Fix audio resampling functionality #7858
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sakertooth
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AudioResampler
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…cate buffers unnecessarily
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Appreciate the review 👍 |
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Keeping some of the review unresolved because I consider them to bring up outstanding issues we should eventually resolve, maybe before merge. |
Was not used
… output sample rates
To avoid lifetime issues
Adding a disclaimer that certain plugins may not work well (or at all) with certain sample rates is a better option than doing the resampling directly in LADSPA. Most times it wont be necessary and adds more complexity.
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Pretty much done |
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This PR improves the usage of libsamplerate when resampling.
Changes:
Redirect all libsamplerate usage to
AudioResampler. This ensures we are always using libsamplerate with the same resampling logic, preventing bugs.Call
src_processas many times as necessary to resample all of the source audio we need to fill up the destination buffer. Before, LMMS was only callingsrc_processonce, and assumed that libsamplerate always read all of the input data it gave it, which isn't necessarily true and can cause input frames to be dropped. Any input frames not read in the current iteration are stored within a small array inAudioResamplerand will be used on the next call tosrc_process.Remove the use of buffer margins. This was needed to accommodate for libsamplerate's transport delay, but this involved expensive copies and allocations, and the margin added may not be adequate depending on how long the transport delay needs to be, which wasn't accounted for. To fix this, we allow the transport delay to occur on the onset of when
AudioResampleris first used, which then the delay will be removed on subsequent resampling.