Difference between revisions of "Arbitrary Sampling Rate Converter in VHDL"
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== The wrong way of upsampling: Linear interpolation == | == The wrong way of upsampling: Linear interpolation == | ||
| − | With a linear interpolation, you look at two consecutive input samples. Take the number of intermediate you want between these two, and gradually change the output | + | With a linear interpolation, you look at two consecutive input samples. Take the number of intermediate samples you want between these two, and gradually change the output value from one to the next sample. Although this can easily be implemented, this is not a good way to interpolate. This is because the original signal is very unlikely to change in a linear fashion, and if you interpolate like this, you introduce a distortion into the output signal. It is better than the copy sample method, but not usable for digital audio. |
Revision as of 01:03, 4 March 2014
| Project: Arbitrary Sampling Rate Converter in VHDL | |
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| Featured: | |
| State | Active |
| Members | Danny Witberg |
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| Description | ASRC's: What, how, why? |
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Contents
Summary
This project describes the theory of an Arbitrary Sampling Rate Converter, and a practical lightweight implementation in VHDL. ASRC's are ofthen used in digital audio, to adapt an input to an output which differ in sampling rates to each other. If sampling rates are not matched, clicking noises and distortion can be heard and it could ruin your signal.
Hardware solutions
Standalone hardware SRC IC's for converting digital audio exist, but they are not cheap. To name some:
- Analog Devices AD1895, AD1896 and AD1893
- Cirrus Logic CS8420, CS8421, and CS8422
- Texas Instruments SRC4190, SRC4382, SRC4392, SRC4190, SRC4184, SRC4194, SRC4190, SRC4192 and SRC4193
- AKM AK4120, AK4121, AK4122, AK4127, AK4128 and AK4129
Theory of operation
Conversion between sampling rates of a digital (audio) signal is all about interpolation. Between the actual input signal and the desired output signal, additional samples must be created, or omitted. There are a number of techniques that can be used for this, but the most effective is to convert or "upsample" the input signal to a very high sampling rate, and then resample that signal to the desired output sampling rate. There are several ways to upsample a signal:
- Copy input samples
- Linear interpolation
- Sinc function interpolarion
The wrong way of upsampling: Copy samples
If you want a higher sampling rate, the easiest way to do this is to resample the input signal at a higher rate, thus creating copies of input samples. However, is is not the way to properly do this, because the interpolated samples do not differ from the actual input samples, and thus not improving any clicking noise and distortion. The output signal will sound just as bad, if not worse, than no conversion at all. Some other manipulation of the signal must be applied to improve quality.
The wrong way of upsampling: Linear interpolation
With a linear interpolation, you look at two consecutive input samples. Take the number of intermediate samples you want between these two, and gradually change the output value from one to the next sample. Although this can easily be implemented, this is not a good way to interpolate. This is because the original signal is very unlikely to change in a linear fashion, and if you interpolate like this, you introduce a distortion into the output signal. It is better than the copy sample method, but not usable for digital audio.
