Redirect all libsamplerate usage to AudioResampler. This ensures we are always using libsamplerate with the same resampling logic, preventing bugs.

Call src_process as many times as necessary to resample all of the source audio we need to fill up the destination buffer. Before, LMMS was only calling src_process once, 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 and will be used on the next call to src_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 AudioResampler is first used, which then the delay will be removed on subsequent resampling.

Remove the choice to choose an interpolation mode when exporting. This can conflict with the interpolation mode specified by the instruments and their APIs, so it was removed.

Redirect all libsamplerate usage to AudioResampler. This ensures we are always using libsamplerate with the same resampling logic, preventing bugs.

Call src_process as many times as necessary to resample all of the source audio we need to fill up the destination buffer. Before, LMMS was only calling src_process once, 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 and will be used on the next call to src_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 AudioResampler is first used, which then the delay will be removed on subsequent resampling.

Remove the choice to choose an interpolation mode when exporting. This can conflict with the interpolation mode specified by the instruments and their APIs, so it was removed.

Redirect all libsamplerate usage to AudioResampler. This ensures we are always using libsamplerate with the same resampling logic, preventing bugs.

Call src_process as many times as necessary to resample all of the source audio we need to fill up the destination buffer. Before, LMMS was only calling src_process once, 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 and will be used on the next call to src_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 AudioResampler is first used, which then the delay will be removed on subsequent resampling.

Remove the choice to choose an interpolation mode when exporting. This can conflict with the interpolation mode specified by the instruments and their APIs, so it was removed.