|
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.
|
|
|
f3e76ea1
|
2019-05-23T13:47:30
|
|
Use the OpenSL ES audio driver by default on Android, as it has the lowest latency.
|
|
|
02f9667a
|
2019-05-23T13:47:27
|
|
Fixed static and buzzing when trying to use floating point audio on the OpenSL ES audio driver.
|
|
|
abcfe804
|
2019-05-14T14:20:54
|
|
[SDL] iOS fix bug with audio interrupted by a phone call not restoring.
|
|
|
2fbfe8b9
|
2019-03-25T12:59:30
|
|
coreaudio: Set audio callback thread priority.
Fixes Bugzilla #4155.
|
|
|
6a3356ab
|
2019-03-25T12:24:38
|
|
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.
|
|
|
35255342
|
2019-03-16T18:48:21
|
|
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
|
|
|
03cbac40
|
2019-02-05T15:14:15
|
|
Android/openslES: fix warnings, comment out un-used interface
|
|
|
614c8aea
|
2019-02-05T15:09:41
|
|
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
|
|
|
bf823bf2
|
2019-02-05T15:05:32
|
|
Android/openslES: prevent to run out of buffers if Enqueue() fails.
|
|
3b4e3693
|
2019-01-29T12:21:22
|
|
Emscripten: No need for Runtime. for dynCalls
|
|
|
53ead95e
|
2019-01-29T12:19:36
|
|
Emscripten: Avoid SDL2 in JS global scope
After this fix, closure works with the LLVM wasm backend on SDL2.
|
|
|
1b24b2ec
|
2019-01-14T22:56:57
|
|
Android/openslES: fix Pause/ResumeDevices when openslES is not used
|
|
|
647b1f6a
|
2019-01-14T14:36:13
|
|
Android/openslES: check for non NULL variable, some intialization.
use the previous naming
|
|
|
7b1cc441
|
2019-01-14T14:31:06
|
|
Android/openslES: start playing, after creating ressources
|
|
|
955d8789
|
2019-01-14T12:33:29
|
|
Android/openslES: set audio in paused/resumed state for Android event loop
And also in "stopped" state before closing the device.
|
|
|
59c8c7b6
|
2019-01-14T10:58:57
|
|
Android/openslES: move a few static variables to SDL_PrivateAudioData structure
|
|
|
5aeeaaab
|
2019-01-14T10:16:26
|
|
Android/openslES: register and use CloseDevice function.
|
|
|
365fd9c6
|
2019-01-14T10:04:54
|
|
Android/openslES: some space and indentation to match SDL conventions
|
|
|
7dc92a76
|
2019-01-12T12:18:44
|
|
Initial Android OpenSL ES implementation, contributed by ANTA
|
|
|
d23c2f07
|
2019-01-10T18:05:56
|
|
Fixed bug 3930 - Android, set thread priorities and names
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).
|
|
|
5e13087b
|
2019-01-04T22:01:14
|
|
Updated copyright for 2019
|
|
|
aea7e56a
|
2018-12-04T12:34:45
|
|
android: use __ARM_NEON instead of __ARM_NEON__ to include <arm_neon.h>
Only __ARM_NEON is defined with Android NDK and arm64-v8a
Tested on ndk-r18, ndk-r13 and also Xcode.
(Visual Studio needs a different fix).
Fixes Bugzilla #4409.
|
|
1f6bd951
|
2018-11-15T18:22:30
|
|
Emscripten: make CloseAudio actually close audio
cf. https://bugzilla.libsdl.org/show_bug.cgi?id=4176
|
|
91820998
|
2018-10-28T21:36:48
|
|
Add and update include guards
Include guards in most changed files were missing, I added them keeping
the same style as other SDL files. In some cases I moved the include
guards around to be the first thing the header has to take advantage of
any possible improvements compiler may have for inclusion guards.
|
|
|
4a50a042
|
2018-10-21T22:40:17
|
|
wasapi/win32: Sort initial device lists by device GUID.
This makes an unchanged set of hardware always report devices in the same
order on each run.
|
|
|
04cbf132
|
2018-10-10T15:20:56
|
|
audio: All device names reported by SDL must be unique.
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.
|
|
|
0378529e
|
2018-10-10T14:55:24
|
|
audio: clean_out_device_list() already sets this flag to false for us.
|
|
|
f5a21ebf
|
2018-10-09T20:12:43
|
|
Added support for surround sound and float audio on Android
|
|
|
b2518761
|
2018-10-01T09:47:10
|
|
commit c6b28f46b8116552ec2b38d1d3c8535df28ba7a1
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
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56ec349d
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2018-09-29T16:48:15
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audio: disable NEON converters for now.
To be revisited after 2.0.9 ships!
(doesn't fix Bugzilla #4186, but stops the regression for the time being.)
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7f9854b9
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2018-09-25T01:45:12
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WinRT: Wait until audio device activation is complete and PrepDevice during OpenAudio
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5febdfce
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2018-09-24T11:49:25
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Fixed whitespace
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623a6def
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2018-08-07T16:49:18
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alsa: optionally run entire pipeline non-blocking.
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56f44cfa
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2018-08-07T13:04:15
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audio: Deal with device shutdown more carefully.
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.
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ff8c62f2
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2018-07-02T03:53:57
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Fixed bug 4210 - SSE2-based converter makes junk result of S32 -> Float
At the HG state abdd17144682, 64-bit assemblies are using SSE2-based resampler, produces junk sound when converting the S32 -> Float32 -> S16 chain. The `NEED_SCALAR_CONVERTER_FALLBACKS` thing works perfectly.
If I will find a reason that caused this mistake, I'll send a patch by myself.
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4773690d
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2018-06-25T12:55:23
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Deal with possible malloc(0) calls, as pointed out by static analysis.
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1d25135b
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2018-06-01T19:43:53
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Fixed bug 4184 - jack audio driver fails in presence of midi ports
Martin ?irokov
Launching an SDL application with SDL_AUDIODRIVER=jack, and then calling SDL_OpenAudioDevice() with whatever parameters fails with an error like this one:
SDL_OpenAudioDevice: Couldn't connect JACK ports: SDL:sdl_jack_output_0 => system:midi_playback_1
This happens because JACK_OpenDevice in src/audio/jack/SDL_jackaudio.c blindly tries to connect to all input ports without checking whether they are for audio or midi.
The fix is to check port types and ignore all non audio ports. Also I removed devports field from struct SDL_PrivateAudioData, because it's never really used and removing unused ports from it would be PITA.
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8325df25
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2018-05-24T07:30:24
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Fixed bug 4169 - Crash due to audio session observer race condition
Jona
The following explains why this bug was happening:
This crash was caused because the audio session was being set as active [session setActive:YES error:&err] when the audio device was actually being CLOSED. Certain cases the audio session being set to active would fail and the method would return right away. Because of the way the error was handled we never removed the SDLInterruptionListener thus leaking it. Later when an interruption was received the THIS_ object would contain a pointer to an already released device causing the crash.
The fix:
When only one device remained open and it was being closed we needed to set the audio session as NOT active and completely ignore the returned error to successfully release the SDLInterruptionListener. I think the user assumed that the open_playback_devices and open_capture_devices would equal 0 when all of them where closed but the truth is that at the end of the closing process that the open devices count is decremented.
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101544d6
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2018-05-21T12:05:17
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audio: Needed to fix two more instances for Visual Studio.
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49881861
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2018-05-21T11:54:09
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audio: Patched to compile on Visual Studio.
(It gets upset at the -2147483648, thinking this should be an unsigned value
because 2147483648 is too large for an int32, so the negative sign upsets the
compiler.)
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b7e88aaa
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2018-05-16T02:03:06
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audio: Added ARM NEON versions of audio converters.
These are _much_ faster than the scalar equivalents on the Raspberry Pi that
I tested on. Often 3x to 4x as fast!
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cb0e614f
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2018-05-15T02:29:35
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audio: SSE2 float-to-int converters should clamp input.
The scalar versions already do this.
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