Age | Commit message (Collapse) | Author | Files | Lines |
|
- Remove the "log-like" parts, following the same logic as the previous
commit
- Unify format
- Add missing major contributors, including myself
- Sort entries in order of first contribution (Creative comes last for
optical reasons; they don't appear to have directly contributed
anyway)
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230715160839.326978-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Empty BUGS and TODO sections don't really help anyone, so remove them.
Version information is chronically outdated, and not really useful in a
git world anyway, so remove it as well.
Also remove duplicated (and outdated, of course) status section from
p16v.h (the one in p16v.c is in better shape).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230715160839.326978-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The mixer, PCM prepare, MIDI, synth driver, and procfs callbacks are all
always invoked with IRQs enabled, so there is no point in saving the
state.
snd_emu1010_load_firmware_entry() is called from emu1010_firmware_work()
and snd_emu10k1_emu1010_init(); the latter from snd_emu10k1_create() and
snd_emu10k1_resume(), all of which have IRQs enabled.
The voice and memory functions are called from mixed contexts, so they
keep the state saving.
The low-level functions all keep the state saving, because it's not
feasible to keep track of what is called where.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230712145750.125086-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Fixes a tentative FIXME. Because we can.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-9-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Now that we know the actual word clock, we can:
- Put the resulting rate into the hardware info
- At 44.1 kHz word clock shift the rate for the pitch calculations,
which presume a 48 kHz word clock
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
So far, we set the fallback as a side effect of setting the source. But
the fallback makes no sense at all when an internal clock is selected.
Defaulting to 48k for S/PDIF & ADAT makes sense, but as that is the
global default and we're not changing it automatically any more, it's
just fine to leave it entirely to the explicit setting.
This changes the name of the pre-existing control to something more
appropriate (regardless of the split), so users will need to adjust
their mixer settings.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
On SB cards the number of captured channels is derived from the voice
mask mixer control. But for E-MU cards this wasn't actually "wired up",
so changing the mask would simply mess up the recording.
We could fix that, but the channel routing through the FPGA makes the
masking redundant. So instead we hide the control, and let the user
specify the PCM channel count the traditional way.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The hardware can deal with primes up to 7 and power-of-two multiples
thereof; the limitation is reflected by the possible buffer sizes.
Note that setting the voice mask will not allow more than 16 channels
even on Sound Blaster Audigy anymore, as 32 seems a bit excessive (the
code overall appears to think so, just not in this case).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We need to specify that the hardware supports non-standard rates, as
otherwise the sound core creates a constraint which limits the rate to
the specified standard rates. That also made the rate constraint we were
already adding meaningless.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The buffer size register sets the size of the whole buffer, not just one
period. We actually handled it like that, except that the constraint was
set on the wrong parameter. The period size is implicitly constrained by
the buffer size and the fixed period count of 2.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236059-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
There is no reason to nail it to 16 channels.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236023-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We use independent voices for the channels, so we need to make an effort
to ensure that they are actually in sync.
The hardware doesn't provide atomicity, so we may need to retry a few
times, due to NMIs, PCI contention, and the wrong phase of the moon.
Solution inspired by kX-project.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236023-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
For unclear reasons, the extra voice was set up with half the buffer
size instead of the period size. Commit 27ae958cf6 ("emu10k1 driver -
add multichannel device hw:x,3 [2-8/8]") mentions half-loop interrupts,
so maybe this was an artifact of an earlier iteration of the patch.
While at it, also fix periods_min of the regular playback - one period
makes just no sense.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523200709.236023-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
... instead of passing in a high-level mixer struct. Let the
higher-level functions handle the differences between the voice types.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523104612.198884-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This adds snd_emu10k1_pcm_init_{voices,extra_voice}() and
snd_emu10k1_playback_{un,}mute_voices() to slightly abstract by voice
function and potential stereo property.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230523104612.198884-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Instead of separate voices, we now allocate non-interleaved channels,
which may in turn contain two interleaved voices each. The higher-level
code keeps only one pointer per channel. The channels are not allocated
in one block any more, as there is no reason to do that. As a
consequence of that, and because it is cleaner regardless, we now let
the allocator store these pointers at a specified location, rather than
returning only the first one and having the calling code deduce the
remaining ones.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140947.3725394-8-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The voice allocator clearly knows about the field (it resets it), so
it's more consistent (and leads to less duplicated code) to have the
constructor take it as a parameter.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140947.3725394-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This saves some code duplication; no functional change.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230519184122.3808185-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This allows us to drop the code that tries to preserve already allocated
voices upon repeated hw_param callback invocations. Getting it right for
multi-channel voices would otherwise get a bit hairy.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140947.3725394-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Eliminate the MIDI type, as there is no such thing - the MPU401 port
doesn't have anything to do with voices.
For clarity, differentiate between regular and extra voices.
Don't atomize the enum into bits in the table display.
Simplify/optimize the storage.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140947.3725394-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
On Audigy, the send amounts are merely targets, presumably to avoid
sound distortion due to sudden changes, which the EMU8K docu explicitly
warns about.
However, that "soft-start" would prevent bit-for-bit reproduction, so
we now force the current send amounts to their final values at PCM
playback init.
One might want to do that for the MIDI synthesizer as well, though it
seems mostly pointless due to the attack phase each note has anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140339.3722279-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
CPF_CURRENTPITCH starts swerving towards PTRX_PITCHTARGET as soon as
that is set. In practice this means that CPF_FRACADDRESS may acquire a
non-zero value before we manage to force CPF_CURRENTPITCH to the final
value, which would prevent bit-for-bit reproduction.
To avoid that this state persists, we now reset CPF_FRACADDRESS when
setting CPF_CURRENTPITCH, and to (mostly) avoid that it progresses too
far in the first place (possibly even reaching CCCA_CURRADDR), we write
PTRX and CPF in one critical section (though NMIs, etc. still make this
unreliable).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140339.3722279-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
While this nicely denoises the code, the real intent is being able to
write many registers pseudo-atomically, which will come in handy later.
Idea stolen from kX-project.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518093134.3697955-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The period_bytes_min parameter and the buffer_bytes minimum constraint
made no sense at all, as they didn't reflect any hardware limitation.
Instead, apply a frame-based period_size minimum constraint, which is
derived from the cache size (it would be actually possible to go below
that, but it would require special handling, and it would be practically
impossible to keep up with the IRQ rate anyway).
Sync up the constraints of the EFX playback with those of the regular
playback, as there is no reason for them to diverge.
N.b., the maximum buffer size is actually arbitrary - the hardware could
go waay beyond 128 KiB.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518092224.3696958-9-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Pull the special handling of extra voices out of
snd_emu10k1_pcm_init_voice(), simplify snd_emu10k1_playback_pointer(),
and make the logic overall clearer. Also, add verbose comments.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-9-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Rename snd_emu10k1_playback_invalidate_cache() to the more apt
snd_emu10k1_playback_fill_cache(), and factor out
snd_emu10k1_playback_prepare_voices(), which calls the former for all
channels.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-8-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Originally, there was a 1:1 relationship between the PCM streams' and
the low-level voices' parameters. The addition of multi-channel playback
partially invalidated that, but didn't introduce proper layering, so
things kept working only by virtue of the multi-channel device never
having two channels (yet). The upcoming addition of 32-bit playback
would complete upending the relationships.
So this patch detaches the low-level parameters from the high-level
ones: we pass pre-calculated bit width and stereo flags to the low-level
manipulation functions instead of calculating them in-place from the
stream parameters.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The cache causes a fixed delay regardless of stream parameters.
Consequently, all that "cache invalidate size" calculation stuff was
garbage (which can be traced right back to Creative's OSS driver).
This also removes the definitions of registers CD1..CDF, because they
are accessed only relative to CD0 anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Given that the data is going to be ignored anyway, and that the cache
does not influence interrupt timing (which is the purpose of the extra
voices), it's pointless to pre-fill the cache.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The idea is to make the extra voice lag behind the "real" voices, but
moving the buffer address around doesn't contribute to that, as the CCCA
write below uses the same address. The exact address is unimportant, as
the data is discarded anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This workaround fails to address the underlying problem, which is
actually wholly self-made. Subsequent patches will fix it.
This reverts commit 56385a12d9bb9e173751f74b6c430742018cafc0.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
... to snd_emu10k1_pcm_init_voice(). This makes the code arguably less
convoluted.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Saves a bit of code duplication.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536451-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Many registers are meaningless for stereo slaves and the extra voices.
This patch cleans up these unnecessary register writes.
snd_emu10k1_playback_{trigger,stop}_voice() is not called for stereo
slaves any more.
snd_emu10k1_playback_prepare_voice() is renamed to
snd_emu10k1_playback_unmute_voice(), as this better reflects its
remaining function. It's not called for the extra voices any more.
Accordingly, snd_emu10k1_playback_mute_voice() is factored out from
snd_emu10k1_playback_stop_voice(), and is called selectively as well.
This doesn't add conditionals which would avoid initializing
sub-registers, as that wouldn't pull its weight.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536451-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We now enable ints even before triggering, and disable them only after
stopping - otherwise there is a race condition we may plausibly run into
when we pause/resume near the end of the buffer.
Updating the epcm->running flag is moved the same way, as it affects the
*_pointer() functions, which are called by the interrupt handler.
Also, factor these out to own functions, for clarity.
For multi-channel, the extra voice is now triggered after all regular
voices - we wouldn't want to receive an int before all channels have
passed the period boundary.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536451-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We initialize them at card init and don't touch them later, so there is
no need to reset them again at voice start.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536451-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
We (rightfully) don't enable the envelope engine for PCM voices, so any
related setup is entirely pointless - the EMU8K documentation makes that
very clear, and the fact that the various open drivers all use different
values to no observable detriment pretty much confirms it.
The remaining initializations are regrouped for clarity.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536451-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The mixer structures were filled in two places: on driver init, and when
the devices are opened. The latter made the former pointless, so we
remove the former. This implies that mixer dumps may now return all
zeroes, which is OK, as restoring them is meaningless as well.
Things were even weirder for the (generally unused) secondary sends:
Some of the initialization loops were forgotten when support for Audigy
was added, thus creating the technically illegal state of multiple sends
being routed to the same FX accumulator (though it apparently doesn't
matter when the amount is zero).
The global multi-channel init used some rather bizarre values for the
secondary sends, and the init on open actually forgot to re-initialize
them. We now use a not really more useful, but simpler formula.
The direct register init was also bogus. This doesn't really matter, as
the value is overwritten when a voice comes into use, but still.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536451-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The voice volume is a raw fractional multiplier that can't actually
represent 1.0. To still enable real pass-through, we now set the volume
to 0.5 (which results in no loss of precision, as the FX bus provides
fractional values) and scale up the samples in DSP code.
To maintain backwards compatibility with existing configuration files,
we rescale the values in the mixer controls. The range is extended
upwards from 0xffff to 0x1fffd, which actually introduces the
possibility of specifying an amplification.
There is still a minor incompatibility with user space, namely if
someone loaded custom DSP code. They'll just get half the volume, so
this doesn't seem like a big deal.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230514170323.3408834-8-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Since commit 5bbb1ab5bd ("control: use counting semaphore as write lock
for ELEM_WRITE operation"), mixer values have been fully read-write
locked. This means that it is now unnecessary to apply any additional
locks to values that are accessed solely by mixer callbacks. Values that
are read outside mixer callbacks still need write locking. There are no
cases of mixer values being written outside mixer callbacks, so no read
locks remain in mixer callbacks.
Note that the removed locks refer only to the emu data structure, not
the card's registers as the lock's name suggests.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230428095941.1706278-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Using the *_MASK defines for "maximal value" is debatable. I got the
idea from FreeBSD, and it sorta makes sense to me.
Some hunks look a bit incomplete, because code that is going to be
subsequently removed is not touched here.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230428080732.1697695-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Merge 6.3-devel branch back in order to apply the more Realtek HD-audio
changes cleanly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Firstly, remove the FXWC_* defines - the comment on FXWC implies that
the relevant defines are the (A_)EXTOUT_* ones. It's unclear where this
came from - it was in the initial ALSA import, but neither the driver
from Creative nor kX-project have these defines.
Secondly, remove A_HR, which made plain no sense (was unused, and
clashed with FXRT). Amends commit cbb7d8f9b7b ("emu10k1: Update
registers defines for the Audigy 2/emu10k2.5").
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230422161021.1143903-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Move comments to better locations, de-duplicate, fix/remove incorrect/
outdated ones, add new ones, and unify spacing somewhat.
While at it, also add testing credits for Jonathan Dowland (SB Live!
Platinum) and myself (E-MU 0404b).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230422161021.1143903-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Amends historic commit 27ae958cf6 ("emu10k1 driver - add multichannel
device hw:x,3 [2-8/8]").
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230422132430.1057468-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
It's just an (outdated) copy of snd_emu10k1_playback_pointer().
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230421141006.1005452-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Or actually, replace snd_emu10k1_playback_hw_free() with it, as that is
a subset.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230421141006.1005452-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
It could have never worked, as snd_emu10k1_fx8010_playback_prepare() and
snd_emu10k1_fx8010_playback_hw_free() assume the emu10k1 offset for the
ETRAM, and the default DSP code includes no handler for it. It also
wouldn't make a lot of sense to make it work, as Audigy has an own, much
simpler, pass-through mechanism. So just skip creation of the device.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201220.2197938-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Due to two copy/pastos, closing the MIC or EFX capture device would
make a running ADC capture hang due to unsetting its interrupt handler.
In principle, this would have also allowed dereferencing dangling
pointers, but we're actually rather thorough at disabling and flushing
the ints.
While it may sound like one, this actually wasn't a hypothetical bug:
PortAudio will open a capture stream at startup (and close it right
away) even if not asked to. If the first device is busy, it will just
proceed with the next one ... thus killing a concurrent capture.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20230405201220.2197923-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The voice allocator sometimes begins allocating from near the end of the
array and then wraps around, however snd_emu10k1_pcm_channel_alloc()
accesses the newly allocated voices as if it never wrapped around.
This results in out of bounds access if the first voice has a high enough
index so that first_voice + requested_voice_count > NUM_G (64).
The more voices are requested, the more likely it is for this to occur.
This was initially discovered using PipeWire, however it can be reproduced
by calling aplay multiple times with 16 channels:
aplay -r 48000 -D plughw:CARD=Live,DEV=3 -c 16 /dev/zero
UBSAN: array-index-out-of-bounds in sound/pci/emu10k1/emupcm.c:127:40
index 65 is out of range for type 'snd_emu10k1_voice [64]'
CPU: 1 PID: 31977 Comm: aplay Tainted: G W IOE 6.0.0-rc2-emu10k1+ #7
Hardware name: ASUSTEK COMPUTER INC P5W DH Deluxe/P5W DH Deluxe, BIOS 3002 07/22/2010
Call Trace:
<TASK>
dump_stack_lvl+0x49/0x63
dump_stack+0x10/0x16
ubsan_epilogue+0x9/0x3f
__ubsan_handle_out_of_bounds.cold+0x44/0x49
snd_emu10k1_playback_hw_params+0x3bc/0x420 [snd_emu10k1]
snd_pcm_hw_params+0x29f/0x600 [snd_pcm]
snd_pcm_common_ioctl+0x188/0x1410 [snd_pcm]
? exit_to_user_mode_prepare+0x35/0x170
? do_syscall_64+0x69/0x90
? syscall_exit_to_user_mode+0x26/0x50
? do_syscall_64+0x69/0x90
? exit_to_user_mode_prepare+0x35/0x170
snd_pcm_ioctl+0x27/0x40 [snd_pcm]
__x64_sys_ioctl+0x95/0xd0
do_syscall_64+0x5c/0x90
? do_syscall_64+0x69/0x90
? do_syscall_64+0x69/0x90
entry_SYSCALL_64_after_hwframe+0x63/0xcd
Signed-off-by: Tasos Sahanidis <tasos@tasossah.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/3707dcab-320a-62ff-63c0-73fc201ef756@tasossah.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|