After 91eb50ec2f, we would get random crashes/asserts from the pulseaudio library like:
`Assertion 'e->mainloop->n_enabled_defer_events > 0' failed at ../pulseaudio-17.0/src/pulse/mainloop.c:261, function mainloop_defer_enable(). Aborting.`
It seems like we would run into a race condition if we didn't guard the
pulseaudio flush command with the pulseaudio mutex.
This is probably because the pulseaudio thread would try to read the
buffer for a tiny bit even after pausing the playback.
Sadly the only way to reproduce this is to playback any scene (seem to happen more often if A/V sync is on) and spam play/pause.
Note that I could not reproduce this on every computer I tested this on.
But by expanding the main pulseaudio mutex lock, I can't seem to reproduce this anymore.
So I think that is the correct solution.
Pull Request: https://projects.blender.org/blender/blender/pulls/120072
In ffmpeg 5.0, several variables were made const to try to prevent bad API usage.
Removed some dead code that wasn't used anymore as well.
Reviewed By: Richard Antalik
Differential Revision: http://developer.blender.org/D14063
On the blender side this commit fixes importing video files with audio
and video streams that do not share the same start time and duration.
Differential Revision: https://developer.blender.org/D12353
The duration and start time for audio strips were not correctly read in
audaspace.
Some video files have a "lead in" section of audio that plays before the
video starts playing back. Before this patch, we would play this lead in
audio at the same time as the video started and thus the audio would not
be in sync anymore.
Now the lead in audio is cut off and the duration should be correctly
calculated with this in mind.
If the audio starts after the video, the audio strip is shifted to
account for this, but it will also lead to cut off audio which might not
be wanted. However we don't have a simple way to solve this at this
point.
Differential Revision: http://developer.blender.org/D11917
- Changing API for time values from float to double for better precision.
- Fixing minor mistakes in the documentation.
- Fixing minor unnecessary large memory allocation.
Due to some floating point errors the last frame of a VSE audio strip can
cause integer overflow and crash Blender. This overflow was caused by a
cast from `int64_t` to `int` without prior check. The crash is fixed by
keeping the variable as `int64_t` for as long as possible.
Deleting the old internal audaspace.
Major changes from there are:
- The whole library was refactored to use C++11.
- Many stability and performance improvements.
- Major Python API refactor:
- Most requested: Play self generated sounds using numpy arrays.
- For games: Sound list, random sounds and dynamic music.
- Writing sounds to files.
- Sequencing API.
- Opening sound devices, eg. Jack.
- Ability to choose different OpenAL devices in the user settings.
