|
db56526f
|
2021-06-10T13:07:27
|
|
audiocvt: Replace divisions
Division is expensive and the compiler can not replace it themselves.
Though we now we will be ok in terms of floating point edge behaviour.
|
|
|
b2c8d3e9
|
2021-06-11T22:14:54
|
|
audiocvt: 5.1 to Stereo conversion utilizing AVX
|
|
|
638befc1
|
2021-06-10T17:22:39
|
|
audiocvt: 5.1 to Stereo conversion utilizing SSE
|
|
86ee383f
|
2021-05-25T09:59:45
|
|
Removed duplicate SDL_AtomicGet()
|
|
db146e66
|
2021-05-25T10:34:04
|
|
Fixed warnings building with Visual Studio
|
|
58884e4c
|
2021-05-04T00:23:40
|
|
SDL_audiocvt.c: fixed MSVC double->float conversion warnings.
|
|
8ac0fb52
|
2021-04-29T09:29:02
|
|
OpenSLES: CloseDevice() is called at higher level, if OpenDevice() fails
- explicit initialization of static variables
|
|
|
fcbf19b7
|
2021-04-28T21:04:47
|
|
AAudio: make sure stream is not null to prevent crash in RequestStop (see #3710)
|
|
|
f8695185
|
2021-04-27T11:07:51
|
|
Audio: normalize conversion Stereo to 5.1, Quad to 7.1, 5.1 to 7.1 (bug #4104)
|
|
|
21349901
|
2021-04-27T10:57:48
|
|
Audio: convert 5.1 to 7.1, use right-surround for r-front and r-back (see #4104)
|
|
058bbe02
|
2021-04-24T10:17:03
|
|
Set volume on device open
|
|
|
f1fab24e
|
2021-04-15T21:00:00
|
|
AAudio: add bootstrap in SDL_audio.c
|
|
|
04b2f5f6
|
2021-04-15T20:54:58
|
|
Android: add AAudio back-end, with playback and capture (see #3710)
https://developer.android.com/ndk/guides/audio/aaudio/aaudio
|
|
|
4118fe62
|
2021-04-15T20:52:43
|
|
Android: OpenSLES, explicitly initialise the global variable 'bqPlayerPlay',
it may be read even if OpenSLES back-end hasn't been intialized
|
|
0f4aba7b
|
2021-04-06T18:34:53
|
|
audio: Fixed assertion failure if trying to use dummy backend.
|
|
|
64853b73
|
2021-04-06T18:34:17
|
|
audio: Changed a disk and dummy backends to use _this instead of this.
|
|
9d294f1f
|
2021-03-27T00:53:10
|
|
audio: Allow AudioStreamGet to return 0 in RunAudio.
While we should normally expect _something_ from the stream based on the
AudioStreamAvailable check, it's possible for a device change to flush the
stream at an inconvenient time, causing this function to return 0.
Thing is, this is harmless. Either data will be NULL and the result won't matter
anyway, or the data buffer will be zeroed out and the output will just be
silence for the brief moment that the device change is occurring. Both scenarios
work themselves out, and testing on Windows shows that this behavior is safe.
|
|
|
9b7babf9
|
2021-03-27T00:47:54
|
|
wasapi: Remove assert added by 67e8522d
|
|
5f9effaa
|
2021-03-28T17:45:41
|
|
audio: pipewire: Block while waiting on stream state info
Initializing streams, particularly capture streams, can take many milliseconds, which is a bit much for a busy wait. Use a blocking wait instead.
|
|
|
8deb4063
|
2021-03-28T17:22:59
|
|
audio: pipewire: Avoid redundant locking
The pw_thread_loop already locks and unlocks the thread mutex at the start and end of each loop iteration, so these locks are unnecessary.
|
|
|
5bb2bbd4
|
2021-03-28T17:17:00
|
|
audio: pipewire: Don't use uninitialized variables in callbacks
Some of the SDL_AudioDevice struct members aren't initialized until after returning from the OpenDevice function. Since Pipewire uses it's own processing threads, the callbacks can be entered before all members of SDL_AudioDevice are initialized, such as work_buffer, callbackspec and the processing stream, which creates a race condition. Don't use these members when in the paused state to avoid potentially using uninitialized values and memory.
|
|
|
e7e519a4
|
2021-03-17T13:04:05
|
|
dsp: Refuse to initialize if there aren't any Open Sound System devices.
This prevents the dsp target from stealing the audio subsystem but not
being able to produce sound, so other audio targets further down the list
can make an attempt instead.
Thanks to Frank Praznik who did a lot of the research on this problem!
|
|
|
b98b5adc
|
2021-03-15T10:21:36
|
|
wasapi: Don't use the system's resampler.
|
|
a5f3ea14
|
2021-03-10T09:36:46
|
|
netbsdaudio: Handle ioctls failing
A user reported that the mpv video player hangs after attempting to
set an unsupported number of channels with the SDL audio output,
because it thinks it's successfully opened the device. This makes
the failure graceful.
|
|
|
4fbd60b8
|
2021-03-09T12:22:48
|
|
audio: pipewire: Remove the nickname portion of sink/source names
Removes the node nickname from sink/source nodes as it doesn't provide any useful information and names now match those used in Pulseaudio, so any stored configuration data will be compatible between the two audio backends.
|
|
|
7d89f09f
|
2020-12-18T14:28:09
|
|
ISO C90 fixes
|
|
|
e7edb06e
|
2020-12-04T00:06:15
|
|
Audio fix
|
|
|
2d64e37e
|
2020-11-02T18:09:43
|
|
Initial rebase of xerpi's port
|
|
|
a4ddb175
|
2021-03-08T19:28:58
|
|
Formatting
|
|
391bb80b
|
2021-03-05T16:53:06
|
|
Replace duplicate functions and lstrlen/lstrcat with SDL string functions
|
|
|
67e8522d
|
2021-02-27T17:37:25
|
|
Add SDL_GetAudioDeviceSpec.
This API is supported by pipewire, pulseaudio, coreaudio, wasapi, and disk.
|
|
|
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
|
|
|
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.
|
|
|
2f0b99a7
|
2021-02-13T11:56:05
|
|
audio: Add Pipewire playback/capture sink
|
|
|
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
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Fixed re-setting the audio session category when closing an audio device
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14bf532d
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2020-02-13T16:10:52
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Fixed opening audio on Android from the Steam Link shell activity
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4bb95e84
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2020-02-11T16:14:02
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Implemented OpenSL-ES audio recording on Android
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b1c6e7c2
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2020-01-23T00:32:34
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Fixed compile warning
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f30ef6ed
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2020-01-21T17:40:16
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audio: Fixed a '//' style comment.
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dbe5c14b
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2020-01-21T15:49:37
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audio: Calculate a legitimate SDL_AudioSpec::silence in SDL_LoadWAV_RW().
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