Abstract:
A digital-to-analog converting (DAC) circuit is utilized for converting a 1-bit stream into an analog output signal. The DAC includes an N-bit encoder, a multiplexer, a low-pass filter, and a digital-to-analog conversion circuit. The N-bit encoder is utilized for receiving the 1-bit stream and encoding the 1-bit stream to generate an N-bit stream, where N is larger than 1; the multiplexer is utilized for selectively outputting the N-bit stream or a zero signal as an output signal according to a selection signal; the low-pass filter is utilized to generate a filtered output signal according to the output signal; and the digital to analog conversion circuit is utilized to generate the analog output signal according to the filtered output signal.

Description:
BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The present invention relates to a digital-to-analog converter, and more particularly, to a digital-to-analog converter for reducing pop noise and harmonic tone and related converting method. 
     2. Description of the Prior Art 
     In audio processing, an over-sampling 1-bit sigma-delta modulation signal is generally used for data transmission. When using this data transmission technique, however, a “pop” will be heard due to the occurrence of discontinuous points on the waveform of the audio signal (e.g., starting/finishing displaying the audio signal), causing discomfort to the listener. In addition, when finishing displaying the audio signal and under the condition that the audio signal has a lower sampling rate, if the last audio data has a higher period (i.e., more taps of a low-pass filter in a digital-to-analog converter are used) and is still in a display loop, unexpected harmonic tones are produced and a frequency of a fundamental harmonic is decreased. The frequencies of the harmonic tones are shown as follows: 
                 f   tone     =         f   s       N   tap       ×   n       ,     n   =   1     ,   2   ,     3   ⁢   …     ⁢           ,         
where f tone  is the frequency of the harmonic tone, f s  is an over-sampling rate of the audio signal, N tap  is a number of the taps which the low-pass filter uses.
 
     When the lowest frequency of the harmonic tones is reduced, even if the frequency of the fundamental harmonic is not within an audio band (generally defined within 20 Hz-20 kHz), the post-processing of the audio signal may be influenced when the fundamental harmonic is within 20 kHz-100 kHz. 
     SUMMARY OF THE INVENTION 
     It is therefore an objective of the present invention to provide a digital-to-analog converter for reducing the pop noise and the unexpected harmonic tones. 
     According to one embodiment of the present invention, an apparatus for converting a digital audio signal into an analog audio signal comprises a first circuit, a multiplexer, a low-pass filter and a digital-to-analog converter. The first circuit is configured to receiving the digital audio signal and converting the digital audio signal into an N-bit sigma-delta modulation signal, wherein N is larger than 1; The multiplexer is utilized for receiving the N-bit sigma-delta modulation signal and a zero signal, and selectively outputting the N-bit sigma-delta modulation signal or the zero signal as an output signal according to a selection signal. The low-pass filter is utilized for generating a filtered output signal according to the output signal. The digital-to-analog converter is utilized for generating the analog audio signal according to the filtered output signal. 
     According to one embodiment of the present invention, a digital-to-analog converting circuit for converting a 1-bit stream into an analog output signal is disclosed. The digital-to-analog converter includes an N-bit encoder, a multiplexer, a low-pass filter, and a digital-to-analog conversion circuit. The N-bit encoder is utilized for receiving the 1-bit stream and encoding the 1-bit stream to generate an N-bit stream, where N is larger than 1; the multiplexer is utilized for selectively outputting the N-bit stream or a zero signal as an output signal according to a selection signal; the low-pass filter is utilized to generate a filtered output signal according to the output signal; and the digital to analog conversion circuit is utilized to generate the analog output signal according to the filtered output signal. 
     According to one embodiment of the present invention, a method for processing a digital audio signal to generate an analog signal is disclosed. The method comprises: processing the digital audio signal to generate an N-bit stream; selectively outputting the N-bit stream or a zero signal as an output signal according to a selection signal; filtering the output signal to generate a filtered output signal; and converting the filtered output signal into the analog output signal. 
     These and other objectives of the present invention will no doubt become obvious to those of ordinary skill in the art after reading the following detailed description of the preferred embodiment that is illustrated in the various figures and drawings. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  is a digital-to-analog converter according to a first embodiment of the present invention. 
         FIG. 2  is a digital-to-analog converter according to a second embodiment of the present invention. 
     
    
    
     DETAILED DESCRIPTION 
     To prevent the above-mentioned “pop” issue from occurring when finishing playing the audio signal, the present invention provides a digital-to-analog converter in which, when finishing playing the audio signal, an exactly zero signal is inputted into a low-pass filter of the digital-to-analog converter. Therefore, when finishing playing the audio signal, discontinuous points on the waveform of the audio signal are smoothed, and the pop issue is reduced. In the conventional 1-bit sigma-delta modulation signal, a digital signal “0” represents a value “−1” and a digital signal “1” represents a value “+1”. Therefore, the conventional 1-bit sigma-delta modulation signal cannot output an exact zero. In the present invention, a 1-bit sigma-delta modulation signal is encoded into a 2-bit sigma-delta modulation (SDM) signal, and the 2-bit sigma-delta modulation signal can represent four values at least including the values “0”, “+1”, “−1”. Therefore, the digital-to-analog converter can output an exactly zero signal when finishing playing the audio signal, to solve the “pop” issue. 
     Please refer to  FIG. 1 .  FIG. 1  is a digital-to-analog converter according to a first embodiment of the present invention. As shown in  FIG. 1 , the digital-to-analog converter  100  includes a circuit  105 , a multiplexer  130 , a low-pass filter  140 , and a digital-to-analog conversion circuit  150 . In an embodiment, the circuit  105  includes a First-In-First-Out (FIFO) unit  110 , a 2-bit encoder  120 . The multiplexer  130  is controlled by the selection signal corresponding to the data amount of the audio signal in the circuit  105 . The operations of the digital-to-analog converter  100  are described in the following. 
     In the embodiment, the circuit  105  is used for generating 2-bit sigma-delta modulation signal SDM 2-bit . In an embodiment, the circuit  105  includes a FIFO unit  110  and a 2-bit encoder  120 . First, a 1-bit sigma-delta modulation signal SDM 1-bit  is inputted into the FIFO unit  110  of the circuit  105 . Then, the 2-bit encoder  120  of the circuit  105  encodes the 1-bit sigma-delta modulation signal SDM 1-bit  to generate a 2-bit sigma-delta modulation signal SDM 2-bit . In this embodiment, the digital signals “0” and “1” in the 1-bit sigma-delta modulation signal SDM 1-bit  are respectively encoded as “11” (representing the value “−1”) and “01” (representing the value “+1”). Additionally, the value “0” is encoded as a digital signal “00”. In another embodiment, the 2-bit encoder  120  receives 1-bit SDM signal and converters the 1-bit SDM signal into the 2-bit SDM signal. The FIFO  110  temporarily stores the 2-bit SDM signal from the 2-bit encoder  120  and outputs the 2-bit SDM signal to the multiplexer  130 . The multiplexer  130  selectively outputs the 2-bit sigma-delta modulation signal SDM 2-bit  or the exactly zero signal S zero  as an output signal S out , where, when the FIFO unit  110  is not empty (i.e. the audio signal is continuously played), the multiplexer  130  outputs the 2-bit sigma-delta modulation signal SDM 2-bit  according to a selection signal Emp; and when the FIFO unit  110  is empty (i.e. the audio signal is stopped playing), the multiplexer  130  outputs the exactly zero signal S zero  according to the selection signal Emp. That is, the multiplexer  130  outputs the 2-bit sigma-delta modulation signal SDM 2-bit  into the low-pass filter  140  when the audio signal is playing and the multiplexer  130  outputs the exactly zero signal S zero  into the low-pass filter  140  when the audio signal is stopped playing. Then, the low-pass filter  140  generates an m-bit pulse-code modulation (PCM) signal S PCM  according to the output signal S out . In one embodiment, the low-pass filter  140  can be implemented by a finite impulse response (FIR) digital filter which can be implemented by delay units, multipliers and adders, or implemented by multiplexers or a look-up table. Finally, the digital-to-analog conversion circuit  150  generates an analog output signal S ou   _   ana  according to the PCM signal S PCM . 
     In this embodiment, when the audio signal is stopped playing such that the FIFO unit  110  is empty, the multiplexer  130  outputs the exactly zero signal S zero  into the low-pass filter  140  according to the selection signal Emp from the FIFO unit  110 . After filtering by the low-pass filter  140 , the filtered output signal (i.e., PCM signal S PCM ) gradually approaches a zero point. Therefore, discontinuous points on the waveform of the output audio signal (filtered output signal) can be smoothed to prevent the occurrence of the pop. In addition, because the exactly zero signal S zero  is encoded as the digital signal “00”, the output of the low-pass filter  140  will be zero after a period (N tap *f s ), harmonic tones will not be produced and the post-processing of the audio signal will therefore not be influenced. 
     The digital-to-analog converter  100  can apply to a direct stream digital (DSD) technique of a super audio compact disc (SACD), where the DSD techniques provide a frequency response up to 100 kHz.  FIG. 2  is a digital-to-analog converter  200  according to a second embodiment of the present invention. As shown in  FIG. 2 , the digital-to-analog converter  200  includes a circuit  205 , a multiplexer  230 , a low-pass filter  240 , a volume control unit  250 , a modulator  260 , and a digital-to-analog conversion circuit  270 . The circuit  205  comprises a FIFO unit  210 , a 2-bit encoder  220 . The operations of the digital-to-analog converter  200  are described in the following. 
     First, audio data undergoes an analog-to-digital conversion and is over-sampled at a sampling rate that is 64 times a sampling rate of the compact disc (i.e., sampling rate f s =64×44.1 kHz) to generate a 1-bit sigma-delta modulation signal SDM 1-bit . Then, the 1-bit sigma-delta modulation signal SDM 1-bit  is inputted into the FIFO unit  210  of the circuit  205 . The 2-bit encoder  220  of the circuit  205  then encodes the 1-bit sigma-delta modulation signal SDM 1-bit  to generate a 2-bit sigma-delta modulation signal SDM 2-bit . The multiplexer  230  selectively outputs the 2-bit sigma-delta modulation signal SDM 2-bit  or the exactly zero signal S zero  as an output signal S out , where, when the FIFO unit  210  is not empty (i.e., the audio signal is continuously played), the multiplexer  230  outputs the 2-bit sigma-delta modulation signal SDM2-bit according to a selection signal Emp; and when the FIFO unit  210  is empty (i.e., the audio signal is stopped playing), the multiplexer  230  outputs the exactly zero signal S zero  according to the selection signal Emp. That is, the multiplexer  230  outputs the 2-bit sigma-delta modulation signal SDM 2-bit  into the low-pass filter  240  when the audio signal is playing and the multiplexer  230  outputs the exactly zero signal S zero  into the low-pass filter  240  when the audio signal is stopped playing Then, the low-pass filter  240  generates an m-bit PCM signal S PCM  according to the output signal S out . In this embodiment, the low-pass filter  240  can be implemented by a FIR digital filter which can be implemented by delay units, multipliers and adders, or implemented by multiplexers or a look-up table. In addition, the volume control unit  250  is used to adjust the PCM signal S PCM  to generate an adjusted output signal S VC . Then, the modulator  260  modulates the adjusted output signal S VC  to generate a 4-bit modulated output signal S m . The digital-to-analog conversion circuit  270  generates an analog output signal S out   _   ana  according to the modulated output signal S m . Additionally, the low-pass filter  240  and the volume control unit  250  can also be implemented by software. 
     Those skilled in the art will readily observe that numerous modifications and alterations of the device and method may be made while retaining the teachings of the invention.