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.
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.
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
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).
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.
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.
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.
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
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.
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.
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.
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.
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.
(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.)
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.
SDL now builds with gcc 7.2 with the following command line options:
-Wall -pedantic-errors -Wno-deprecated-declarations -Wno-overlength-strings --std=c99
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
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.
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.
(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.
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.
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.
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.
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.
"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.
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:
manuel.montezelo
Original bug report (note that it was against 2.0.0, it might have been fixed in between): http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=733015
--------------------------------------------------------
Package: libsdl2-2.0-0
Version: 2.0.0+dfsg1-3
Severity: normal
Tags: patch
I have occasional crashes here caused by the X11 backend of SDL2. It seems to
be caused by the X11_Pending function trying to add a high number (> 1024)
file descriptor to a fd_set before doing a select on it to avoid busy waiting
on X11 events. This causes a buffer overflow because the file descriptor is
larger (or equal) than the limit FD_SETSIZE.
Attached is a possible workaround patch.
Please also keep in mind that fd_set are also used in following files which
may have similar problems.
src/audio/bsd/SDL_bsdaudio.c
src/audio/paudio/SDL_paudio.c
src/audio/qsa/SDL_qsa_audio.c
src/audio/sun/SDL_sunaudio.c
src/joystick/linux/SDL_sysjoystick.c
--------------------------------------------------------
On Tuesday 24 December 2013 00:43:13 Sven Eckelmann wrote:
> I have occasional crashes here caused by the X11 backend of SDL2. It seems
> to be caused by the X11_Pending function trying to add a high number (>
> 1024) file descriptor to a fd_set before doing a select on it to avoid busy
> waiting on X11 events. This causes a buffer overflow because the file
> descriptor is larger (or equal) than the limit FD_SETSIZE.
I personally experienced this problem while hacking on the python bindings
package for SDL2 [1] (while doing make runtest). But it easier to reproduce in
a smaller, synthetic testcase.
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.
kdrakehp
The attached patch adds capture support to the sndio backend.
The patch also allows the `OpenDevice' function to accept arbitrary device names.
Simon Hug
There's a chance that an audio conversion from many channels to a few can use more than 9 audio filters. SDL_AudioCVT has 10 SDL_AudioFilter pointers of which one has to be the terminating NULL pointer. The SDL code has no checks for this limit. If it overflows there can be stack or heap corruption or a call to 0xa.
Attached patch adds a function that checks for this limit and throws an error if it is reached. Also adds some documentation.
Test parameters that trigger this issue:
AUDIO_U16MSB with 224 channels at 46359 Hz
V
AUDIO_S16MSB with 6 channels at 27463 Hz
The fuzzer program I uploaded in bug 3667 has more of them.
We don't fill buffers just to throw them away during shutdown now, we let the
AudioQueue free its own buffers during disposal (which fixes possible warnings
getting printed to stderr by CoreAudio), and we stop the queue after running
any queued audio during shutdown, which prevents dropping the end of the
audio playback if you opened the device with an enormous sample buffer.
Fixes Bugzilla #3555.
We need more than two buffers to flip between if they are small, or CoreAudio
won't make any sound; apparently it needs X milliseconds of audio queued when
it needs to play more or it drops any queued buffers. We are currently
guessing 50 milliseconds as a minimum, but there's probably a more proper
way to get the minimum time period from the system.
Fixes Bugzilla #3656.
This gracefully recovers when a device format is changed, and will switch
to the new default device if the current one is unplugged, etc.
This does not handle when a new default device is added; it only notices
if the current default goes away. That will be fixed by implementing the
stubbed-out MMNotificationClient_OnDefaultDeviceChanged() function.
* alsa hotplug thread is low priority
* give a chance for other threads to catch up when audio playback is not progressing
* use nonblocking for alsa audio capture
There is a bug with SDL hanging when an audio capture USB device is removed, because poll never returns
We will throw away the data anyhow, but some apps depend on the callback
firing to make progress; testmultiaudio.c, if nothing else, is an example
of this.
Capture also will now fire the callback in these conditions, offering nothing
but silence.
Apps can check SDL_GetAudioDeviceStatus() or listen for the
SDL_AUDIODEVICEREMOVED event if they want to gracefully deal with
an opened audio device that has been unexpectedly lost.
This is just enough to get you through a file that just used the extended
header for float or int data. It doesn't handle all the other things that
you expect from this header, like 24-bit samples inside a 32-bit container
or speaker masks.
This should remain binary compatible with Windows XP, as we dynamically
load anything we need and fall back to DirectSound/WinMM/XAudio2 if not
available.
Walter van Niftrik
We have found that since SDL 2.0.5 the audio callback thread is created with a very small stack size. In our application this is leading to stack overflows.
We believe there is a bug at http://hg.libsdl.org/SDL/file/391fd532f79e/src/audio/SDL_audio.c#l1132, where the is_internal_thread flag appears to be inverted.
This defaults to the internal SDL resampler, since that's the likely default
without a system-wide install of libsamplerate, but those that need more can
tweak this.
This currently favors libsamplerate over the fast path (quality over speed),
but I'm not sure that's the correct approach, as there may be surprising
changes in performance metrics depending on what packages are available on
a user's system. That being said, currently, the only thing with access to
SDL_AudioStream is an SDL audio device's thread, and it might be mostly idle
otherwise, so maybe this is generally good.
Turns out that iterating from 0 to channels-1 was a serious performance hit!
These cases now tend to match or beat the original audio resampler's speed!
This allows us to avoid an extra copy, allocate less memory and reduce cache
pressure. On the downside: we have to do a lot of tapdancing to resample the
buffer in reverse when the output is growing.
It's expensive and (hopefully) unnecessary. If this becomes an overflow
problem, we could multiply both values by 0.5f before adding them, but let's
see if we can get by without the extra multiplication first.
We never seem to overflow the source buffer now; this might have been a
leftover from a bug that was covered by Vitaly's fixes?
Removing this conditional makes the resampler 10-20% faster. Left an
assert in there for debug builds, in case this still happens.
Removed some needless things ("len / sizeof (Uint8)"), and made sure the
int32 -> float code uses doubles to avoid working with large integer values
in a 32-bit float.
Lukasz Biel
Tried to compile SDL2 using newest version of VS.
Got:
SDL_audiocvt.obj : error LNK2019: unresolved external symbol memcpy referenced in function SDL_ResampleCVT
1>E:\Users\dotPo\Lib\SDL\VisualC\x64\Release\SDL2.dll : fatal error LNK1120: 1 unresolved externals
whole compilation process: http://pastebin.com/eWDAvBce
Steps to reproduce:
clone http://hg.libsdl.org/SDL using tortoise hg,
open SDL\VisualC\SDL.sln,
when promted if should retarget solution click ok,
select release x64 build type,
Build/Build Solution
attempt 2, using Visual Studio cmake support:
open folder SDL\
select release x64 build type,
run CMake\Build CMakeLists.txt
build fails
When switched to debug build type, buils succeeds in both cases.
VS 2017 is still beta.
It causes audio pops if you're converting in chunks (and needs to
allocate/initialize/free on each convert). We'll either adjust this interface
when we break ABI for 2.1 to make this usable, or publish the SDL_AudioStream
API for those that want a streaming solution.
In the meantime, the "simple" resampler produces "good enough" audio without
pops and doesn't have to be initialized, so that'll do for now on the
SDL_AudioCVT interface.
There was a draft of this where it did audio conversion into the final buffer,
if there was enough room available past what you asked for, but that interface
got removed, so the parameters didn't make sense (and we were using the
wrong one in any case, too!).
Coriiander
This notice is about a misplaced comment.
Often times when we use an #if #endif sequence, the #endif is followed by a comment to indicate what #if statement it belonged to. The SDL_xaudio2.c file contains a misplaced comment, as follows (I stripped the other comments):
#ifdef __GNUC__
# define SDL_XAUDIO2_HAS_SDK 1
#elif defined(__WINRT__)
# define SDL_XAUDIO2_HAS_SDK
#include "SDL_xaudio2.h"
#else
#if 0
#include <dxsdkver.h>
#if (!defined(_DXSDK_BUILD_MAJOR) || (_DXSDK_BUILD_MAJOR < 1284))
# pragma message("Your DirectX SDK is too old. Disabling XAudio2 support.")
#else
# define SDL_XAUDIO2_HAS_SDK 1
#endif
#endif
#endif /* 0 */
That final /* 0 */ should be moved one line up. Like this (I tabbed it out for you to make it more clear):
This was a leftover of simplifying the resamplers down from autogenerated
code; I forgot to make something that the generator hardcoded into something
variable.
Fixes Bugzilla #3507.
Vitaly Novichkov
Okay, when I researched code and algorithm, I tried to replace condition "while(dst >= target)" with "while(dst > target)" and crashes are gone.
Seems on some moments it tries to write into the place before memory block begin, therefore phantom crashes appearing after some moments.
This no longer uses a script to generate code for every possible type
conversion or resampler. This caused a bloat in binary size and and compile
times. Now we use a handful of more generic functions and assume staying in
the CPU cache is the most important thing anyhow.
This shrinks the size of the final build (in this case: macOS X amd64, -Os to
optimize for size) by 15%. When compiling on a single core, build times drop
by about 15% too (although the previous cost was largely hidden by multicore
builds).
Apparently some systems see "hw:", some see "default:" and some see
"sysdefault:" (and maybe others!). My workstation sees both "hw:" and
"sysdefault:" ...
Try to find a prefix we like and prioritize the prefixes we (think) we want
most. If everything else fails, if there's a "default" (not a prefix) device
name, list that by itself so the user gets _something_ here.
If we can't find a prefix we like _and_ there's no "default" device, report
no hardware found at all.
Simon Sandstr?m
As stated in Summary. The switch statement will execute the default case and set a SDL error message: "SDL_MixAudio(): unknown audio format".
There are atleast two more problems here:
1. SDL_MixAudioFormat does not notify the user that an error has occured and that a SDL error message was set. It took me awhile to understand why I couldn't mix down the volume on my AUDIO_U16LSB formatted audio stream.. until I started digging in the SDL source code.
2. The error message is incorrect, it should read: "SDL_MixAudioFormat(): unknown audio format".
This tends to be a frequent spot where drivers hang, and the waits were
often unreliable in any case.
Instead, our audio thread now alerts the driver that we're done streaming audio
(which currently XAudio2 uses to alert the system not to warn about the
impending underflow) and then SDL_Delay()'s for a duration that's reasonable
to drain the DMA buffers before closing the device.
This tries to make SDL robust against device drivers that have hung up,
apps don't freeze in catastrophic (but not necessarily uncommon) conditions.
Now we detach the audio thread and let it clean up and don't care if it
never actually runs to completion.
James Zipperer
The problem I was seeing was that the the ALSA hotplug thread would call SDL_RemoveAudioDevice, but my application code was not seeing an SDL_AUDIODEVICEREMOVED event to go along with it. To fix it, I added some code into SDL_RemoveAudioDevice to call SDL_OpenedAudioDeviceDisconnected on the corresponding open audio device. There didn't appear to be a way to cross reference the handle that SDL_RemoveAudioDevice gets and the SDL_AudioDevice pointer that SDL_OpenedAudioDeviceDisconnected needs, so I ended up adding a void *handle field to struct SDL_AudioDevice so that I could do the cross reference.
Is there some other way beside adding a void *handle field to the struct to get the proper information for SDL_OpenedAudioDeviceDisconnected?
James Zipperer
Close the audio device before waiting for the audio thread to complete, which fixes a situation where the audio thread never completes
Add an additional check in the audio thread to see if the device is enabled and bail out if the device is no longer enabled
The Apple TV remote is currently exposed as a joystick with its touch surface treated as two axes. Key presses are also generated when its buttons and touch surface are used.
A new hint has been added to help deal with deciding whether to background the app when the remote's menu button is pressed: SDL_HINT_APPLE_TV_CONTROLLER_UI_EVENTS.
AudioQueues are available in Mac OS X 10.5 and later (and iOS 2.0 and later).
Their API is much more clear (and if you don't mind the threading tapdance
to get its own CFRunLoop) much easier to use in general for our purposes.
As an added benefit: they seemlessly deal with format conversion in ways
AudioUnits don't: for example, my MacBook Pro's built-in microphone won't
capture at 8000Hz and the AudioUnit version wouldn't resample to hide this
fact; the AudioQueue version, however, can handle this.
Moved this code from winmm into core so both can use it.
DirectSound (at least on Win10) also returns truncated device names, even
though it's handed in as a string pointer and not a static-sized buffer. :/
Sam Lantinga
Ryan C. Gordon
Sylvain Becker
Alon Zakai
Sylvain Beucler
Micha? Janiszewski
Ethan Lee
Wohlstand
Anthony Pesch
sezero
Philipp Wiesemann
Alex Szpakowski
Juha Kuikka