JPH06209296A - Sound digital sampling rate conversion system - Google Patents

Abstract

PURPOSE:To enable sound data conversion with small distortion by providing an interpolation digital filter, a DELTASIGMA modulator, first and second FF and a thinning digital filter. CONSTITUTION:An interpolation digital filter 101 receives the data of digital sound data Di by a clock signal C101 and prepares (n)-fold sound data E101. ADELTASIGMA modulator 102 modulates the data E101 into pulse strings by a clock signal C102 and output a pulse width modulation signal E102. An FF 103 receives the signal E102 by a clock signal 103, holds it and imparts a pulse width modulation signal E103 to an FF 104. The FF 104 receives the signal E103 by a clock signal C104 and holds it. By holding the data E104 held by the FF 104 by a clock C104, pulse width is held as the pulse width performing a phase synchronizing with a necessary second sampling frequency Fso. The data E104 is imparted to a digital filter 105 and sound data Do is generated.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an audio digital sampling rate conversion system, which is particularly required without recognizing the first sampling frequency of an input signal, and without recognizing audibility by minimizing conversion distortion. The present invention relates to an audio digital sampling rate conversion system for converting audio data of a second sampling frequency.

[0002]

2. Description of the Related Art In a conventional audio digital sampling conversion system, an input digital signal is once converted into an analog signal and then converted into a digital signal again at a required second sampling frequency, or with a first sampling frequency of the input signal. Data interpolation is performed at the frequency of the least common multiple of the required second sampling frequency, and the interpolation data is output.
Alternatively, the audio data of the input signal is subjected to high-order data interpolation using DF or the like, and data close to the required data phase of the second sampling frequency is used.

[0003]

The object of the present invention is to use a converter such as an AD converter or a DA converter, not to be aware of the first sampling frequency of the input signal, and to use a ΔΣ modulator. It is an object of the present invention to provide an audio digital sampling rate conversion system capable of converting to audio data with less distortion in order to obtain a required second sampling frequency based on a pulse width modulation signal by using.

[0004]

According to the present invention, in an audio digital sampling rate conversion system for converting input audio data sampled at a first sampling frequency into necessary audio data at a second sampling frequency, An interpolating digital filter that generates n times as much audio data as the input audio data, a ΔΣ modulator that modulates the audio data from the interpolating digital filter into a pulse width modulation signal, and a clock signal that is phase-locked with the input signal A first flip-flop for holding the pulse width modulation signal from the ΔΣ modulator, and a first flip-flop for holding the pulse width modulation signal of the first flip-flop with a clock signal phase-locked to a required second sampling frequency. 2 flip-flops and the pulse width from this second flip-flop Second audio digital sampling modulation system characterized by comprising a decimation digital filter for generating a speech data required from the tone signal.

[0005]

The present invention will be described below with reference to the drawings.

FIG. 1 is a block diagram showing an embodiment of the present invention. As shown in FIG. 1, the audio digital sampling rate conversion system of the present invention comprises an interpolation digital filter 101, a ΔΣ modulator 102, and a flip-flop 10.
3, 104, clock generators 106 and 107, and a thinning digital filter 105.

The interpolation digital filter 101, the ΔΣ modulator 102, and the flip-flop 103 are connected to the input sampling frequency Fsi from the clock generator 106.
Clock signals C101, C102, C1 phase-locked to
03 is given respectively. The interpolation digital filter 101 converts the data of the digital audio data Di sampled at the sampling frequency Fsi into the clock signal C.
101 to generate n times as many voice data. The ΔΣ modulator 102 modulates the n-fold audio data E101 calculated by the interpolation digital filter 101 into a 1-bit (or several-bit) pulse train by the clock signal C102 and outputs a pulse width modulation signal E102.
The flip-flop 103 receives the pulse width modulation signal E102 from the ΔΣ modulator 102 by the clock signal C103, holds the pulse width modulation signal E102, and supplies the pulse width modulation signal E103 to the flip flop 104.

The clock generator 107 has a clock signal C10 phase-locked with the output sampling frequency Fso.
4 and C105 to generate flip-flops 10
4 and the thinned-out digital filter 105. The flip-flop 104 receives and holds the pulse width modulation signal E103 by the clock signal C104. The data E104 held in the flip-flop 104 is the clock C
Since the pulse width is held by 104, the pulse width is held at the pulse width phase-locked with the required second sampling frequency Fso. Flip flop 10
The pulse width modulation signal E104 from 4 is given to the thinning digital filter 105. The decimation digital filter 105 has an integration calculator, receives the pulse width modulation signal E104 by the clock signal C105, and generates the audio data Do sampled at the required second sampling frequency Fso.

[0009]

According to the present invention, a converter such as an AD converter or a DA converter is not used, the first sampling frequency of an input signal is not taken into consideration, and a pulse width modulation signal is used using a ΔΣ modulator. Since the required second sampling frequency is obtained based on the above, it can be converted into audio data with less distortion.

[Brief description of drawings]

FIG. 1 is a block diagram showing a first embodiment of the present invention.

[Explanation of symbols]

 101 Interpolation Digital Filter 102 ΔΣ Modulator 103, 104 Flip-Flop 105 Decimation Digital Filter 106, 107 Clock Generator

Claims (1)

[Claims] 1. An audio digital sampling rate conversion system for converting input audio data sampled at a first sampling frequency into audio data of a required second sampling frequency, wherein the input audio data is n times as large as the input audio data. An interpolation digital filter that generates audio data, a ΔΣ modulator that modulates the audio data from the interpolation digital filter into a pulse width modulation signal, and a pulse width modulation signal from the ΔΣ modulator with a clock signal that is phase-locked with the input signal. Holding a first flip-flop, a second flip-flop holding the pulse width modulation signal of the first flip-flop with a clock signal phase-locked to a second sampling frequency requiring the second flip-flop, and a second flip-flop holding the second flip-flop. Second audio required from pulse width modulated signal from flip-flop Audio digital sampling conversion system characterized by comprising a decimation digital filter for generating a chromatography data.