Cameron Gutman
I was trying to use SDL_GetQueuedAudioSize() to ensure my audio latency didn't get too high while streaming data in from the network. If I get more than N frames of audio queued, I know that the network is giving me more data than I can play and I need to drop some to keep latency low.
This doesn't work well on WASAPI out of the box, due to the addition of GetPendingBytes() to the amount of queued data. As a terrible hack, I loop 100 times calling SDL_Delay(10) and SDL_GetQueuedAudioSize() before I ever call SDL_QueueAudio() to get a "baseline" amount that I then subtract from SDL_GetQueuedAudioSize() later. However, because this value isn't actually a constant, this hack can cause SDL_GetQueuedAudioSize() - baselineSize to be < 0. This means I have no accurate way of determining how much data is actually queued in SDL's audio buffer queue.
The SDL_GetQueuedAudioSize() documentation says: "This is the number of bytes that have been queued for playback with SDL_QueueAudio(), but have not yet been sent to the hardware." Yet, SDL_GetQueuedAudioSize() returns > 0 value when SDL_QueueAudio() has never been called.
Based on that documentation, I believe the current behavior contradicts the documented behavior of this function and should be changed in line with Boris's patch.
I understand that exposing the IAudioClient::GetCurrentPadding() value is useful, but a solution there needs to take into account what of that data is silence inserted by SDL and what is actual data queued by the user with SDL_QueueAudio(). Until that happens, I think the best approach is to remove the GetPendingBytes() call until SDL is able to keep track of queued data to make sense of it. This would make SDL_GetQueuedAudioSize() possible to use accurately with WASAPI.
This was meant to migrate CoreAudio onto the same SDL_RunAudio() path that
most other audio drivers are on, but it introduced a bug because it doesn't
deal with dropped audio buffers...and fixing that properly just introduces
latency.
I might revisit this later, perhaps by reworking SDL_RunAudio to allow for
this sort of API better, or redesigning the whole subsystem or something, I
don't know. I'm not super-thrilled that this has to exist outside of the usual
codepaths, though.
Fixes Bugzilla #4481.
SDLActivity thread priority is unchanged, by default -10 (THREAD_PRIORITY_VIDEO).
SDLAudio thread priority was -4 (SDL_SetThreadPriority was ignored) and is now -16 (THREAD_PRIORITY_AUDIO).
SDLThread thread priority was 0 (THREAD_PRIORITY_DEFAULT) and is -4 (THREAD_PRIORITY_DISPLAY).
This means that if you have two devices named "Soundblaster Pro" in your
machine, one will be reported as "Soundblaster Pro" and the other as
"Soundblaster Pro (2)".
This makes it so you can't into a position where one of your devices can't
be opened because another is sitting on the same name.
Author: Anthony Pesch <inolen@gmail.com>
Date: Fri May 4 20:21:21 2018 -0400
Added SDL_AUDIO_ALLOW_SAMPLES_CHANGE flag enabling users of SDL_OpenAudioDevice to get
the sample size of the actual hardware buffer vs having a stream created to handle the
delta
This would cause problems in various ways, but specifically triggers an
assert when you close a WASAPI capture device in an app running over RDP.
Related to (but not the actual bug) in Bugzilla #3924.
SDL now builds with gcc 7.2 with the following command line options:
-Wall -pedantic-errors -Wno-deprecated-declarations -Wno-overlength-strings --std=c99
XAudio2 doesn't have capture support, so WASAPI was to replace it; the holdout
was WinRT, which still needed it as its primary audio target until the WASAPI
code code be made to work.
The support matrix now looks like:
WinXP: directsound by default, winmm as a fallback for buggy drivers.
Vista+: WASAPI (directsound and winmm as fallbacks for debugging).
WinRT: WASAPI
This time it's using real math from a real whitepaper instead of my previous
amateur, fast-but-low-quality attempt. The new resampler does "bandlimited
interpolation," as described here: https://ccrma.stanford.edu/~jos/resample/
The output appears to sound cleaner, especially at high frequencies, and of
course works with non-power-of-two rate conversions.
There are some obvious optimizations to be done to this still, and there is
other fallout: this doesn't resample a buffer in-place, the 2-channels-Sint16
fast path is gone because this resampler does a _lot_ of floating point math.
There is a nasty hack to make it work with SDL_AudioCVT.
It's possible these issues are solvable, but they aren't solved as of yet.
Still, I hope this effort is slouching in the right direction.
Simon Hug
This issue actually raises the question if this API change (requirement of initialized audio subsystem) is breaking backwards compatibility. I don't see the documentation saying it is needed in 2.0.5.
David Ludwig
I've created a new set of patches. I am happy to create more, if it would help.
One version only copies 'size'.
A second version copies both 'size' and 'silence'. When looking over the documentation for SDL_OpenAudio in SDL_audio.h, it mentioned that both 'size' and 'silence' were things that SDL_OpenAudio would calculate.
Regarding *both* patches, I did notice that SDL 1.2 appears to have always modified desired's size and silence fields. The SDL wiki, at https://wiki.libsdl.org/SDL_OpenAudio#Remarks , does note:
Simon Hug
Some code in SDL loads libraries with SDL_LoadObject to get more information or use newer APIs. SDL_LoadObject may fail, set an error message and SDL will continue with some fallback code. Since SDL will overwrite the error or exit the function with a return value that indicates success, the error form SDL_LoadObject for the optional stuff might as well be cleared right away.
This gracefully recovers when a device format is changed, and will switch
to the new default device if the current one is unplugged, etc.
This does not handle when a new default device is added; it only notices
if the current default goes away. That will be fixed by implementing the
stubbed-out MMNotificationClient_OnDefaultDeviceChanged() function.
We will throw away the data anyhow, but some apps depend on the callback
firing to make progress; testmultiaudio.c, if nothing else, is an example
of this.
Capture also will now fire the callback in these conditions, offering nothing
but silence.
Apps can check SDL_GetAudioDeviceStatus() or listen for the
SDL_AUDIODEVICEREMOVED event if they want to gracefully deal with
an opened audio device that has been unexpectedly lost.
This should remain binary compatible with Windows XP, as we dynamically
load anything we need and fall back to DirectSound/WinMM/XAudio2 if not
available.
Walter van Niftrik
We have found that since SDL 2.0.5 the audio callback thread is created with a very small stack size. In our application this is leading to stack overflows.
We believe there is a bug at http://hg.libsdl.org/SDL/file/391fd532f79e/src/audio/SDL_audio.c#l1132, where the is_internal_thread flag appears to be inverted.
This defaults to the internal SDL resampler, since that's the likely default
without a system-wide install of libsamplerate, but those that need more can
tweak this.
There was a draft of this where it did audio conversion into the final buffer,
if there was enough room available past what you asked for, but that interface
got removed, so the parameters didn't make sense (and we were using the
wrong one in any case, too!).
