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.
* alsa hotplug thread is low priority
* give a chance for other threads to catch up when audio playback is not progressing
* use nonblocking for alsa audio capture
There is a bug with SDL hanging when an audio capture USB device is removed, because poll never returns
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 is just enough to get you through a file that just used the extended
header for float or int data. It doesn't handle all the other things that
you expect from this header, like 24-bit samples inside a 32-bit container
or speaker masks.
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.
This currently favors libsamplerate over the fast path (quality over speed),
but I'm not sure that's the correct approach, as there may be surprising
changes in performance metrics depending on what packages are available on
a user's system. That being said, currently, the only thing with access to
SDL_AudioStream is an SDL audio device's thread, and it might be mostly idle
otherwise, so maybe this is generally good.
Turns out that iterating from 0 to channels-1 was a serious performance hit!
These cases now tend to match or beat the original audio resampler's speed!
This allows us to avoid an extra copy, allocate less memory and reduce cache
pressure. On the downside: we have to do a lot of tapdancing to resample the
buffer in reverse when the output is growing.
It's expensive and (hopefully) unnecessary. If this becomes an overflow
problem, we could multiply both values by 0.5f before adding them, but let's
see if we can get by without the extra multiplication first.
We never seem to overflow the source buffer now; this might have been a
leftover from a bug that was covered by Vitaly's fixes?
Removing this conditional makes the resampler 10-20% faster. Left an
assert in there for debug builds, in case this still happens.
Removed some needless things ("len / sizeof (Uint8)"), and made sure the
int32 -> float code uses doubles to avoid working with large integer values
in a 32-bit float.
Lukasz Biel
Tried to compile SDL2 using newest version of VS.
Got:
SDL_audiocvt.obj : error LNK2019: unresolved external symbol memcpy referenced in function SDL_ResampleCVT
1>E:\Users\dotPo\Lib\SDL\VisualC\x64\Release\SDL2.dll : fatal error LNK1120: 1 unresolved externals
whole compilation process: http://pastebin.com/eWDAvBce
Steps to reproduce:
clone http://hg.libsdl.org/SDL using tortoise hg,
open SDL\VisualC\SDL.sln,
when promted if should retarget solution click ok,
select release x64 build type,
Build/Build Solution
attempt 2, using Visual Studio cmake support:
open folder SDL\
select release x64 build type,
run CMake\Build CMakeLists.txt
build fails
When switched to debug build type, buils succeeds in both cases.
VS 2017 is still beta.
It causes audio pops if you're converting in chunks (and needs to
allocate/initialize/free on each convert). We'll either adjust this interface
when we break ABI for 2.1 to make this usable, or publish the SDL_AudioStream
API for those that want a streaming solution.
In the meantime, the "simple" resampler produces "good enough" audio without
pops and doesn't have to be initialized, so that'll do for now on the
SDL_AudioCVT interface.