JasonMArcher Ledio Berdellima Ledio Berdellima 1 1 gold badge 5 5 silver badges 10 10 bronze badges. I don't know enough about sound processing to know if that's even near sufficient, but it'll keep you busy until someone comes along who knows their stuff :- You may also need to anti-alias the downsample. Add a comment. Active Oldest Votes. Improve this answer.
Thank you. I don't know anything about filters, but I do need runtime performance, as I'm making a kind of music player. I honestly didn't think it would be this complicated to convert from one sampling rate to a higher one. X-N2O : It's only complicated if you want the results to sound good. If a standalone program suffices, SoX sox. It has sample rate conversion built in. Easy as sox file1 -r [newsamplerate] file2 — mtrw. A few comments, although I'm only guessing at your actual intent: You are up-sampling at a rate times the original sample rate.
For example, if your input was at 10kHz your intermediate cbuf[] would be at MHz which is a tad high for most audio analysis. You are incrementing z twice in the up-sampling loop. This results in all odd elements of cbuf[] to have their original values. I believe this ultimately results in the final buf[] having invalid odd elements which may be the source of your noise. There is also a potential buffer overflow in cbuf if you didn't create it with at least twice the required number of elements.
As mentioned by Steve a linear interpolation is generally the simplest that creates a good result when up-sampling. More complicated up-sampling can be done if desired polynomials, splines, etc Similarly, when down-sampling you may wish to average samples instead of just truncating.
Thank you for pointing that out. Use FFMpeg and avcodec directly. Fippo Fippo 21 1 1 silver badge 3 3 bronze badges. Digitally mixing different sources sampled at different rates will require resampling to a common rate and resolution. Many PC audio cards most notably the 10k1 and 10k2 based Creative Labs ones and AC97 codecs can only input, output or internally process audio data at 48kHz and forcefully resample any digital audio data at one stage or another.
Sometimes the audio software or the drivers will add a resampling step e. ALSA when software mixing, see this thread. Jump to: navigation , search. Resampling or Sample Rate Conversion [ edit ] Digital audio is always sampled, which means that any digital audio file is created with a fixed sample rate and resolution. Understanding Digital Signal Processing.
Category : Signal Processing. Navigation menu Personal tools Create account Log in. The tricky part is figuring out which polyphase filters to apply to which inputs, to calculate the desired outputs, as a function of L and M. Just download and install the ScopeFIR distribution file.
As hinted at above: Determine the cutoff frequency of the decimation filter as explained in Part 2: Decimation. Determine the cutoff frequency of the interpolation filter as explained in Part 3: Interpolation Use the lower of the two cutoff frequencies to design the resampling filter.
You must preserve the Nyquist criteria at each stage or else aliasing will result. That is, no stage can have an output rate which is less than twice the highest frequency of interest.
But why bother with all that? When decimating by a factor of M, you only have to calculate one output for every M decimator inputs. Therefore, if you analyze the spectral content of the sine, be sure to skip past the ramp-up portion. Depending on the resampling factor, resampling can be thought of as a general case of other types of multirate filtering.
It can be: Interpolation: The interpolation factor, L, is greater than one, and the decimation factor, M, is one.
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