Audio signal processing circuit

An oversampling filter oversamples a digital audio signal. A ΔΣ modulator delta-sigma modulates a signal output from the oversampling filter. A D/A converter converts a signal output from the ΔΣ modulator into an analog audio signal. The oversampling filter includes a processor configured to run firmware and a computational algorithm is configurable based on the firmware.

BACKGROUND

1. Technical Field

The present disclosure relates to an audio signal processing circuit.

2. Description of the Related Art

In some cases of audio signal processing, an oversampling filter is inserted upstream of a D/A converter to improve sound quality. The oversampling filter is also referred to as an interpolator, an interpolation filter, or the like and is used to sample an input signal at a frequency substantially higher than the Nyquist frequency. Oversampling is capable of improving S/N ratio and resolution as well as of relaxing requirements for an anti-aliasing filter inserted downstream of the D/A converter.

FIG.1shows graphs illustrating an oversampling process. Oversampling by a factor of K involves inserting K−1 pieces of zeros between samples of an input signal SIN(upsampling).FIG.1shows an example given when K=4. Then, the audio signal after upsampling is passed through a linear phase digital filter to generate an oversampled signal.

The digital filter is a finite impulse response (FIR) filter or an infinite impulse response (IIR) filter and properties of the filter have a significant influence on sound quality. Some conventional audio signal processor ICs allow the selection of a set of coefficients for the FIR filter or allow the loading of a set of coefficients made by a set designer. Accordingly, the set designer is required to design a set of coefficients as a parameter and make sound quality of the audio signal more similar to desired quality.

SUMMARY

The sound quality of audio signals is significantly influenced by a computational algorithm as well as a set of coefficients of a digital filter. However, conventional audio signal processor ICs provide a fixed computational algorithm for a digital filter and leave no room for the set designer to change the computational algorithm, putting a constraint on flexibility in sound quality design. Such a problem was independently recognized by the inventors of the present disclosure.

The present disclosure has been made in view of the above problem.

An embodiment of the present disclosure relates to an audio signal processing circuit. The audio signal processing circuit includes: an oversampling filter to oversample a digital audio signal; a ΔΣ modulator to delta-sigma modulate a signal output from the oversampling filter; and a D/A converter to convert a signal output from the ΔΣ modulator into an analog audio signal. The oversampling filter includes a processor configured to run firmware and a computational algorithm is configurable based on the firmware.

The firmware (software) can be rewritten, and the computational algorithm of a digital filter can be thereby changed. This enables a set designer to make sound quality more similar to desired quality.

The audio signal processing circuit may be capable of loading a plurality of sets of coefficients for the oversampling filter, and the audio signal processing circuit may enable the set of coefficients that is to be used to be selected in response to a register setting.

Each of the sets of the coefficients may be loaded in an encrypted state. The audio signal processing circuit may further include a decoder to decode each of the encrypted sets of coefficients. The sets of coefficients are confidential information containing a wealth of know-how for the set designer. Encrypting the sets of coefficients prevents a malicious third party from secretly looking at the sets of coefficients.

Another embodiment of the present disclosure is also an audio signal processing circuit. The audio signal processing circuit includes: an oversampling filter to oversample a digital audio signal; an interface circuit to receive a set of coefficients for the oversampling filter, the set of coefficients being encrypted; and a decoder to decode the set of coefficients received by the interface circuit and put the decoded set of coefficients in the oversampling filter.

It is to be noted that any arbitrary combination or rearrangement of the above-described structural components and so forth is effective as and encompassed by the present embodiments. Moreover, all of the features described in this summary are not necessarily required by embodiments so that the embodiment may also be a sub-combination of these described features. In addition, embodiments may have other features not described above.

DETAILED DESCRIPTION

A preferred embodiment will now be described with reference to the drawings. Identical reference marks are assigned to identical or equivalent components, members, processes illustrated in the drawings, and the repeated description thereof is omitted as appropriate. The embodiment is an exemplification and should not limit the disclosure. All the features described in the embodiment and a combination thereof are not necessarily essential to the disclosure.

In the present specification, a “state in which a member A is connected to a member B” includes not only a case in which the member A and the member B are physically and directly connected to each other but also a case in which the member A and the member B are indirectly connected to each other through another member that does not have a substantial influence on electrical connection between these members or that does not impair a function or an effect produced by a coupling between these members.

Similarly, a “state in which a member C is disposed between a member A and a member B” includes not only a case in which the members A and C or the members B and C are directly connected to each other but also a case in which the two members are indirectly connected to each other through the other member that does not have a substantial influence on electrical connection between the two members or that does not impair a function or an effect produced by a coupling between the two members.

FIG.2is a block diagram of an audio signal processing circuit100according to an embodiment of the present disclosure. The audio signal processing circuit100primarily includes an oversampling filter110, a ΔΣ modulator120, and a D/A converter130, which are monolithically integrated on a single semiconductor substrate. The audio signal processing circuit100may be, for example, a DAC (D/A converter) chip or a digital signal processor (DSP) that is more advanced than the DAC chip.

The oversampling filter110oversamples a digital audio signal S1. The ΔΣ modulator120delta-sigma modulates a signal S2output from the oversampling filter110. The D/A converter130converts a signal S3output from the ΔΣ modulator120into an analog audio signal S4.

The oversampling filter110includes a processor112and memory114. The processor112is configured to run firmware (a software program) stored in the memory114. The firmware contains written code to implement a function of the oversampling filter110. In other words, a computational algorithm in the oversampling filter110is editable. Read-only memory (ROM) used to store the firmware may be built in or attached to the audio signal processing circuit100or, alternatively, the firmware may be loaded from ROM via an external host processor (a microcontroller).

A configuration of the audio signal processing circuit100has been described above. According to the audio signal processing circuit100, the firmware (software) can be rewritten and the computational algorithm of a digital filter can be thereby changed. This enables a set designer to make sound quality more similar to desired quality.

A key parameter of the digital filter is a set of filter coefficients COEFF. The digital filter is a FIR filter, and the set of coefficients COEFF is understood as impulse response waveforms. In one example, the audio signal processing circuit100is configured to be able to load a set of filter coefficients from outside. The firmware may contain default values of the set of filter coefficients COEFF and may be designed to selectively use either the default values or loaded values.

The audio signal processing circuit100further includes an interface circuit140such as an Inter-Integrated Circuit (I2C) and a memory150. The interface circuit140, which is connected to an external host processor200, receives the set of coefficients COEFF from the host processor200and stores the set of coefficients COEFF in the memory150. The memory150may be a register or may be random-access memory (RAM) if a number of taps is large. As a result, in addition to the computational algorithm, the set of coefficients COEFF can be changed, allowing the set designer to determine sound quality with improved flexibility.

The audio signal processing circuit100is capable of loading a plurality of sets of coefficients COEFF1to COEFFN(N≥2). For instance, when the audio signal processing circuit100starts, the audio signal processing circuit100loads the plurality of the sets of coefficients COEFF1to COEFFNfrom the host processor200and stores the loaded sets of coefficients in the memory150. The host processor200sends selection data SEL to the interface circuit140to specify a set of coefficients COEFF1(1≤i≤N) that is to be actually used and writes the selection data SEL onto the memory150. Out of the plurality of the sets of coefficients COEFF1to COEFFN, one set that corresponds to the selection data SEL written onto the memory150is put in the oversampling filter110.

If only one set of coefficients can be loaded, changing the set of coefficients requires another set of coefficients to be reloaded, causing a substantial delay. In contrast to this, the audio signal processing circuit100inFIG.2enables switching between sets of coefficients simply by sending of the selection signal SEL. This provides improved responsiveness.

FIG.3is a block diagram of an audio signal processing circuit100A according to a modification of the embodiment of the present disclosure. The audio signal processing circuit100A further includes a decoder160.

The set of coefficients COEFF is confidential information containing a wealth of know-how for the set designer. If the set of coefficients COEFF can be loaded, it is feared that a malicious third party could secretly look at communications between the audio signal processing circuit100and the host process200, resulting in a leakage of the set of coefficients COEFF.

The set of coefficients COEFF is sent from the host processor200to the interface circuit140in an encrypted state. The audio signal processing circuit100A includes the decoder160to decode the encrypted set of coefficients COEFF. The set of decoded coefficients COEFF is put in the digital filter of the oversampling filter110.

According to the modification of the embodiment, even if a malicious third party secretly looks at communications between the host process200and the interface circuit140, a leakage of the set of coefficients COEFF can be prevented unless an encryption protocol or a secret key is leaked.

The embodiment is intended only for illustration of the principle and applications of the present disclosure. It should be understood that various modifications or altered arrangements may be made to the embodiment within the scope of the present disclosure as defined by the appended claims.