Low-complexity sampling rate conversion method and apparatus for audio processing
View Patent ↗A low-complexity sampling rate conversion (SRC) method and apparatus for the processing of digital audio signals. A first stage upsamples an input audio signal to generate an upsampled audio signal. For example, the first stage may perform 1:2 upsampling using a halfband filter. A second stage re-samples the upsampled audio signal from the first stage at a target sampling rate. For example, re-sampling may be achieved using linear interpolation.
1. A method for converting a digital audio signal from a first sampling rate to a second sampling rate, comprising:
upsampling the audio signal by a factor that is unrelated to either the first sampling rate or the second sampling rate to generate an upsampled audio signal; and
re-sampling the upsampled audio signal at the second sampling rate using interpolation.
2. The method of claim 1 , wherein the upsampling comprises upsampling the audio signal by a factor of two.
3. The method of claim 2 , wherein the upsampling the audio signal by a factor of two comprises:
zero padding the audio signal to generate a zero-padded audio signal; and
lowpass filtering the zero-padded signal.
4. The method of claim 3 , wherein the zero padding the audio signal comprises performing 1:2 zero padding of the audio signal and wherein the lowpass filtering the zero-padded signal comprises halfband filtering the zero-padded signal.
5. The method of claim 2 , wherein the upsampling the audio signal by a factor of two comprises:
generating first and second samples corresponding to a sample in the audio signal, wherein the first sample is a copy of the audio signal.
6. The method of claim 5 , wherein the second sample is determined by multiplying a filter coefficient sequence by an audio signal sequence.
7. The method of claim 5 , wherein the second sample is determined based on the equation:
x
u
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n
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=
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k
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1
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3
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5
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7
h
[
k
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z
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z
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wherein x u [n] is the second sample corresponding to the audio signal sample, x z [n+k] and x z [n−k] are samples from the audio signal, and h[k] are filter coefficients.
8. The method of claim 1 , wherein the re-sampling the upsampled audio signal at the second sampling rate using interpolation comprises using at least one of linear interpolation, quadratic interpolation, cubic interpolation, and spline interpolation.
9. The method of claim 1 , wherein the first sampling rate is greater than the second sampling rate.
10. The method of claim 1 , wherein the second sampling rate is greater than the first sampling rate.
11. The method of claim 1 , wherein the upsampling the audio signal to generate an upsampled audio signal comprises performing 1:3 upsampling.
12. The method of claim 1 , wherein the upsampling the audio signal to generate an upsampled audio signal comprises using a lowpass filter.
13. The method of claim 12 , wherein the lowpass filter comprises a third-band filter.
14. The method of claim 1 , wherein the upsampling the audio signal to generate an upsampled audio signal comprises using a bandpass filter.
15. The method of claim 1 , wherein the upsampling the audio signal to generate an upsampled audio signal comprises using a filter that substantially eliminates stop-band leakage.
16. A sampling rate converter for converting a digital audio signal from a first sampling rate to a second sampling rate, comprising:
a first stage configured to upsample the audio signal by a factor that is unrelated to either the first sampling rate or the second sampling rate to generate an upsampled audio signal; and
a second stage configured to re-sample the upsampled audio signal at the second sampling rate using interpolation.
17. The sampling rate converter of claim 16 , wherein the first stage is configured to upsample the audio signal by a factor of two.
18. The sampling rate converter of claim 17 , wherein the first stage is configured to zero pad the audio signal to generate a zero-padded audio signal and to lowpass filter the zero-padded signal.
19. The sampling rate converter of claim 18 , wherein the first stage is configured to perform 1:2 zero padding of the audio signal and to lowpass filter the zero-padded signal using a halfband filter.
20. The sampling rate converter of claim 17 , wherein the first stage is configured to upsample the audio signal by a factor of two by generating first and second samples corresponding to a sample in the audio signal, wherein the first sample is a copy of the audio signal.
21. The sampling rate converter of claim 20 , wherein the first stage generates the second sample by multiplying a filter coefficient sequence by an audio signal sequence.
22. The sampling rate converter of claim 20 , wherein the first stage generates the second sample based on the equation:
x
u
[
n
]
=
∑
k
=
1
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3
,
5
,
7
h
[
k
]
(
x
z
[
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+
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+
x
z
[
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-
k
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)
wherein x u [n] is the second sample corresponding to the audio signal sample, x z [n+k] and x z [n−k] are samples from the audio signal, and h[k] are filter coefficients.
23. The sampling rate converter of claim 16 , wherein the second stage is configured to re-sample the upsampled audio signal at the second sampling rate using at least one of linear interpolation, quadratic interpolation, cubic interpolation, and spline interpolation.
24. The sampling rate converter of claim 16 , wherein the first sampling rate is greater than the second sampling rate.
25. The sampling rate converter of claim 16 , wherein the second sampling rate is greater than the first sampling rate.
26. The sampling rate converter of claim 16 , wherein the first stage is configured to generate an upsampled audio signal by performing 1:3 upsampling.
27. The sampling rate converter of claim 16 , wherein the first stage is configured to generate an upsampled audio signal using a lowpass filter.
28. The sampling rate converter of claim 27 , wherein the lowpass filter comprises a third-band filter.
29. The sampling rate converter of claim 16 , wherein the first stage is configured to generate an upsampled audio signal using a bandpass filter.
30. The sampling rate converter of claim 16 , wherein the first stage is configured to generate an upsampled audio signal using a filter that substantially eliminates stop-band leakage.
31. A computer program product comprising a computer usable medium having computer readable program code stored therein for causing a computer to convert a digital audio signal from a first sampling rate to a second sampling rate, the computer readable program code comprising:
first computer readable program code for upsampling the audio signal by a factor that is unrelated to either the first sampling rate or the second sampling rate to generate an upsampled audio signal; and
second computer readable program code for re-sampling the upsampled audio signal at the second sampling rate using interpolation.
32. The computer program product of claim 31 , wherein the first computer readable program code comprises code for upsampling the audio signal by a factor of two.
33. The computer program product of claim 32 , wherein the first computer readable program code further comprises:
code for zero padding the audio signal to generate a zero-padded audio signal; and
code for lowpass filtering the zero-padded signal.
34. The computer program product of claim 33 , wherein the code for zero padding the audio signal comprises code for performing 1:2 zero padding of the audio signal and wherein the code for lowpass filtering the zero-padded signal comprises code for halfband filtering the zero-padded signal.
35. The computer program product of claim 32 , wherein the code for upsampling the audio signal by a factor of two comprises:
code for generating first and second samples corresponding to a sample in the audio signal, wherein the first sample is a copy of the audio signal.
36. The computer program product of claim 35 , wherein the second sample is determined by multiplying a filter coefficient sequence by an audio signal sequence.
37. The computer program product of claim 35 , wherein the second sample is determined based on the equation:
x
u
[
n
]
=
∑
k
=
1
,
3
,
5
,
7
h
[
k
]
(
x
z
[
n
+
k
]
+
x
z
[
n
-
k
]
)
wherein x u [n] is the second sample corresponding to the audio signal sample, x z [n+k] and x z [n−k] are samples from the audio signal, and h[k] are filter coefficients.
38. The computer program product of claim 31 , wherein the second computer readable program code comprises code for re-sampling the upsampled audio signal at the second sampling rate using at least one of linear interpolation, quadratic interpolation, cubic interpolation, and spline interpolation.
39. The computer program product of claim 31 , wherein the first sampling rate is greater than the second sampling rate.
40. The computer program product of claim 31 , wherein the second sampling rate is greater than the first sampling rate.
41. The computer program product of claim 31 , wherein the first computer readable program code for upsampling the audio signal comprises code for performing 1:3 upsampling.
42. The computer program product of claim 31 , wherein the first computer readable program code for upsampling the audio signal comprises code for lowpass filtering the audio signal.
43. The computer program product of claim 42 , wherein the code for lowpass filtering the audio signal comprises code for third-band filtering the audio signal.
44. The computer program product of claim 31 , wherein the first computer readable program code for upsampling the audio signal comprises code for bandpass filtering the audio signal.
45. The computer program product of claim 31 , wherein the first computer readable program code for upsampling the audio signal comprises code for filtering to substantially eliminate stop-band leakage.