remove prototypes with no matching function; ok miod@

Make audio(9) get_props() optional, remove it from duplex drivers

The property bits of audio(9) are obsolete and ought to be removed
completely.

sys/dev/audio.c:audio_open() currently uses get_props() to bail out if
read *and* write was requested on a non-duplex driver.

Drivers that currently support playing but not recording need adjustment
before the API can be cleaned up.

Drivers that advertise themselves as full duplex, i.e. those that always
return AUDIO_PROP_FULLDUPLEX unconditionally in their get_props() currently
always succeed this check.

As this is the only property, losen audio_open()'s DIAGNOSTIC check and only
do the duplex check if the driver provides get_props().

This allows for simple removal of get_props() from full-duplex drivers
without adding any other code or without changing functionality.

This includes all audio drivers under sys/dev/pci/ (maestro(4) being the
only unfinished exception here).

Other drivers as well as the API change can then follow in smaller diffs.

This builds on amd64, arm64, i386, macppc and sparc64.
amd64 with azalia(4) still plays, records as well as plays and records
at the same time on my X230 as tested with

$ aucat -i play.wav [-o rec.wav]

alpha and hppa tests by miod
OK ratchov miod

Use C99 struct init for struct audio_hw_if

This audio(9) struct will lose a member, but drivers init their struct quite
inconsistently, most pre-C99 style.

Use C99 style everywhere, to get rid of all annoying differences and allow
for easy member removals/additions:
- don't change current order of members
- no explicit NULL members
- no comments or blank lines
- trailing comma in last member line

GENERIC.MP builds fine with this on arm64, amd64, i386 and sparc64.

macppc and alpha build-tested by miod
OK ratchov miod

Constify struct {audio,midi,radio,video}_hw_if. No functional change.

ok mpi@ ratchov@ "More const is good" deraadt@

Constify struct cfattach.

ok miod@

Make room for a cookie argument passed to audio_attach_mi(). Currently
unused but intended to be used to correlate audio and wskbd devices.

ok ratchov@

dev/rndvar.h no longer has statistical interfaces (removed during various
conversion steps). it only contains kernel prototypes for 4 interfaces,
all of which legitimately belong in sys/systm.h, which are already included
by all enqueue_randomness() users.

replace add_*_randomness with enqueue_randomness()

this gets rid of the source annotation which doesn't really add
anything other than adding complexitiy. randomess is generally
good enough that the few extra bits that the source type would
add are not worth it.

ok mikeb@ deraadt@

Remove unused getdev() audio driver functions.

Remove drain(), query_encoding(), mappage() and get_default_params()
methods from all audio drivers and from the audio_if structure as they
are never called.

sizes for free(), when known; ok semarie

Remove all audio format conversion code from the kernel (btw holding
the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi

ok armani@

add a size argument to free. will be used soon, but for now default to 0.
after discussions with beck deraadt kettenis.

Introduce a global interrupt-aware mutex protecting data
structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).

ok and help from kettenis, a lot of work from armani

add two new members to structs audio_encoding and audio_prinfo.
for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.

discussed with ratchov, deraadt

allow low level audio drivers to specify a default sample format,
instead of 8-bit mono mulaw @ 8kHz.

this is just the infrastructure; no drivers are specifying a default
yet.

ok ratchov@, deanna@

do not even match on unsupported hw (instead of return from attach); from 710

round -- not truncate the buffer size

order irq,pri args to gsc_intr_establish() the way less confusing compared to other trolley busses around ...

Pass -Wformat, fix a few uninitialized variables as well.

bitneaks anonymous randomisation

make all intr_establish take the same last arg for name

nuke clause 3 & 4

string cleaning; krw ok

branches: 1.17.4;
reclaim more resources on failures.
test the harmony id to filter out unsupported yet
audio types and detect the teleshare.
detect input overload and report to userland.

Add knob for getting at the "output level bit", supposedly changes the
full scale outputs between 2.8Vpp and 2.0Vpp for line.

Get rid of the pops:
- interrupt is called when the block has been DMA'd... since LASI seems to
include a FIFO this is -not- the same as block completion which we have
to poll for. Luckily the FIFO is deep enough that polling in the
interrupt handler seems to work.
- This probably breaks capture, but I'll fix that when I get up tomorrow.
(committed to the tune of Fugazi's "Waiting Room")

reread the dma status reg after playing part of the dma got programmed.
according to the doc, the "play next" bit and therefore the device
interrupt may get raised earlier than the "record next" is set,
thus we might not program the record dma and get another harmony
interrupt right after servicing this one. this little optimization
allows to deal w/ the situation more efficiently, should it occur.
jason@ ok

real meat in audio_device; discussed w/ jason@

Check the codec revision... if it's >=CS4215E or >=AD1849K (same value
strangely =) the make ulinear (native) and slinear (emulated) available.
Also, while here, make ulinear_le:16 available via emulation.
(Btw, the LASI docs say that the format code for ulinear is a
reserved value... Probably just wasn't available in the early CS4215/AD1849's)

print out the codec revision in dmesg

Ok, you can only set the HE/SE/LE/IS bits in GAINCTL when reset is asserted (why? I have no clue, but not abding this causes the chip to wedge).
So, you can now switch input/output ports to your hearts content...
oh did I mention several commits ago that capture works?

Move device driver specific stuff to *var.h and add most of the code for
dealing with output source selection (doesn't work right yet, so its disabled).

Stupid stupid stupid... DMA will not start until the capture next register
is written. I'm not sure how to -correctly- handle this for the fullduplex
case, but this seems to make playback sound MUCH better...

As promised, I got some sleep and rewrote most of the driver... playback
is still a bit spotty, but at least interrupts are not enabled constantly
any more.
Also discovered a feature: messing with the LINE/MIC bit in GAINCTL has
the side effect of wedging the chip under some circumstances, but I think
this can be worked around by resetting the codec... later.

separate register defns and revise them based on the LASI info... a HUGE
chunk of this driver will be rewritten once I've gotten some zzz's.

I declare defeat... once interrupts are enabled, just leave them enabled.
This is annoying because the chip will keep dma'n, but everytime I disable
interrupts in halt_*, they never come back. Grr.

Write to the cntl register with the c flag set.

Ok, playback more or less works, now for capture. (I forgot to save the pintr/parg)

Ok, this now produces some noise (that's the best way I can describe it),
but the timing appears to get off some how and buffers just stay on the
chip. Oh well, at least its keeping me from playing in the traffic.

Start on the harmony audio driver. It doesn't play or capture anything
yet, but the knobs and interrupt routine seem to work. More to come.

Make audio(9) get_props() optional, remove it from duplex drivers

The property bits of audio(9) are obsolete and ought to be removed
completely.

sys/dev/audio.c:audio_open() currently uses get_props() to bail out if
read *and* write was requested on a non-duplex driver.

Drivers that currently support playing but not recording need adjustment
before the API can be cleaned up.

Drivers that advertise themselves as full duplex, i.e. those that always
return AUDIO_PROP_FULLDUPLEX unconditionally in their get_props() currently
always succeed this check.

As this is the only property, losen audio_open()'s DIAGNOSTIC check and only
do the duplex check if the driver provides get_props().

This allows for simple removal of get_props() from full-duplex drivers
without adding any other code or without changing functionality.

This includes all audio drivers under sys/dev/pci/ (maestro(4) being the
only unfinished exception here).

Other drivers as well as the API change can then follow in smaller diffs.

This builds on amd64, arm64, i386, macppc and sparc64.
amd64 with azalia(4) still plays, records as well as plays and records
at the same time on my X230 as tested with

$ aucat -i play.wav [-o rec.wav]

alpha and hppa tests by miod
OK ratchov miod

Use C99 struct init for struct audio_hw_if

This audio(9) struct will lose a member, but drivers init their struct quite
inconsistently, most pre-C99 style.

Use C99 style everywhere, to get rid of all annoying differences and allow
for easy member removals/additions:
- don't change current order of members
- no explicit NULL members
- no comments or blank lines
- trailing comma in last member line

GENERIC.MP builds fine with this on arm64, amd64, i386 and sparc64.

macppc and alpha build-tested by miod
OK ratchov miod

Constify struct {audio,midi,radio,video}_hw_if. No functional change.

ok mpi@ ratchov@ "More const is good" deraadt@

Constify struct cfattach.

ok miod@

Make room for a cookie argument passed to audio_attach_mi(). Currently
unused but intended to be used to correlate audio and wskbd devices.

ok ratchov@

dev/rndvar.h no longer has statistical interfaces (removed during various
conversion steps). it only contains kernel prototypes for 4 interfaces,
all of which legitimately belong in sys/systm.h, which are already included
by all enqueue_randomness() users.

replace add_*_randomness with enqueue_randomness()

this gets rid of the source annotation which doesn't really add
anything other than adding complexitiy. randomess is generally
good enough that the few extra bits that the source type would
add are not worth it.

ok mikeb@ deraadt@

Remove unused getdev() audio driver functions.

Remove drain(), query_encoding(), mappage() and get_default_params()
methods from all audio drivers and from the audio_if structure as they
are never called.

sizes for free(), when known; ok semarie

Remove all audio format conversion code from the kernel (btw holding
the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi

ok armani@

add a size argument to free. will be used soon, but for now default to 0.
after discussions with beck deraadt kettenis.

Introduce a global interrupt-aware mutex protecting data
structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).

ok and help from kettenis, a lot of work from armani

add two new members to structs audio_encoding and audio_prinfo.
for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.

discussed with ratchov, deraadt

allow low level audio drivers to specify a default sample format,
instead of 8-bit mono mulaw @ 8kHz.

this is just the infrastructure; no drivers are specifying a default
yet.

ok ratchov@, deanna@

do not even match on unsupported hw (instead of return from attach); from 710

round -- not truncate the buffer size

order irq,pri args to gsc_intr_establish() the way less confusing compared to other trolley busses around ...

Pass -Wformat, fix a few uninitialized variables as well.

bitneaks anonymous randomisation

make all intr_establish take the same last arg for name

nuke clause 3 & 4

string cleaning; krw ok

branches: 1.17.4;
reclaim more resources on failures.
test the harmony id to filter out unsupported yet
audio types and detect the teleshare.
detect input overload and report to userland.

Add knob for getting at the "output level bit", supposedly changes the
full scale outputs between 2.8Vpp and 2.0Vpp for line.

Get rid of the pops:
- interrupt is called when the block has been DMA'd... since LASI seems to
include a FIFO this is -not- the same as block completion which we have
to poll for. Luckily the FIFO is deep enough that polling in the
interrupt handler seems to work.
- This probably breaks capture, but I'll fix that when I get up tomorrow.
(committed to the tune of Fugazi's "Waiting Room")

reread the dma status reg after playing part of the dma got programmed.
according to the doc, the "play next" bit and therefore the device
interrupt may get raised earlier than the "record next" is set,
thus we might not program the record dma and get another harmony
interrupt right after servicing this one. this little optimization
allows to deal w/ the situation more efficiently, should it occur.
jason@ ok

real meat in audio_device; discussed w/ jason@

Check the codec revision... if it's >=CS4215E or >=AD1849K (same value
strangely =) the make ulinear (native) and slinear (emulated) available.
Also, while here, make ulinear_le:16 available via emulation.
(Btw, the LASI docs say that the format code for ulinear is a
reserved value... Probably just wasn't available in the early CS4215/AD1849's)

print out the codec revision in dmesg

Ok, you can only set the HE/SE/LE/IS bits in GAINCTL when reset is asserted (why? I have no clue, but not abding this causes the chip to wedge).
So, you can now switch input/output ports to your hearts content...
oh did I mention several commits ago that capture works?

Move device driver specific stuff to *var.h and add most of the code for
dealing with output source selection (doesn't work right yet, so its disabled).

Stupid stupid stupid... DMA will not start until the capture next register
is written. I'm not sure how to -correctly- handle this for the fullduplex
case, but this seems to make playback sound MUCH better...

As promised, I got some sleep and rewrote most of the driver... playback
is still a bit spotty, but at least interrupts are not enabled constantly
any more.
Also discovered a feature: messing with the LINE/MIC bit in GAINCTL has
the side effect of wedging the chip under some circumstances, but I think
this can be worked around by resetting the codec... later.

separate register defns and revise them based on the LASI info... a HUGE
chunk of this driver will be rewritten once I've gotten some zzz's.

I declare defeat... once interrupts are enabled, just leave them enabled.
This is annoying because the chip will keep dma'n, but everytime I disable
interrupts in halt_*, they never come back. Grr.

Write to the cntl register with the c flag set.

Ok, playback more or less works, now for capture. (I forgot to save the pintr/parg)

Ok, this now produces some noise (that's the best way I can describe it),
but the timing appears to get off some how and buffers just stay on the
chip. Oh well, at least its keeping me from playing in the traffic.

Start on the harmony audio driver. It doesn't play or capture anything
yet, but the knobs and interrupt routine seem to work. More to come.

Use C99 struct init for struct audio_hw_if

This audio(9) struct will lose a member, but drivers init their struct quite
inconsistently, most pre-C99 style.

Use C99 style everywhere, to get rid of all annoying differences and allow
for easy member removals/additions:
- don't change current order of members
- no explicit NULL members
- no comments or blank lines
- trailing comma in last member line

GENERIC.MP builds fine with this on arm64, amd64, i386 and sparc64.

macppc and alpha build-tested by miod
OK ratchov miod

Constify struct {audio,midi,radio,video}_hw_if. No functional change.

ok mpi@ ratchov@ "More const is good" deraadt@

Constify struct cfattach.

ok miod@

Make room for a cookie argument passed to audio_attach_mi(). Currently
unused but intended to be used to correlate audio and wskbd devices.

ok ratchov@

dev/rndvar.h no longer has statistical interfaces (removed during various
conversion steps). it only contains kernel prototypes for 4 interfaces,
all of which legitimately belong in sys/systm.h, which are already included
by all enqueue_randomness() users.

replace add_*_randomness with enqueue_randomness()

this gets rid of the source annotation which doesn't really add
anything other than adding complexitiy. randomess is generally
good enough that the few extra bits that the source type would
add are not worth it.

ok mikeb@ deraadt@

Remove unused getdev() audio driver functions.

Remove drain(), query_encoding(), mappage() and get_default_params()
methods from all audio drivers and from the audio_if structure as they
are never called.

sizes for free(), when known; ok semarie

Remove all audio format conversion code from the kernel (btw holding
the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi

ok armani@

add a size argument to free. will be used soon, but for now default to 0.
after discussions with beck deraadt kettenis.

Introduce a global interrupt-aware mutex protecting data
structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).

ok and help from kettenis, a lot of work from armani

add two new members to structs audio_encoding and audio_prinfo.
for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.

discussed with ratchov, deraadt

allow low level audio drivers to specify a default sample format,
instead of 8-bit mono mulaw @ 8kHz.

this is just the infrastructure; no drivers are specifying a default
yet.

ok ratchov@, deanna@

do not even match on unsupported hw (instead of return from attach); from 710

round -- not truncate the buffer size

order irq,pri args to gsc_intr_establish() the way less confusing compared to other trolley busses around ...

Pass -Wformat, fix a few uninitialized variables as well.

bitneaks anonymous randomisation

make all intr_establish take the same last arg for name

nuke clause 3 & 4

string cleaning; krw ok

branches: 1.17.4;
reclaim more resources on failures.
test the harmony id to filter out unsupported yet
audio types and detect the teleshare.
detect input overload and report to userland.

Add knob for getting at the "output level bit", supposedly changes the
full scale outputs between 2.8Vpp and 2.0Vpp for line.

Get rid of the pops:
- interrupt is called when the block has been DMA'd... since LASI seems to
include a FIFO this is -not- the same as block completion which we have
to poll for. Luckily the FIFO is deep enough that polling in the
interrupt handler seems to work.
- This probably breaks capture, but I'll fix that when I get up tomorrow.
(committed to the tune of Fugazi's "Waiting Room")

reread the dma status reg after playing part of the dma got programmed.
according to the doc, the "play next" bit and therefore the device
interrupt may get raised earlier than the "record next" is set,
thus we might not program the record dma and get another harmony
interrupt right after servicing this one. this little optimization
allows to deal w/ the situation more efficiently, should it occur.
jason@ ok

real meat in audio_device; discussed w/ jason@

Check the codec revision... if it's >=CS4215E or >=AD1849K (same value
strangely =) the make ulinear (native) and slinear (emulated) available.
Also, while here, make ulinear_le:16 available via emulation.
(Btw, the LASI docs say that the format code for ulinear is a
reserved value... Probably just wasn't available in the early CS4215/AD1849's)

print out the codec revision in dmesg

Ok, you can only set the HE/SE/LE/IS bits in GAINCTL when reset is asserted (why? I have no clue, but not abding this causes the chip to wedge).
So, you can now switch input/output ports to your hearts content...
oh did I mention several commits ago that capture works?

Move device driver specific stuff to *var.h and add most of the code for
dealing with output source selection (doesn't work right yet, so its disabled).

Stupid stupid stupid... DMA will not start until the capture next register
is written. I'm not sure how to -correctly- handle this for the fullduplex
case, but this seems to make playback sound MUCH better...

As promised, I got some sleep and rewrote most of the driver... playback
is still a bit spotty, but at least interrupts are not enabled constantly
any more.
Also discovered a feature: messing with the LINE/MIC bit in GAINCTL has
the side effect of wedging the chip under some circumstances, but I think
this can be worked around by resetting the codec... later.

separate register defns and revise them based on the LASI info... a HUGE
chunk of this driver will be rewritten once I've gotten some zzz's.

I declare defeat... once interrupts are enabled, just leave them enabled.
This is annoying because the chip will keep dma'n, but everytime I disable
interrupts in halt_*, they never come back. Grr.

Write to the cntl register with the c flag set.

Ok, playback more or less works, now for capture. (I forgot to save the pintr/parg)

Ok, this now produces some noise (that's the best way I can describe it),
but the timing appears to get off some how and buffers just stay on the
chip. Oh well, at least its keeping me from playing in the traffic.

Start on the harmony audio driver. It doesn't play or capture anything
yet, but the knobs and interrupt routine seem to work. More to come.

Constify struct {audio,midi,radio,video}_hw_if. No functional change.

ok mpi@ ratchov@ "More const is good" deraadt@

Constify struct cfattach.

ok miod@

Make room for a cookie argument passed to audio_attach_mi(). Currently
unused but intended to be used to correlate audio and wskbd devices.

ok ratchov@

dev/rndvar.h no longer has statistical interfaces (removed during various
conversion steps). it only contains kernel prototypes for 4 interfaces,
all of which legitimately belong in sys/systm.h, which are already included
by all enqueue_randomness() users.

replace add_*_randomness with enqueue_randomness()

this gets rid of the source annotation which doesn't really add
anything other than adding complexitiy. randomess is generally
good enough that the few extra bits that the source type would
add are not worth it.

ok mikeb@ deraadt@

Remove unused getdev() audio driver functions.

Remove drain(), query_encoding(), mappage() and get_default_params()
methods from all audio drivers and from the audio_if structure as they
are never called.

sizes for free(), when known; ok semarie

Remove all audio format conversion code from the kernel (btw holding
the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi

ok armani@

add a size argument to free. will be used soon, but for now default to 0.
after discussions with beck deraadt kettenis.

Introduce a global interrupt-aware mutex protecting data
structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).

ok and help from kettenis, a lot of work from armani

add two new members to structs audio_encoding and audio_prinfo.
for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.

discussed with ratchov, deraadt

allow low level audio drivers to specify a default sample format,
instead of 8-bit mono mulaw @ 8kHz.

this is just the infrastructure; no drivers are specifying a default
yet.

ok ratchov@, deanna@

do not even match on unsupported hw (instead of return from attach); from 710

round -- not truncate the buffer size

order irq,pri args to gsc_intr_establish() the way less confusing compared to other trolley busses around ...

Pass -Wformat, fix a few uninitialized variables as well.

bitneaks anonymous randomisation

make all intr_establish take the same last arg for name

nuke clause 3 & 4

string cleaning; krw ok

branches: 1.17.4;
reclaim more resources on failures.
test the harmony id to filter out unsupported yet
audio types and detect the teleshare.
detect input overload and report to userland.

Add knob for getting at the "output level bit", supposedly changes the
full scale outputs between 2.8Vpp and 2.0Vpp for line.

Get rid of the pops:
- interrupt is called when the block has been DMA'd... since LASI seems to
include a FIFO this is -not- the same as block completion which we have
to poll for. Luckily the FIFO is deep enough that polling in the
interrupt handler seems to work.
- This probably breaks capture, but I'll fix that when I get up tomorrow.
(committed to the tune of Fugazi's "Waiting Room")

reread the dma status reg after playing part of the dma got programmed.
according to the doc, the "play next" bit and therefore the device
interrupt may get raised earlier than the "record next" is set,
thus we might not program the record dma and get another harmony
interrupt right after servicing this one. this little optimization
allows to deal w/ the situation more efficiently, should it occur.
jason@ ok

real meat in audio_device; discussed w/ jason@

Check the codec revision... if it's >=CS4215E or >=AD1849K (same value
strangely =) the make ulinear (native) and slinear (emulated) available.
Also, while here, make ulinear_le:16 available via emulation.
(Btw, the LASI docs say that the format code for ulinear is a
reserved value... Probably just wasn't available in the early CS4215/AD1849's)

print out the codec revision in dmesg

Ok, you can only set the HE/SE/LE/IS bits in GAINCTL when reset is asserted (why? I have no clue, but not abding this causes the chip to wedge).
So, you can now switch input/output ports to your hearts content...
oh did I mention several commits ago that capture works?

Move device driver specific stuff to *var.h and add most of the code for
dealing with output source selection (doesn't work right yet, so its disabled).

Stupid stupid stupid... DMA will not start until the capture next register
is written. I'm not sure how to -correctly- handle this for the fullduplex
case, but this seems to make playback sound MUCH better...

As promised, I got some sleep and rewrote most of the driver... playback
is still a bit spotty, but at least interrupts are not enabled constantly
any more.
Also discovered a feature: messing with the LINE/MIC bit in GAINCTL has
the side effect of wedging the chip under some circumstances, but I think
this can be worked around by resetting the codec... later.

separate register defns and revise them based on the LASI info... a HUGE
chunk of this driver will be rewritten once I've gotten some zzz's.

I declare defeat... once interrupts are enabled, just leave them enabled.
This is annoying because the chip will keep dma'n, but everytime I disable
interrupts in halt_*, they never come back. Grr.

Write to the cntl register with the c flag set.

Ok, playback more or less works, now for capture. (I forgot to save the pintr/parg)

Ok, this now produces some noise (that's the best way I can describe it),
but the timing appears to get off some how and buffers just stay on the
chip. Oh well, at least its keeping me from playing in the traffic.

Start on the harmony audio driver. It doesn't play or capture anything
yet, but the knobs and interrupt routine seem to work. More to come.

Constify struct cfattach.

ok miod@

Make room for a cookie argument passed to audio_attach_mi(). Currently
unused but intended to be used to correlate audio and wskbd devices.

ok ratchov@

dev/rndvar.h no longer has statistical interfaces (removed during various
conversion steps). it only contains kernel prototypes for 4 interfaces,
all of which legitimately belong in sys/systm.h, which are already included
by all enqueue_randomness() users.

replace add_*_randomness with enqueue_randomness()

this gets rid of the source annotation which doesn't really add
anything other than adding complexitiy. randomess is generally
good enough that the few extra bits that the source type would
add are not worth it.

ok mikeb@ deraadt@

Remove unused getdev() audio driver functions.

Remove drain(), query_encoding(), mappage() and get_default_params()
methods from all audio drivers and from the audio_if structure as they
are never called.

sizes for free(), when known; ok semarie

Remove all audio format conversion code from the kernel (btw holding
the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi

ok armani@

add a size argument to free. will be used soon, but for now default to 0.
after discussions with beck deraadt kettenis.

Introduce a global interrupt-aware mutex protecting data
structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).

ok and help from kettenis, a lot of work from armani

add two new members to structs audio_encoding and audio_prinfo.
for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.

discussed with ratchov, deraadt

allow low level audio drivers to specify a default sample format,
instead of 8-bit mono mulaw @ 8kHz.

this is just the infrastructure; no drivers are specifying a default
yet.

ok ratchov@, deanna@

do not even match on unsupported hw (instead of return from attach); from 710

round -- not truncate the buffer size

order irq,pri args to gsc_intr_establish() the way less confusing compared to other trolley busses around ...

Pass -Wformat, fix a few uninitialized variables as well.

bitneaks anonymous randomisation

make all intr_establish take the same last arg for name

nuke clause 3 & 4

string cleaning; krw ok

branches: 1.17.4;
reclaim more resources on failures.
test the harmony id to filter out unsupported yet
audio types and detect the teleshare.
detect input overload and report to userland.

Add knob for getting at the "output level bit", supposedly changes the
full scale outputs between 2.8Vpp and 2.0Vpp for line.

Get rid of the pops:
- interrupt is called when the block has been DMA'd... since LASI seems to
include a FIFO this is -not- the same as block completion which we have
to poll for. Luckily the FIFO is deep enough that polling in the
interrupt handler seems to work.
- This probably breaks capture, but I'll fix that when I get up tomorrow.
(committed to the tune of Fugazi's "Waiting Room")

reread the dma status reg after playing part of the dma got programmed.
according to the doc, the "play next" bit and therefore the device
interrupt may get raised earlier than the "record next" is set,
thus we might not program the record dma and get another harmony
interrupt right after servicing this one. this little optimization
allows to deal w/ the situation more efficiently, should it occur.
jason@ ok

real meat in audio_device; discussed w/ jason@

Check the codec revision... if it's >=CS4215E or >=AD1849K (same value
strangely =) the make ulinear (native) and slinear (emulated) available.
Also, while here, make ulinear_le:16 available via emulation.
(Btw, the LASI docs say that the format code for ulinear is a
reserved value... Probably just wasn't available in the early CS4215/AD1849's)

print out the codec revision in dmesg

Ok, you can only set the HE/SE/LE/IS bits in GAINCTL when reset is asserted (why? I have no clue, but not abding this causes the chip to wedge).
So, you can now switch input/output ports to your hearts content...
oh did I mention several commits ago that capture works?

Move device driver specific stuff to *var.h and add most of the code for
dealing with output source selection (doesn't work right yet, so its disabled).

Stupid stupid stupid... DMA will not start until the capture next register
is written. I'm not sure how to -correctly- handle this for the fullduplex
case, but this seems to make playback sound MUCH better...

As promised, I got some sleep and rewrote most of the driver... playback
is still a bit spotty, but at least interrupts are not enabled constantly
any more.
Also discovered a feature: messing with the LINE/MIC bit in GAINCTL has
the side effect of wedging the chip under some circumstances, but I think
this can be worked around by resetting the codec... later.

separate register defns and revise them based on the LASI info... a HUGE
chunk of this driver will be rewritten once I've gotten some zzz's.

I declare defeat... once interrupts are enabled, just leave them enabled.
This is annoying because the chip will keep dma'n, but everytime I disable
interrupts in halt_*, they never come back. Grr.

Write to the cntl register with the c flag set.

Ok, playback more or less works, now for capture. (I forgot to save the pintr/parg)

Ok, this now produces some noise (that's the best way I can describe it),
but the timing appears to get off some how and buffers just stay on the
chip. Oh well, at least its keeping me from playing in the traffic.

Start on the harmony audio driver. It doesn't play or capture anything
yet, but the knobs and interrupt routine seem to work. More to come.

Make room for a cookie argument passed to audio_attach_mi(). Currently
unused but intended to be used to correlate audio and wskbd devices.

ok ratchov@

dev/rndvar.h no longer has statistical interfaces (removed during various
conversion steps). it only contains kernel prototypes for 4 interfaces,
all of which legitimately belong in sys/systm.h, which are already included
by all enqueue_randomness() users.

replace add_*_randomness with enqueue_randomness()

this gets rid of the source annotation which doesn't really add
anything other than adding complexitiy. randomess is generally
good enough that the few extra bits that the source type would
add are not worth it.

ok mikeb@ deraadt@

Remove unused getdev() audio driver functions.

Remove drain(), query_encoding(), mappage() and get_default_params()
methods from all audio drivers and from the audio_if structure as they
are never called.

sizes for free(), when known; ok semarie

Remove all audio format conversion code from the kernel (btw holding
the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi

ok armani@

add a size argument to free. will be used soon, but for now default to 0.
after discussions with beck deraadt kettenis.

Introduce a global interrupt-aware mutex protecting data
structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).

ok and help from kettenis, a lot of work from armani

add two new members to structs audio_encoding and audio_prinfo.
for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.

discussed with ratchov, deraadt

allow low level audio drivers to specify a default sample format,
instead of 8-bit mono mulaw @ 8kHz.

this is just the infrastructure; no drivers are specifying a default
yet.

ok ratchov@, deanna@

do not even match on unsupported hw (instead of return from attach); from 710

round -- not truncate the buffer size

order irq,pri args to gsc_intr_establish() the way less confusing compared to other trolley busses around ...

Pass -Wformat, fix a few uninitialized variables as well.

bitneaks anonymous randomisation

make all intr_establish take the same last arg for name

nuke clause 3 & 4

string cleaning; krw ok

branches: 1.17.4;
reclaim more resources on failures.
test the harmony id to filter out unsupported yet
audio types and detect the teleshare.
detect input overload and report to userland.

Add knob for getting at the "output level bit", supposedly changes the
full scale outputs between 2.8Vpp and 2.0Vpp for line.

Get rid of the pops:
- interrupt is called when the block has been DMA'd... since LASI seems to
include a FIFO this is -not- the same as block completion which we have
to poll for. Luckily the FIFO is deep enough that polling in the
interrupt handler seems to work.
- This probably breaks capture, but I'll fix that when I get up tomorrow.
(committed to the tune of Fugazi's "Waiting Room")

reread the dma status reg after playing part of the dma got programmed.
according to the doc, the "play next" bit and therefore the device
interrupt may get raised earlier than the "record next" is set,
thus we might not program the record dma and get another harmony
interrupt right after servicing this one. this little optimization
allows to deal w/ the situation more efficiently, should it occur.
jason@ ok

real meat in audio_device; discussed w/ jason@

Check the codec revision... if it's >=CS4215E or >=AD1849K (same value
strangely =) the make ulinear (native) and slinear (emulated) available.
Also, while here, make ulinear_le:16 available via emulation.
(Btw, the LASI docs say that the format code for ulinear is a
reserved value... Probably just wasn't available in the early CS4215/AD1849's)

print out the codec revision in dmesg

Ok, you can only set the HE/SE/LE/IS bits in GAINCTL when reset is asserted (why? I have no clue, but not abding this causes the chip to wedge).
So, you can now switch input/output ports to your hearts content...
oh did I mention several commits ago that capture works?

Move device driver specific stuff to *var.h and add most of the code for
dealing with output source selection (doesn't work right yet, so its disabled).

Stupid stupid stupid... DMA will not start until the capture next register
is written. I'm not sure how to -correctly- handle this for the fullduplex
case, but this seems to make playback sound MUCH better...

As promised, I got some sleep and rewrote most of the driver... playback
is still a bit spotty, but at least interrupts are not enabled constantly
any more.
Also discovered a feature: messing with the LINE/MIC bit in GAINCTL has
the side effect of wedging the chip under some circumstances, but I think
this can be worked around by resetting the codec... later.

separate register defns and revise them based on the LASI info... a HUGE
chunk of this driver will be rewritten once I've gotten some zzz's.

I declare defeat... once interrupts are enabled, just leave them enabled.
This is annoying because the chip will keep dma'n, but everytime I disable
interrupts in halt_*, they never come back. Grr.

Write to the cntl register with the c flag set.

Ok, playback more or less works, now for capture. (I forgot to save the pintr/parg)

Ok, this now produces some noise (that's the best way I can describe it),
but the timing appears to get off some how and buffers just stay on the
chip. Oh well, at least its keeping me from playing in the traffic.

Start on the harmony audio driver. It doesn't play or capture anything
yet, but the knobs and interrupt routine seem to work. More to come.

dev/rndvar.h no longer has statistical interfaces (removed during various
conversion steps). it only contains kernel prototypes for 4 interfaces,
all of which legitimately belong in sys/systm.h, which are already included
by all enqueue_randomness() users.

replace add_*_randomness with enqueue_randomness()

this gets rid of the source annotation which doesn't really add
anything other than adding complexitiy. randomess is generally
good enough that the few extra bits that the source type would
add are not worth it.

ok mikeb@ deraadt@

Remove unused getdev() audio driver functions.

Remove drain(), query_encoding(), mappage() and get_default_params()
methods from all audio drivers and from the audio_if structure as they
are never called.

sizes for free(), when known; ok semarie

Remove all audio format conversion code from the kernel (btw holding
the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi

ok armani@

add a size argument to free. will be used soon, but for now default to 0.
after discussions with beck deraadt kettenis.

Introduce a global interrupt-aware mutex protecting data
structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).

ok and help from kettenis, a lot of work from armani

add two new members to structs audio_encoding and audio_prinfo.
for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.

discussed with ratchov, deraadt

allow low level audio drivers to specify a default sample format,
instead of 8-bit mono mulaw @ 8kHz.

this is just the infrastructure; no drivers are specifying a default
yet.

ok ratchov@, deanna@

do not even match on unsupported hw (instead of return from attach); from 710

round -- not truncate the buffer size

order irq,pri args to gsc_intr_establish() the way less confusing compared to other trolley busses around ...

Pass -Wformat, fix a few uninitialized variables as well.

bitneaks anonymous randomisation

make all intr_establish take the same last arg for name

nuke clause 3 & 4

string cleaning; krw ok

branches: 1.17.4;
reclaim more resources on failures.
test the harmony id to filter out unsupported yet
audio types and detect the teleshare.
detect input overload and report to userland.

Add knob for getting at the "output level bit", supposedly changes the
full scale outputs between 2.8Vpp and 2.0Vpp for line.

Get rid of the pops:
- interrupt is called when the block has been DMA'd... since LASI seems to
include a FIFO this is -not- the same as block completion which we have
to poll for. Luckily the FIFO is deep enough that polling in the
interrupt handler seems to work.
- This probably breaks capture, but I'll fix that when I get up tomorrow.
(committed to the tune of Fugazi's "Waiting Room")

reread the dma status reg after playing part of the dma got programmed.
according to the doc, the "play next" bit and therefore the device
interrupt may get raised earlier than the "record next" is set,
thus we might not program the record dma and get another harmony
interrupt right after servicing this one. this little optimization
allows to deal w/ the situation more efficiently, should it occur.
jason@ ok

real meat in audio_device; discussed w/ jason@

Check the codec revision... if it's >=CS4215E or >=AD1849K (same value
strangely =) the make ulinear (native) and slinear (emulated) available.
Also, while here, make ulinear_le:16 available via emulation.
(Btw, the LASI docs say that the format code for ulinear is a
reserved value... Probably just wasn't available in the early CS4215/AD1849's)

print out the codec revision in dmesg

Ok, you can only set the HE/SE/LE/IS bits in GAINCTL when reset is asserted (why? I have no clue, but not abding this causes the chip to wedge).
So, you can now switch input/output ports to your hearts content...
oh did I mention several commits ago that capture works?

Move device driver specific stuff to *var.h and add most of the code for
dealing with output source selection (doesn't work right yet, so its disabled).

Stupid stupid stupid... DMA will not start until the capture next register
is written. I'm not sure how to -correctly- handle this for the fullduplex
case, but this seems to make playback sound MUCH better...

As promised, I got some sleep and rewrote most of the driver... playback
is still a bit spotty, but at least interrupts are not enabled constantly
any more.
Also discovered a feature: messing with the LINE/MIC bit in GAINCTL has
the side effect of wedging the chip under some circumstances, but I think
this can be worked around by resetting the codec... later.

separate register defns and revise them based on the LASI info... a HUGE
chunk of this driver will be rewritten once I've gotten some zzz's.

I declare defeat... once interrupts are enabled, just leave them enabled.
This is annoying because the chip will keep dma'n, but everytime I disable
interrupts in halt_*, they never come back. Grr.

Write to the cntl register with the c flag set.

Ok, playback more or less works, now for capture. (I forgot to save the pintr/parg)

Ok, this now produces some noise (that's the best way I can describe it),
but the timing appears to get off some how and buffers just stay on the
chip. Oh well, at least its keeping me from playing in the traffic.

Start on the harmony audio driver. It doesn't play or capture anything
yet, but the knobs and interrupt routine seem to work. More to come.

replace add_*_randomness with enqueue_randomness()

this gets rid of the source annotation which doesn't really add
anything other than adding complexitiy. randomess is generally
good enough that the few extra bits that the source type would
add are not worth it.

ok mikeb@ deraadt@

Remove unused getdev() audio driver functions.

Remove drain(), query_encoding(), mappage() and get_default_params()
methods from all audio drivers and from the audio_if structure as they
are never called.

sizes for free(), when known; ok semarie

Remove all audio format conversion code from the kernel (btw holding
the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi

ok armani@

add a size argument to free. will be used soon, but for now default to 0.
after discussions with beck deraadt kettenis.

Introduce a global interrupt-aware mutex protecting data
structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).

ok and help from kettenis, a lot of work from armani

add two new members to structs audio_encoding and audio_prinfo.
for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.

discussed with ratchov, deraadt

allow low level audio drivers to specify a default sample format,
instead of 8-bit mono mulaw @ 8kHz.

this is just the infrastructure; no drivers are specifying a default
yet.

ok ratchov@, deanna@

do not even match on unsupported hw (instead of return from attach); from 710

round -- not truncate the buffer size

order irq,pri args to gsc_intr_establish() the way less confusing compared to other trolley busses around ...

Pass -Wformat, fix a few uninitialized variables as well.

bitneaks anonymous randomisation

make all intr_establish take the same last arg for name

nuke clause 3 & 4

string cleaning; krw ok

branches: 1.17.4;
reclaim more resources on failures.
test the harmony id to filter out unsupported yet
audio types and detect the teleshare.
detect input overload and report to userland.

Add knob for getting at the "output level bit", supposedly changes the
full scale outputs between 2.8Vpp and 2.0Vpp for line.

Get rid of the pops:
- interrupt is called when the block has been DMA'd... since LASI seems to
include a FIFO this is -not- the same as block completion which we have
to poll for. Luckily the FIFO is deep enough that polling in the
interrupt handler seems to work.
- This probably breaks capture, but I'll fix that when I get up tomorrow.
(committed to the tune of Fugazi's "Waiting Room")

reread the dma status reg after playing part of the dma got programmed.
according to the doc, the "play next" bit and therefore the device
interrupt may get raised earlier than the "record next" is set,
thus we might not program the record dma and get another harmony
interrupt right after servicing this one. this little optimization
allows to deal w/ the situation more efficiently, should it occur.
jason@ ok

real meat in audio_device; discussed w/ jason@

Check the codec revision... if it's >=CS4215E or >=AD1849K (same value
strangely =) the make ulinear (native) and slinear (emulated) available.
Also, while here, make ulinear_le:16 available via emulation.
(Btw, the LASI docs say that the format code for ulinear is a
reserved value... Probably just wasn't available in the early CS4215/AD1849's)

print out the codec revision in dmesg

Ok, you can only set the HE/SE/LE/IS bits in GAINCTL when reset is asserted (why? I have no clue, but not abding this causes the chip to wedge).
So, you can now switch input/output ports to your hearts content...
oh did I mention several commits ago that capture works?

Move device driver specific stuff to *var.h and add most of the code for
dealing with output source selection (doesn't work right yet, so its disabled).

Stupid stupid stupid... DMA will not start until the capture next register
is written. I'm not sure how to -correctly- handle this for the fullduplex
case, but this seems to make playback sound MUCH better...

As promised, I got some sleep and rewrote most of the driver... playback
is still a bit spotty, but at least interrupts are not enabled constantly
any more.
Also discovered a feature: messing with the LINE/MIC bit in GAINCTL has
the side effect of wedging the chip under some circumstances, but I think
this can be worked around by resetting the codec... later.

separate register defns and revise them based on the LASI info... a HUGE
chunk of this driver will be rewritten once I've gotten some zzz's.

I declare defeat... once interrupts are enabled, just leave them enabled.
This is annoying because the chip will keep dma'n, but everytime I disable
interrupts in halt_*, they never come back. Grr.

Write to the cntl register with the c flag set.

Ok, playback more or less works, now for capture. (I forgot to save the pintr/parg)

Ok, this now produces some noise (that's the best way I can describe it),
but the timing appears to get off some how and buffers just stay on the
chip. Oh well, at least its keeping me from playing in the traffic.

Start on the harmony audio driver. It doesn't play or capture anything
yet, but the knobs and interrupt routine seem to work. More to come.

Remove unused getdev() audio driver functions.

Remove drain(), query_encoding(), mappage() and get_default_params()
methods from all audio drivers and from the audio_if structure as they
are never called.

sizes for free(), when known; ok semarie

Remove all audio format conversion code from the kernel (btw holding
the kernel_lock), as we already do better conversions in
user-mode. Yet, no need for every single driver to fiddle with the
conversion code as they are done transparently by common MI code. With
help from armani and miod, support from mpi

ok armani@

add a size argument to free. will be used soon, but for now default to 0.
after discussions with beck deraadt kettenis.

Introduce a global interrupt-aware mutex protecting data
structures (including sound-card registers) from concurent
access by syscall and interrupt code-paths. Since critical
sections remain the same, calls to splraise/spllower can be
safely replaced by calls to mtx_enter/mtx_leave with two
exceptions: (1) mutexes are not reentrant (the inner splraise
is thus removed), and (2) we're not allowed to sleep with a
mutex (either msleep is used or the mutex is released before
sleeping).

ok and help from kettenis, a lot of work from armani

add two new members to structs audio_encoding and audio_prinfo.
for both structs, the new members are 'bps' and 'msb', which
describe the number of bytes per sample and data alignment in the
sample, respectively. drivers must properly set these fields in
the 'query_encoding', 'set_parameters' and 'get_default_params'
hardware interface methods.

discussed with ratchov, deraadt

allow low level audio drivers to specify a default sample format,
instead of 8-bit mono mulaw @ 8kHz.

this is just the infrastructure; no drivers are specifying a default
yet.

ok ratchov@, deanna@

do not even match on unsupported hw (instead of return from attach); from 710

round -- not truncate the buffer size

order irq,pri args to gsc_intr_establish() the way less confusing compared to other trolley busses around ...

Pass -Wformat, fix a few uninitialized variables as well.

bitneaks anonymous randomisation

make all intr_establish take the same last arg for name

nuke clause 3 & 4

string cleaning; krw ok

branches: 1.17.4;
reclaim more resources on failures.
test the harmony id to filter out unsupported yet
audio types and detect the teleshare.
detect input overload and report to userland.

Add knob for getting at the "output level bit", supposedly changes the
full scale outputs between 2.8Vpp and 2.0Vpp for line.

Get rid of the pops:
- interrupt is called when the block has been DMA'd... since LASI seems to
include a FIFO this is -not- the same as block completion which we have
to poll for. Luckily the FIFO is deep enough that polling in the
interrupt handler seems to work.
- This probably breaks capture, but I'll fix that when I get up tomorrow.
(committed to the tune of Fugazi's "Waiting Room")

reread the dma status reg after playing part of the dma got programmed.
according to the doc, the "play next" bit and therefore the device
interrupt may get raised earlier than the "record next" is set,
thus we might not program the record dma and get another harmony
interrupt right after servicing this one. this little optimization
allows to deal w/ the situation more efficiently, should it occur.
jason@ ok

real meat in audio_device; discussed w/ jason@

Check the codec revision... if it's >=CS4215E or >=AD1849K (same value
strangely =) the make ulinear (native) and slinear (emulated) available.
Also, while here, make ulinear_le:16 available via emulation.
(Btw, the LASI docs say that the format code for ulinear is a
reserved value... Probably just wasn't available in the early CS4215/AD1849's)

print out the codec revision in dmesg

Ok, you can only set the HE/SE/LE/IS bits in GAINCTL when reset is asserted (why? I have no clue, but not abding this causes the chip to wedge).
So, you can now switch input/output ports to your hearts content...
oh did I mention several commits ago that capture works?

Move device driver specific stuff to *var.h and add most of the code for
dealing with output source selection (doesn't work right yet, so its disabled).

Stupid stupid stupid... DMA will not start until the capture next register
is written. I'm not sure how to -correctly- handle this for the fullduplex
case, but this seems to make playback sound MUCH better...

As promised, I got some sleep and rewrote most of the driver... playback
is still a bit spotty, but at least interrupts are not enabled constantly
any more.
Also discovered a feature: messing with the LINE/MIC bit in GAINCTL has
the side effect of wedging the chip under some circumstances, but I think
this can be worked around by resetting the codec... later.

separate register defns and revise them based on the LASI info... a HUGE
chunk of this driver will be rewritten once I've gotten some zzz's.

I declare defeat... once interrupts are enabled, just leave them enabled.
This is annoying because the chip will keep dma'n, but everytime I disable
interrupts in halt_*, they never come back. Grr.

Write to the cntl register with the c flag set.

Ok, playback more or less works, now for capture. (I forgot to save the pintr/parg)

Ok, this now produces some noise (that's the best way I can describe it),
but the timing appears to get off some how and buffers just stay on the
chip. Oh well, at least its keeping me from playing in the traffic.

Start on the harmony audio driver. It doesn't play or capture anything
yet, but the knobs and interrupt routine seem to work. More to come.