-
-
Notifications
You must be signed in to change notification settings - Fork 1.1k
Fix audio resampling functionality #7858
New issue
Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.
By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.
Already on GitHub? Sign in to your account
base: master
Are you sure you want to change the base?
Conversation
AudioResampler
6bb4aed
to
0ede71a
Compare
Appreciate the review 👍 |
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.
Pretty much done |
The function srccpy is used for different audio streams, so the resampler sttate has to be reset on each call to it.
Might be useful in the future, for e.g. exporting audio while resampling
Let me not get ahead of myself. It can be added when its actually going to be used, if ever.
Hey @messmerd, I added a |
Co-authored-by: Dalton Messmer <[email protected]>
There was a problem hiding this comment.
Choose a reason for hiding this comment
The reason will be displayed to describe this comment to others. Learn more.
Works great!
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_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 callingsrc_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 inAudioResampler
and 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
AudioResampler
is first used, which then the delay will be removed on subsequent resampling.