|
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*
|
|
|
ed39f2f3
|
2021-01-04T01:23:50
|
|
SDL_wasapi_win32.c (WASAPI_PlatformThreadInit): use L instead of TEXT()
because AvSetMmThreadCharacteristicsW specifically accepts WCHAR* input
cf. bug #5435.
|
|
|
01a2f276
|
2021-01-04T01:23:50
|
|
consistently use TEXT() macro with LoadLibrary() and GetModuleHandle()
cf. bug #5435.
|
|
|
9130f7c3
|
2021-01-02T10:25:38
|
|
Updated copyright for 2021
|
|
|
8476df3e
|
2020-12-30T23:55:10
|
|
SDL_mixer.c: remove calls to non-existing m68k asm code.
|
|
|
cb361896
|
2020-12-09T07:16:22
|
|
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.
|
|
|
53b16679
|
2020-11-11T12:33:55
|
|
SIZE_MAX need not be defined in limits.h
it can be in limits.h (windows) or stdint.h.
|
|
1b8dee7c
|
2020-10-31T11:32:40
|
|
coreaudio: Remove unnecessary include of CoreServices.h
|
|
|
a4040293
|
2020-10-25T10:10:02
|
|
os2: misc build fixes
|
|
|
bfc80d83
|
2020-10-25T03:55:02
|
|
minor coding style cleanup
|
|
|
a90f0400
|
2020-10-15T21:37:30
|
|
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.
|
|
|
d2723875
|
2020-10-14T23:01:06
|
|
os2: integrate the port into main tree.
|
|
|
1d9cf23e
|
2020-10-14T23:01:05
|
|
os2: updated copyright dates for 2020. header guard fixes.
|
|
|
a3d7913c
|
2020-10-14T23:01:05
|
|
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.
|
|
|
e112b776
|
2020-10-14T23:01:05
|
|
SDL_os2audio.c (OS2_OpenDevice): change {0} initializers to SDL_zero()
|
|
|
72594e25
|
2020-10-14T23:01:04
|
|
SDL_os2audio.c (OS2_OpenDevice): remove assignment to wrong spec member
Correct assignment to 'format' member is done below, already.
|
|
|
222f0268
|
2020-10-14T23:01:03
|
|
os/2: port from SDL2-2.0.4 to SDL2-2.0.5:
changes to SDL_os2audio.c, SDL_os2video.c, os2/SDL_systhread.c in order
to accomodate SDL2-2.0.5 changes.
- audio: WaitDone() is gone, CloseDevice() interface changes.
- events / video: DropFile() changes:
SDL_DROPBEGIN and SDL_DROPCOMPLETE events, window IDs for drops.
- thread: struct SDL_Thread->stacksize
|
|
|
aa790837
|
2020-10-14T23:01:02
|
|
os2: several warning fixes.
mostly those "W007: '&array' may not produce intended result" warnings
from Watcom, visible only in C++ mode. one or two others here & there.
|
|
|
c2188619
|
2020-10-14T23:01:01
|
|
os2: added a 2-byte padding to os2 SDL_PrivateAudioData
|
|
|
74cfb81d
|
2020-10-14T23:01:00
|
|
os2: add port files for SDL2-2.0.4 from Andrey Vasilkin
only geniconv/iconv.h (was from LGPL libiconv) is replaced with a generic
minimal iconv.h based on public knowledge.
|
|
|
003a1698
|
2020-10-06T11:07:50
|
|
wav: Make sure the data size is a multiple of blockalign, not an exact match.
I _think_ this is a right thing to do; it fixes a .wav file I have here that
has blockalign==2 when channels==2 and bitspersample==16, which otherwise
would fail.
|
|
554037a6
|
2020-09-26T09:30:08
|
|
audio: fix popping sounds caused by signed/unsigned conversion
When converting audio from signed to unsigned values of vice-versa
the silence value chosen by SDL was the value of the device, not
of the stream that the data was being put into. After conversion
this would lead to a very high or low value, making the speaker
jump to a extreme positon, leading to an audible noise whenever
creating, destroying or playing scilence on a device that reqired
such conversion.
|
|
|
7ef188a1
|
2020-09-19T14:01:57
|
|
jack: Fixed memory leak on device close.
|
|
a69c61fb
|
2020-08-14T12:08:58
|
|
Only assign context and mainloop once we have connected successfully
If we fail to connect to the the pa server, we have an assigned context
and mainloop that isn't connected. So, when PULSEAUDIO_pa_context_disconnect
is called, pa asserts and crashes the application.
Assertion 'pa_atomic_load(&(c)->_ref) >= 1' failed at pulse/context.c:1055, function pa_context_disconnect(). Aborting.
|
|
|
ff53521b
|
2020-06-04T12:26:57
|
|
Fixed Bluetooth audio output on Apple TV
|
|
|
68777406
|
2020-05-20T16:58:33
|
|
windows: Fix calls to CoCreateInstance() so last parameter is a LPVOID *.
|
|
|
8601996f
|
2020-05-03T22:13:48
|
|
hints: Allow specifying audio device metadata.
This is only supported on PulseAudio. You can set a description when opening
your audio device that will show up in pauvcontrol, which lets you set
per-stream volume levels.
Fixes Bugzilla #4801.
|
|
|
a990a34a
|
2020-04-14T22:26:02
|
|
Cleanly switch between audio recording, playback, and both, on iOS
|
|
|
2ae1c0f5
|
2020-04-14T09:52:27
|
|
Allow Bluetooth headphones for iOS playandrecord mode
|
|
|
fba081e4
|
2020-04-07T14:51:08
|
|
wasapi: Patched to compile on C89 systems, and use SDL_ceilf instead of ceilf.
|
|
|
4c2be472
|
2020-04-07T14:37:24
|
|
wasapi: Improve WASAPI audio backend latency (thanks, Anthony!).
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.
|
|
|
b6afbe63
|
2020-04-07T09:38:57
|
|
Added SDL_log.h to SDL_internal.h so logging is available everywhere
|
|
|
9525f972
|
2020-04-05T10:44:51
|
|
Fixed bug 5076 - SDL_netbsdaudio: Add support for 32-bit LPCM
Nia Alarie
The kernel supports this, make SDL expose it so it can be used.
|
|
|
f3e60967
|
2020-04-02T12:27:29
|
|
Fixed setting the "playandrecord" audio hint on Apple TV
The Apple TV doesn't have record capability by default, so activating the audio session with AVAudioSessionCategoryPlayAndRecord fails.
|
|
|
55b4f18e
|
2020-03-29T01:54:00
|
|
coreaudio: The default SDL audio device now tracks the system output default.
So if you go into System Preferences on a MacBook and toggle between a pair of
connected bluetooth headphones and built-in internal speakers, SDL will
switch the device it is playing sound through, to match this setting, on the
fly.
Likewise if the default output device is a USB thing and is unplugged; as the
default device changes at the system level, SDL will pick this up and carry
on with the new default. This is different from our unplug detection for
specific devices, as in those cases we want to send the app a disconnect
notification, instead of migrating transparently as we now do for default
devices.
Note that this should also work for capture devices; if the device changes,
SDL will start recording from the new default.
Fixes Bugzilla #4851.
|
|
|
abdc5cbf
|
2020-03-26T19:30:17
|
|
Allow background music to play in the "play and record" case on iOS
|
|
27889d02
|
2020-03-03T12:31:41
|
|
winrt: Wait for EnumerationCompleted before leaving WASAPI_EnumerateEndpoints
|
|
|
e3b0713e
|
2020-02-24T12:07:18
|
|
Don't call setPreferredOutputNumberOfChannels on iOS, it breaks audio output
|
|
|
2c9871a4
|
2020-02-24T10:25:57
|
|
Fixed surround sound support on Apple TV
|
|
|
f4e23553
|
2020-02-14T15:19:34
|
|
Fixed audio not coming out of the phone speakers while recording on iOS
|
|
|
922b3dc3
|
2020-02-14T14:18:12
|
|
Fixed re-setting the audio session category when closing an audio device
|
|
|
14bf532d
|
2020-02-13T16:10:52
|
|
Fixed opening audio on Android from the Steam Link shell activity
|
|
|
4bb95e84
|
2020-02-11T16:14:02
|
|
Implemented OpenSL-ES audio recording on Android
|
|
|
b1c6e7c2
|
2020-01-23T00:32:34
|
|
Fixed compile warning
|
|
|
f30ef6ed
|
2020-01-21T17:40:16
|
|
audio: Fixed a '//' style comment.
|
|
|
dbe5c14b
|
2020-01-21T15:49:37
|
|
audio: Calculate a legitimate SDL_AudioSpec::silence in SDL_LoadWAV_RW().
|
|
|
a8780c6a
|
2020-01-16T20:49:25
|
|
Updated copyright date for 2020
|
|
|
3ce56f62
|
2020-01-13T08:12:10
|
|
Fixed error formatting
|
|
|
e3cedf96
|
2020-01-11T04:38:13
|
|
Add the destination format to the error when conversion isn't possible
|
|
|
3da6a0b2
|
2019-12-03T03:53:06
|
|
pulseaudio: don't let FlushCapture get stuck in an infinite loop on shutdown.
Fixes Bugzilla #4645.
|
|
60d3965e
|
2019-10-30T15:36:17
|
|
Readability: remove redundant return, continue, enum declaration
|
|
|
b458d7a2
|
2019-10-30T15:13:55
|
|
Readability: remove redundant cast to the same type
|
|
|
ed469fa5
|
2019-10-23T09:36:41
|
|
Fixed bug 4842 - Redundant condition in MS_ADPCM_Decode and IMA_ADPCM_Decode
(Thanks!)
|
|
|
aef1ed4a
|
2019-09-25T15:40:27
|
|
audio: Set (something close to) the correct silence value for U16 audio.
Partially fixes Bugzilla #4805.
|
|
|
693755f0
|
2019-09-25T15:07:07
|
|
coreaudio: Apple doesn't support U16 data, so convert in that case.
|
|
|
70dc8d16
|
2019-08-30T08:55:20
|
|
Android: fix corresponding warnings
|
|
|
455944c8
|
2019-08-22T16:12:16
|
|
Fixed whitespace
|
|
|
b521df66
|
2019-08-22T16:09:42
|
|
[SDL][IOS] Audio fix - applies stream to sound data when resampling or reformatting is required.
|
|
|
05f35c24
|
2019-08-19T21:23:47
|
|
Fix audio conversion U16_to_F32_SSE2 (bug 4186)
|
|
|
1d220401
|
2019-08-19T20:35:02
|
|
Fixed bug 4186 - ARM/NEON audio converters cause strange clicking noises
reverse the order when storing ouput buffer
|
|
|
c0fc94f2
|
2019-08-19T16:57:15
|
|
Fixed bug 4186 - ARM/NEON audio converters cause strange clicking noises
reverse the order when storing ouput buffer
|
|
|
4953e050
|
2019-07-31T05:11:40
|
|
use SDL_zeroa at more places where the argument is an array.
|
|
|
7a47c292
|
2019-07-31T01:22:02
|
|
Fix bug 4746 - introduce SDL_zeroa macro.
|
|
|
fdc67c3c
|
2019-07-31T00:10:00
|
|
MS_ADPCM_Decode: fix assigning an array to a pointer (lose '&').
|
|
|
680e7937
|
2019-07-07T09:10:56
|
|
Fixed bug 4710 - audio/alsa: avoid configuring hardware parameters with only a single period
Anthony Pesch
The previous code first configured the period size using snd_pcm_hw_par-
ams_set_period_size_near. Then, it further narrowed the configuration
space by calling snd_pcm_hw_params_set_buffer_size_near using a buffer
size of 2 times the _requested_ period size in order to try and get a
configuration with only 2 periods. If the configured period size was
larger than the requested size, the second call could inadvertently
narrow the configuration space to contain only a single period.
Rather than fixing the call to snd_pcm_hw_params_set_buffer_size_near
to use a size of 2 times the configured period size, the code has been
changed to use snd_pcm_hw_params_set_periods_min in order to more
clearly explain the intent.
|
|
|
14e8b93e
|
2019-06-18T14:24:24
|
|
Fixed compiler warning
|
|
|
90e2dc98
|
2019-06-14T18:23:51
|
|
A few minor changes to placate static analysis.
|
|
|
289d1092
|
2019-06-14T16:52:42
|
|
audio: Attempt to fix build on ARM versions of Visual Studio.
|
|
|
33b235f4
|
2019-06-14T15:52:48
|
|
audio: Fix ARM NEON audio converter bugs.
(Patch from Sylvain, I'm just applying it.)
Fixes Bugzilla #4186.
|
|
|
5c56c888
|
2019-06-14T15:47:32
|
|
audio: patched to compile.
|
|
|
5bd9b8b1
|
2019-06-14T09:51:22
|
|
Check src alignment for S32_to_F32 conversions
|
|
|
2fa33d6f
|
2019-06-12T15:43:08
|
|
wave: Fixed static analysis warning about dead assignment.
(technically, this function never returns an error at this point, but since
it _does_ have an "uhoh, is this corrupt data?" comment that it ignores, we
should probably make sure we handle error cases in the future. :) )
|
|
|
cd011bb1
|
2019-06-12T10:42:02
|
|
SDL_Wave: missing field 'length' initializer
|
|
|
254eb677
|
2019-06-11T02:08:31
|
|
windows: Don't let Visual Studio insert an implicit dependency on memset().
Fixes Bugzilla #4662.
|
|
|
d8da33c0
|
2019-06-10T08:49:26
|
|
Fixed bug 4662 - SDL failed to build due to error LNK2019: unresolved external symbol _memset referenced in function _IMA_ADPCM_Decode with MSVC on Windows
LinGao
We build SDL with Visual studio 2017 compiler on Windows Server 2016, but it failed to build due to error LNK2019: unresolved external symbol _memset referenced in function _IMA_ADPCM_Decode on latest default branch. And we found that it can be first reproduced on ca7283111ad0 changeset. Could you please help have a look about this issue? Thanks in advance!
|
|
|
762b788f
|
2019-06-09T12:46:10
|
|
Cleanup on bug 3894 - Fuzzing crashes for SDL_LoadWAV
Simon Hug
Attached is a minor cleanup patch. It changes the option name of one hint to something better, puts one or two more checks in, and adds explicit casting where warnings could appear otherwise.
I hope the naming of the hints and their options is acceptable. It would be kind of awkward to change them after they get released with an official SDL version.
|
|
|
a21b5b30
|
2019-06-08T19:09:43
|
|
Fixed build
|
|
|
990e166a
|
2019-06-08T19:02:42
|
|
Fixed bug 3894 - Fuzzing crashes for SDL_LoadWAV
Simon Hug
I had a look at this and made some additions to SDL_wave.c.
The attached patch adds many checks and error messages. For some reason I also added A-law and ?-law decoders. Forgot exactly why... but hey, they're small.
The WAVE format is seriously underspecified (at least by the documents that are publicly available on the internet) and it's a shame Microsoft never put something better out there. The language used in them is so loose at times, it's not surprising the encoders and decoders behave very differently. The Windows Media Player doesn't even support MS ADPCM correctly.
The patch also adds some hints to make the decoder more strict at the cost of compatibility with weird WAVE files.
I still think it needs a bit of cleaning up (Not happy with the MultiplySize function. Don't like the name and other SDL code may want to use something like this too.) and some duplicated code may be folded together. It does work in this state and I have thrown all kinds of WAVE files at it. The AFL files also pass with it and some even play (obviously just noise). Crafty little fuzzer.
Any critique would be welcome. I have a fork of SDL with a audio-loadwav branch over here if someone wants to use the commenting feature of Bitbucket:
https://bitbucket.org/ChliHug/SDL
I also cobbled some Lua scripts together to create WAVE test files:
https://bitbucket.org/ChliHug/gendat
|
|
|
31765242
|
2019-06-08T18:22:18
|
|
Fixed bug 4294 - Audio: perform more validation on conversion request
janisozaur
There are many cases which are not able to be handled by SDL's audio conversion routines, including too low (negative) rate, too high rate (impossible to allocate).
This patch aims to report such issues early and handle others in a graceful manner. The "INT32_MAX / RESAMPLER_SAMPLES_PER_ZERO_CROSSING" value is the conservative approach in terms of what can _technically_ be supported, but its value is 4'194'303, or just shy of 4.2MHz. I highly doubt any sane person would use such rates, especially in SDL2, so I would like to drive this limit further down, but would need some assistance to do that, as doing so would have to introduce an arbitrary value. Are you OK with such approach? What would a good value be? Wikipedia (https://en.wikipedia.org/wiki/High-resolution_audio) lists 96kHz as the highest sampling rate in use, even if I quadruple it for a good measure, to 384kHz it's still an order of magnitude lower than 4MHz.
|
|
|
3f19a6d5
|
2019-06-08T18:07:58
|
|
CVE-2019-7578: Fix a buffer overread in InitIMA_ADPCM
If IMA ADPCM format chunk was too short, InitIMA_ADPCM() parsing it
could read past the end of chunk data. This patch fixes it.
CVE-2019-7578
https://bugzilla.libsdl.org/show_bug.cgi?id=4494
Signed-off-by: Petr P?sa? <ppisar@redhat.com>
|
|
|
8a37848d
|
2019-06-08T13:41:46
|
|
Fixed bug 4605 - WASAPI_WaitDevice hang
Matt Brocklehurst
We've noticed that if you are playing audio on Windows via the WASAPI interface and you unplug and reconnect the device a few times the program hangs.
We've debugged the problem down to
static void
WASAPI_WaitDevice(_THIS)
{
... snip ...
if (WaitForSingleObjectEx(this->hidden->event, INFINITE, FALSE) == WAIT_OBJECT_0) {
... snip ...
}
This WaitForSingleObjectEx does not havbe a time out defined, so it hangs there forever.
Our suggested fix we found was to include a time out of say 200mSec
We have done quite a bit of testing with this fix in place on various hardware configurations and it seems to have resolved the issue.
|
|
|
15bae953
|
2019-06-08T13:03:36
|
|
Fixed bug 4642 - Rework SDL_netbsdaudio to improve performance
Nia Alarie
The NetBSD audio driver has a few problems. Lots of obsolete code, and extremely bad performance and stuttering.
I have a patch in NetBSD's package system to improve it. This is my attempt to upstream it.
The changes include:
* Removing references to defines which are never used.
* Using the correct structures for playback and recording, previously they were the wrong way around.
* Using the correct types ('struct audio_prinfo' in contrast to 'audio_prinfo')
* Removing the use of non-blocking I/O, as suggested in #3177.
* Removing workarounds for driver bugs on systems that don't exist or use this driver any more.
* Removing all usage of SDL_Delay(1)
* Removing pointless use of AUDIO_INITINFO and tests that expect AUDIO_SETINFO to fail when it can't.
These changes bring its performance in line with the DSP audio driver.
|
|
|
03cf2416
|
2019-06-08T10:21:38
|
|
OpenSL ES audio cleanup and added a note with low latency audio discussion
|
|
|
166d15fd
|
2019-06-07T15:09:15
|
|
Fixed surround sound channel setup for Android OpenSL ES audio driver
|
|
|
723d0143
|
2019-06-04T17:32:15
|
|
Fixed bug 4171 - SDL_GetQueuedAudioSize is broken with WASAPI
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.
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f3e76ea1
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2019-05-23T13:47:30
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Use the OpenSL ES audio driver by default on Android, as it has the lowest latency.
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02f9667a
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2019-05-23T13:47:27
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Fixed static and buzzing when trying to use floating point audio on the OpenSL ES audio driver.
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abcfe804
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2019-05-14T14:20:54
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[SDL] iOS fix bug with audio interrupted by a phone call not restoring.
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2fbfe8b9
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2019-03-25T12:59:30
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coreaudio: Set audio callback thread priority.
Fixes Bugzilla #4155.
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6a3356ab
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2019-03-25T12:24:38
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Backed out changeset cec31de4e126
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.
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35255342
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2019-03-16T18:48:21
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Fixed bug 4525 - Fix crash in ALSA_HotplugThread caused by bad return value check
Anthony Pesch
Fix snd_device_name_hint return value check
According to the ALSA documentation, snd_device_name_hint returns 0 on
success, otherwise a negative error code. The code previously only
considered -1 to be an error, which let other error codes through
resulting in a segfault when hints (which was NULL) was dereferenced
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03cbac40
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2019-02-05T15:14:15
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Android/openslES: fix warnings, comment out un-used interface
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614c8aea
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2019-02-05T15:09:41
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Android/openslES: set number of buffers of DATALOCATOR to internal NUM_BUFFER
If we increase NUM_BUFFER, Enqueue won't fail with SL_RESULT_BUFFER_INSUFFICIENT
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bf823bf2
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2019-02-05T15:05:32
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Android/openslES: prevent to run out of buffers if Enqueue() fails.
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3b4e3693
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2019-01-29T12:21:22
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Emscripten: No need for Runtime. for dynCalls
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53ead95e
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2019-01-29T12:19:36
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Emscripten: Avoid SDL2 in JS global scope
After this fix, closure works with the LLVM wasm backend on SDL2.
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1b24b2ec
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2019-01-14T22:56:57
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Android/openslES: fix Pause/ResumeDevices when openslES is not used
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647b1f6a
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2019-01-14T14:36:13
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Android/openslES: check for non NULL variable, some intialization.
use the previous naming
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7b1cc441
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2019-01-14T14:31:06
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Android/openslES: start playing, after creating ressources
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