*-objs suffix is reserved rather for (user-space) host programs while
usually *-y suffix is used for kernel drivers (although *-objs works
for that purpose for now).
Let's correct the old usages of *-objs in Makefiles.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20240507135513.14919-3-tiwai@suse.de
... into snd_emu1010_load_firmware_entry(). This makes it clearer that
these steps belong together tightly, as implied by prior commits.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-5-oswald.buddenhagen@gmx.de>
It is a low-level I/O access function, so io.c is the natural place for
it.
While we're moving the code, reduce the scope of some variables, use
compound assignment operators, and add/adjust some comments.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-4-oswald.buddenhagen@gmx.de>
There is only one call site, and there we already know that we actually
have a firmware.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-3-oswald.buddenhagen@gmx.de>
Firstly, it is pointless to explicitly disable the power to the dock
prior to resetting the FPGA, as the latter will do the former anyway.
Secondly, it doesn't make much sense to check whether the FPGA is
already programmed. It's much simpler to just presume it is, and issue
the self-reset command. If it isn't, the effect isn't worse than the
checks themselves. As a side effect, we lose the info if the reset
fails, but there is no plausible way how that could happen unless the
card burns out while operating, and in that case we'll detect a firmware
upload failure a bit later anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093717.3198716-2-oswald.buddenhagen@gmx.de>
We did not delay after the second strobe signal, so another immediately
following access could potentially corrupt the written value.
This is a purely speculative fix with no supporting evidence, but after
taking out the spinlocks around the writes, it seems plausible that a
modern processor could be actually too fast. Also, it's just cleaner to
be consistent.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-7-oswald.buddenhagen@gmx.de>
A side effect of making the dock monitoring interrupt-driven was that
we'd be very quick to program a freshly connected dock. However, for
unclear reasons, the dock does not work when we do that - despite the
FPGA netlist upload going just fine. We work around this by adding a
delay before programming the dock; for safety, the value is several
times as much as was determined empirically.
Note that a badly timed dock hot-plug would have triggered the problem
even before the referenced commit - but now it would happen 100% instead
of about 3% of the time, thus making it impossible to work around by
re-plugging.
Fixes: fbb64eedf5 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-6-oswald.buddenhagen@gmx.de>
The FPGA access through the GPIO port does not interfere with other
sound processor register access, so there is no need to subject it to
emu_lock. And after moving all FPGA access out of the interrupt handler,
it does not need to be IRQ-safe, either.
What's more, attaching the dock causes a firmware upload, which takes
several seconds. We really don't want to disable IRQs for this long, and
even less also have someone else spin with IRQs disabled waiting for us.
Therefore, use a mutex for FPGA access locking.
This makes the code somewhat more noisy, as we need to wrap bigger
sections into the mutex, as it needs to enclose the spinlocks.
The latter has the "side effect" of fixing dock FPGA programming in a
corner case: a really badly timed mixer access right between entering
FPGA programming mode and uploading the netlist would mess up the
protocol.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-5-oswald.buddenhagen@gmx.de>
The actual event processing was already done by workqueue items. We can
move the event dispatching there as well, rather than doing it already
in the interrupt handler callback.
This change has a rather profound "side effect" on the reliability of
the FPGA programming: once we enter programming mode, we must not issue
any snd_emu1010_fpga_{read,write}() calls until we're done, as these
would badly mess up the programming protocol. But exactly that would
happen when trying to program the dock, as that triggers GPIO interrupts
as a side effect. This is mitigated by deferring the actual interrupt
handling, as workqueue items are not re-entrant.
To avoid scheduling the dispatcher on non-events, we now explicitly
ignore GPIO IRQs triggered by "uninteresting" pins, which happens a lot
as a side effect of calling snd_emu1010_fpga_{read,write}().
Fixes: fbb64eedf5 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-4-oswald.buddenhagen@gmx.de>
Pulled out of the next patch to improve its legibility.
As the function is now available, call it directly from
snd_emu10k1_emu1010_init(), thus making the MicroDock firmware loading
synchronous - there isn't really a reason not to. Note that this does
not affect the AudioDocks of rev1 cards, as these have no independent
power supplies, and thus come up only a while after the main card is
initialized.
As a drive-by, adjust the priorities of two messages to better reflect
their impact.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-3-oswald.buddenhagen@gmx.de>
While there are two separate IRQ status bits for dock attach and detach,
the hardware appears to mix them up more or less randomly, making them
useless for tracking what actually happened. It is much safer to check
the dock status separately and proceed based on that, as the old polling
code did.
Note that the code assumes that only the dock can be hot-plugged - if
other option card bits changed, the logic would break.
Fixes: fbb64eedf5 ("ALSA: emu10k1: make E-MU dock monitoring interrupt-driven")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218584
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Message-ID: <20240428093716.3198666-2-oswald.buddenhagen@gmx.de>
There is no need for it to be 32 samples - 3 will do just fine (which is
the interpolator's epsilon). The old size was presumably meant to
compensate for the cache's presence, but we're now handling that
properly.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-17-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Compensate for the cache lag of 64 frames, and actually populate the
cache. Without these, the playback would start with garbage (which
would be (mostly?) masqueraded by the note's attack phase).
Note that we set the starting address only 61 frames ahead, to
compensate for the interpolator's epsilon. Unlike for PCM playback, we
don't even need to manually silence-fill the first frames in the cache,
because we insert some silence in front of each sample anyway.
A challenge are extremely short samples with a loop end below the cache
size, because a) we'd have to wrap the current address to be within the
loop and b) automatic pre-filling of the cache with the right data does
not work in this case.
We could pre-fill the cache manually, but that's slow, requires
additional code for each sample width, and is made even more complex by
the driver's virtual address space having no contiguous mapping for the
CPU.
We could have the engine fill the cache piece-wise (which is really what
happens when playback is running), but that would also be complex, and
we'd need to wait for the engine to handle each piece, so it wouldn't be
that much faster than the manual fill.
For the case of requiring only one loop iteration prior to reaching the
cache size, we could leverage the engine's looping mechanism around
CCR_CACHELOOPFLAG, but this special case doesn't seem worth the
complexity.
So we just unroll the loop as far as necessary to be able to play back
the sample without any fiddling.
Pedantically, this would be incorrect for loop-until-release samples
with a low loop end which are released very quickly, but that would be
relatively harmless, is not a plausible use case in the first place, and
SoundFont sample mode 3 isn't actually implemented anyway (it's
conflated with mode 1, infinite looping).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-16-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The offsets are counted in samples, not in bytes.
While the code block is being rewritten, also move it up a bit, to avoid
churn in a subsequent patch.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-13-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This de-duplicates the code slightly. But the real reason is that it
moves the code up, which the next patch will depend on.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-12-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Samples are byte-sized in this mode, and thus the offset calculation
needs no shifting.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-11-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hardware supports S16LE and U8 samples, while U16LE and S8 (which
the driver implicitly claims to support) require sign flipping.
Note that this matters only for the GUS patch loader, as the implemented
SoundFont v2.01 spec is limited to S16LE.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-10-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert some checks in snd_emu10k1_sample_new() back into assertions (as
they were prior to da3cec35dd (ALSA: Kill snd_assert() in sound/pci/*,
2008-08-08)), and move them into the low-level memory access functions
they protect.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-9-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This does several closely related things:
- Move the code from the drivers into the SoundFont loader, which
de-duplicates it.
- Sort of explain the weird "recalculate address offset" feature. Note
that I don't think it actually makes any sense - the calling user
space code should do that. The background is certainly that the source
data (the SoundFont format) uses pointers into a single wave block
(and the API allows doing the same for on-board ROM), but the API
expects the wave data from user space to be pre-chopped into
individual patches anyway.
- Make sure that the specified offsets actually lie within the supplied
wave data. Note that we don't validate ROM offsets, so one can play
back anything within the sound card's address space.
- In load_guspatch(), don't call the sample_new callback anymore when
the patch size is zero, as was already the case in load_data(). The
callbacks would instantly return in that case anyway; these checks are
now removed.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-7-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is required only to implement WAVE_BIDIR_LOOP and WAVE_LOOP_BACK in
the GUS patch loader. It has not worked on emu10k1 since before ALSA hit
mainline, yet nobody appears to have complained. And as it isn't super
easy to implement, just admit defeat and clean up the code.
If somebody wanted to resurrect the feature, the emu8k driver could
serve as a template, but the code would be quite different. But
arguably, this should be done in user space in the first place, as this
doesn't represent a hardware feature (somewhat ironically, the actual
GUS driver has no synth support, and therefore no GUS patch loader).
Note that instead of properly rejecting affected samples, we continue to
just pretend that the feature wasn't requested. This is extremely
questionable behavior, but avoids that possibly unused instruments
suddenly prevent loading the entire file, which would break backwards
compatibility. But at least we log a warning now.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240406064830.1029573-6-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As already anticipated in the original commit, playback was broken for
very short samples. I just didn't expect it to be an actual problem,
because we're talking about less than 1.5 milliseconds here. But clearly
such wavetable samples do actually exist.
The problem was that for such short samples we'd set the current
position beyond the end of the loop, so we'd run off the end of the
sample and play garbage.
This is a bigger (more audible) problem than the original one, which was
that we'd start playback with garbage (whatever was still in the cache),
which would be mostly masked by the note's attack phase.
So revert to the old behavior for now. We'll subsequently fix it
properly with a bigger patch series.
Note that this isn't a full revert - the dead code is not re-introduced,
because that would be silly.
Fixes: df335e9a8b ("ALSA: emu10k1: fix synthesizer sample playback position and caching")
Link: https://bugzilla.kernel.org/show_bug.cgi?id=218625
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Message-ID: <20240401145805.528794-1-oswald.buddenhagen@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few ALSA control API helpers like snd_ctl_rename(), snd_ctl_remove()
and snd_ctl_find_*() suppose the callers taking card->controls_rwsem.
But it's error-prone and fragile. This patch set tries to change
those API functions to take the card->controls>rwsem internally by
themselves, so that the drivers don't need to take care of lockings.
After applying this patch set, only a couple of places still touch
card->controls_rwsem (which are OK-ish as they need for traversing the
control linked list).
Link: https://lore.kernel.org/r/20230718141304.1032-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'tags/ctl-lock-fixes-6.6' into for-next
ALSA: Make control API taking controls_rwsem consistently
A few ALSA control API helpers like snd_ctl_rename(), snd_ctl_remove()
and snd_ctl_find_*() suppose the callers taking card->controls_rwsem.
But it's error-prone and fragile. This patch set tries to change
those API functions to take the card->controls>rwsem internally by
themselves, so that the drivers don't need to take care of lockings.
After applying this patch set, only a couple of places still touch
card->controls_rwsem (which are OK-ish as they need for traversing the
control linked list).
Link: https://lore.kernel.org/r/20230718141304.1032-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that snd_ctl_find_id() takes the locking itself, we can get rid of
the messy locking in the caller side in snd_emu10k1_verify_controls().
Link: https://lore.kernel.org/r/20230718141304.1032-12-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For reducing the unnecessary use of controls_rwsem in the drivers,
this patch adds a new variant for snd_ctl_find_*() helpers:
snd_ctl_find_id_locked() and snd_ctl_find_numid_locked() look for a
kctl element inside the card->controls_rwsem -- that is, doing the
very same as what snd_ctl_find_id() and snd_ctl_find_numid() did until
now. snd_ctl_find_id() and snd_ctl_find_numid() remain same,
i.e. still unlocked version, but they will be switched to locked
version once after all callers are replaced.
The patch also replaces the calls of snd_ctl_find_id() and
snd_ctl_find_numid() in a few places; all of those are places where we
know that the functions are called properly with controls_rwsem held.
All others are without rwsem (although they should have been).
After this patch, we'll turn on the locking in snd_ctl_find_id() and
snd_ctl_find_numid() to be more race-free.
Link: https://lore.kernel.org/r/20230718141304.1032-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, snd_ctl_remove() requires its caller to take
card->controls_rwsem manually before the call for avoiding possible
races. However, many callers don't care and miss the locking.
Basically it's cumbersome and error-prone to enforce it to each
caller. Moreover, card->controls_rwsem is a field that should be used
only by internal or proper helpers, and it's not to be touched at
random external places.
This patch is an attempt to make those calls more consistent: now
snd_ctl_remove() takes the card->controls_rwsem internally, just like
other API functions for kctls. Since a few callers already take the
controls_rwsem locks, the patch removes those locks at the same time,
too.
Link: https://lore.kernel.org/r/20230718141304.1032-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- 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>
85;95;0c
This uses IRQs to track spontaneous changes to the word clock source
register.
FWIW, that this can happen in the first place is the reason why it is
futile to lock the clock source mixer setting while the device is open -
we can't consistently control the rate anyway. Though arguably, we
should reset any open streams when that happens, as they become
corrupted anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230715160738.326832-1-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>
The file is called spdif-in, but we abused it to show only sample rates
from various sources. Rectify it as far as possible (the FPGA doesn't
give us a lot of information).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-10-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is only a very partial fix - the frequency-dependent envelope & LFO
register values aren't adjusted.
But I'm not sure they were even correct at 48 kHz to start with, as most
of them are precalculated by common code which assumes an EMU8K-specific
44.1 kHz word clock, and it seems somewhat unlikely that the hardware's
register interpretation was adjusted to compensate for the different
word clock.
In any case I'm not going to spend time on fixing that, as this code is
unlikely to be actually used by anyone today.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-6-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>
The value isn't used yet; the subsequent commits will do that.
This ignores the existence of rates above 48 kHz, which is fine, as the
hardware will just switch to the fallback clock source when fed with a
rate which is incompatible with the base clock multiplier, which
currently is always x1.
The sample rate display in /proc spdif-in is adjusted to reflect our
understanding of the input rates.
This is tested only with an 0404b card without sync card, so there is a
lot of room for improvement.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The actually available clock sources depend on the available audio input
ports and dedicated clock input ports.
This includes refactoring the code to be data-driven to remain
manageable.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230612191325.1315854-3-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>
- Include the FX bus map, without which the already present send routing
info would require looking up the documentation.
- Include the physical I/O channels as known to the driver
- Make the multi-channel capture map actually name the mapped input
channels rather than "FXBUS" (Audigy) or even "???" (SbLive)
- The latter two are omitted for E-MU cards, as their physical I/O is
routed through the FPGA
- While at it, make the "Card" field somewhat more useful
This includes de-duplicating the label tables between emuproc and emufx,
updating/improving the FX bus label table, and making the SB Live! 5.1
multi-track capture channel mapping hack data-driven.
Tested-by: Jonathan Dowland <jon@dow.land>
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Include the routing information, which can be actually read back.
Somewhat as a drive-by, make the register dump format less obscure - the
previous one made no sense at all.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems to make little sense to include the FX send routing, but not
the amounts.
This also simplifies the code somewhat, and lines up the output.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The limits were appropriate only for the 2nd set.
FWIW, the channel count 4 for the 2nd set is suspicious as well - at
least P17V_PLAYBACK_FIFO_PTR actually has 8 channels, and comments on
HCFG2 hint at that as well. But all bitmasks are documented only for 4
channels. Anyway, rectifying that is out of scope for this patch.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 2nd register set belongs to the P16V chip (or embedded P17V module),
so there is nothing to show when no such part is present. Gen2 E-MU
cards have a P17V, but it's entirely unused, so we hide it there as
well.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230526101659.437969-2-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>
