![]() The 24 and 16 bit to float conversions? Something like float Was the conversion code you were referring to adding to lmms_math.h To mind such a corner case as long as it works up to, say, 192k. In any case 192x8 is a prettyĪbsurd sample rate to render in any case, I think we don't really need IIRC it had some kind of limit for resampling Have you tried other sample rates? Could be that 192x8 is just too bigįor libsrc to handle. Took a quick look but can't really find any fault in the code so far. Oversampling and best interpolation, I get no output frames for oneīuffer per export (with margin = 128). Still when converting 44.1k to 192k at 8x Sample positions by how many frames it actually used (apparentlyĮither 256 or 0 since margin is a multiple of 2), which I believe is I now provide a margin and then increment the I fixed the "note" variable name issues, now use stack arrays asīuffers, and fixed the resampling issues (again). I would expect input_frames_used ~= input_frames - margin, but that is not the case even for enormous margins. when oversampling by 8x, and then input_frames_used = 0 since the margin is too small). In the link below (at the end) Erik Lopo says input_frames_used < input_frames for the case I'm describing where you provide more samples than needed, but then Jean-Yves Avenard says input_frames_used = input_frames for this case, as I'm seeing (except in certain circumstances, e.g. This may be related to why input_frames_used is always the number of frames provided. It's as if the libsamplerate ratio also depends on the number of new samples I'm providing. I was providing an extra margin and saving that margin to use at the beginning of the next period, so I always ended up rendering this same number of new samples (except for the first period). But, on second thought, I'm not sure why this appeared to also fix the issue in my code that provides more samples than needed. The ratio is (double)processRate/sample'sRate. Rounding the number of samples to input to src_process, not the ratio. You'll probably want to look at AudioFilePlayer and SampleBuffer to see Have that bit of an extra margin when resampling. Larger chunk from the sample than you need for the period, so you'll Well, with samples, you can sort of cheat. The file is one sample rate and we need the output to be another yetĬan't you get libgig to do the resampling for you? Or does it only do I'm not quite sure how to get around this resampling since the note in IIRC the sf2player only resamples if Fluidsynth is unable to output the Which seems to be how the SF2 player is doing it (unless I'm reading Sample rate so I was using using src_process (libsamplerate) toĬonvert the rendered buffer (44.1k) to the mixer's sample rate (48k), If I run Jack at 48k or try exporting it as 48k, thenĮngine::mixer()->processingSampleRate() is different than the sample's But, I'm aįor example with this piano, the samples in the GIG file are at 44.1 Yes, I can correct the coding style and rename those files. So even the solutions to the problem contain problems. Somewhat complex to implement (requires ringbuffers and such). This is also problematic because weĭon't have any kind of latency compensation mechanism, and is also However problematic because it produces poor sound quality.Ģ- add enough latency to the resampled stream so we can look aheadĮnough to use proper interpolation. Interpolation - as it doesn't require almost any lookahead. The only way to avoid these problems is either:ġ- use a very simple, 1st-order resampling algorithm, such as linear Inevitably cause some clicks and disturbances in every period. We getĪway with resampling samples in the AFP because that doesn't have to beĭone in realtime (we can always look up enough frames from the audioįile), but if you don't have any lookahead, then it's going to Resampling algorithm requires some lookahead to work properly. Resampling audio in realtime is problematic, because every good I second this - better to avoid resampling. LMMS can also connect to any of your MIDI devices and allow you to perform live, all in a user-friendly and modern interface.Concerning the resampling: couldn't you try to render the data at theĬurrent processing sample rate and try to avoid resampling at all?Ĭonnect to Mixer's sampleRateChanged() signals if neccessary. This includes the creation of melodies and beats, the synthesis and mixing of sounds, and arranging of samples. LMMS is a free cross-platform alternative to commercial program like FL Studio, which allow you to produce music with your computer.
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