kmx git

Commit	Date	Message
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.
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).

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.

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).

kc3-lang/SDL/src/audio

src/audio

Log