|
072e3fdf
|
2021-10-14T23:17:08
|
|
Fixed bug #4534: NEON implementation of Convert51ToStereo (Thanks Ryan!)
|
|
2423c514
|
2021-10-13T09:33:51
|
|
Work around hang in AAudioStream_write() during extended shared object loading while running in a debugger. Observed on a OnePlus 8T (KB2005) running Oxygen OS 11.0.10.10.KB05AA.
The observed behavior is that any nonzero timeout value would hang until the device was paused and resumed. And a zero timeout value would always return 0 frames written even when audio fragments could be heard. Making a manual timeout system unworkable.
None of the straightforward systems imply that there's a detectable problem before the call to AAudioStream_write(). And the callback set within AAudioStreamBuilder_setErrorCallback() does not get called as we enter the hang state.
I've found that AAudioStream_getTimestamp() will report an error state from another thread. So this change codifies that behavior a bit until a better fix or more root cause can be found.
|
|
e92a639b
|
2021-10-04T23:00:28
|
|
replaced many uses of libc calls with SDL_ counterparts in os2 sources.
TODO: core/os2 and geniconv sources _must_ be updated, as well..
|
|
|
345c161f
|
2021-09-22T09:06:45
|
|
Fixed some accidental uses of external C runtime functions
|
|
ce11caa8
|
2021-09-21T16:41:29
|
|
alsa: Map 7.1 audio channels to match what Windows and macOS expect.
This matches what we did a long time ago for 5.1 audio.
Fixes #55.
(FIFTY FIVE. Bug reported 15 years, 3 months, and 11 days ago! lol)
|
|
|
c45facf2
|
2021-09-21T11:13:46
|
|
alsa: clean up macro salsa a little.
|
|
|
bf97c5a2
|
2021-09-08T14:47:40
|
|
Make sure SDL file descriptors don't leak into child processes
|
|
a1ffeda0
|
2021-08-28T22:52:13
|
|
Add SDL_HINT_APP_NAME and DBUS inhibition hint
See SDL bug #4703. This implements two new hints:
- SDL_APP_NAME
- SDL_SCREENSAVER_INHIBIT_ACTIVITY_NAME
The former is the successor to SDL_HINT_AUDIO_DEVICE_APP_NAME, and acts
as a generic "application name" used both by audio drivers and DBUS
screensaver inhibition. If SDL_AUDIO_DEVICE_APP_NAME is set, it will
still take priority over SDL_APP_NAME.
The second allows the "activity name" used by
org.freedesktop.ScreenSavver's Inhibit method, which are often shown in
the UI as the reason the screensaver (and/or suspend/other
power-managment features) are disabled.
|
|
7d90df0e
|
2021-08-29T15:24:23
|
|
Restore previous behavior of empty SDL_AUDIODRIVER trying all drivers.
The recent change to make SDL_AUDIODRIVER support comma-separated lists
broke the previous behavior where an SDL_AUDIODRIVER that was empty
behaved the same as if it was not set at all. This old behavior was
necessary to paper over differences in platforms where SDL_setenv may
or may not actually delete the env var if an empty string is specified.
This patch just adds a simple check to ensure SDL_AUDIODRIVER is not
empty before using it, restoring the old interpretation of the empty
var.
|
|
|
3261f7f6
|
2021-08-26T16:15:30
|
|
audio: Support "pulse" as an alias for "pulseaudio"
Originally, SDL 1.2 used "pulse" as the name for its PulseAudio driver.
While it now supports "pulseaudio" as well for compatibility with SDL
2.0 [1], there are still scripts and distro packages which set
SDL_AUDIODRIVER=pulse [2]. While it's possible to remove this in most
circumstances or replace it with "pulseaudio" or a comma-separated list,
this may still conflict if the environment variable is set globally and
old binary builds of SDL 1.2 (e.g. packaged with older games) are being
used.
To fix this on SDL 2.0, add a hardcoded check for "pulse" as an audio
driver name, and replace it with "pulseaudio". This mimics what SDL 1.2
does (but in reverse). Note that setting driver_attempt{,_len} is safe
here as they're reset correctly based on driver_attempt_end on the next
loop.
[1] https://github.com/libsdl-org/SDL-1.2/commit/d9514097846381cd30fe08df65dbdd48de92a058
[2] https://bugzilla.opensuse.org/show_bug.cgi?id=1189778
|
|
|
9504bb12
|
2021-08-23T00:47:25
|
|
pulseaudio: Fix some function signatures that expect const pointers.
This might have changed at some point in the Pulse API, or this might have
always been wrong, but we didn't notice because the dynamic loading code
hides it by casting things to void *. The static path, where it
assigns the function pointer directly, puts out a clear compiler warning,
though.
|
|
70c23b25
|
2021-08-18T13:18:11
|
|
audio: pipewire: Reset hotplug atomic variables on deinit.
|
|
|
bfa15931
|
2021-08-18T12:04:38
|
|
audio: pipewire: Set PW_KEY_NODE_RATE to suggest a rate.
This can be used by recent pipewire to avoid resampling.
|
|
07a2d71e
|
2021-03-30T11:56:38
|
|
emscripten: Feed silence to device when paused
|
|
1e921352
|
2021-08-13T07:58:48
|
|
Improve portability of SDL_Convert51ToStereo_AVX
Don't rely on checking __clang_major__ since it is not comparable
between different vendors. Don't use "#pragma clang attribute" since it
is only available in relatively recent versions, there's no obvious way
to check if it's supported, and just using __attribute__ directly (for
gcc as well) results in simpler code anyway.
|
|
|
ed6eb07e
|
2021-08-12T01:40:50
|
|
SDL_wasapi.c: fixed build against older SDKs.
|
|
|
44a76710
|
2021-08-11T12:55:50
|
|
Don't disable clang avx instrinsics on win32 if __AVX__ is defined.
C.f.: https://github.com/libsdl-org/SDL/issues/4533
|
|
|
ac32c522
|
2021-08-10T18:11:09
|
|
Try using the built-in WASAPI audio rate conversion
Fixes https://github.com/libsdl-org/SDL/issues/4608
|
|
|
a0af7ce7
|
2021-08-10T15:05:49
|
|
OSS is no longer the preferred audio backend on modern UNIX systems
Fixes https://github.com/libsdl-org/SDL/issues/4207
|
|
25f9ed87
|
2021-08-10T13:10:36
|
|
audio: Fix false positives in driver name comparison
Without this change, driver names don't get matched correctly;
for example "a" can get matched with "alsa" since it only checks
whether the string matches up to the length of the requested
driver name.
|
|
|
bec78357
|
2021-08-06T14:20:55
|
|
Better fix for compiling using clang on Windows
This fixes https://github.com/libsdl-org/SDL/issues/4533
|
|
|
cae7bd9b
|
2021-08-06T12:01:24
|
|
Don't use AVX with clang if the compiler isn't building with AVX instructions
Fixes https://github.com/libsdl-org/SDL/issues/4533
|
|
|
659e1f0a
|
2021-07-29T17:49:52
|
|
audiocvt: The to-5.1 converters now soften FL and FR channels more.
This is experimental and might be tweaked further.
Reference #4104.
Also reference:
https://github.com/Keriew/augustus/issues/194#issuecomment-847655049
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|
8f38ba4d
|
2021-07-29T18:02:47
|
|
Fix casts that should be using uintptr_t
This is needed to support CHERI, and thus Arm's experimental Morello
prototype, where pointers are implemented using unforgeable capabilities
that include bounds and permissions metadata to provide fine-grained
spatial and referential memory safety, as well as revocation by sweeping
memory to provide heap temporal memory safety.
On most systems (anything with a flat memory hierarchy rather than using
segment-based addressing), size_t and uintptr_t are the same type.
However, on CHERI, size_t is just an integer offset, whereas uintptr_t
is still a capability as described above. Casting a pointer to size_t
will strip the metadata and validity tag, and casting from size_t to a
pointer will result in a null-derived capability whose validity tag is
not set, and thus cannot be dereferenced without faulting.
The audio and cursor casts were harmless as they intend to stuff an
integer into a pointer, but using uintptr_t is the idiomatic way to do
that and silences our compiler warnings (which our build tool makes
fatal by default as they often indicate real problems). The iconv and
egl casts were true positives as SDL_iconv_t and iconv_t are pointer
types, as is NativeDisplayType on most OSes, so this would have trapped
at run time when using the round-tripped pointers. The gles2 casts were
also harmless; the OpenGL API defines this argument to be a pointer type
(and uses the argument name "pointer"), but it in fact represents an
integer offset, so like audio and cursor the additional idiomatic cast
is needed to silence the warning.
|
|
53987e9b
|
2021-07-28T21:03:42
|
|
Optimized SDL_Convert51ToStereo_AVX
|
|
|
cf85710c
|
2021-07-28T22:55:10
|
|
SDL_audiocvt.c: disable AVX for clang < 5 and gcc < 4.9
See: https://github.com/libsdl-org/SDL/issues/4533
|
|
c72aef26
|
2021-05-01T23:56:23
|
|
Fixed microphone randomly stop working
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.
|
|
|
4ef8674d
|
2021-07-27T14:18:44
|
|
Revert "wasapi: Open capture devices the way we used to."
This reverts commit 0d0fee7569803ddc41985bfc249418b02dd8cd97.
|
|
|
0d0fee75
|
2021-07-27T14:12:18
|
|
wasapi: Open capture devices the way we used to.
This should work around the regression in #3234, since it basically reverts
the problem change, but only for capture devices.
Fixes #3234.
|
|
|
736a424f
|
2021-07-27T13:12:57
|
|
pulseaudio: Optionally add "monitor" sources to device list.
There's a new hint to enable this.
Fixes #2917.
|
|
|
8d790b10
|
2021-07-27T12:23:46
|
|
audiocvt: stereo-to-mono SSE3 now uses unaligned accesses.
On modern CPUs, there's no penalty for using the unaligned instruction on
aligned memory, but now it can vectorize unaligned data too, which even if
it's not optimal, is still going to be faster than the scalar fallback.
Fixes #4532.
|
|
45de0a1d
|
2021-04-02T06:28:10
|
|
Support comma-separated lists in SDL_AUDIODRIVER
|
|
b2c8d3e9
|
2021-06-11T22:14:54
|
|
audiocvt: 5.1 to Stereo conversion utilizing AVX
|
|
|
638befc1
|
2021-06-10T17:22:39
|
|
audiocvt: 5.1 to Stereo conversion utilizing SSE
|
|
|
db56526f
|
2021-06-10T13:07:27
|
|
audiocvt: Replace divisions
Division is expensive and the compiler can not replace it themselves.
Though we now we will be ok in terms of floating point edge behaviour.
|
|
86ee383f
|
2021-05-25T09:59:45
|
|
Removed duplicate SDL_AtomicGet()
|
|
|
db146e66
|
2021-05-25T10:34:04
|
|
Fixed warnings building with Visual Studio
|
|
|
58884e4c
|
2021-05-04T00:23:40
|
|
SDL_audiocvt.c: fixed MSVC double->float conversion warnings.
|
|
|
8ac0fb52
|
2021-04-29T09:29:02
|
|
OpenSLES: CloseDevice() is called at higher level, if OpenDevice() fails
- explicit initialization of static variables
|
|
|
fcbf19b7
|
2021-04-28T21:04:47
|
|
AAudio: make sure stream is not null to prevent crash in RequestStop (see #3710)
|
|
|
f8695185
|
2021-04-27T11:07:51
|
|
Audio: normalize conversion Stereo to 5.1, Quad to 7.1, 5.1 to 7.1 (bug #4104)
|
|
|
21349901
|
2021-04-27T10:57:48
|
|
Audio: convert 5.1 to 7.1, use right-surround for r-front and r-back (see #4104)
|
|
058bbe02
|
2021-04-24T10:17:03
|
|
Set volume on device open
|
|
|
f1fab24e
|
2021-04-15T21:00:00
|
|
AAudio: add bootstrap in SDL_audio.c
|
|
|
04b2f5f6
|
2021-04-15T20:54:58
|
|
Android: add AAudio back-end, with playback and capture (see #3710)
https://developer.android.com/ndk/guides/audio/aaudio/aaudio
|
|
|
4118fe62
|
2021-04-15T20:52:43
|
|
Android: OpenSLES, explicitly initialise the global variable 'bqPlayerPlay',
it may be read even if OpenSLES back-end hasn't been intialized
|
|
|
0f4aba7b
|
2021-04-06T18:34:53
|
|
audio: Fixed assertion failure if trying to use dummy backend.
|
|
|
64853b73
|
2021-04-06T18:34:17
|
|
audio: Changed a disk and dummy backends to use _this instead of this.
|
|
9d294f1f
|
2021-03-27T00:53:10
|
|
audio: Allow AudioStreamGet to return 0 in RunAudio.
While we should normally expect _something_ from the stream based on the
AudioStreamAvailable check, it's possible for a device change to flush the
stream at an inconvenient time, causing this function to return 0.
Thing is, this is harmless. Either data will be NULL and the result won't matter
anyway, or the data buffer will be zeroed out and the output will just be
silence for the brief moment that the device change is occurring. Both scenarios
work themselves out, and testing on Windows shows that this behavior is safe.
|
|
|
9b7babf9
|
2021-03-27T00:47:54
|
|
wasapi: Remove assert added by 67e8522d
|
|
5f9effaa
|
2021-03-28T17:45:41
|
|
audio: pipewire: Block while waiting on stream state info
Initializing streams, particularly capture streams, can take many milliseconds, which is a bit much for a busy wait. Use a blocking wait instead.
|
|
|
8deb4063
|
2021-03-28T17:22:59
|
|
audio: pipewire: Avoid redundant locking
The pw_thread_loop already locks and unlocks the thread mutex at the start and end of each loop iteration, so these locks are unnecessary.
|
|
|
5bb2bbd4
|
2021-03-28T17:17:00
|
|
audio: pipewire: Don't use uninitialized variables in callbacks
Some of the SDL_AudioDevice struct members aren't initialized until after returning from the OpenDevice function. Since Pipewire uses it's own processing threads, the callbacks can be entered before all members of SDL_AudioDevice are initialized, such as work_buffer, callbackspec and the processing stream, which creates a race condition. Don't use these members when in the paused state to avoid potentially using uninitialized values and memory.
|
|
|
e7e519a4
|
2021-03-17T13:04:05
|
|
dsp: Refuse to initialize if there aren't any Open Sound System devices.
This prevents the dsp target from stealing the audio subsystem but not
being able to produce sound, so other audio targets further down the list
can make an attempt instead.
Thanks to Frank Praznik who did a lot of the research on this problem!
|
|
|
b98b5adc
|
2021-03-15T10:21:36
|
|
wasapi: Don't use the system's resampler.
|
|
a5f3ea14
|
2021-03-10T09:36:46
|
|
netbsdaudio: Handle ioctls failing
A user reported that the mpv video player hangs after attempting to
set an unsupported number of channels with the SDL audio output,
because it thinks it's successfully opened the device. This makes
the failure graceful.
|
|
|
4fbd60b8
|
2021-03-09T12:22:48
|
|
audio: pipewire: Remove the nickname portion of sink/source names
Removes the node nickname from sink/source nodes as it doesn't provide any useful information and names now match those used in Pulseaudio, so any stored configuration data will be compatible between the two audio backends.
|
|
|
a4ddb175
|
2021-03-08T19:28:58
|
|
Formatting
|
|
|
7d89f09f
|
2020-12-18T14:28:09
|
|
ISO C90 fixes
|
|
|
e7edb06e
|
2020-12-04T00:06:15
|
|
Audio fix
|
|
|
2d64e37e
|
2020-11-02T18:09:43
|
|
Initial rebase of xerpi's port
|
|
391bb80b
|
2021-03-05T16:53:06
|
|
Replace duplicate functions and lstrlen/lstrcat with SDL string functions
|
|
|
67e8522d
|
2021-02-27T17:37:25
|
|
Add SDL_GetAudioDeviceSpec.
This API is supported by pipewire, pulseaudio, coreaudio, wasapi, and disk.
|
|
|
ac8a3fda
|
2021-03-03T20:33:20
|
|
fix prepare_audiospec() possibly missing a bad SDL_AUDIO_CHANNELS env.
|
|
|
4de0c74a
|
2021-03-02T10:02:59
|
|
audio: pipewire: Add the application name to the stream properties
|
|
|
9ed01da7
|
2021-03-02T09:47:47
|
|
audio: pipewire: Constify and clarify period size calculations
Constify the min/max period variables, use a #define for the base clock rate used in the calculations and note that changing the upper limit can have dire side effects as it's a hard limit in Pipewire.
|
|
|
d7ca855c
|
2021-03-02T09:33:11
|
|
audio: pipewire: Add missing static qualifiers to globals
|
|
|
84c44e01
|
2021-03-01T12:39:52
|
|
audio: pipewire: fix uninitialized variable warnings
|
|
|
7001b531
|
2021-02-27T12:53:08
|
|
audio: pipewire: Add vim format lines to files and fix indentation
Increase indentation spacing from 2 to 4 to comply with style standards.
|
|
|
2fcba50e
|
2021-02-27T12:08:15
|
|
audio: pipewire: Code and comment cleanups
Replace "magic numbers" with #defines, explain the requirements when using the userdata pointer in the node_object struct and a few other minor code and comment cleanups.
|
|
|
4eadd147
|
2021-02-25T14:00:23
|
|
audio: pipewire: Fix outdated comment
|
|
|
cd56f1b3
|
2021-02-24T14:36:58
|
|
audio: pipewire: Use "rear" designation for rear channels
Use the 'R' (rear) prefixed designations for the rear audio channels instead of 'S' (surround). Surround designated channels are only used in the 8 channel configuration.
|
|
|
adc0a931
|
2021-02-24T14:08:08
|
|
audio: Move Pipewire bootstrap after Jack
Move the Pipewire audio driver below others in the list so it won't be mistakenly initialized when it's not the system mixer.
|
|
|
21adec93
|
2021-02-24T12:02:54
|
|
audio: pipewire: Make enumeration structure and function names more descriptive
Rename the add/remove/clear list functions and rename connected_device to io_node, as a sink/source node isn't necessarily a device.
|
|
|
a07f5434
|
2021-02-21T13:24:20
|
|
audio: pipewire: Report default devices first
Further refactor the device enumeration code to retrieve the default sink/source node IDs from the metadata node. Use the retrieved IDs to sort the device list so that the default devices are at the beginning and thus are the first reported to SDL.
|
|
|
9afd7570
|
2021-02-20T13:33:12
|
|
audio: pipewire: Always buffer source audio
The latency of source nodes can change depending on the overall latency of the processing graph. Incoming audio must therefore always be buffered to ensure uninterrupted delivery.
The SDL_AudioStream path was removed in the input callback as the only thing it was used for was buffering audio outside of Pipewire's min/max period sizes, and that case is now handled by the omnipresent buffer.
|
|
106dc009
|
2021-02-19T17:18:36
|
|
audio: pipewire: Pass proper parameter to user audio callback
The audio callbacks should pass the callbackspec.userdata parameter to the callback, not spec.userdata
Co-authored-by: Oschowa <Oschowa@web.de>
|
|
|
f3ebbc06
|
2021-02-19T16:02:20
|
|
audio: pipewire: Retrieve the channel count and default sample rate for sinks/sources
Extend device enumeration to retrieve the channel count and default sample rate for sink and source nodes. This required a fairly significant rework of the enumeration procedure as multiple callbacks are involved now. Sink/source nodes are tracked in a separate list during the enumeration process so they can be cleaned up if a device is removed before completion. These changes also simplify any future efforts that may be needed to retrieve additional configuration information from the nodes.
|
|
|
2f0b99a7
|
2021-02-13T11:56:05
|
|
audio: Add Pipewire playback/capture sink
|
|
|
08547adb
|
2021-02-20T09:28:03
|
|
pulseaudio: Add "zerocopy" playback path
|
|
ef85ed93
|
2021-02-17T20:53:35
|
|
pulseaudio: Initialize fragsize to fix mic recording
fragsize wasn't initialized, and it is used for recording.
If the value was 0 or -1, pulseaudio configures it itself. But sometimes
we can get a random (and large) value that makes pulseaudio give us
large sample at a very low frequency.
https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/blob/master/src/pulse/def.h#L453
https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/blob/v13.0/src/pulsecore/protocol-native.c#L409
|
|
20be1d63
|
2020-07-15T09:13:03
|
|
emscripten: Automatically resume audio contexts
This uses the mechanism added in emscripten-core/emscripten#10843
which was applied to SDL1 and OpenAL. This adds the same for SDL2.
This also reverts commit 865eaddffed50dbd13e6564c3f73902472cf74e8
which did something similar, but the new mechanism is more effective.
|
|
f443a6fc
|
2021-02-11T02:05:02
|
|
Fix format string warnings for width-based integers
The DJGPP compiler emits many warnings for conflicts between print
format specifiers and argument types. To fix the warnings, I added
`SDL_PRIx32` macros for use with `Sint32` and `Uint32` types. The macros
alias those found in <inttypes.h> or fallback to a reasonable default.
As an alternative, print arguments could be cast to plain old integers.
I opted slightly for the current solution as it felt more technically correct,
despite making the format strings more verbose.
|
|
|
b852590b
|
2021-01-24T17:02:40
|
|
minor clean-up in SDL_os2audio.c
|
|
|
8f102589
|
2021-01-23T17:28:10
|
|
os2audio: changed backend name from MMOS2 to DART (like SDL-1.2)
|
|
|
50ea3b77
|
2021-01-08T10:09:37
|
|
Fixed bug 5080 - SDL_netbsdaudio: Always use the device's preferred frequency
Nia Alarie
The NetBSD kernel's audio resampling code is much simpler and lower quality than libsamplerate.
Presumably, if SDL always performs I/O on the audio device in its native frequency, we can avoid resampling audio in the kernel and let SDL do it with libsamplerate instead.
|
|
|
265a1cc9
|
2021-01-05T15:50:02
|
|
use WIN_StringToUTF8W instead of WIN_StringToUTF8 where needed (#2)
cf. bug #5435.
- SDL_wasapi_win32.c (GetWasapiDeviceName): pwszVal is WCHAR*
- windows/SDL_sysfilesystem.c (SDL_GetBasePath, SDL_GetPrefPath)
- windows/SDL_sysurl.c (SDL_SYS_OpenURL): wurl is WCHAR*
- SDL_windowssensor.c (ConnectSensor): bstr_name is WCHAR*
- windows/SDL_systhread.c (SDL_SYS_SetupThread): strw is WCHAR*
|
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ed39f2f3
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2021-01-04T01:23:50
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SDL_wasapi_win32.c (WASAPI_PlatformThreadInit): use L instead of TEXT()
because AvSetMmThreadCharacteristicsW specifically accepts WCHAR* input
cf. bug #5435.
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01a2f276
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2021-01-04T01:23:50
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consistently use TEXT() macro with LoadLibrary() and GetModuleHandle()
cf. bug #5435.
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9130f7c3
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2021-01-02T10:25:38
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Updated copyright for 2021
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8476df3e
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2020-12-30T23:55:10
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SDL_mixer.c: remove calls to non-existing m68k asm code.
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cb361896
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2020-12-09T07:16:22
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Fixed bug 5235 - All internal sources should include SDL_assert.h
Ryan C. Gordon
We should really stick this in SDL_internal.h or something so it's always available.
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53b16679
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2020-11-11T12:33:55
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SIZE_MAX need not be defined in limits.h
it can be in limits.h (windows) or stdint.h.
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1b8dee7c
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2020-10-31T11:32:40
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coreaudio: Remove unnecessary include of CoreServices.h
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a4040293
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2020-10-25T10:10:02
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os2: misc build fixes
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bfc80d83
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2020-10-25T03:55:02
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minor coding style cleanup
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a90f0400
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2020-10-15T21:37:30
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os2: a _lot_ of coding style cleanup, sot that they match the SDL style.
also renamed the 'debug' macro to debug_os2: the former was dangerously
a common name.
the binary (dll) output is precisely the same as before.
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d2723875
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2020-10-14T23:01:06
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os2: integrate the port into main tree.
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1d9cf23e
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2020-10-14T23:01:05
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os2: updated copyright dates for 2020. header guard fixes.
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a3d7913c
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2020-10-14T23:01:05
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SDL_os2audio.c (OS2_OpenDevice): change spec->samples assignment:
Original code assigned MCIMixSetup.ulSamplesPerSec value to it, but it
is just the freq... We now change spec->samples only either if it is 0
or we changed the frequency, by picking a default of ~46 ms at desired
frequency (code taken from SDL_audio.c:prepare_audiospec()).
With this, the crashes I have been experiencing are gone.
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