distinct hardware playback channels. DAC configuration can be
accessed through kernel hint - hint.pcm.<unit>.dac="val" with
following possible values:
0 = Enable both DACs (default)
1 = Enable single DAC (DAC1)
2 = Enable single DAC (DAC2)
3 = Enable both DACs, swap position (DAC2 comes first instead
of DAC1)
Special case for ES1370:
Unlike ES1371,2,3/CT5880, volume for each DAC 1 and 2 can be
controlled indepedently (synth for DAC1, pcm for DAC2). It is
possible that user will confuse by this behaviour, since both
DACs are enabled by default. Thus, provide a knob through sysctl
hw.snd.pcm<unit>.single_pcm_mixer:
0 = each DACs will be controlled separately (synth/pcm).
1 = combine both DACs volume mixer controller into a single
"pcm" (default)
As a side note, fixed rate operation (provided by previous
commit) is not a mandatory if the configuration space does not
involve DAC2 (perhaps disabled by user through the above kernel
hint). Unlike DAC2, DAC1 has its own register / control space,
not affected by the speed settings of ADC.
Tested by: multimedia@
Approved by: netchild (mentor)
sampling rate between playback and recording. This can be
disabled / enabled via kernel hints
(hint.pcm.<unit>.fixed_rate=0/4000-48000) or sysctl
hw.snd.pcm<unit>.fixed_rate=0/4000-48000). Default to 48khz
fixed rate. [1]
* Basic cleanup. *_es1371x_* -> *_es137x_*.
* Some locking fixes. [2]
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Discussed with: yongari [2]
See also: http://lists.freebsd.org/pipermail/freebsd-multimedia/2005-September/002758.html [1]
Reported by: Jos Backus <jos at catnook.com> [1]
especially for CT4730 / EV1938 chip, causing misconfigured mixer
(David Xu), crippled after power cycle (Kevin Oberman). Fixed.
* Incorporate locking/spdif patches from Jon Noack / matk. Not all
es137x can really do spdif, clean it up a bit to only let few capable
chip. This adds a "hw.snd.pcm<unit>.spdif_enabled" sysctl until
a more generic way of handling this from userland (by an ordinary
user) is designed/implemented.
* Convert all bus_space_(read|write) to use es_rd/es_wr, simmilar
with other drivers.
* Add tunable hw.snd.pcm<unit>.latency_timer sysctl to toggle pci
latency timer value on the fly. Much noise / pop / crackling
issues can be solved by increasing its value. Other people have
pointed out to use pciconf instead, but this is just an added
value specific for CT4730/EV1938.
* Remove es137x specific debug sysctl/code.
Several PRs can now be closed.
Submitted by: Ariff Abdullah <skywizard@MyBSD.org.my>
Submitted by: Jon Noack <noackjr@alumni.rice.edu> (implicit)
Submitted by: matk (implicit)
PR: 59349, 68594, 73498
Tested by: multimedia@
assign DMA address to the wrong address. It can cause system lockup
or other mysterious errors. Since most sound cards requires low DMA
address(BUS_SPACE_MAXADDR_24BIT) sndbuf_alloc() would fail when the
audio driver is loaded after long running of operations.
Approved by: jake (mentor)
Reviewed by: truckman, matk
- `sound'
The generic sound driver, always required.
- `snd_*'
Device-dependent drivers, named after the sound module names.
Configure accordingly to your hardware.
In addition, rename the `snd_pcm' module to `sound' in order to sync
with the driver names.
Suggested by: cg
Add two new arguments to bus_dma_tag_create(): lockfunc and lockfuncarg.
Lockfunc allows a driver to provide a function for managing its locking
semantics while using busdma. At the moment, this is used for the
asynchronous busdma_swi and callback mechanism. Two lockfunc implementations
are provided: busdma_lock_mutex() performs standard mutex operations on the
mutex that is specified from lockfuncarg. dftl_lock() is a panic
implementation and is defaulted to when NULL, NULL are passed to
bus_dma_tag_create(). The only time that NULL, NULL should ever be used is
when the driver ensures that bus_dmamap_load() will not be deferred.
Drivers that do not provide their own locking can pass
busdma_lock_mutex,&Giant args in order to preserve the former behaviour.
sparc64 and powerpc do not provide real busdma_swi functions, so this is
largely a noop on those platforms. The busdma_swi on is64 is not properly
locked yet, so warnings will be emitted on this platform when busdma
callback deferrals happen.
If anyone gets panics or warnings from dflt_lock() being called, please
let me know right away.
Reviewed by: tmm, gibbs
es137x.c: In function `es1371_rdcd':
es137x.c:598: warning: `x' might be used uninitialized in this function
PR: kern/35408
Submitted by: Thomas Quinot <thomas@cuivre.fr.eu.org>
* add new channels to the end of the list so channels used in order of
addition
* de-globalise definition of struct snddev_info and provide accessor
functions where necessary.
* move the $FreeBSD$ tag in each .c file into a macro and allow the
/dev/sndstat handler to display these when set to maximum verbosity to aid
debugging.
* allow each device to register its own sndstat handler to reduce the amount
of groping sndstat must do in foreign structs.
worked before.
mixer, dsp and sndstat are seperate devices - give them their own cdevsws
instead of demuxing requests sent to a single cdevsw.
use the si_drv1/si_drv2 fields in dev_t structures for holding information
specific to an open instance of mixer/dsp.
nuke /dev/{dsp,dspW,audio}[0-9]* links - this functionality is now provided
using cloning.
various locking fixes.
this introduces a new buffering mechanism which results in dramatic
simplification of the channel manager.
as several structures have changed, we take the opportunity to move their
definitions into the source files where they are used, make them private and
de-typedef them.
the sound drivers are updated to use snd_setup_intr instead of
bus_setup_intr, and to comply with the de-typedefed structures.
the ac97, mixer and channel layers have been updated with finegrained
locking, as have some drivers- not all though. the rest will follow soon.
this apparently fixes problems initialising certain es1371/es1373/ct5880
revisions.
Confirmed working by: Richard J Kuhns <rjk@grauel.com>
PR: i386/25944
modify chn_setblocksize() to pick a default soft-blocksize appropriate to the
sample rate and format in use. it will aim for a power of two size small
enough to generate block sizes of at most 20ms. it will also set the
hard-blocksize taking into account rate/format conversions in use.
update drivers to implement setblocksize correctly:
updated, tested: sb16, emu10k1, maestro, solo
updated, untested: ad1816, ess, mss, sb8, csa
not updated: ds1, es137x, fm801, neomagic, t4dwave, via82c686
i lack hardware to test: ad1816, csa, fm801, neomagic
others will be updated/tested in the next few days.
this gives us several benefits, including:
* easier extensibility- new optional methods can be added to
ac97/mixer/channel classes without having to fixup every driver.
* forward compatibility for drivers, provided no new mandatory methods are
added.
change channel interface - kobj implementation coming soonish
make pcm_makelinks not panic if modular
add pcm_unregister()
these changes support newpcm kld unloading, but this is only implemented
by ds1.c
modify driver capability reporting format to list every audio format
seperately- required for above and because we could not previously indicate
that mono was unsupported.
there should be no functional impact.
* add a callback for initialising the mixer interface
* support ac97 2.1 variable rate audio feature
fix ac97-using drivers for the above
add suspend/resume support for neomagic
