 
         [0013]    One of the advantages of the solution is to reduce the number of DACs to process signals without necessitating any additional complex process for these signals. 
     
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         [0014]    Reference will now be made, by way of example, to the accompanying drawings, in which: 
           [0015]      FIG. 1  shows a schematic architecture of a prior art with three DACs 
           [0016]      FIG. 2  shows a schematic architecture in accordance with one embodiment of the invention, given by way of example; 
           [0017]      FIG. 3  shows a schematic architecture with two DACs for playing the playback of mono voice band stream in accordance with one embodiment of the invention, given by way of example; 
           [0018]      FIG. 4  shows a schematic architecture with two DACs for playing the playback of stereo wide band stream in accordance with one embodiment of the invention, given by way of example; 
           [0019]      FIG. 5  shows a schematic architecture for simultaneously playing voice band and wide band streams in accordance with one embodiment of the invention, given by way of example; 
           [0020]      FIG. 6  shows a schematic diagram of a digital analog converter in accordance with one embodiment of the invention, given by way of example. 
       
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
       [0021]      FIG. 2  shows a circuit which relates to the present invention. As described in prior art the circuit  200  comprises three inputs  202 ,  204  and  206  for three signals  208 ,  210  and  212 . The signal  208  may be an audio digital signal with a first type of sample rate such as a voice signal. The sample rate of such a signal is usually low. The signal  208  is a narrow band signal. This voice signal  208  relates to a phone call which the user receives on a mobile phone. A voice signal usually comprises a 13-bits or 14-bits coded structure. The signals  210  and  212  may be other audio digital signal with a second type of sample rate such as a music signal. The sample rate for this music signal is usually higher than a voice signal. The signals  210  and  212  are wide band signals. The combination of these two signals  210  and  212  represent a stereo music signal. These wide band signals  210  and  212  represent a music signal. A music signal usually comprises a 16-bits (or more) coded structure. This means that 16 bits belong to one signal. A music signal relates for instance to a signal corresponding to an mp3 file already registered on storage means of the mobile phone for instance. The circuit  200  also comprises three corresponding serial parallel interface or interface module  214  for each signal. Differing with the prior art, a multiplexing module  216  is located after the interface modules  214 . The multiplexing module  216  receives each signal coming either from interface modules  214  related to the first input  202  or related to the second and third input  204  and  206  or to all entries  202 ,  204  and  206  in order to pass them to further digital analog converters  224  and  226 . A SPI (Serial Parallel Interface) bus register module  220  passes specific information to the multiplexing module  216 . The SPI bus register module is a module which may be programmed in advance during the phone operation. This SPI bus register module  220  carries out selecting functions and determining functions in order to send specific information to the multiplexing module  216 . This specific information relates to the number of the input signals. The SPI bus register module  220  generates a number equal to one if there is only signal  208  as an input signal, a number equal to two if there are both input signals  210  and  212 ; and a number equal to three if there are input signals  208 ,  210  and  212 . The SPI bus register module  220  also transmits information relating to the type of the input signals i.e. voice type or music type. The SPI bus register module  220  detects the sample rate of each input signal  208 ,  210  or  212 . Thus knowing these both pieces of information concerning the number of the signals and the type of the signals, the multiplexing module  216  is able to pass one or more input signals on one or more corresponding connection lines. Then the multiplexing module  216  determines to which digital analog converters  224  and  226  to send the audio digital signals  208 ,  210 ,  212  using the connection lines  232 ,  234 ,  236 ,  240 . 
         [0022]    Also differing from the prior art, the circuit  200  comprises a combining module  228 . This combining module allows combining both audio digital stereo signals  210  and  212  into an audio digital mono signal  230 . This combining module  228  comprises a first function to add the instantaneous amplitudes of signal  210  and signal  212  and a second function to divide by two the total resulting amplitude in order to avoid an overflow of the component  300  which comprises a digital filter. This overflow relates to a hardware limitation of such a component. The combination of both functions addition and division provides a stereo to mono function. This means that the stereo input signal becomes a mono signal after the combination process. 
         [0023]    From the multiplexing module  216  to the digital analog converters  224  and  226 , the circuit  200  comprises different connection lines. Connection line  232  connects the multiplexing module  216  and the digital analog converter  224 . Connection line  232  refers to the conversion line for the voice signal  208  and also for one of the two stereo signals  210  and  212  as signal  210  for instance. Connection line  234  connects the multiplexing module  216  and the combining module  228 . Connection line  234  refers to the connection line for one of the two stereo signals  210  and  212  as signal  210  for instance. Connection line  236  also connects the multiplexing module  216  and the combining module  228 . Connection line  236  refers to the conversion line for the other of the two stereo signals  210  and  212  as signal  212  for instance. Connection line  238  connects the combining module  228  to the digital to audio converter  226  and refers to the conversion line for the audio combined mono signal  230 . Connection line  240  connects the multiplexing module  216  to the digital to audio converter  226  and refers to the other of the two stereo signals  210  and  212  as for instance signal  212 . 
         [0024]    The use of these different connection lines depends on the number and type of input signals the SPI bus register module  220  sends to the multiplexing module  216 . This will now be explained in more detail. 
         [0025]    Three situations may occur in the circuit  200 . As described in  FIG. 3 , the circuit  200  only processes a mono voice signal  208  to the multiplexing module  216 . Therefore the SPI register module  220  sets the number of digital audio input signals register to one referring to signal  208 . In the same way the SPI register module  200  sets the type of bandwidth to narrow band as the signal  208  is a voice signal. Thus the multiplexing module  216  transmits the signal  208  to the digital analog converter  224  through the connection line  232 . In this situation there is one resulting analog signal  242  representing analog voice signal. 
         [0026]    As described in  FIG. 4 , another situation may occur where the circuit  200  only processes stereo signals  210  and  212  to the multiplexing module  216 . Therefore the SPI register module  220  sets the number of digital audio input signals to two referring to signal  210  and  212 . In the same way, the SPI register module  220  sets the type of bandwidth to wide band as both signals relate to a music signal. As the SPI register module  220  does not select any other signal, the multiplexing module  216  determines that the connection line  232  is available. Thus the multiplexing module  216  transmits signal  210  i.e. one of the two stereo signals to the digital analog converter  224  through the connection line  232 . The multiplexing module  216  sends the other stereo signal  212  to the digital to audio converter  226  through the connection line  240 . In this situation there are two resulting signals  244  and  246  representing analog stereo music signals. 
         [0027]    As described in  FIG. 5 , another situation may occur where the circuit  200  processes three signals  208 ,  210  and  212  to the multiplexing module  216 . Therefore the SPI register module  220  sets the number of digital audio input signals to three referring to signal  208 ,  210  and  212 . In this situation, the SPI register module  220  selects different types of bandwidth. The signal  208  has a narrow bandwidth and signals  210  and  212  have a wide bandwidth. In order to convert simultaneously the three different signals, the multiplexing module  216  transmits in a different way all these three signals. The multiplexing module  216  transmits the voice signal  208  to the digital audio converter  224  through connection line  232 . Simultaneously the multiplexing module transmits the first stereo signal  210  to the combining module  228  through the connection line  234  and the second stereo signal  212  to the combining module  228  through the connection line  236 . The combining module  228  processes both signals  210  and  212  to provide a mono signal  230 . This mono signal uses connection line  238  to reach digital audio converter  226 . In this situation there are two resulting signals,  242  and  248 . The signal  242  represents the analog mono voice signal and the signal  248  represents the analog mono music signal resulting from the digital stereo-to-mono conversion of the signals  210  and  212 . 
         [0028]    Digital analog converters  224  and  226  comprise the same elements. These elements are detailed on  FIG. 6  for DAC  224 . The same description is valid for DAC  226 . In  FIG. 6 , DAC  224  comprises a digital filter  300 , a sigma delta modulator  302 , a D-to-A filter  304  and smoothing filter  306 . The components  300  and  302  process a digital transformation of the signal to be converted. The components  304  and  306  process an analog transformation of the signal. According to situations described in  FIG. 3  and in  FIG. 5 , the different components of the DAC  224  have to be adaptive in order to manage and process both voice signal  208  and music signal  210  according to one of the three above mentioned situations that may occur in the whole circuit  200 . In case of a narrow band signal processing, the different components of the DAC  224  are adapted in order to minimize the power consumption. In case of a wide band signal processing, the different components of the DAC  224  are adapted in order to maximize the audio performances defined as signal-to-noise ratio and total harmonic distortion. 
         [0029]    The process of the combining module  228  as shown in  FIG. 5  will now be described. In the situation described for  FIG. 5 , three signals enter the circuit  200 . The multiplexing module  216  receives these threes signals and then as described above in the description it transmits two digital stereo signals having the same sample rate to the combining module  228 . This combining module  228  processes two transforming functions on the two signals  210  and  212 . The first function is to add both instantaneous amplitudes of the two signals to obtain resulting amplitude. The second function is to divide by two the resulting amplitude. So the amplitude of the resulting signal  230  is an average amplitude from the two signals  210  and  212 . The second function is mandatory to avoid an overflow of the digital filters  300  when both signals  210  and  212  have a full scale amplitude. Additionally the signal  230  is now a mono digital signal. 
         [0030]    It will be appreciated the examples described above are just that. Other alternatives may exist which fall within the scope of the present invention. 
         [0031]    In particular it will be appreciated that this invention can be implemented in software. Also the invention can be adapted to occur with any number of input signals, with the objective of reducing the number of converters, to be less than the number of input signals.