kmx git

Commit	Date	Message
ac8a3fda	2021-03-03T20:33:20	fix prepare_audiospec() possibly missing a bad SDL_AUDIO_CHANNELS env.
4de0c74a	2021-03-02T10:02:59	audio: pipewire: Add the application name to the stream properties
9ed01da7	2021-03-02T09:47:47	audio: pipewire: Constify and clarify period size calculations Constify the min/max period variables, use a #define for the base clock rate used in the calculations and note that changing the upper limit can have dire side effects as it's a hard limit in Pipewire.
d7ca855c	2021-03-02T09:33:11	audio: pipewire: Add missing static qualifiers to globals
84c44e01	2021-03-01T12:39:52	audio: pipewire: fix uninitialized variable warnings
2f0b99a7	2021-02-13T11:56:05	audio: Add Pipewire playback/capture sink
7001b531	2021-02-27T12:53:08	audio: pipewire: Add vim format lines to files and fix indentation Increase indentation spacing from 2 to 4 to comply with style standards.
2fcba50e	2021-02-27T12:08:15	audio: pipewire: Code and comment cleanups Replace "magic numbers" with #defines, explain the requirements when using the userdata pointer in the node_object struct and a few other minor code and comment cleanups.
4eadd147	2021-02-25T14:00:23	audio: pipewire: Fix outdated comment
cd56f1b3	2021-02-24T14:36:58	audio: pipewire: Use "rear" designation for rear channels Use the 'R' (rear) prefixed designations for the rear audio channels instead of 'S' (surround). Surround designated channels are only used in the 8 channel configuration.
adc0a931	2021-02-24T14:08:08	audio: Move Pipewire bootstrap after Jack Move the Pipewire audio driver below others in the list so it won't be mistakenly initialized when it's not the system mixer.
21adec93	2021-02-24T12:02:54	audio: pipewire: Make enumeration structure and function names more descriptive Rename the add/remove/clear list functions and rename connected_device to io_node, as a sink/source node isn't necessarily a device.
a07f5434	2021-02-21T13:24:20	audio: pipewire: Report default devices first Further refactor the device enumeration code to retrieve the default sink/source node IDs from the metadata node. Use the retrieved IDs to sort the device list so that the default devices are at the beginning and thus are the first reported to SDL.
9afd7570	2021-02-20T13:33:12	audio: pipewire: Always buffer source audio The latency of source nodes can change depending on the overall latency of the processing graph. Incoming audio must therefore always be buffered to ensure uninterrupted delivery. The SDL_AudioStream path was removed in the input callback as the only thing it was used for was buffering audio outside of Pipewire's min/max period sizes, and that case is now handled by the omnipresent buffer.
106dc009	2021-02-19T17:18:36	audio: pipewire: Pass proper parameter to user audio callback The audio callbacks should pass the callbackspec.userdata parameter to the callback, not spec.userdata Co-authored-by: Oschowa <Oschowa@web.de>
f3ebbc06	2021-02-19T16:02:20	audio: pipewire: Retrieve the channel count and default sample rate for sinks/sources Extend device enumeration to retrieve the channel count and default sample rate for sink and source nodes. This required a fairly significant rework of the enumeration procedure as multiple callbacks are involved now. Sink/source nodes are tracked in a separate list during the enumeration process so they can be cleaned up if a device is removed before completion. These changes also simplify any future efforts that may be needed to retrieve additional configuration information from the nodes.
08547adb	2021-02-20T09:28:03	pulseaudio: Add "zerocopy" playback path
ef85ed93	2021-02-17T20:53:35	pulseaudio: Initialize fragsize to fix mic recording fragsize wasn't initialized, and it is used for recording. If the value was 0 or -1, pulseaudio configures it itself. But sometimes we can get a random (and large) value that makes pulseaudio give us large sample at a very low frequency. https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/blob/master/src/pulse/def.h#L453 https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/blob/v13.0/src/pulsecore/protocol-native.c#L409
20be1d63	2020-07-15T09:13:03	emscripten: Automatically resume audio contexts This uses the mechanism added in emscripten-core/emscripten#10843 which was applied to SDL1 and OpenAL. This adds the same for SDL2. This also reverts commit 865eaddffed50dbd13e6564c3f73902472cf74e8 which did something similar, but the new mechanism is more effective.
f443a6fc	2021-02-11T02:05:02	Fix format string warnings for width-based integers The DJGPP compiler emits many warnings for conflicts between print format specifiers and argument types. To fix the warnings, I added `SDL_PRIx32` macros for use with `Sint32` and `Uint32` types. The macros alias those found in <inttypes.h> or fallback to a reasonable default. As an alternative, print arguments could be cast to plain old integers. I opted slightly for the current solution as it felt more technically correct, despite making the format strings more verbose.
b852590b	2021-01-24T17:02:40	minor clean-up in SDL_os2audio.c
8f102589	2021-01-23T17:28:10	os2audio: changed backend name from MMOS2 to DART (like SDL-1.2)
50ea3b77	2021-01-08T10:09:37	Fixed bug 5080 - SDL_netbsdaudio: Always use the device's preferred frequency Nia Alarie The NetBSD kernel's audio resampling code is much simpler and lower quality than libsamplerate. Presumably, if SDL always performs I/O on the audio device in its native frequency, we can avoid resampling audio in the kernel and let SDL do it with libsamplerate instead.
265a1cc9	2021-01-05T15:50:02	use WIN_StringToUTF8W instead of WIN_StringToUTF8 where needed (#2) cf. bug #5435. - SDL_wasapi_win32.c (GetWasapiDeviceName): pwszVal is WCHAR* - windows/SDL_sysfilesystem.c (SDL_GetBasePath, SDL_GetPrefPath) - windows/SDL_sysurl.c (SDL_SYS_OpenURL): wurl is WCHAR* - SDL_windowssensor.c (ConnectSensor): bstr_name is WCHAR* - windows/SDL_systhread.c (SDL_SYS_SetupThread): strw is WCHAR*
ed39f2f3	2021-01-04T01:23:50	SDL_wasapi_win32.c (WASAPI_PlatformThreadInit): use L instead of TEXT() because AvSetMmThreadCharacteristicsW specifically accepts WCHAR* input cf. bug #5435.
01a2f276	2021-01-04T01:23:50	consistently use TEXT() macro with LoadLibrary() and GetModuleHandle() cf. bug #5435.
9130f7c3	2021-01-02T10:25:38	Updated copyright for 2021
8476df3e	2020-12-30T23:55:10	SDL_mixer.c: remove calls to non-existing m68k asm code.
cb361896	2020-12-09T07:16:22	Fixed bug 5235 - All internal sources should include SDL_assert.h Ryan C. Gordon We should really stick this in SDL_internal.h or something so it's always available.
53b16679	2020-11-11T12:33:55	SIZE_MAX need not be defined in limits.h it can be in limits.h (windows) or stdint.h.
1b8dee7c	2020-10-31T11:32:40	coreaudio: Remove unnecessary include of CoreServices.h
a4040293	2020-10-25T10:10:02	os2: misc build fixes
bfc80d83	2020-10-25T03:55:02	minor coding style cleanup
a90f0400	2020-10-15T21:37:30	os2: a _lot_ of coding style cleanup, sot that they match the SDL style. also renamed the 'debug' macro to debug_os2: the former was dangerously a common name. the binary (dll) output is precisely the same as before.
d2723875	2020-10-14T23:01:06	os2: integrate the port into main tree.
1d9cf23e	2020-10-14T23:01:05	os2: updated copyright dates for 2020. header guard fixes.
a3d7913c	2020-10-14T23:01:05	SDL_os2audio.c (OS2_OpenDevice): change spec->samples assignment: Original code assigned MCIMixSetup.ulSamplesPerSec value to it, but it is just the freq... We now change spec->samples only either if it is 0 or we changed the frequency, by picking a default of ~46 ms at desired frequency (code taken from SDL_audio.c:prepare_audiospec()). With this, the crashes I have been experiencing are gone.
e112b776	2020-10-14T23:01:05	SDL_os2audio.c (OS2_OpenDevice): change {0} initializers to SDL_zero()
72594e25	2020-10-14T23:01:04	SDL_os2audio.c (OS2_OpenDevice): remove assignment to wrong spec member Correct assignment to 'format' member is done below, already.
222f0268	2020-10-14T23:01:03	os/2: port from SDL2-2.0.4 to SDL2-2.0.5: changes to SDL_os2audio.c, SDL_os2video.c, os2/SDL_systhread.c in order to accomodate SDL2-2.0.5 changes. - audio: WaitDone() is gone, CloseDevice() interface changes. - events / video: DropFile() changes: SDL_DROPBEGIN and SDL_DROPCOMPLETE events, window IDs for drops. - thread: struct SDL_Thread->stacksize
aa790837	2020-10-14T23:01:02	os2: several warning fixes. mostly those "W007: '&array' may not produce intended result" warnings from Watcom, visible only in C++ mode. one or two others here & there.
c2188619	2020-10-14T23:01:01	os2: added a 2-byte padding to os2 SDL_PrivateAudioData
74cfb81d	2020-10-14T23:01:00	os2: add port files for SDL2-2.0.4 from Andrey Vasilkin only geniconv/iconv.h (was from LGPL libiconv) is replaced with a generic minimal iconv.h based on public knowledge.
003a1698	2020-10-06T11:07:50	wav: Make sure the data size is a multiple of blockalign, not an exact match. I _think_ this is a right thing to do; it fixes a .wav file I have here that has blockalign==2 when channels==2 and bitspersample==16, which otherwise would fail.
554037a6	2020-09-26T09:30:08	audio: fix popping sounds caused by signed/unsigned conversion When converting audio from signed to unsigned values of vice-versa the silence value chosen by SDL was the value of the device, not of the stream that the data was being put into. After conversion this would lead to a very high or low value, making the speaker jump to a extreme positon, leading to an audible noise whenever creating, destroying or playing scilence on a device that reqired such conversion.
7ef188a1	2020-09-19T14:01:57	jack: Fixed memory leak on device close.
a69c61fb	2020-08-14T12:08:58	Only assign context and mainloop once we have connected successfully If we fail to connect to the the pa server, we have an assigned context and mainloop that isn't connected. So, when PULSEAUDIO_pa_context_disconnect is called, pa asserts and crashes the application. Assertion 'pa_atomic_load(&(c)->_ref) >= 1' failed at pulse/context.c:1055, function pa_context_disconnect(). Aborting.
ff53521b	2020-06-04T12:26:57	Fixed Bluetooth audio output on Apple TV
68777406	2020-05-20T16:58:33	windows: Fix calls to CoCreateInstance() so last parameter is a LPVOID *.
8601996f	2020-05-03T22:13:48	hints: Allow specifying audio device metadata. This is only supported on PulseAudio. You can set a description when opening your audio device that will show up in pauvcontrol, which lets you set per-stream volume levels. Fixes Bugzilla #4801.
a990a34a	2020-04-14T22:26:02	Cleanly switch between audio recording, playback, and both, on iOS
2ae1c0f5	2020-04-14T09:52:27	Allow Bluetooth headphones for iOS playandrecord mode
fba081e4	2020-04-07T14:51:08	wasapi: Patched to compile on C89 systems, and use SDL_ceilf instead of ceilf.
4c2be472	2020-04-07T14:37:24	wasapi: Improve WASAPI audio backend latency (thanks, Anthony!). Anthony Pesch's notes on his patch: "Currently, the WASAPI backend creates a stream in shared mode and sets the device's callback size to be half of the shared stream's total buffer size. This works, but doesn't coordinate will with the actual hardware. The hardware will raise an interrupt after every period which in turn will signal the object being waited on inside of WaitDevice. From my empirical testing, the callback size was often larger than the period size and not a multiple of it, which resulted in poor latency when trying to time an application based on the audio callback. The reason for this looked something like: * The device's callback would be called and and the audio buffer was filled. * WaitDevice would be called. * The hardware would raise an interrupt after one period. * WaitDevice would resume, see that a a full callback had not been played and then wait again. * The hardware would raise an interrupt after another period. * WaitDevice would resume, see that a full callback + some extra amount had been played and then it would again call our callback and this process would repeat. The effect of this is that the pacing between subsequent callbacks is poor - sometimes it's called very quickly, sometimes it's called very late. By matching the callback's size to the stream's period size, the pacing of calls to the user callback is improved substantially. I didn't write an actual test for this, but my use case for this was my Dreamcast emulator (https://redream.io) which uses the audio callback to help drive the emulation speed. Without this change and with the default shared stream buffer (which has a period of ~10ms) I would get frame times that were between ~3-30 milliseconds; after this change I get frame times of ~11-22 milliseconds. Note, this patch also has a change that removes passing a duration to the Initialize call. It seems that the default duration used (when 0 is passed) does typically match up with the duration returned by GetDevicePeriod, however the Initialize docs say: > To set the buffer to the minimum size required by the engine thread, the > client should call Initialize with the hnsBufferDuration parameter set to 0. > Following the Initialize call, the client can get the size of the resulting > buffer by calling IAudioClient::GetBufferSize. This change isn't strictly required, but I made it to hopefully rule out another source of unexpected latency." Fixes Bugzilla #4592.
b6afbe63	2020-04-07T09:38:57	Added SDL_log.h to SDL_internal.h so logging is available everywhere
9525f972	2020-04-05T10:44:51	Fixed bug 5076 - SDL_netbsdaudio: Add support for 32-bit LPCM Nia Alarie The kernel supports this, make SDL expose it so it can be used.
f3e60967	2020-04-02T12:27:29	Fixed setting the "playandrecord" audio hint on Apple TV The Apple TV doesn't have record capability by default, so activating the audio session with AVAudioSessionCategoryPlayAndRecord fails.
55b4f18e	2020-03-29T01:54:00	coreaudio: The default SDL audio device now tracks the system output default. So if you go into System Preferences on a MacBook and toggle between a pair of connected bluetooth headphones and built-in internal speakers, SDL will switch the device it is playing sound through, to match this setting, on the fly. Likewise if the default output device is a USB thing and is unplugged; as the default device changes at the system level, SDL will pick this up and carry on with the new default. This is different from our unplug detection for specific devices, as in those cases we want to send the app a disconnect notification, instead of migrating transparently as we now do for default devices. Note that this should also work for capture devices; if the device changes, SDL will start recording from the new default. Fixes Bugzilla #4851.
abdc5cbf	2020-03-26T19:30:17	Allow background music to play in the "play and record" case on iOS
27889d02	2020-03-03T12:31:41	winrt: Wait for EnumerationCompleted before leaving WASAPI_EnumerateEndpoints
e3b0713e	2020-02-24T12:07:18	Don't call setPreferredOutputNumberOfChannels on iOS, it breaks audio output
2c9871a4	2020-02-24T10:25:57	Fixed surround sound support on Apple TV
f4e23553	2020-02-14T15:19:34	Fixed audio not coming out of the phone speakers while recording on iOS
922b3dc3	2020-02-14T14:18:12	Fixed re-setting the audio session category when closing an audio device
14bf532d	2020-02-13T16:10:52	Fixed opening audio on Android from the Steam Link shell activity
4bb95e84	2020-02-11T16:14:02	Implemented OpenSL-ES audio recording on Android
b1c6e7c2	2020-01-23T00:32:34	Fixed compile warning
f30ef6ed	2020-01-21T17:40:16	audio: Fixed a '//' style comment.
dbe5c14b	2020-01-21T15:49:37	audio: Calculate a legitimate SDL_AudioSpec::silence in SDL_LoadWAV_RW().
a8780c6a	2020-01-16T20:49:25	Updated copyright date for 2020
3ce56f62	2020-01-13T08:12:10	Fixed error formatting
e3cedf96	2020-01-11T04:38:13	Add the destination format to the error when conversion isn't possible
3da6a0b2	2019-12-03T03:53:06	pulseaudio: don't let FlushCapture get stuck in an infinite loop on shutdown. Fixes Bugzilla #4645.
60d3965e	2019-10-30T15:36:17	Readability: remove redundant return, continue, enum declaration
b458d7a2	2019-10-30T15:13:55	Readability: remove redundant cast to the same type
ed469fa5	2019-10-23T09:36:41	Fixed bug 4842 - Redundant condition in MS_ADPCM_Decode and IMA_ADPCM_Decode (Thanks!)
aef1ed4a	2019-09-25T15:40:27	audio: Set (something close to) the correct silence value for U16 audio. Partially fixes Bugzilla #4805.
693755f0	2019-09-25T15:07:07	coreaudio: Apple doesn't support U16 data, so convert in that case.
70dc8d16	2019-08-30T08:55:20	Android: fix corresponding warnings
455944c8	2019-08-22T16:12:16	Fixed whitespace
b521df66	2019-08-22T16:09:42	[SDL][IOS] Audio fix - applies stream to sound data when resampling or reformatting is required.
05f35c24	2019-08-19T21:23:47	Fix audio conversion U16_to_F32_SSE2 (bug 4186)
1d220401	2019-08-19T20:35:02	Fixed bug 4186 - ARM/NEON audio converters cause strange clicking noises reverse the order when storing ouput buffer
c0fc94f2	2019-08-19T16:57:15	Fixed bug 4186 - ARM/NEON audio converters cause strange clicking noises reverse the order when storing ouput buffer
4953e050	2019-07-31T05:11:40	use SDL_zeroa at more places where the argument is an array.
7a47c292	2019-07-31T01:22:02	Fix bug 4746 - introduce SDL_zeroa macro.
fdc67c3c	2019-07-31T00:10:00	MS_ADPCM_Decode: fix assigning an array to a pointer (lose '&').
680e7937	2019-07-07T09:10:56	Fixed bug 4710 - audio/alsa: avoid configuring hardware parameters with only a single period Anthony Pesch The previous code first configured the period size using snd_pcm_hw_par- ams_set_period_size_near. Then, it further narrowed the configuration space by calling snd_pcm_hw_params_set_buffer_size_near using a buffer size of 2 times the _requested_ period size in order to try and get a configuration with only 2 periods. If the configured period size was larger than the requested size, the second call could inadvertently narrow the configuration space to contain only a single period. Rather than fixing the call to snd_pcm_hw_params_set_buffer_size_near to use a size of 2 times the configured period size, the code has been changed to use snd_pcm_hw_params_set_periods_min in order to more clearly explain the intent.
14e8b93e	2019-06-18T14:24:24	Fixed compiler warning
90e2dc98	2019-06-14T18:23:51	A few minor changes to placate static analysis.
289d1092	2019-06-14T16:52:42	audio: Attempt to fix build on ARM versions of Visual Studio.
33b235f4	2019-06-14T15:52:48	audio: Fix ARM NEON audio converter bugs. (Patch from Sylvain, I'm just applying it.) Fixes Bugzilla #4186.
5c56c888	2019-06-14T15:47:32	audio: patched to compile.
5bd9b8b1	2019-06-14T09:51:22	Check src alignment for S32_to_F32 conversions
2fa33d6f	2019-06-12T15:43:08	wave: Fixed static analysis warning about dead assignment. (technically, this function never returns an error at this point, but since it _does_ have an "uhoh, is this corrupt data?" comment that it ignores, we should probably make sure we handle error cases in the future. :) )
cd011bb1	2019-06-12T10:42:02	SDL_Wave: missing field 'length' initializer
254eb677	2019-06-11T02:08:31	windows: Don't let Visual Studio insert an implicit dependency on memset(). Fixes Bugzilla #4662.
d8da33c0	2019-06-10T08:49:26	Fixed bug 4662 - SDL failed to build due to error LNK2019: unresolved external symbol _memset referenced in function _IMA_ADPCM_Decode with MSVC on Windows LinGao We build SDL with Visual studio 2017 compiler on Windows Server 2016, but it failed to build due to error LNK2019: unresolved external symbol _memset referenced in function _IMA_ADPCM_Decode on latest default branch. And we found that it can be first reproduced on ca7283111ad0 changeset. Could you please help have a look about this issue? Thanks in advance!
762b788f	2019-06-09T12:46:10	Cleanup on bug 3894 - Fuzzing crashes for SDL_LoadWAV Simon Hug Attached is a minor cleanup patch. It changes the option name of one hint to something better, puts one or two more checks in, and adds explicit casting where warnings could appear otherwise. I hope the naming of the hints and their options is acceptable. It would be kind of awkward to change them after they get released with an official SDL version.
a21b5b30	2019-06-08T19:09:43	Fixed build

ac8a3fda

2021-03-03T20:33:20

fix prepare_audiospec() possibly missing a bad SDL_AUDIO_CHANNELS env.

4de0c74a

2021-03-02T10:02:59

audio: pipewire: Add the application name to the stream properties

9ed01da7

2021-03-02T09:47:47

audio: pipewire: Constify and clarify period size calculations Constify the min/max period variables, use a #define for the base clock rate used in the calculations and note that changing the upper limit can have dire side effects as it's a hard limit in Pipewire.

d7ca855c

2021-03-02T09:33:11

audio: pipewire: Add missing static qualifiers to globals

84c44e01

2021-03-01T12:39:52

audio: pipewire: fix uninitialized variable warnings

2f0b99a7

2021-02-13T11:56:05

audio: Add Pipewire playback/capture sink

7001b531

2021-02-27T12:53:08

audio: pipewire: Add vim format lines to files and fix indentation Increase indentation spacing from 2 to 4 to comply with style standards.

2fcba50e

2021-02-27T12:08:15

audio: pipewire: Code and comment cleanups Replace "magic numbers" with #defines, explain the requirements when using the userdata pointer in the node_object struct and a few other minor code and comment cleanups.

4eadd147

2021-02-25T14:00:23

audio: pipewire: Fix outdated comment

cd56f1b3

2021-02-24T14:36:58

audio: pipewire: Use "rear" designation for rear channels Use the 'R' (rear) prefixed designations for the rear audio channels instead of 'S' (surround). Surround designated channels are only used in the 8 channel configuration.

adc0a931

2021-02-24T14:08:08

audio: Move Pipewire bootstrap after Jack Move the Pipewire audio driver below others in the list so it won't be mistakenly initialized when it's not the system mixer.

21adec93

2021-02-24T12:02:54

audio: pipewire: Make enumeration structure and function names more descriptive Rename the add/remove/clear list functions and rename connected_device to io_node, as a sink/source node isn't necessarily a device.

a07f5434

2021-02-21T13:24:20

audio: pipewire: Report default devices first Further refactor the device enumeration code to retrieve the default sink/source node IDs from the metadata node. Use the retrieved IDs to sort the device list so that the default devices are at the beginning and thus are the first reported to SDL.

9afd7570

2021-02-20T13:33:12

audio: pipewire: Always buffer source audio The latency of source nodes can change depending on the overall latency of the processing graph. Incoming audio must therefore always be buffered to ensure uninterrupted delivery. The SDL_AudioStream path was removed in the input callback as the only thing it was used for was buffering audio outside of Pipewire's min/max period sizes, and that case is now handled by the omnipresent buffer.

106dc009

2021-02-19T17:18:36

audio: pipewire: Pass proper parameter to user audio callback The audio callbacks should pass the callbackspec.userdata parameter to the callback, not spec.userdata Co-authored-by: Oschowa <Oschowa@web.de>

f3ebbc06

2021-02-19T16:02:20

audio: pipewire: Retrieve the channel count and default sample rate for sinks/sources Extend device enumeration to retrieve the channel count and default sample rate for sink and source nodes. This required a fairly significant rework of the enumeration procedure as multiple callbacks are involved now. Sink/source nodes are tracked in a separate list during the enumeration process so they can be cleaned up if a device is removed before completion. These changes also simplify any future efforts that may be needed to retrieve additional configuration information from the nodes.

08547adb

2021-02-20T09:28:03

pulseaudio: Add "zerocopy" playback path

ef85ed93

2021-02-17T20:53:35

pulseaudio: Initialize fragsize to fix mic recording fragsize wasn't initialized, and it is used for recording. If the value was 0 or -1, pulseaudio configures it itself. But sometimes we can get a random (and large) value that makes pulseaudio give us large sample at a very low frequency. https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/blob/master/src/pulse/def.h#L453 https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/blob/v13.0/src/pulsecore/protocol-native.c#L409

20be1d63

2020-07-15T09:13:03

emscripten: Automatically resume audio contexts This uses the mechanism added in emscripten-core/emscripten#10843 which was applied to SDL1 and OpenAL. This adds the same for SDL2. This also reverts commit 865eaddffed50dbd13e6564c3f73902472cf74e8 which did something similar, but the new mechanism is more effective.

f443a6fc

2021-02-11T02:05:02

Fix format string warnings for width-based integers The DJGPP compiler emits many warnings for conflicts between print format specifiers and argument types. To fix the warnings, I added `SDL_PRIx32` macros for use with `Sint32` and `Uint32` types. The macros alias those found in <inttypes.h> or fallback to a reasonable default. As an alternative, print arguments could be cast to plain old integers. I opted slightly for the current solution as it felt more technically correct, despite making the format strings more verbose.

b852590b

2021-01-24T17:02:40

minor clean-up in SDL_os2audio.c

8f102589

2021-01-23T17:28:10

os2audio: changed backend name from MMOS2 to DART (like SDL-1.2)

50ea3b77

2021-01-08T10:09:37

Fixed bug 5080 - SDL_netbsdaudio: Always use the device's preferred frequency Nia Alarie The NetBSD kernel's audio resampling code is much simpler and lower quality than libsamplerate. Presumably, if SDL always performs I/O on the audio device in its native frequency, we can avoid resampling audio in the kernel and let SDL do it with libsamplerate instead.

265a1cc9

2021-01-05T15:50:02

use WIN_StringToUTF8W instead of WIN_StringToUTF8 where needed (#2) cf. bug #5435. - SDL_wasapi_win32.c (GetWasapiDeviceName): pwszVal is WCHAR* - windows/SDL_sysfilesystem.c (SDL_GetBasePath, SDL_GetPrefPath) - windows/SDL_sysurl.c (SDL_SYS_OpenURL): wurl is WCHAR* - SDL_windowssensor.c (ConnectSensor): bstr_name is WCHAR* - windows/SDL_systhread.c (SDL_SYS_SetupThread): strw is WCHAR*

ed39f2f3

2021-01-04T01:23:50

SDL_wasapi_win32.c (WASAPI_PlatformThreadInit): use L instead of TEXT() because AvSetMmThreadCharacteristicsW specifically accepts WCHAR* input cf. bug #5435.

01a2f276

2021-01-04T01:23:50

consistently use TEXT() macro with LoadLibrary() and GetModuleHandle() cf. bug #5435.

9130f7c3

2021-01-02T10:25:38

Updated copyright for 2021

8476df3e

2020-12-30T23:55:10

SDL_mixer.c: remove calls to non-existing m68k asm code.

cb361896

2020-12-09T07:16:22

Fixed bug 5235 - All internal sources should include SDL_assert.h Ryan C. Gordon We should really stick this in SDL_internal.h or something so it's always available.

53b16679

2020-11-11T12:33:55

SIZE_MAX need not be defined in limits.h it can be in limits.h (windows) or stdint.h.

1b8dee7c

2020-10-31T11:32:40

coreaudio: Remove unnecessary include of CoreServices.h

a4040293

2020-10-25T10:10:02

os2: misc build fixes

bfc80d83

2020-10-25T03:55:02

minor coding style cleanup

a90f0400

2020-10-15T21:37:30

os2: a _lot_ of coding style cleanup, sot that they match the SDL style. also renamed the 'debug' macro to debug_os2: the former was dangerously a common name. the binary (dll) output is precisely the same as before.

d2723875

2020-10-14T23:01:06

os2: integrate the port into main tree.

1d9cf23e

2020-10-14T23:01:05

os2: updated copyright dates for 2020. header guard fixes.

a3d7913c

2020-10-14T23:01:05

SDL_os2audio.c (OS2_OpenDevice): change spec->samples assignment: Original code assigned MCIMixSetup.ulSamplesPerSec value to it, but it is just the freq... We now change spec->samples only either if it is 0 or we changed the frequency, by picking a default of ~46 ms at desired frequency (code taken from SDL_audio.c:prepare_audiospec()). With this, the crashes I have been experiencing are gone.

e112b776

2020-10-14T23:01:05

SDL_os2audio.c (OS2_OpenDevice): change {0} initializers to SDL_zero()

72594e25

2020-10-14T23:01:04

SDL_os2audio.c (OS2_OpenDevice): remove assignment to wrong spec member Correct assignment to 'format' member is done below, already.

222f0268

2020-10-14T23:01:03

os/2: port from SDL2-2.0.4 to SDL2-2.0.5: changes to SDL_os2audio.c, SDL_os2video.c, os2/SDL_systhread.c in order to accomodate SDL2-2.0.5 changes. - audio: WaitDone() is gone, CloseDevice() interface changes. - events / video: DropFile() changes: SDL_DROPBEGIN and SDL_DROPCOMPLETE events, window IDs for drops. - thread: struct SDL_Thread->stacksize

aa790837

2020-10-14T23:01:02

os2: several warning fixes. mostly those "W007: '&array' may not produce intended result" warnings from Watcom, visible only in C++ mode. one or two others here & there.

c2188619

2020-10-14T23:01:01

os2: added a 2-byte padding to os2 SDL_PrivateAudioData

74cfb81d

2020-10-14T23:01:00

os2: add port files for SDL2-2.0.4 from Andrey Vasilkin only geniconv/iconv.h (was from LGPL libiconv) is replaced with a generic minimal iconv.h based on public knowledge.

003a1698

2020-10-06T11:07:50

wav: Make sure the data size is a multiple of blockalign, not an exact match. I _think_ this is a right thing to do; it fixes a .wav file I have here that has blockalign==2 when channels==2 and bitspersample==16, which otherwise would fail.

554037a6

2020-09-26T09:30:08

audio: fix popping sounds caused by signed/unsigned conversion When converting audio from signed to unsigned values of vice-versa the silence value chosen by SDL was the value of the device, not of the stream that the data was being put into. After conversion this would lead to a very high or low value, making the speaker jump to a extreme positon, leading to an audible noise whenever creating, destroying or playing scilence on a device that reqired such conversion.

7ef188a1

2020-09-19T14:01:57

jack: Fixed memory leak on device close.

a69c61fb

2020-08-14T12:08:58

Only assign context and mainloop once we have connected successfully If we fail to connect to the the pa server, we have an assigned context and mainloop that isn't connected. So, when PULSEAUDIO_pa_context_disconnect is called, pa asserts and crashes the application. Assertion 'pa_atomic_load(&(c)->_ref) >= 1' failed at pulse/context.c:1055, function pa_context_disconnect(). Aborting.

ff53521b

2020-06-04T12:26:57

Fixed Bluetooth audio output on Apple TV

68777406

2020-05-20T16:58:33

windows: Fix calls to CoCreateInstance() so last parameter is a LPVOID *.

8601996f

2020-05-03T22:13:48

hints: Allow specifying audio device metadata. This is only supported on PulseAudio. You can set a description when opening your audio device that will show up in pauvcontrol, which lets you set per-stream volume levels. Fixes Bugzilla #4801.

a990a34a

2020-04-14T22:26:02

Cleanly switch between audio recording, playback, and both, on iOS

2ae1c0f5

2020-04-14T09:52:27

Allow Bluetooth headphones for iOS playandrecord mode

fba081e4

2020-04-07T14:51:08

wasapi: Patched to compile on C89 systems, and use SDL_ceilf instead of ceilf.

4c2be472

2020-04-07T14:37:24

wasapi: Improve WASAPI audio backend latency (thanks, Anthony!). Anthony Pesch's notes on his patch: "Currently, the WASAPI backend creates a stream in shared mode and sets the device's callback size to be half of the shared stream's total buffer size. This works, but doesn't coordinate will with the actual hardware. The hardware will raise an interrupt after every period which in turn will signal the object being waited on inside of WaitDevice. From my empirical testing, the callback size was often larger than the period size and not a multiple of it, which resulted in poor latency when trying to time an application based on the audio callback. The reason for this looked something like: * The device's callback would be called and and the audio buffer was filled. * WaitDevice would be called. * The hardware would raise an interrupt after one period. * WaitDevice would resume, see that a a full callback had not been played and then wait again. * The hardware would raise an interrupt after another period. * WaitDevice would resume, see that a full callback + some extra amount had been played and then it would again call our callback and this process would repeat. The effect of this is that the pacing between subsequent callbacks is poor - sometimes it's called very quickly, sometimes it's called very late. By matching the callback's size to the stream's period size, the pacing of calls to the user callback is improved substantially. I didn't write an actual test for this, but my use case for this was my Dreamcast emulator (https://redream.io) which uses the audio callback to help drive the emulation speed. Without this change and with the default shared stream buffer (which has a period of ~10ms) I would get frame times that were between ~3-30 milliseconds; after this change I get frame times of ~11-22 milliseconds. Note, this patch also has a change that removes passing a duration to the Initialize call. It seems that the default duration used (when 0 is passed) does typically match up with the duration returned by GetDevicePeriod, however the Initialize docs say: > To set the buffer to the minimum size required by the engine thread, the > client should call Initialize with the hnsBufferDuration parameter set to 0. > Following the Initialize call, the client can get the size of the resulting > buffer by calling IAudioClient::GetBufferSize. This change isn't strictly required, but I made it to hopefully rule out another source of unexpected latency." Fixes Bugzilla #4592.

b6afbe63

2020-04-07T09:38:57

Added SDL_log.h to SDL_internal.h so logging is available everywhere

9525f972

2020-04-05T10:44:51

Fixed bug 5076 - SDL_netbsdaudio: Add support for 32-bit LPCM Nia Alarie The kernel supports this, make SDL expose it so it can be used.

f3e60967

2020-04-02T12:27:29

Fixed setting the "playandrecord" audio hint on Apple TV The Apple TV doesn't have record capability by default, so activating the audio session with AVAudioSessionCategoryPlayAndRecord fails.

55b4f18e

2020-03-29T01:54:00

coreaudio: The default SDL audio device now tracks the system output default. So if you go into System Preferences on a MacBook and toggle between a pair of connected bluetooth headphones and built-in internal speakers, SDL will switch the device it is playing sound through, to match this setting, on the fly. Likewise if the default output device is a USB thing and is unplugged; as the default device changes at the system level, SDL will pick this up and carry on with the new default. This is different from our unplug detection for specific devices, as in those cases we want to send the app a disconnect notification, instead of migrating transparently as we now do for default devices. Note that this should also work for capture devices; if the device changes, SDL will start recording from the new default. Fixes Bugzilla #4851.

abdc5cbf

2020-03-26T19:30:17

Allow background music to play in the "play and record" case on iOS

27889d02

2020-03-03T12:31:41

winrt: Wait for EnumerationCompleted before leaving WASAPI_EnumerateEndpoints

e3b0713e

2020-02-24T12:07:18

Don't call setPreferredOutputNumberOfChannels on iOS, it breaks audio output

2c9871a4

2020-02-24T10:25:57

Fixed surround sound support on Apple TV

f4e23553

2020-02-14T15:19:34

Fixed audio not coming out of the phone speakers while recording on iOS

922b3dc3

2020-02-14T14:18:12

Fixed re-setting the audio session category when closing an audio device

14bf532d

2020-02-13T16:10:52

Fixed opening audio on Android from the Steam Link shell activity

4bb95e84

2020-02-11T16:14:02

Implemented OpenSL-ES audio recording on Android

b1c6e7c2

2020-01-23T00:32:34

Fixed compile warning

f30ef6ed

2020-01-21T17:40:16

audio: Fixed a '//' style comment.

dbe5c14b

2020-01-21T15:49:37

audio: Calculate a legitimate SDL_AudioSpec::silence in SDL_LoadWAV_RW().

a8780c6a

2020-01-16T20:49:25

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

kc3-lang/SDL/src/audio

src/audio

Log