Each emulated Wiimote can have its speaker routed from left to right via the "Speaker Pan" setting in the emulated wiimote settings dialog. Use any value from -127 for leftmost to 127 for rightmost with 0 being the centre.
Added code in the InputConfig to use a spin control for non-boolean values.
Defaulted the setting of "Enable Speaker Data" to disabled.
The Wiimotes are positioned as follows:
Wiimote 0 = Center
Wiimote 1 = Left
Wiimote 2 = Right
Wiimote 3 = Center
The Wiimote speaker output can be disabled via the "Enable Speaker Data" checkbox in the Wiimote settings.
It was only used for Windows XP and lower.
This also bumps the _WIN32_WINNT define in the stdafx precompiled headers to set the minimum version as Windows Vista.
The two instances of this class were sharing a frac variable causing
audio glitches when both were running (which is now all the time).
Fixes issue 7463 (Since DTK merge, audio has staic in it).
The code actually handles this case correctly; the algorithm is linear
interpolation between the two closest samples, and the way it is written
should work correctly with any ratio.
The primary motivation here is to make sure we submit samples from the
CPU thread. This makes sure the timing of related interrupts accurate,
and generally keeps the different kinds of audio synchronized. This will also
allow improvements to audio dumping functionality.
The new code is also more concise because it gets rid of some duplicated
audio mixing code.
This reverts commit 4990b8910b.
The commit is causing substantial performance issues for the DSound
backend which I somehow didn't catch during testing.
Pretty straightforward; IDirectSoundNotify lets you register for
notifications after a certain amount of sound has played, so use that
instead of depending on Update() notifications from the CPU thread.
Also, while I'm here, reduce the buffer size by a factor of 4; this seems
to reduce the latency, although the difference is sort of subtle.
This is good for a couple of reasons: one, it gets rid of duplicated code,
and two, DSP emulation shouldn't need to interact with audio in the first
place.
- remove unused variables
- reduce the scope where it makes sense
- correct limits (did you know that strcat()'s last parameter does not
include the \0 that is always added?)
- set some free()'d pointers to NULL
The default async api allow us to set some latency options. The old one (simple API) was the lazy way to go for usual audio where latency doesn't matter.
This also streams audio, so it should be a bit faster then the old one.
* Currently there is no DEBUGFAST configuration. Defining DEBUGFAST as a preprocessor definition in Base.props (or a global header) enables it for now, pending a better method. This was done to make managing the build harder to screw up. However it may not even be an issue anymore with the new .props usage.
* D3DX11SaveTextureToFile usage is dropped and not replaced.
* If you have $(DXSDK_DIR) in your global property sheets (Microsoft.Cpp.$(PlatformName).user), you need to remove it. The build will error out with a message if it's configured incorrectly.
* If you are on Windows 8 or above, you no longer need the June 2010 DirectX SDK installed to build dolphin. If you are in this situation, it is still required if you want your built binaries to be able to use XAudio2 and XInput on previous Windows versions.
* GLew updated to 1.10.0
* compiler switches added: /volatile:iso, /d2Zi+
* LTCG available via msbuild property: DolphinRelease
* SDL updated to 2.0.0
* All Externals (excl. OpenAL and SDL) are built from source.
* Now uses STL version of std::{mutex,condition_variable,thread}
* Now uses Build as root directory for *all* intermediate files
* Binary directory is populated as post-build msbuild action
* .gitignore is simplified
* UnitTests project is no longer compiled
This fix the 1h32 audio bug which outputs static sound after 1h32.
The mixer is used for 32->48kHz resampling and as output buffer for the async audio backends.
So this buffer was indiced by a writing and a reading pointer and the count of samples in it.
As this is redundant and the sample count isn't accurate calculateable because of the interpolation,
both indices gets out of sync. So after some time (~92min), this buffer overflows and return only garbage.
thx @ moosehunter + delroth for debugging on this issue. You did the most work :-)
