I updated .clang-format and ran clang-format 14 over the src and test directories to standardize the code base.
In general I let clang-format have it's way, and added markup to prevent formatting of code that would break or be completely unreadable if formatted.
The script I ran for the src directory is added as build-scripts/clang-format-src.sh
This fixes:
#6592#6593#6594
* Add braces after if conditions
* More add braces after if conditions
* Add braces after while() conditions
* Fix compilation because of macro being modified
* Add braces to for loop
* Add braces after if/goto
* Move comments up
* Remove extra () in the 'return ...;' statements
* More remove extra () in the 'return ...;' statements
* More remove extra () in the 'return ...;' statements after merge
* Fix inconsistent patterns are xxx == NULL vs !xxx
* More "{}" for "if() break;" and "if() continue;"
* More "{}" after if() short statement
* More "{}" after "if () return;" statement
* More fix inconsistent patterns are xxx == NULL vs !xxx
* Revert some modificaion on SDL_RLEaccel.c
* SDL_RLEaccel: no short statement
* Cleanup 'if' where the bracket is in a new line
* Cleanup 'while' where the bracket is in a new line
* Cleanup 'for' where the bracket is in a new line
* Cleanup 'else' where the bracket is in a new line
I ran this script in the include directory:
```sh
sed -i '' -e 's,#include "\(SDL.*\)",#include <SDL3/\1>,' *.h
```
I ran this script in the src directory:
```sh
for i in ../include/SDL3/SDL*.h
do hdr=$(basename $i)
if [ x"$(echo $hdr | egrep 'SDL_main|SDL_name|SDL_test|SDL_syswm|SDL_opengl|SDL_egl|SDL_vulkan')" != x ]; then
find . -type f -exec sed -i '' -e 's,#include "\('$hdr'\)",#include <SDL3/\1>,' {} \;
else
find . -type f -exec sed -i '' -e '/#include "'$hdr'"/d' {} \;
fi
done
```
Fixes https://github.com/libsdl-org/SDL/issues/6575
- reorganize the loop which checks for the right wave-format
- use the return value of UpdateAudioStream
- ensure SetError is called in SDL_NewAudioStream
- use SDL_bool if possible
- assume NULL/SDL_FALSE filled impl
- skip zfill of current_audio at the beginning of SDL_AudioInit (done before the init() calls)
WASAPI_WaitDevice is used for audio playback and capture, but needs to
behave slighty different.
For playback `GetCurrentPadding` returns the padding which is already
queued, so WaitDevice should return when buffer length falls below the
buffer threshold (`maxpadding`).
For capture `GetCurrentPadding` returns the available data which can be
read, so WaitDevice can return as soon as any data is available.
In the old implementation WaitDevice could suddenly hang. This is
because on many capture devices the buffer (`padding`) wasn't filled
fast enough to surpass `maxpadding`. But if at one point (due to unlucky
timing) more than maxpadding frames were available, WaitDevice would not
return anymore.
Issue #3234 is probably related to this.
Anthony Pesch's notes on his patch:
"Currently, the WASAPI backend creates a stream in shared mode and sets the
device's callback size to be half of the shared stream's total buffer size.
This works, but doesn't coordinate will with the actual hardware. The hardware
will raise an interrupt after every period which in turn will signal the
object being waited on inside of WaitDevice. From my empirical testing, the
callback size was often larger than the period size and not a multiple of it,
which resulted in poor latency when trying to time an application based on the
audio callback. The reason for this looked something like:
* The device's callback would be called and and the audio buffer was filled.
* WaitDevice would be called.
* The hardware would raise an interrupt after one period.
* WaitDevice would resume, see that a a full callback had not been played and
then wait again.
* The hardware would raise an interrupt after another period.
* WaitDevice would resume, see that a full callback + some extra amount had
been played and then it would again call our callback and this process would
repeat.
The effect of this is that the pacing between subsequent callbacks is poor -
sometimes it's called very quickly, sometimes it's called very late.
By matching the callback's size to the stream's period size, the pacing of
calls to the user callback is improved substantially. I didn't write an actual
test for this, but my use case for this was my Dreamcast emulator
(https://redream.io) which uses the audio callback to help drive the emulation
speed. Without this change and with the default shared stream buffer (which
has a period of ~10ms) I would get frame times that were between ~3-30
milliseconds; after this change I get frame times of ~11-22 milliseconds.
Note, this patch also has a change that removes passing a duration to the
Initialize call. It seems that the default duration used (when 0 is passed)
does typically match up with the duration returned by GetDevicePeriod, however
the Initialize docs say:
> To set the buffer to the minimum size required by the engine thread, the
> client should call Initialize with the hnsBufferDuration parameter set to 0.
> Following the Initialize call, the client can get the size of the resulting
> buffer by calling IAudioClient::GetBufferSize.
This change isn't strictly required, but I made it to hopefully rule out
another source of unexpected latency."
Fixes Bugzilla #4592.
