kmx git

Commit	Date	Message
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
56ec349d	2018-09-29T16:48:15	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.)
7f9854b9	2018-09-25T01:45:12	WinRT: Wait until audio device activation is complete and PrepDevice during OpenAudio
5febdfce	2018-09-24T11:49:25	Fixed whitespace
623a6def	2018-08-07T16:49:18	alsa: optionally run entire pipeline non-blocking.
56f44cfa	2018-08-07T13:04:15	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.
ff8c62f2	2018-07-02T03:53:57	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.
4773690d	2018-06-25T12:55:23	Deal with possible malloc(0) calls, as pointed out by static analysis.
1d25135b	2018-06-01T19:43:53	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.
8325df25	2018-05-24T07:30:24	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.
101544d6	2018-05-21T12:05:17	audio: Needed to fix two more instances for Visual Studio.
49881861	2018-05-21T11:54:09	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.)
b7e88aaa	2018-05-16T02:03:06	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!
cb0e614f	2018-05-15T02:29:35	audio: SSE2 float-to-int converters should clamp input. The scalar versions already do this.
a07e5815	2018-05-15T01:40:05	audio: Fix range on float-to-int data clamping. I can't tell if there was a good reason for this or it was just me getting numbers wrong due to exhaustion.
7832cb65	2018-05-15T01:35:53	audio: float to int converters should clamp inclusively. If we have to test if a sample is > 1.0f anyhow, we might as well use this to avoid the unnecessary multiplication when it's == 1.0f, too. (etc).
e2ec1eb1	2018-05-15T01:04:11	audio: converting int32 to/from float shouldn't use doubles. The concern is that a massive int sample, like 0x7FFFFFFF, won't fit in a float32, which doesn't have enough bits to hold a whole number this large, just to divide it to get a value between 0 and 1. Previously we would convert to double, to get more bits, do the division, and cast back to a float, but this is expensive. Casting to double is more accurate, but it's 2x to 3x slower. Shifting out the least significant byte of an int32, so it'll definitely fit in a float, and dividing by 0x7FFFFF is still accurate to about 5 decimal places, and the difference doesn't appear to be perceptable.
c5914295	2018-05-04T21:21:32	alsa: avoid hardware parameters with an excessive number of periods. The previous code attempted to use set_buffer_size / set_period_size discretely, favoring the parameters which generated a buffer size that was exactly 2x the requested buffer size. This solution ultimately prioritizes only the buffer size, which comes at a large performance cost on some machines where this results in an excessive number of periods. In my case, for a 4096 sample buffer, this configured the device to use 37 periods with a period size of 221 samples and a buffer size of 8192 samples. With 37 periods, the SDL Audio thread was consuming 25% of the CPU. This code has been refactored to use set_period_size and set_buffer_size together. set_period_size is called first to attempt to set the period to exactly match the requested buffer size, and set_buffer_size is called second to further refine the parameters to attempt to use only 2 periods. The fundamental change here is that the period size / count won't go to extreme values if the buffer size can't be exactly matched, the buffer size should instead just increase to the next closest multiple of the target period size that is supported. After changing this, for a 4096 sample buffer, the device is configured to use 3 periods with a period size of 4096 samples and a buffer size of 12288 samples. With only 3 periods, the SDL Audio thread doesn't even show up when profiling. Fixes Bugzilla #4156.
f521b22e	2018-04-23T22:07:56	Added SDL_THREAD_PRIORITY_TIME_CRITICAL
dc8b55e5	2018-04-16T02:11:09	coreaudio: Use the standard SDL audio thread instead of spinning a new one. Fixes corner cases, like the audio callback not firing if the device is disconnected, etc.
99a0c0f0	2018-02-24T08:23:44	Fixed MinGW-w64 build
7e1fa0ce	2018-02-21T21:34:35	wasapi: fixed typo in an assert message.
c7e43665	2018-02-21T21:34:06	wasapi: let Windows do the resampling for us if possible.
97494f53	2018-02-17T18:30:21	pulseaudio: Just read/dump captured data in FlushCapture. Apparently pa_stream_flush() doesn't work as expected: https://lists.freedesktop.org/archives/pulseaudio-discuss/2012-April/013328.html Fixes Bugzilla #4087.
40b27fd5	2018-02-12T17:00:00	revert the recent typecast assignment changes (see bug #4079) also change the void* typedefs for the two vulkan function pointers added in vulkan_internal.h into generic function pointer typedefs.
ba0ecc67	2018-02-12T10:47:00	fix building SDL_audiotypecvt.c with gcc < 4.0
90e72bf4	2018-01-30T18:08:34	Fixed ISO C99 compatibility SDL now builds with gcc 7.2 with the following command line options: -Wall -pedantic-errors -Wno-deprecated-declarations -Wno-overlength-strings --std=c99
48882401	2018-01-22T09:36:40	wasapi: Fixed some compiler warnings.
e3cc5b2c	2018-01-03T10:03:25	Updated copyright for 2018
77bb49b7	2017-12-31T03:34:16	wasapi: Patched to compile on non-UWP WinRT builds.
ab4695f4	2017-12-13T14:35:55	wasapi: switched to event-driven interface. This reduces latency and improves battery life.
351d6d47	2017-12-06T12:24:32	audio: Port WASAPI to WinRT, remove XAudio2 backend. 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
e830ef34	2017-10-20T16:53:42	Fixed typo converting 4 channel audio to 2 channel
9a291c1e	2017-10-20T14:51:22	Added a note about adjusting channel weights when converting to fewer channels
72932906	2017-10-19T18:05:42	audio: Added SDL_AudioStreamFlush().
e98920f5	2017-10-18T23:49:46	Check correct variable for malloc() results.
afefcbfe	2017-10-18T19:30:47	Fixed bug 3876 - Resampling of certain sounds adds heavy distortion Simon Hug Patch that adds [-1, 1] clamping to the scalar audio type conversions. This may come from the SDL_Convert_F32_to_X_Scalar functions. They don't clamp the float value to [-1, 1] and when they cast it to the target integer it may be too large or too small for the type and get truncated, causing horrible noise. The attached patch throws clamping in, but I don't know if that's the preferred way to fix this. For x86 (without SSE) the compiler (I tested MSVC) seems to throw a horrible amount of x87 code in it. It's a bit better with SSE, but probably still quite the performance hit. And SSE2 uses a branchless approach with maxss and minss.

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

56ec349d

2018-09-29T16:48:15

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.)

7f9854b9

2018-09-25T01:45:12

WinRT: Wait until audio device activation is complete and PrepDevice during OpenAudio

5febdfce

2018-09-24T11:49:25

Fixed whitespace

623a6def

2018-08-07T16:49:18

alsa: optionally run entire pipeline non-blocking.

56f44cfa

2018-08-07T13:04:15

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.

ff8c62f2

2018-07-02T03:53:57

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.

4773690d

2018-06-25T12:55:23

Deal with possible malloc(0) calls, as pointed out by static analysis.

1d25135b

2018-06-01T19:43:53

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.

8325df25

2018-05-24T07:30:24

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.

101544d6

2018-05-21T12:05:17

audio: Needed to fix two more instances for Visual Studio.

49881861

2018-05-21T11:54:09

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.)

b7e88aaa

2018-05-16T02:03:06

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!

cb0e614f

2018-05-15T02:29:35

audio: SSE2 float-to-int converters should clamp input. The scalar versions already do this.

a07e5815

2018-05-15T01:40:05

audio: Fix range on float-to-int data clamping. I can't tell if there was a good reason for this or it was just me getting numbers wrong due to exhaustion.

7832cb65

2018-05-15T01:35:53

audio: float to int converters should clamp inclusively. If we have to test if a sample is > 1.0f anyhow, we might as well use this to avoid the unnecessary multiplication when it's == 1.0f, too. (etc).

e2ec1eb1

2018-05-15T01:04:11

audio: converting int32 to/from float shouldn't use doubles. The concern is that a massive int sample, like 0x7FFFFFFF, won't fit in a float32, which doesn't have enough bits to hold a whole number this large, just to divide it to get a value between 0 and 1. Previously we would convert to double, to get more bits, do the division, and cast back to a float, but this is expensive. Casting to double is more accurate, but it's 2x to 3x slower. Shifting out the least significant byte of an int32, so it'll definitely fit in a float, and dividing by 0x7FFFFF is still accurate to about 5 decimal places, and the difference doesn't appear to be perceptable.

c5914295

2018-05-04T21:21:32

alsa: avoid hardware parameters with an excessive number of periods. The previous code attempted to use set_buffer_size / set_period_size discretely, favoring the parameters which generated a buffer size that was exactly 2x the requested buffer size. This solution ultimately prioritizes only the buffer size, which comes at a large performance cost on some machines where this results in an excessive number of periods. In my case, for a 4096 sample buffer, this configured the device to use 37 periods with a period size of 221 samples and a buffer size of 8192 samples. With 37 periods, the SDL Audio thread was consuming 25% of the CPU. This code has been refactored to use set_period_size and set_buffer_size together. set_period_size is called first to attempt to set the period to exactly match the requested buffer size, and set_buffer_size is called second to further refine the parameters to attempt to use only 2 periods. The fundamental change here is that the period size / count won't go to extreme values if the buffer size can't be exactly matched, the buffer size should instead just increase to the next closest multiple of the target period size that is supported. After changing this, for a 4096 sample buffer, the device is configured to use 3 periods with a period size of 4096 samples and a buffer size of 12288 samples. With only 3 periods, the SDL Audio thread doesn't even show up when profiling. Fixes Bugzilla #4156.

f521b22e

2018-04-23T22:07:56

Added SDL_THREAD_PRIORITY_TIME_CRITICAL

dc8b55e5

2018-04-16T02:11:09

coreaudio: Use the standard SDL audio thread instead of spinning a new one. Fixes corner cases, like the audio callback not firing if the device is disconnected, etc.

99a0c0f0

2018-02-24T08:23:44

Fixed MinGW-w64 build

7e1fa0ce

2018-02-21T21:34:35

wasapi: fixed typo in an assert message.

c7e43665

2018-02-21T21:34:06

wasapi: let Windows do the resampling for us if possible.

97494f53

2018-02-17T18:30:21

pulseaudio: Just read/dump captured data in FlushCapture. Apparently pa_stream_flush() doesn't work as expected: https://lists.freedesktop.org/archives/pulseaudio-discuss/2012-April/013328.html Fixes Bugzilla #4087.

40b27fd5

2018-02-12T17:00:00

revert the recent typecast assignment changes (see bug #4079) also change the void* typedefs for the two vulkan function pointers added in vulkan_internal.h into generic function pointer typedefs.

ba0ecc67

2018-02-12T10:47:00

fix building SDL_audiotypecvt.c with gcc < 4.0

90e72bf4

2018-01-30T18:08:34

Fixed ISO C99 compatibility SDL now builds with gcc 7.2 with the following command line options: -Wall -pedantic-errors -Wno-deprecated-declarations -Wno-overlength-strings --std=c99

48882401

2018-01-22T09:36:40

wasapi: Fixed some compiler warnings.

e3cc5b2c

2018-01-03T10:03:25

77bb49b7

2017-12-31T03:34:16

wasapi: Patched to compile on non-UWP WinRT builds.

ab4695f4

2017-12-13T14:35:55

wasapi: switched to event-driven interface. This reduces latency and improves battery life.

351d6d47

2017-12-06T12:24:32

audio: Port WASAPI to WinRT, remove XAudio2 backend. 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

e830ef34

2017-10-20T16:53:42

Fixed typo converting 4 channel audio to 2 channel

9a291c1e

2017-10-20T14:51:22

Added a note about adjusting channel weights when converting to fewer channels

72932906

2017-10-19T18:05:42

audio: Added SDL_AudioStreamFlush().

e98920f5

2017-10-18T23:49:46

Check correct variable for malloc() results.

afefcbfe

2017-10-18T19:30:47

Fixed bug 3876 - Resampling of certain sounds adds heavy distortion Simon Hug Patch that adds [-1, 1] clamping to the scalar audio type conversions. This may come from the SDL_Convert_F32_to_X_Scalar functions. They don't clamp the float value to [-1, 1] and when they cast it to the target integer it may be too large or too small for the type and get truncated, causing horrible noise. The attached patch throws clamping in, but I don't know if that's the preferred way to fix this. For x86 (without SSE) the compiler (I tested MSVC) seems to throw a horrible amount of x87 code in it. It's a bit better with SSE, but probably still quite the performance hit. And SSE2 uses a branchless approach with maxss and minss.

kc3-lang/SDL/src/audio

src/audio

Log