Commit graph

587 commits

Author SHA1 Message Date
Frank Praznik 21adec93b9 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.
2021-02-28 19:40:09 -08:00
Frank Praznik a07f543436 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.
2021-02-28 19:40:09 -08:00
Frank Praznik 9afd7570d6 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.
2021-02-28 19:40:09 -08:00
Frank Praznik 106dc009ac 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>
2021-02-28 19:40:09 -08:00
Frank Praznik f3ebbc06d3 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.
2021-02-28 19:40:09 -08:00
Frank Praznik 2f0b99a774 audio: Add Pipewire playback/capture sink 2021-02-28 19:40:09 -08:00
Oschowa 08547adb52 pulseaudio: Add "zerocopy" playback path 2021-02-20 12:50:36 -05:00
Romain Roffé ef85ed9352 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
2021-02-18 09:51:35 -05:00
Alon Zakai 20be1d634b 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.
2021-02-13 11:56:01 -05:00
Jay Petacat f443a6fc7a 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.
2021-02-11 19:41:41 -08:00
Ozkan Sezer b852590ba5 minor clean-up in SDL_os2audio.c 2021-02-10 10:22:16 -05:00
Ozkan Sezer 8f1025899a os2audio: changed backend name from MMOS2 to DART (like SDL-1.2) 2021-01-24 00:51:25 -05:00
Sam Lantinga 50ea3b77f1 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.
2021-01-08 10:09:37 -08:00
Ozkan Sezer 265a1cc97a 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*
2021-01-05 15:50:02 +03:00
Ozkan Sezer ed39f2f3f9 SDL_wasapi_win32.c (WASAPI_PlatformThreadInit): use L instead of TEXT()
because AvSetMmThreadCharacteristicsW specifically accepts WCHAR* input
cf. bug #5435.
2021-01-04 01:23:50 +03:00
Ozkan Sezer 01a2f27679 consistently use TEXT() macro with LoadLibrary() and GetModuleHandle()
cf. bug #5435.
2021-01-04 01:23:50 +03:00
Sam Lantinga 9130f7c377 Updated copyright for 2021 2021-01-02 10:25:38 -08:00
Ozkan Sezer 8476df3e31 SDL_mixer.c: remove calls to non-existing m68k asm code. 2020-12-30 23:55:10 +03:00
Sam Lantinga cb36189692 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.
2020-12-09 07:16:22 -08:00
Alistair Leslie-Hughes a69c61fbfd 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.
2020-08-14 12:08:58 +10:00
Ozkan Sezer 53b166797d SIZE_MAX need not be defined in limits.h
it can be in limits.h (windows) or stdint.h.
2020-11-11 12:33:55 +03:00
Ryan C. Gordon 1b8dee7caf coreaudio: Remove unnecessary include of CoreServices.h 2020-10-31 11:32:40 -04:00
Ozkan Sezer a4040293dd os2: misc build fixes 2020-10-25 10:10:02 +03:00
Ozkan Sezer bfc80d83c2 minor coding style cleanup 2020-10-25 03:55:02 +03:00
Manuel V?gele 554037a6f7 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.
2020-09-26 09:30:08 +02:00
Ozkan Sezer a90f0400a5 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.
2020-10-15 21:37:30 +03:00
Ozkan Sezer d27238751f os2: integrate the port into main tree. 2020-10-14 23:01:06 +03:00
Ozkan Sezer 1d9cf23e4c os2: updated copyright dates for 2020. header guard fixes. 2020-10-14 23:01:05 +03:00
Ozkan Sezer a3d7913c07 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.
2020-10-14 23:01:05 +03:00
Ozkan Sezer e112b776fc SDL_os2audio.c (OS2_OpenDevice): change {0} initializers to SDL_zero() 2020-10-14 23:01:05 +03:00
Ozkan Sezer 72594e255a SDL_os2audio.c (OS2_OpenDevice): remove assignment to wrong spec member
Correct assignment to 'format' member is done below, already.
2020-10-14 23:01:04 +03:00
Ozkan Sezer 222f026899 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
2020-10-14 23:01:03 +03:00
Ozkan Sezer aa790837eb 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.
2020-10-14 23:01:02 +03:00
Ozkan Sezer c218861946 os2: added a 2-byte padding to os2 SDL_PrivateAudioData 2020-10-14 23:01:01 +03:00
Ozkan Sezer 74cfb81dbb 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.
2020-10-14 23:01:00 +03:00
Ryan C. Gordon 003a16980c 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.
2020-10-06 11:07:50 -04:00
Ryan C. Gordon 7ef188a1fb jack: Fixed memory leak on device close. 2020-09-19 14:01:57 -04:00
Sam Lantinga ff53521bc6 Fixed Bluetooth audio output on Apple TV 2020-06-04 12:26:57 -07:00
Ryan C. Gordon 68777406e5 windows: Fix calls to CoCreateInstance() so last parameter is a LPVOID *. 2020-05-20 16:58:33 -04:00
Ryan C. Gordon 8601996fbc 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.
2020-05-03 22:13:48 -04:00
Sam Lantinga a990a34ac4 Cleanly switch between audio recording, playback, and both, on iOS 2020-04-14 22:26:02 -07:00
Sam Lantinga 2ae1c0f5d0 Allow Bluetooth headphones for iOS playandrecord mode 2020-04-14 09:52:27 -07:00
Ryan C. Gordon fba081e489 wasapi: Patched to compile on C89 systems, and use SDL_ceilf instead of ceilf. 2020-04-07 14:51:08 -04:00
Ryan C. Gordon 4c2be47207 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.
2020-04-07 14:37:24 -04:00
Sam Lantinga b6afbe6317 Added SDL_log.h to SDL_internal.h so logging is available everywhere 2020-04-07 09:38:57 -07:00
Sam Lantinga 9525f9729a 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.
2020-04-05 10:44:51 -07:00
Sam Lantinga f3e609679d 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.
2020-04-02 12:27:29 -07:00
Ryan C. Gordon 55b4f18e1a 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.
2020-03-29 01:54:00 -04:00
Sam Lantinga abdc5cbf24 Allow background music to play in the "play and record" case on iOS 2020-03-26 19:30:17 -07:00
Ethan Lee 27889d0261 winrt: Wait for EnumerationCompleted before leaving WASAPI_EnumerateEndpoints 2020-03-03 12:31:41 -05:00
Sam Lantinga e3b0713e40 Don't call setPreferredOutputNumberOfChannels on iOS, it breaks audio output 2020-02-24 12:07:18 -08:00
Sam Lantinga 2c9871a4a8 Fixed surround sound support on Apple TV 2020-02-24 10:25:57 -08:00
Sam Lantinga f4e23553d7 Fixed audio not coming out of the phone speakers while recording on iOS 2020-02-14 15:19:34 -08:00
Sam Lantinga 922b3dc3e7 Fixed re-setting the audio session category when closing an audio device 2020-02-14 14:18:12 -08:00
Sam Lantinga 14bf532df3 Fixed opening audio on Android from the Steam Link shell activity 2020-02-13 16:10:52 -08:00
Sam Lantinga 4bb95e8403 Implemented OpenSL-ES audio recording on Android 2020-02-11 16:14:02 -08:00
Sam Lantinga b1c6e7c244 Fixed compile warning 2020-01-23 00:32:34 -08:00
Ryan C. Gordon f30ef6ed3d audio: Fixed a '//' style comment. 2020-01-21 17:40:16 -05:00
Ryan C. Gordon dbe5c14b33 audio: Calculate a legitimate SDL_AudioSpec::silence in SDL_LoadWAV_RW(). 2020-01-21 15:49:37 -05:00
Sam Lantinga a8780c6a28 Updated copyright date for 2020 2020-01-16 20:49:25 -08:00
Sam Lantinga 3ce56f621c Fixed error formatting 2020-01-13 08:12:10 -08:00
Sam Lantinga e3cedf967d Add the destination format to the error when conversion isn't possible 2020-01-11 04:38:13 -08:00
Ryan C. Gordon 3da6a0b20e pulseaudio: don't let FlushCapture get stuck in an infinite loop on shutdown.
Fixes Bugzilla #4645.
2019-12-03 03:53:06 -05:00
Sylvain Becker 60d3965ece Readability: remove redundant return, continue, enum declaration 2019-10-30 15:36:17 +01:00
Sylvain Becker b458d7a28f Readability: remove redundant cast to the same type 2019-10-30 15:13:55 +01:00
Sylvain Becker ed469fa586 Fixed bug 4842 - Redundant condition in MS_ADPCM_Decode and IMA_ADPCM_Decode
(Thanks!)
2019-10-23 09:36:41 +02:00
Ryan C. Gordon aef1ed4ac6 audio: Set (something close to) the correct silence value for U16 audio.
Partially fixes Bugzilla #4805.
2019-09-25 15:40:27 -04:00
Ryan C. Gordon 693755f0b2 coreaudio: Apple doesn't support U16 data, so convert in that case. 2019-09-25 15:07:07 -04:00
Sylvain Becker 70dc8d1648 Android: fix corresponding warnings 2019-08-30 08:55:20 +02:00
Sam Lantinga 455944c870 Fixed whitespace 2019-08-22 16:12:16 -07:00
Sam Lantinga b521df66c3 [SDL][IOS] Audio fix - applies stream to sound data when resampling or reformatting is required. 2019-08-22 16:09:42 -07:00
Sylvain Becker 05f35c2420 Fix audio conversion U16_to_F32_SSE2 (bug 4186) 2019-08-19 21:23:47 +02:00
Sylvain Becker 1d220401ce Fixed bug 4186 - ARM/NEON audio converters cause strange clicking noises
reverse the order when storing ouput buffer
2019-08-19 20:35:02 +02:00
Sylvain Becker c0fc94f2de Fixed bug 4186 - ARM/NEON audio converters cause strange clicking noises
reverse the order when storing ouput buffer
2019-08-19 16:57:15 +02:00
Ozkan Sezer 4953e050f5 use SDL_zeroa at more places where the argument is an array. 2019-07-31 05:11:40 +03:00
Ozkan Sezer 7a47c292c0 Fix bug 4746 - introduce SDL_zeroa macro. 2019-07-31 01:22:02 +03:00
Ozkan Sezer fdc67c3c60 MS_ADPCM_Decode: fix assigning an array to a pointer (lose '&'). 2019-07-31 00:10:00 +03:00
Sam Lantinga 680e7937e0 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.
2019-07-07 09:10:56 -07:00
Sam Lantinga 14e8b93e37 Fixed compiler warning 2019-06-18 14:24:24 -07:00
Ryan C. Gordon 90e2dc9891 A few minor changes to placate static analysis. 2019-06-14 18:23:51 -04:00
Ryan C. Gordon 289d109245 audio: Attempt to fix build on ARM versions of Visual Studio. 2019-06-14 16:52:42 -04:00
Ryan C. Gordon 33b235f4c3 audio: Fix ARM NEON audio converter bugs.
(Patch from Sylvain, I'm just applying it.)

Fixes Bugzilla #4186.
2019-06-14 15:52:48 -04:00
Ryan C. Gordon 5c56c88824 audio: patched to compile. 2019-06-14 15:47:32 -04:00
Ethan Lee 5bd9b8b167 Check src alignment for S32_to_F32 conversions 2019-06-14 09:51:22 -04:00
Ryan C. Gordon 2fa33d6f98 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.  :)  )
2019-06-12 15:43:08 -04:00
Sylvain Becker cd011bb1e7 SDL_Wave: missing field 'length' initializer 2019-06-12 10:42:02 +02:00
Ryan C. Gordon 254eb67775 windows: Don't let Visual Studio insert an implicit dependency on memset().
Fixes Bugzilla #4662.
2019-06-11 02:08:31 -04:00
Sam Lantinga d8da33c03f 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!
2019-06-10 08:49:26 -07:00
Sam Lantinga 762b788f67 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.
2019-06-09 12:46:10 -07:00
Sam Lantinga a21b5b3018 Fixed build 2019-06-08 19:09:43 -07:00
Sam Lantinga 990e166a3b Fixed bug 3894 - Fuzzing crashes for SDL_LoadWAV
Simon Hug

I had a look at this and made some additions to SDL_wave.c.

The attached patch adds many checks and error messages. For some reason I also added A-law and ?-law decoders. Forgot exactly why... but hey, they're small.

The WAVE format is seriously underspecified (at least by the documents that are publicly available on the internet) and it's a shame Microsoft never put something better out there. The language used in them is so loose at times, it's not surprising the encoders and decoders behave very differently. The Windows Media Player doesn't even support MS ADPCM correctly.

The patch also adds some hints to make the decoder more strict at the cost of compatibility with weird WAVE files.

I still think it needs a bit of cleaning up (Not happy with the MultiplySize function. Don't like the name and other SDL code may want to use something like this too.) and some duplicated code may be folded together. It does work in this state and I have thrown all kinds of WAVE files at it. The AFL files also pass with it and some even play (obviously just noise). Crafty little fuzzer.

Any critique would be welcome. I have a fork of SDL with a audio-loadwav branch over here if someone wants to use the commenting feature of Bitbucket:

https://bitbucket.org/ChliHug/SDL

I also cobbled some Lua scripts together to create WAVE test files:

https://bitbucket.org/ChliHug/gendat
2019-06-08 19:02:42 -07:00
Sam Lantinga 31765242d6 Fixed bug 4294 - Audio: perform more validation on conversion request
janisozaur

There are many cases which are not able to be handled by SDL's audio conversion routines, including too low (negative) rate, too high rate (impossible to allocate).

This patch aims to report such issues early and handle others in a graceful manner. The "INT32_MAX / RESAMPLER_SAMPLES_PER_ZERO_CROSSING" value is the conservative approach in terms of what can _technically_ be supported, but its value is 4'194'303, or just shy of 4.2MHz. I highly doubt any sane person would use such rates, especially in SDL2, so I would like to drive this limit further down, but would need some assistance to do that, as doing so would have to introduce an arbitrary value. Are you OK with such approach? What would a good value be? Wikipedia (https://en.wikipedia.org/wiki/High-resolution_audio) lists 96kHz as the highest sampling rate in use, even if I quadruple it for a good measure, to 384kHz it's still an order of magnitude lower than 4MHz.
2019-06-08 18:22:18 -07:00
Sam Lantinga 3f19a6d5e8 CVE-2019-7578: Fix a buffer overread in InitIMA_ADPCM
If IMA ADPCM format chunk was too short, InitIMA_ADPCM() parsing it
could read past the end of chunk data. This patch fixes it.

CVE-2019-7578
https://bugzilla.libsdl.org/show_bug.cgi?id=4494

Signed-off-by: Petr P?sa? <ppisar@redhat.com>
2019-06-08 18:07:58 -07:00
Sam Lantinga 8a37848de9 Fixed bug 4605 - WASAPI_WaitDevice hang
Matt Brocklehurst

We've noticed that if you are playing audio on Windows via the WASAPI interface and you unplug and reconnect the device a few times the program hangs.

We've debugged the problem down to

static void
WASAPI_WaitDevice(_THIS)
{

   ... snip ...
 if (WaitForSingleObjectEx(this->hidden->event, INFINITE, FALSE) == WAIT_OBJECT_0) {
   ... snip ...
}

This WaitForSingleObjectEx does not havbe a time out defined, so it hangs there forever.

Our suggested fix we found was to include a time out of say 200mSec

We have done quite a bit of testing with this fix in place on various hardware configurations and it seems to have resolved the issue.
2019-06-08 13:41:46 -07:00
Sam Lantinga 15bae953b1 Fixed bug 4642 - Rework SDL_netbsdaudio to improve performance
Nia Alarie

The NetBSD audio driver has a few problems. Lots of obsolete code, and extremely bad performance and stuttering.

I have a patch in NetBSD's package system to improve it. This is my attempt to upstream it.

The changes include:

* Removing references to defines which are never used.
* Using the correct structures for playback and recording, previously they were the wrong way around.
* Using the correct types ('struct audio_prinfo' in contrast to 'audio_prinfo')
* Removing the use of non-blocking I/O, as suggested in #3177.
* Removing workarounds for driver bugs on systems that don't exist or use this driver any more.
* Removing all usage of SDL_Delay(1)
* Removing pointless use of AUDIO_INITINFO and tests that expect AUDIO_SETINFO to fail when it can't.

These changes bring its performance in line with the DSP audio driver.
2019-06-08 13:03:36 -07:00
Sam Lantinga 03cf24162f OpenSL ES audio cleanup and added a note with low latency audio discussion 2019-06-08 10:21:38 -07:00
Sam Lantinga 166d15fd75 Fixed surround sound channel setup for Android OpenSL ES audio driver 2019-06-07 15:09:15 -07:00
Sam Lantinga 723d014336 Fixed bug 4171 - SDL_GetQueuedAudioSize is broken with WASAPI
Cameron Gutman

I was trying to use SDL_GetQueuedAudioSize() to ensure my audio latency didn't get too high while streaming data in from the network. If I get more than N frames of audio queued, I know that the network is giving me more data than I can play and I need to drop some to keep latency low.

This doesn't work well on WASAPI out of the box, due to the addition of GetPendingBytes() to the amount of queued data. As a terrible hack, I loop 100 times calling SDL_Delay(10) and SDL_GetQueuedAudioSize() before I ever call SDL_QueueAudio() to get a "baseline" amount that I then subtract from SDL_GetQueuedAudioSize() later. However, because this value isn't actually a constant, this hack can cause SDL_GetQueuedAudioSize() - baselineSize to be < 0. This means I have no accurate way of determining how much data is actually queued in SDL's audio buffer queue.

The SDL_GetQueuedAudioSize() documentation says: "This is the number of bytes that have been queued for playback with SDL_QueueAudio(), but have not yet been sent to the hardware." Yet, SDL_GetQueuedAudioSize() returns > 0 value when SDL_QueueAudio() has never been called.

Based on that documentation, I believe the current behavior contradicts the documented behavior of this function and should be changed in line with Boris's patch.

I understand that exposing the IAudioClient::GetCurrentPadding() value is useful, but a solution there needs to take into account what of that data is silence inserted by SDL and what is actual data queued by the user with SDL_QueueAudio(). Until that happens, I think the best approach is to remove the GetPendingBytes() call until SDL is able to keep track of queued data to make sense of it. This would make SDL_GetQueuedAudioSize() possible to use accurately with WASAPI.
2019-06-04 17:32:15 -07:00
Sam Lantinga f3e76ea1d0 Use the OpenSL ES audio driver by default on Android, as it has the lowest latency. 2019-05-23 13:47:30 -07:00
Sam Lantinga 02f9667a08 Fixed static and buzzing when trying to use floating point audio on the OpenSL ES audio driver. 2019-05-23 13:47:27 -07:00
Sam Lantinga abcfe80480 [SDL] iOS fix bug with audio interrupted by a phone call not restoring. 2019-05-14 14:20:54 -07:00
Ryan C. Gordon 2fbfe8b912 coreaudio: Set audio callback thread priority.
Fixes Bugzilla #4155.
2019-03-25 12:59:30 -04:00
Ryan C. Gordon 6a3356ab3f Backed out changeset cec31de4e126
This was meant to migrate CoreAudio onto the same SDL_RunAudio() path that
most other audio drivers are on, but it introduced a bug because it doesn't
deal with dropped audio buffers...and fixing that properly just introduces
latency.

I might revisit this later, perhaps by reworking SDL_RunAudio to allow for
this sort of API better, or redesigning the whole subsystem or something, I
don't know. I'm not super-thrilled that this has to exist outside of the usual
codepaths, though.

Fixes Bugzilla #4481.
2019-03-25 12:24:38 -04:00
Sam Lantinga 35255342cd Fixed bug 4525 - Fix crash in ALSA_HotplugThread caused by bad return value check
Anthony Pesch

Fix snd_device_name_hint return value check

According to the ALSA documentation, snd_device_name_hint returns 0 on
success, otherwise a negative error code. The code previously only
considered -1 to be an error, which let other error codes through
resulting in a segfault when hints (which was NULL) was dereferenced
2019-03-16 18:48:21 -07:00
Sylvain Becker 03cbac4040 Android/openslES: fix warnings, comment out un-used interface 2019-02-05 15:14:15 +01:00
Sylvain Becker 614c8aea20 Android/openslES: set number of buffers of DATALOCATOR to internal NUM_BUFFER
If we increase NUM_BUFFER, Enqueue won't fail with SL_RESULT_BUFFER_INSUFFICIENT
2019-02-05 15:09:41 +01:00
Sylvain Becker bf823bf2dc Android/openslES: prevent to run out of buffers if Enqueue() fails. 2019-02-05 15:05:32 +01:00
Alon Zakai 3b4e369365 Emscripten: No need for Runtime. for dynCalls 2019-01-29 12:21:22 +00:00
Alon Zakai 53ead95e1d Emscripten: Avoid SDL2 in JS global scope
After this fix, closure works with the LLVM wasm backend on SDL2.
2019-01-29 12:19:36 +00:00
Sylvain Becker 1b24b2eca5 Android/openslES: fix Pause/ResumeDevices when openslES is not used 2019-01-14 22:56:57 +01:00
Sylvain Becker 647b1f6a6d Android/openslES: check for non NULL variable, some intialization.
use the previous naming
2019-01-14 14:36:13 +01:00
Sylvain Becker 7b1cc441dd Android/openslES: start playing, after creating ressources 2019-01-14 14:31:06 +01:00
Sylvain Becker 955d87894b Android/openslES: set audio in paused/resumed state for Android event loop
And also in "stopped" state before closing the device.
2019-01-14 12:33:29 +01:00
Sylvain Becker 59c8c7b684 Android/openslES: move a few static variables to SDL_PrivateAudioData structure 2019-01-14 10:58:57 +01:00
Sylvain Becker 5aeeaaab70 Android/openslES: register and use CloseDevice function. 2019-01-14 10:16:26 +01:00
Sylvain Becker 365fd9c602 Android/openslES: some space and indentation to match SDL conventions 2019-01-14 10:04:54 +01:00
Sam Lantinga 7dc92a7669 Initial Android OpenSL ES implementation, contributed by ANTA 2019-01-12 12:18:44 -08:00
Sylvain Becker d23c2f07e3 Fixed bug 3930 - Android, set thread priorities and names
SDLActivity thread priority is unchanged, by default -10 (THREAD_PRIORITY_VIDEO).

SDLAudio thread priority was -4 (SDL_SetThreadPriority was ignored) and is now -16 (THREAD_PRIORITY_AUDIO).

SDLThread thread priority was 0 (THREAD_PRIORITY_DEFAULT) and is -4 (THREAD_PRIORITY_DISPLAY).
2019-01-10 18:05:56 +01:00
Sam Lantinga 5e13087b0f Updated copyright for 2019 2019-01-04 22:01:14 -08:00
Sylvain Becker aea7e56a24 android: use __ARM_NEON instead of __ARM_NEON__ to include <arm_neon.h>
Only __ARM_NEON is defined with Android NDK and arm64-v8a
Tested on ndk-r18, ndk-r13 and also Xcode.
(Visual Studio needs a different fix).

Fixes Bugzilla #4409.
2018-12-04 12:34:45 +01:00
Sylvain Beucler 1f6bd95110 Emscripten: make CloseAudio actually close audio
cf. https://bugzilla.libsdl.org/show_bug.cgi?id=4176
2018-11-15 18:22:30 +00:00
Micha? Janiszewski 91820998fc 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.
2018-10-28 21:36:48 +01:00
Ryan C. Gordon 4a50a04213 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.
2018-10-21 22:40:17 -04:00
Ryan C. Gordon 04cbf13261 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.
2018-10-10 15:20:56 -04:00
Ryan C. Gordon 0378529e1e audio: clean_out_device_list() already sets this flag to false for us. 2018-10-10 14:55:24 -04:00
Sam Lantinga f5a21ebf0c Added support for surround sound and float audio on Android 2018-10-09 20:12:43 -07:00
Sam Lantinga b251876126 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
2018-10-01 09:47:10 -07:00
Ryan C. Gordon 56ec349d2a 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.)
2018-09-29 16:48:15 -04:00
Ethan Lee 7f9854b9b2 WinRT: Wait until audio device activation is complete and PrepDevice during OpenAudio 2018-09-25 01:45:12 -04:00
Sam Lantinga 5febdfcece Fixed whitespace 2018-09-24 11:49:25 -07:00
Ryan C. Gordon 623a6defd3 alsa: optionally run entire pipeline non-blocking. 2018-08-07 16:49:18 -04:00
Ryan C. Gordon 56f44cfa0f 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.
2018-08-07 13:04:15 -04:00
Wohlstand ff8c62f227 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.
2018-07-02 03:53:57 +03:00
Ryan C. Gordon 4773690d0f Deal with possible malloc(0) calls, as pointed out by static analysis. 2018-06-25 12:55:23 -04:00
Anthony Pesch c591429542 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.
2018-05-04 21:21:32 -04:00
Sam Lantinga 1d25135b71 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.
2018-06-01 19:43:53 -07:00
Sam Lantinga 8325df25aa 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.
2018-05-24 07:30:24 -07:00
Ryan C. Gordon 101544d6f0 audio: Needed to fix two more instances for Visual Studio. 2018-05-21 12:05:17 -04:00
Ryan C. Gordon 49881861b1 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.)
2018-05-21 11:54:09 -04:00
Ryan C. Gordon b7e88aaae0 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!
2018-05-16 02:03:06 -04:00
Ryan C. Gordon cb0e614fb1 audio: SSE2 float-to-int converters should clamp input.
The scalar versions already do this.
2018-05-15 02:29:35 -04:00
Ryan C. Gordon a07e5815a5 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.
2018-05-15 01:40:05 -04:00
Ryan C. Gordon 7832cb652e 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).
2018-05-15 01:35:53 -04:00
Ryan C. Gordon e2ec1eb12e 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.
2018-05-15 01:04:11 -04:00
Sam Lantinga f521b22eb5 Added SDL_THREAD_PRIORITY_TIME_CRITICAL 2018-04-23 22:07:56 -07:00
Ryan C. Gordon dc8b55e50b 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.
2018-04-16 02:11:09 -04:00
Sam Lantinga 99a0c0f0e2 Fixed MinGW-w64 build 2018-02-24 08:23:44 -08:00
Ryan C. Gordon c7e4366530 wasapi: let Windows do the resampling for us if possible. 2018-02-21 21:34:06 -05:00
Ryan C. Gordon 7e1fa0ce53 wasapi: fixed typo in an assert message. 2018-02-21 21:34:35 -05:00
Ryan C. Gordon 97494f5374 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.
2018-02-17 18:30:21 -05:00
sezero ba0ecc6712 fix building SDL_audiotypecvt.c with gcc < 4.0 2018-02-12 10:47:00 +03:00
sezero 40b27fd51b 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.
2018-02-12 17:00:00 +03:00
Sam Lantinga 90e72bf4e2 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
2018-01-30 18:08:34 -08:00
Ryan C. Gordon 488824017a wasapi: Fixed some compiler warnings. 2018-01-22 09:36:40 -05:00
Sam Lantinga e3cc5b2c6b Updated copyright for 2018 2018-01-03 10:03:25 -08:00
Ryan C. Gordon 77bb49b7a7 wasapi: Patched to compile on non-UWP WinRT builds. 2017-12-31 03:34:16 -05:00
Ryan C. Gordon ab4695f48f wasapi: switched to event-driven interface.
This reduces latency and improves battery life.
2017-12-13 14:35:55 -05:00
Ryan C. Gordon 351d6d4784 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
2017-12-06 12:24:32 -05:00
Sam Lantinga e830ef3458 Fixed typo converting 4 channel audio to 2 channel 2017-10-20 16:53:42 -07:00
Sam Lantinga 9a291c1e59 Added a note about adjusting channel weights when converting to fewer channels 2017-10-20 14:51:22 -07:00
Ryan C. Gordon 729329068b audio: Added SDL_AudioStreamFlush(). 2017-10-19 18:05:42 -04:00
Ryan C. Gordon e98920f5f3 Check correct variable for malloc() results. 2017-10-18 23:49:46 -04:00
Sam Lantinga afefcbfeba 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.
2017-10-18 19:30:47 -07:00
Sam Lantinga 653ab5d9c4 Added a staging buffer to the audio stream so that we can accumulate small amounts of data if needed when resampling 2017-10-18 19:26:36 -07:00
Sam Lantinga 80f8464d97 Added audio stream conversion functions:
SDL_NewAudioStream
    SDL_AudioStreamPut
    SDL_AudioStreamGet
    SDL_AudioStreamAvailable
    SDL_AudioStreamClear
    SDL_FreeAudioStream
2017-10-18 15:54:05 -07:00
Ryan C. Gordon fa15674134 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.
2017-10-13 01:15:29 -04:00
Sam Lantinga ba10d2b654 Fixed compiler warning 2017-10-12 13:55:35 -07:00
Ryan C. Gordon 5e5f2290f2 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.
2017-10-11 12:07:43 -04:00
Ryan C. Gordon 9bd2c6b491 audio: Moved the resampler state up to double precision.
Fixes more buffer overflows.
2017-10-11 11:51:14 -04:00
Ryan C. Gordon b2f5123b65 audio: calculate resampling time directly, don't increment (thanks, Eric!).
Fixes buffer overruns as floating point errors accumulate.

Partially fixes Bugzilla #3848.
2017-10-11 11:43:35 -04:00
Ryan C. Gordon 763c387149 audio: clamp resampler interpolation values to prevent buffer overflow.
Partially fixes Bugzilla #3848.
2017-10-11 02:33:55 -04:00
Ryan C. Gordon 0085f917e0 audio: Moved unchanging variable out of loop. 2017-10-11 02:31:58 -04:00
Ryan C. Gordon cb8bf6bbaf audio: Make sure audio stream resampling doesn't overflow buffers. 2017-10-11 02:03:05 -04:00
Ryan C. Gordon 459e2b0bbe audio: Fixed check for minimum audio stream put size. 2017-10-11 01:37:11 -04:00
Ryan C. Gordon 903ff6414e audio: SDL_ResampleCVT() should use memmove instead of memcpy.
This copy can overlap.

Fixes Bugzilla #3849.
2017-10-10 22:31:02 -04:00
Ryan C. Gordon 42fff7ce2b 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.
2017-10-10 22:18:46 -04:00
Ryan C. Gordon 37d89aa10f audio: reworked audio streams to have right-hand resampling padding available.
Fixes Bugzilla #3851.
2017-10-10 16:12:56 -04:00
Ryan C. Gordon 099ae43e81 audio: Fixed compiler warning on Visual Studio. 2017-09-22 22:28:21 -04:00
Sam Lantinga fe6b8f1c31 Fixed Mac OS X build 2017-09-22 11:25:52 -07:00
Sam Lantinga 407e1693ae Fixed audio being silent on older iOS devices
Tested on an iPod running iOS 6.1
2017-09-22 11:15:14 -07:00
Sam Lantinga d74c00e67d Fixed memory leak when HAVE_ALLOCA isn't defined 2017-09-22 08:51:45 -07:00
Ryan C. Gordon 6d206a7b28 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.
2017-09-22 07:42:24 -04:00
Sam Lantinga 8b660c5046 Added some missing "extern" declarations 2017-09-21 00:55:29 -07:00
Ryan C. Gordon f40bd5ee24 audio: removed my perl experiment script. 2017-09-21 02:06:53 -04:00
Ryan C. Gordon 1a3b95a11e 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.
2017-09-21 02:51:14 -04:00
Sam Lantinga c08a7a74a5 Added a hint SDL_HINT_AUDIO_CATEGORY to control the audio category,
determining whether the phone mute switch affects the audio
2017-09-15 17:27:32 -07:00
Ryan C. Gordon 93583d461c 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.
2017-09-09 21:17:46 -04:00
Ryan C. Gordon ca15c7d67f wave: SDL_LoadWAV now supports 24-bit audio. 2017-09-07 10:56:08 -04:00
Ryan C. Gordon 3267398d15 sndio: Patched to compile if SIO_DEVANY isn't defined.
(It isn't in whatever Raspbian is currently shipping.)
2017-09-02 16:41:14 -04:00
Sam Lantinga d619d88560 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.
2017-08-28 21:42:39 -07:00
Ryan C. Gordon b128e8802d 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.
2017-08-29 00:41:45 -04:00
Ryan C. Gordon a0cd7d6bce audio: Converting audio samples from int to float was using wrong equation.
Fixes Bugzilla #3775.
2017-08-29 00:02:04 -04:00
Sam Lantinga 6dd3f55d55 Fixed WinRT build after changing the header guard preprocessor symbol 2017-08-28 01:59:53 -07:00
Sam Lantinga 0d011ec66d Renaming of guard header names to quiet -Wreserved-id-macro 2017-08-28 00:22:23 -07:00
Sam Lantinga 8e7998e19d 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:
2017-08-27 19:10:30 -07:00
Sam Lantinga bcf0e07107 Added WASAPI audio target to autoconf build process 2017-08-18 17:29:44 -07:00
Ryan C. Gordon e3e6b4fd35 audio: better docs on conversion APIs, error if not init'd (thanks, Simon!).
Fixes Bugzilla #3662.
2017-08-18 16:52:19 -04:00
Sam Lantinga fb835f9e3b 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

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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


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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.
2017-08-14 20:22:19 -07:00
Sam Lantinga 96305832bc 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.
2017-08-11 10:21:19 -07:00
Ryan C. Gordon 9dde37eadb sndio: Fix for some platforms (Linux, for example) that don't define INFTIM.
Fixes Bugzilla #3712.
2017-08-07 00:25:18 -04:00