Re: [music-dsp] WSOLA on RealTime
Hi, Could you tell me how to use soundTouch API if I want only to do resampling ? I mean: convert from 48Khz to 8Khz or 8Khz to 48Khz I guess that setRate with SETTING_USE_AA_FILTER. What length should put ? I talk about https://gitlab.com/soundtouch/soundtouch/tree/master/source/SoundTouch Thanks, Alex בתאריך יום ב׳, 1 באוק׳ 2018 ב-19:46 מאת Nuno Santos < nunosan...@imaginando.pt>: > Hey Alex, > > Have you seen this article before? > > https://www.surina.net/article/time-and-pitch-scaling.html > > It has a complete and detailed description on how time and pitch scaling > works. > > Hope it helps! > > Best regards, > > Nuno > > On 1 Oct 2018, at 17:30, Alex Dashevski wrote: > > Hi, > > I got this answer from Robert: > WSOLA fundamentally does time-scaling. time compression or time > stretching without changing pitch. time-scaling is not normally thought of > as real-time because your input and output buffer pointers will collide. > > > My question is: > Why buffers will collide ? > > Thanks, > Alex > ___ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp > > > ___ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Resampling
I have only used libraries for resampling myself. I haven't looked at their source, but it's available. The two libraries I'm aware of are at http://www.mega-nerd.com/SRC/download.html and https://kokkinizita.linuxaudio.org/linuxaudio/zita-resampler/resampler.html perhaps they can give you some insight. On Wed, Oct 3, 2018 at 2:46 PM Alex Dashevski wrote: > I wrote on android ndk and there is fastpath concept. Thus, I think that > resampling can help me. > Can you recommend me code example ? > Can you give me an example of resampling ? for example from 48Khz to 8Khz > and 8Khz to 48Khz. > I found this: > https://dspguru.com/dsp/faqs/multirate/resampling/ > but it is not enough clear for me, > > Thanks, > Alex > > > בתאריך יום ד׳, 3 באוק׳ 2018 ב-20:56 מאת Spencer Jackson < > ssjackso...@gmail.com>: > >> >> >> On Wed, Oct 3, 2018 at 3:17 AM Alex Dashevski wrote: >> >>> >>> if I do resampling before and after processing. for example, 48Khz -> >>> 8Khz and then 8Khz -> 48Khz then will it help ? >>> >> >> Lowering sample rate can help achieve lower latencies by giving you fewer >> samples to process in the same amount of time but just downsampling and >> then upsampling back doesn't really have any effect. >> >> >> >>> I don't understand why I need filter, This is to prevent alias but I >>> can't understand why ? >>> >>> Technically you only need a filter if your signal has information above >> the nyquist frequency of the lowest rate but this is not usually the case. >> I think wikipedia explains aliasing pretty well: >> https://en.wikipedia.org/wiki/Aliasing#Sampling_sinusoidal_functions . >> Once the high frequency information aliases it cannot be recovered by >> resampling back to the higher rate and your lower band information is now >> mixed in with the aliased information. The filter removes this high >> freqency data so that the low band stays clean through the whole process. >> >> >> Is there option to decrease latency or delay ? >>> >> >> The only way to reduce latency in your algorithm (unless there is some >> error in the implementation) is to reduce the block size, so you process >> 128 samples rather than 240. 240 isn't a very large amount of latency for a >> pitch shifter which is typically a CPU intensive process and therefore most >> implementations have relatively high latencies. >> >> I'm not sure I understand what you mean by the pitch duration requiring a >> buffer-resize or sample-rate decrease. WSOLA creates a signal with more >> samples than the input, you must resample that (usually by a non-integer >> amount) to make it the correct number of samples then output that, and >> reload your buffer with the next block of input data. Please clarify if you >> mean some other issue. >> >> _Spencer >> ___ >> dupswapdrop: music-dsp mailing list >> music-dsp@music.columbia.edu >> https://lists.columbia.edu/mailman/listinfo/music-dsp > > ___ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Resampling
I wrote on android ndk and there is fastpath concept. Thus, I think that resampling can help me. Can you recommend me code example ? Can you give me an example of resampling ? for example from 48Khz to 8Khz and 8Khz to 48Khz. I found this: https://dspguru.com/dsp/faqs/multirate/resampling/ but it is not enough clear for me, Thanks, Alex בתאריך יום ד׳, 3 באוק׳ 2018 ב-20:56 מאת Spencer Jackson < ssjackso...@gmail.com>: > > > On Wed, Oct 3, 2018 at 3:17 AM Alex Dashevski wrote: > >> >> if I do resampling before and after processing. for example, 48Khz -> >> 8Khz and then 8Khz -> 48Khz then will it help ? >> > > Lowering sample rate can help achieve lower latencies by giving you fewer > samples to process in the same amount of time but just downsampling and > then upsampling back doesn't really have any effect. > > > >> I don't understand why I need filter, This is to prevent alias but I >> can't understand why ? >> >> Technically you only need a filter if your signal has information above > the nyquist frequency of the lowest rate but this is not usually the case. > I think wikipedia explains aliasing pretty well: > https://en.wikipedia.org/wiki/Aliasing#Sampling_sinusoidal_functions . > Once the high frequency information aliases it cannot be recovered by > resampling back to the higher rate and your lower band information is now > mixed in with the aliased information. The filter removes this high > freqency data so that the low band stays clean through the whole process. > > > Is there option to decrease latency or delay ? >> > > The only way to reduce latency in your algorithm (unless there is some > error in the implementation) is to reduce the block size, so you process > 128 samples rather than 240. 240 isn't a very large amount of latency for a > pitch shifter which is typically a CPU intensive process and therefore most > implementations have relatively high latencies. > > I'm not sure I understand what you mean by the pitch duration requiring a > buffer-resize or sample-rate decrease. WSOLA creates a signal with more > samples than the input, you must resample that (usually by a non-integer > amount) to make it the correct number of samples then output that, and > reload your buffer with the next block of input data. Please clarify if you > mean some other issue. > > _Spencer > ___ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Resampling
On Wed, Oct 3, 2018 at 3:17 AM Alex Dashevski wrote: > > if I do resampling before and after processing. for example, 48Khz -> 8Khz > and then 8Khz -> 48Khz then will it help ? > Lowering sample rate can help achieve lower latencies by giving you fewer samples to process in the same amount of time but just downsampling and then upsampling back doesn't really have any effect. > I don't understand why I need filter, This is to prevent alias but I can't > understand why ? > > Technically you only need a filter if your signal has information above the nyquist frequency of the lowest rate but this is not usually the case. I think wikipedia explains aliasing pretty well: https://en.wikipedia.org/wiki/Aliasing#Sampling_sinusoidal_functions . Once the high frequency information aliases it cannot be recovered by resampling back to the higher rate and your lower band information is now mixed in with the aliased information. The filter removes this high freqency data so that the low band stays clean through the whole process. Is there option to decrease latency or delay ? > The only way to reduce latency in your algorithm (unless there is some error in the implementation) is to reduce the block size, so you process 128 samples rather than 240. 240 isn't a very large amount of latency for a pitch shifter which is typically a CPU intensive process and therefore most implementations have relatively high latencies. I'm not sure I understand what you mean by the pitch duration requiring a buffer-resize or sample-rate decrease. WSOLA creates a signal with more samples than the input, you must resample that (usually by a non-integer amount) to make it the correct number of samples then output that, and reload your buffer with the next block of input data. Please clarify if you mean some other issue. _Spencer ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] Resampling
Hi, I use a sample rate :48Khz and buffer size = 240 samples. I made pitch shifting with WSOLA and resampling. But pitch duration is ~20ms then I need decrease rate sample or increase buffer size. As a result of it, the delay will increase. if I do resampling before and after processing. for example, 48Khz -> 8Khz and then 8Khz -> 48Khz then will it help ? If so, how should do ? I don't understand why I need filter, This is to prevent alias but I can't understand why ? Is there option to decrease latency or delay ? Thanks, Alex ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
