kmx git

Commit	Date	Message
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.
653ab5d9	2017-10-18T19:26:36	Added a staging buffer to the audio stream so that we can accumulate small amounts of data if needed when resampling
80f8464d	2017-10-18T15:54:05	Added audio stream conversion functions: SDL_NewAudioStream SDL_AudioStreamPut SDL_AudioStreamGet SDL_AudioStreamAvailable SDL_AudioStreamClear SDL_FreeAudioStream
fa156741	2017-10-13T01:15:29	coreaudio: changed device close procedure to prevent long hangs in some cases. The audioqueue thread needs to keep running, and processing the CFRunLoop until the AudioQueue is disposed of, otherwise CoreAudio will hang waiting for final data to feed the device. At least, I think this is how it all works. It definitely fixes the bug here! Since AudioQueueDispose() calls AudioQueueStop() internally, there's no need for our thread to handle this, either, which is good because the AudioQueue would be disposed by this point. So now the AudioQueue is disposed first, and then our thread is joined, and everything works out okay. Just in case, we mark the device "paused" before setting everything in motion, so any further callbacks from CoreAudio will write silence and not fire the app's audio callback again. Fixes Bugzilla #3868.
ba10d2b6	2017-10-12T13:55:35	Fixed compiler warning
5e5f2290	2017-10-11T12:07:43	audio: Turns out the accumulation errors sound better. :/ Moving to double fixed the overflows, but using "time = i * incr" instead of "time += incr" causes clicks in the output.
9bd2c6b4	2017-10-11T11:51:14	audio: Moved the resampler state up to double precision. Fixes more buffer overflows.
b2f5123b	2017-10-11T11:43:35	audio: calculate resampling time directly, don't increment (thanks, Eric!). Fixes buffer overruns as floating point errors accumulate. Partially fixes Bugzilla #3848.
763c3871	2017-10-11T02:33:55	audio: clamp resampler interpolation values to prevent buffer overflow. Partially fixes Bugzilla #3848.
0085f917	2017-10-11T02:31:58	audio: Moved unchanging variable out of loop.
cb8bf6bb	2017-10-11T02:03:05	audio: Make sure audio stream resampling doesn't overflow buffers.
459e2b0b	2017-10-11T01:37:11	audio: Fixed check for minimum audio stream put size.
903ff641	2017-10-10T22:31:02	audio: SDL_ResampleCVT() should use memmove instead of memcpy. This copy can overlap. Fixes Bugzilla #3849.
42fff7ce	2017-10-10T22:18:46	audio: Don't stack-allocate resampler padding. (I thought padding size ranged from 5 frames to ~30 frames (based around RESAMPLER_ZERO_CROSSINGS, which is 5), but it's actually between 512 and several thousands (based on RESAMPLER_SAMPLES_PER_ZERO_CROSSING)). It gets big fast when downsampling.
37d89aa1	2017-10-10T16:12:56	audio: reworked audio streams to have right-hand resampling padding available. Fixes Bugzilla #3851.
099ae43e	2017-09-22T22:28:21	audio: Fixed compiler warning on Visual Studio.
fe6b8f1c	2017-09-22T11:25:52	Fixed Mac OS X build
407e1693	2017-09-22T11:15:14	Fixed audio being silent on older iOS devices Tested on an iPod running iOS 6.1
d74c00e6	2017-09-22T08:51:45	Fixed memory leak when HAVE_ALLOCA isn't defined
6d206a7b	2017-09-22T07:42:24	audio: Stream resampling now saves some samples from previous run for padding. Previously, the padding was silence, which was a problem when streaming since you would sample a little bit of this silence between each buffer. We still need a means to get padding data for the right hand side, but this patch makes the resampler output more correct.
8b660c50	2017-09-21T00:55:29	Added some missing "extern" declarations
1a3b95a1	2017-09-21T02:51:14	audio: Replaced the resampler. Again. This time it's using real math from a real whitepaper instead of my previous amateur, fast-but-low-quality attempt. The new resampler does "bandlimited interpolation," as described here: https://ccrma.stanford.edu/~jos/resample/ The output appears to sound cleaner, especially at high frequencies, and of course works with non-power-of-two rate conversions. There are some obvious optimizations to be done to this still, and there is other fallout: this doesn't resample a buffer in-place, the 2-channels-Sint16 fast path is gone because this resampler does a _lot_ of floating point math. There is a nasty hack to make it work with SDL_AudioCVT. It's possible these issues are solvable, but they aren't solved as of yet. Still, I hope this effort is slouching in the right direction.
f40bd5ee	2017-09-21T02:06:53	audio: removed my perl experiment script.
c08a7a74	2017-09-15T17:27:32	Added a hint SDL_HINT_AUDIO_CATEGORY to control the audio category, determining whether the phone mute switch affects the audio
93583d46	2017-09-09T21:17:46	alsa: removed snd_pcm_wait() call before writing to playback device. This would cause playback problems in certain situations, such as on the Raspberry Pi. The device that the wait was added for seems to not benefit from it in modern times, and standard desktop Linux seems to do the right thing when a USB device is unplugged now, without this patch. Fixes Bugzilla #3599.
ca15c7d6	2017-09-07T10:56:08	wave: SDL_LoadWAV now supports 24-bit audio.
3267398d	2017-09-02T16:41:14	sndio: Patched to compile if SIO_DEVANY isn't defined. (It isn't in whatever Raspbian is currently shipping.)
d619d885	2017-08-28T21:42:39	Fixed bug 3662 - Error message when using the audio conversion setup without an initialized audio subsystem is a bit vague Simon Hug This issue actually raises the question if this API change (requirement of initialized audio subsystem) is breaking backwards compatibility. I don't see the documentation saying it is needed in 2.0.5.
b128e880	2017-08-29T00:41:45	audio: A whole bunch of improvements to audio conversion (thanks, Solra!). "Major changes, roughly in order of appearance: - Use float math everywhere, instead of promoting to double and casting back all the time. - Conserve sound energy when downmixing any channel into two other channels. - Add a QuadToStereo filter. (The previous technique of reusing StereoToMono never worked, since it assumed an incorrect channel layout for 4.0.) - Add a 71to51 filter. This removes just under half of the cases the previous code would silently break in. - Add a QuadTo51 filter. More silent breakage fixed. - Add a 51to71 filter, removing another almost-half of the silently broken cases. - Add 8 to the list of values SDL_SupportedChannelCount will accept. - Change SDL_BuildAudioCVT's channel-related logic to handle every case, and to actually fail if it fails instead of silently corrupting sound data and/or crashing down the road." (Note that SDL doesn't otherwise support 7.1 audio yet, but hopefully it will soon and the 7.1 converters are an important piece of that. --ryan.) Fixes Bugzilla #3727.
a0cd7d6b	2017-08-29T00:02:04	audio: Converting audio samples from int to float was using wrong equation. Fixes Bugzilla #3775.
6dd3f55d	2017-08-28T01:59:53	Fixed WinRT build after changing the header guard preprocessor symbol
0d011ec6	2017-08-28T00:22:23	Renaming of guard header names to quiet -Wreserved-id-macro
8e7998e1	2017-08-27T19:10:30	Fixed bug 3710 - SDL_OpenAudio(desired, obtained) doesn't update desired's size when obtained is NULL David Ludwig I've created a new set of patches. I am happy to create more, if it would help. One version only copies 'size'. A second version copies both 'size' and 'silence'. When looking over the documentation for SDL_OpenAudio in SDL_audio.h, it mentioned that both 'size' and 'silence' were things that SDL_OpenAudio would calculate. Regarding both patches, I did notice that SDL 1.2 appears to have always modified desired's size and silence fields. The SDL wiki, at https://wiki.libsdl.org/SDL_OpenAudio#Remarks , does note:
bcf0e071	2017-08-18T17:29:44	Added WASAPI audio target to autoconf build process
e3e6b4fd	2017-08-18T16:52:19	audio: better docs on conversion APIs, error if not init'd (thanks, Simon!). Fixes Bugzilla #3662.
fb835f9e	2017-08-14T20:22:19	Fixed bug 2330 - Debian bug report: SDL2 X11 driver buffer overflow with large X11 file descriptor manuel.montezelo Original bug report (note that it was against 2.0.0, it might have been fixed in between): http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=733015 -------------------------------------------------------- Package: libsdl2-2.0-0 Version: 2.0.0+dfsg1-3 Severity: normal Tags: patch I have occasional crashes here caused by the X11 backend of SDL2. It seems to be caused by the X11_Pending function trying to add a high number (> 1024) file descriptor to a fd_set before doing a select on it to avoid busy waiting on X11 events. This causes a buffer overflow because the file descriptor is larger (or equal) than the limit FD_SETSIZE. Attached is a possible workaround patch. Please also keep in mind that fd_set are also used in following files which may have similar problems. src/audio/bsd/SDL_bsdaudio.c src/audio/paudio/SDL_paudio.c src/audio/qsa/SDL_qsa_audio.c src/audio/sun/SDL_sunaudio.c src/joystick/linux/SDL_sysjoystick.c -------------------------------------------------------- On Tuesday 24 December 2013 00:43:13 Sven Eckelmann wrote: > I have occasional crashes here caused by the X11 backend of SDL2. It seems > to be caused by the X11_Pending function trying to add a high number (> > 1024) file descriptor to a fd_set before doing a select on it to avoid busy > waiting on X11 events. This causes a buffer overflow because the file > descriptor is larger (or equal) than the limit FD_SETSIZE. I personally experienced this problem while hacking on the python bindings package for SDL2 [1] (while doing make runtest). But it easier to reproduce in a smaller, synthetic testcase.
96305832	2017-08-11T10:21:19	Fixed bug 3702 - Clear error messages of SDL_LoadObject for optional libraries Simon Hug Some code in SDL loads libraries with SDL_LoadObject to get more information or use newer APIs. SDL_LoadObject may fail, set an error message and SDL will continue with some fallback code. Since SDL will overwrite the error or exit the function with a return value that indicates success, the error form SDL_LoadObject for the optional stuff might as well be cleared right away.
9dde37ea	2017-08-07T00:25:18	sndio: Fix for some platforms (Linux, for example) that don't define INFTIM. Fixes Bugzilla #3712.
a09efc73	2017-08-04T16:18:34	psp: Force audio channels to stereo if > 2 channels requested (thanks, Solra!). Fixes Bugzilla #3726.
68ca9d9e	2017-07-29T23:00:45	qnx: Fixed error message.
cea33bf5	2017-07-29T23:00:34	aix: Removed unused local variable. Found by Cppcheck.
77ca0f27	2017-07-27T22:55:18	Fixed crash if the WASAPI audio device couldn't be recovered
4a734209	2017-07-27T22:52:19	Fixed infinite recursion if the WASAPI audio device couldn't be recovered
f033ce61	2017-07-27T02:41:58	Fixed typo in WASAPI shutdown code
03eaddca	2017-07-23T19:25:16	Fixed compiler warnings on QNX.
8ac17a2a	2017-07-20T20:40:17	sndio: fixed poll() call (thanks, kdrakehp!). Fixes Bugzilla #3705.
ee9cc324	2017-07-20T18:16:02	sndio: More improvements to the OpenBSD audio target (thanks, kdrakehp!). Fixes Bugzilla #3705.
2cc68064	2017-07-20T10:39:47	Fixed bug 3705 - Add capture support to the sndio backend kdrakehp The attached patch adds capture support to the sndio backend. The patch also allows the `OpenDevice' function to accept arbitrary device names.
fb9c2939	2017-07-07T23:00:10	qnx: Fixed setting a field twice.
1683a0c1	2017-07-05T12:04:37	audio: trying to pacify static analysis.
9f99b3d7	2017-07-02T22:46:49	aix: Fixed audio debug output. DEBUG_AUDIO is checked with #ifdef not #if.
4366721b	2017-07-02T22:46:00	qnx: Removed unused bootstrap declaration. QNX_bootstrap is the VideoBootStrap. QSAAUDIO_bootstrap is still there.
22241ed0	2017-07-01T17:50:47	Support for QNX 7.0 (thanks, Elad!). Fixes Bugzilla #3686.
380e0693	2017-07-01T23:01:49	aix: Fixed compile error.
4c190ce5	2017-07-01T23:00:07	netbsd: Fixed comment.
4c48260c	2017-06-29T23:00:18	netbsd: Removed unused field.
7bb6b402	2017-06-29T23:00:09	netbsd: Fixed compile error.
a509719f	2017-06-12T21:35:24	audio: Converter now checks a strict list of channels and formats we support.
553b3286	2017-06-12T16:39:15	Fixed bug 3668 - Overflow of SDL_AudioCVT.filters with some downmixes Simon Hug There's a chance that an audio conversion from many channels to a few can use more than 9 audio filters. SDL_AudioCVT has 10 SDL_AudioFilter pointers of which one has to be the terminating NULL pointer. The SDL code has no checks for this limit. If it overflows there can be stack or heap corruption or a call to 0xa. Attached patch adds a function that checks for this limit and throws an error if it is reached. Also adds some documentation. Test parameters that trigger this issue: AUDIO_U16MSB with 224 channels at 46359 Hz V AUDIO_S16MSB with 6 channels at 27463 Hz The fuzzer program I uploaded in bug 3667 has more of them.

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.

653ab5d9

2017-10-18T19:26:36

Added a staging buffer to the audio stream so that we can accumulate small amounts of data if needed when resampling

80f8464d

2017-10-18T15:54:05

Added audio stream conversion functions: SDL_NewAudioStream SDL_AudioStreamPut SDL_AudioStreamGet SDL_AudioStreamAvailable SDL_AudioStreamClear SDL_FreeAudioStream

fa156741

2017-10-13T01:15:29

coreaudio: changed device close procedure to prevent long hangs in some cases. The audioqueue thread needs to keep running, and processing the CFRunLoop until the AudioQueue is disposed of, otherwise CoreAudio will hang waiting for final data to feed the device. At least, I think this is how it all works. It definitely fixes the bug here! Since AudioQueueDispose() calls AudioQueueStop() internally, there's no need for our thread to handle this, either, which is good because the AudioQueue would be disposed by this point. So now the AudioQueue is disposed first, and then our thread is joined, and everything works out okay. Just in case, we mark the device "paused" before setting everything in motion, so any further callbacks from CoreAudio will write silence and not fire the app's audio callback again. Fixes Bugzilla #3868.

ba10d2b6

2017-10-12T13:55:35

Fixed compiler warning

5e5f2290

2017-10-11T12:07:43

audio: Turns out the accumulation errors sound better. :/ Moving to double fixed the overflows, but using "time = i * incr" instead of "time += incr" causes clicks in the output.

9bd2c6b4

2017-10-11T11:51:14

audio: Moved the resampler state up to double precision. Fixes more buffer overflows.

b2f5123b

2017-10-11T11:43:35

audio: calculate resampling time directly, don't increment (thanks, Eric!). Fixes buffer overruns as floating point errors accumulate. Partially fixes Bugzilla #3848.

763c3871

2017-10-11T02:33:55

audio: clamp resampler interpolation values to prevent buffer overflow. Partially fixes Bugzilla #3848.

0085f917

2017-10-11T02:31:58

audio: Moved unchanging variable out of loop.

cb8bf6bb

2017-10-11T02:03:05

audio: Make sure audio stream resampling doesn't overflow buffers.

459e2b0b

2017-10-11T01:37:11

audio: Fixed check for minimum audio stream put size.

903ff641

2017-10-10T22:31:02

audio: SDL_ResampleCVT() should use memmove instead of memcpy. This copy can overlap. Fixes Bugzilla #3849.

42fff7ce

2017-10-10T22:18:46

audio: Don't stack-allocate resampler padding. (I thought padding size ranged from 5 frames to ~30 frames (based around RESAMPLER_ZERO_CROSSINGS, which is 5), but it's actually between 512 and several thousands (based on RESAMPLER_SAMPLES_PER_ZERO_CROSSING)). It gets big fast when downsampling.

37d89aa1

2017-10-10T16:12:56

audio: reworked audio streams to have right-hand resampling padding available. Fixes Bugzilla #3851.

099ae43e

2017-09-22T22:28:21

audio: Fixed compiler warning on Visual Studio.

fe6b8f1c

2017-09-22T11:25:52

Fixed Mac OS X build

407e1693

2017-09-22T11:15:14

Fixed audio being silent on older iOS devices Tested on an iPod running iOS 6.1

d74c00e6

2017-09-22T08:51:45

Fixed memory leak when HAVE_ALLOCA isn't defined

6d206a7b

2017-09-22T07:42:24

audio: Stream resampling now saves some samples from previous run for padding. Previously, the padding was silence, which was a problem when streaming since you would sample a little bit of this silence between each buffer. We still need a means to get padding data for the right hand side, but this patch makes the resampler output more correct.

8b660c50

2017-09-21T00:55:29

Added some missing "extern" declarations

1a3b95a1

2017-09-21T02:51:14

audio: Replaced the resampler. Again. This time it's using real math from a real whitepaper instead of my previous amateur, fast-but-low-quality attempt. The new resampler does "bandlimited interpolation," as described here: https://ccrma.stanford.edu/~jos/resample/ The output appears to sound cleaner, especially at high frequencies, and of course works with non-power-of-two rate conversions. There are some obvious optimizations to be done to this still, and there is other fallout: this doesn't resample a buffer in-place, the 2-channels-Sint16 fast path is gone because this resampler does a _lot_ of floating point math. There is a nasty hack to make it work with SDL_AudioCVT. It's possible these issues are solvable, but they aren't solved as of yet. Still, I hope this effort is slouching in the right direction.

f40bd5ee

2017-09-21T02:06:53

audio: removed my perl experiment script.

c08a7a74

2017-09-15T17:27:32

Added a hint SDL_HINT_AUDIO_CATEGORY to control the audio category, determining whether the phone mute switch affects the audio

93583d46

2017-09-09T21:17:46

alsa: removed snd_pcm_wait() call before writing to playback device. This would cause playback problems in certain situations, such as on the Raspberry Pi. The device that the wait was added for seems to not benefit from it in modern times, and standard desktop Linux seems to do the right thing when a USB device is unplugged now, without this patch. Fixes Bugzilla #3599.

ca15c7d6

2017-09-07T10:56:08

wave: SDL_LoadWAV now supports 24-bit audio.

3267398d

2017-09-02T16:41:14

sndio: Patched to compile if SIO_DEVANY isn't defined. (It isn't in whatever Raspbian is currently shipping.)

d619d885

2017-08-28T21:42:39

Fixed bug 3662 - Error message when using the audio conversion setup without an initialized audio subsystem is a bit vague Simon Hug This issue actually raises the question if this API change (requirement of initialized audio subsystem) is breaking backwards compatibility. I don't see the documentation saying it is needed in 2.0.5.

b128e880

2017-08-29T00:41:45

audio: A whole bunch of improvements to audio conversion (thanks, Solra!). "Major changes, roughly in order of appearance: - Use float math everywhere, instead of promoting to double and casting back all the time. - Conserve sound energy when downmixing any channel into two other channels. - Add a QuadToStereo filter. (The previous technique of reusing StereoToMono never worked, since it assumed an incorrect channel layout for 4.0.) - Add a 71to51 filter. This removes just under half of the cases the previous code would silently break in. - Add a QuadTo51 filter. More silent breakage fixed. - Add a 51to71 filter, removing another almost-half of the silently broken cases. - Add 8 to the list of values SDL_SupportedChannelCount will accept. - Change SDL_BuildAudioCVT's channel-related logic to handle every case, and to actually fail if it fails instead of silently corrupting sound data and/or crashing down the road." (Note that SDL doesn't otherwise support 7.1 audio yet, but hopefully it will soon and the 7.1 converters are an important piece of that. --ryan.) Fixes Bugzilla #3727.

a0cd7d6b

2017-08-29T00:02:04

audio: Converting audio samples from int to float was using wrong equation. Fixes Bugzilla #3775.

6dd3f55d

2017-08-28T01:59:53

Fixed WinRT build after changing the header guard preprocessor symbol

0d011ec6

2017-08-28T00:22:23

Renaming of guard header names to quiet -Wreserved-id-macro

8e7998e1

2017-08-27T19:10:30

Fixed bug 3710 - SDL_OpenAudio(desired, obtained) doesn't update desired's size when obtained is NULL David Ludwig I've created a new set of patches. I am happy to create more, if it would help. One version only copies 'size'. A second version copies both 'size' and 'silence'. When looking over the documentation for SDL_OpenAudio in SDL_audio.h, it mentioned that both 'size' and 'silence' were things that SDL_OpenAudio would calculate. Regarding *both* patches, I did notice that SDL 1.2 appears to have always modified desired's size and silence fields. The SDL wiki, at https://wiki.libsdl.org/SDL_OpenAudio#Remarks , does note:

bcf0e071

2017-08-18T17:29:44

Added WASAPI audio target to autoconf build process

e3e6b4fd

2017-08-18T16:52:19

audio: better docs on conversion APIs, error if not init'd (thanks, Simon!). Fixes Bugzilla #3662.

fb835f9e

2017-08-14T20:22:19

Fixed bug 2330 - Debian bug report: SDL2 X11 driver buffer overflow with large X11 file descriptor manuel.montezelo Original bug report (note that it was against 2.0.0, it might have been fixed in between): http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=733015 -------------------------------------------------------- Package: libsdl2-2.0-0 Version: 2.0.0+dfsg1-3 Severity: normal Tags: patch I have occasional crashes here caused by the X11 backend of SDL2. It seems to be caused by the X11_Pending function trying to add a high number (> 1024) file descriptor to a fd_set before doing a select on it to avoid busy waiting on X11 events. This causes a buffer overflow because the file descriptor is larger (or equal) than the limit FD_SETSIZE. Attached is a possible workaround patch. Please also keep in mind that fd_set are also used in following files which may have similar problems. src/audio/bsd/SDL_bsdaudio.c src/audio/paudio/SDL_paudio.c src/audio/qsa/SDL_qsa_audio.c src/audio/sun/SDL_sunaudio.c src/joystick/linux/SDL_sysjoystick.c -------------------------------------------------------- On Tuesday 24 December 2013 00:43:13 Sven Eckelmann wrote: > I have occasional crashes here caused by the X11 backend of SDL2. It seems > to be caused by the X11_Pending function trying to add a high number (> > 1024) file descriptor to a fd_set before doing a select on it to avoid busy > waiting on X11 events. This causes a buffer overflow because the file > descriptor is larger (or equal) than the limit FD_SETSIZE. I personally experienced this problem while hacking on the python bindings package for SDL2 [1] (while doing make runtest). But it easier to reproduce in a smaller, synthetic testcase.

96305832

2017-08-11T10:21:19

Fixed bug 3702 - Clear error messages of SDL_LoadObject for optional libraries Simon Hug Some code in SDL loads libraries with SDL_LoadObject to get more information or use newer APIs. SDL_LoadObject may fail, set an error message and SDL will continue with some fallback code. Since SDL will overwrite the error or exit the function with a return value that indicates success, the error form SDL_LoadObject for the optional stuff might as well be cleared right away.

9dde37ea

2017-08-07T00:25:18

sndio: Fix for some platforms (Linux, for example) that don't define INFTIM. Fixes Bugzilla #3712.

a09efc73

2017-08-04T16:18:34

psp: Force audio channels to stereo if > 2 channels requested (thanks, Solra!). Fixes Bugzilla #3726.

68ca9d9e

2017-07-29T23:00:45

qnx: Fixed error message.

cea33bf5

2017-07-29T23:00:34

aix: Removed unused local variable. Found by Cppcheck.

77ca0f27

2017-07-27T22:55:18

Fixed crash if the WASAPI audio device couldn't be recovered

4a734209

2017-07-27T22:52:19

Fixed infinite recursion if the WASAPI audio device couldn't be recovered

f033ce61

2017-07-27T02:41:58

Fixed typo in WASAPI shutdown code

03eaddca

2017-07-23T19:25:16

Fixed compiler warnings on QNX.

8ac17a2a

2017-07-20T20:40:17

sndio: fixed poll() call (thanks, kdrakehp!). Fixes Bugzilla #3705.

ee9cc324

2017-07-20T18:16:02

sndio: More improvements to the OpenBSD audio target (thanks, kdrakehp!). Fixes Bugzilla #3705.

2cc68064

2017-07-20T10:39:47

Fixed bug 3705 - Add capture support to the sndio backend kdrakehp The attached patch adds capture support to the sndio backend. The patch also allows the `OpenDevice' function to accept arbitrary device names.

fb9c2939

2017-07-07T23:00:10

qnx: Fixed setting a field twice.

1683a0c1

2017-07-05T12:04:37

audio: trying to pacify static analysis.

9f99b3d7

2017-07-02T22:46:49

aix: Fixed audio debug output. DEBUG_AUDIO is checked with #ifdef not #if.

4366721b

2017-07-02T22:46:00

qnx: Removed unused bootstrap declaration. QNX_bootstrap is the VideoBootStrap. QSAAUDIO_bootstrap is still there.

22241ed0

2017-07-01T17:50:47

Support for QNX 7.0 (thanks, Elad!). Fixes Bugzilla #3686.

380e0693

2017-07-01T23:01:49

aix: Fixed compile error.

4c190ce5

2017-07-01T23:00:07

netbsd: Fixed comment.

4c48260c

2017-06-29T23:00:18

netbsd: Removed unused field.

7bb6b402

2017-06-29T23:00:09

netbsd: Fixed compile error.

a509719f

2017-06-12T21:35:24

audio: Converter now checks a strict list of channels and formats we support.

553b3286

2017-06-12T16:39:15

Fixed bug 3668 - Overflow of SDL_AudioCVT.filters with some downmixes Simon Hug There's a chance that an audio conversion from many channels to a few can use more than 9 audio filters. SDL_AudioCVT has 10 SDL_AudioFilter pointers of which one has to be the terminating NULL pointer. The SDL code has no checks for this limit. If it overflows there can be stack or heap corruption or a call to 0xa. Attached patch adds a function that checks for this limit and throws an error if it is reached. Also adds some documentation. Test parameters that trigger this issue: AUDIO_U16MSB with 224 channels at 46359 Hz V AUDIO_S16MSB with 6 channels at 27463 Hz The fuzzer program I uploaded in bug 3667 has more of them.

kc3-lang/SDL/src/audio

src/audio

Log