The widget node 0x21 should be initialized as unmuted/full (0dB)
as default. This will reduce additional manual work by user at the
first time use.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The SPDIF output is toggled at each time any SPDIF status bits are changed
because of the known problems on some codecs. But, this also results in
loosing the sync, and the problem is more obvious on HDMI output over
SPDIF. Since the toggle is necessary only for some codecs, we should
check whether this workaround is needed and skip if unnecessary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in other places, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in sound/pci/*, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in sound/isa/*, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in sound/core/*, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Introduced snd_BUG_ON() macro as a replacement of snd_assert() macro.
snd_assert() is pretty ugly as it has the control flow in its argument.
OTOH, snd_BUG_ON() behaves like a normal conditional, thus it's much
easier to read the flow.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On the Freescale MPC8610, SSI1 is hard-coded to use DMA channels 0 and 1
for playback and capture, and SSI2 is hard-coded to use DMA channels 2 and 3.
This patch fixes the fabric driver so that it uses the right channels.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The WM8580 is an audio CODEC designed for DVD and surround sound
applications, offering three stereo DACs and a stereo ADC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Make snd_ad1848_probe() easier to follow. With an exception for only
trying once as soon as the codec is out of init (which should be all
that's needed) the detection logic should be unchanged.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Acked-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use pin 37 for mono-out as default on ALC203.
Reported-by: george pee <georgepee@gmail.com>
Tested-by: george pee <georgepee@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The support for the SB Extigy's remote seems to be broken in all
recent ALSA versions, including 1.0.17. The driver detects the event
correctly, then submits a URB to query the RC code. On the Extigy, the
URB is submitted with a length of 2 bytes. My hardware, however, only
replies with 1 byte, containing the correct RC button code. The
current implementation discards this as being too short. (line 1784 of
usbmixer.c)
This patch specifies a "minimum packet length" in the remote control
configuration. I've left the values for the Audigy 2/Live! the same as
the packet length, as I'm assuming the existing code works with them.
(I don't have the hardware to confirm) This fixes the Extigy RC
support, e.g. for use with Lirc.
Signed-off-by: Phillip Michael Jordan <phil@philjordan.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The WM8900 is designed for portable multimedia applications requiring
low power consumption, high performance audio and a compact form factor
providing:
- 24 bit stereo ADC and DAC
- Microphone and line inputs
- Line outputs
- Class G headphone amplifier
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Thanks to Sistema Fenix (http://www.sistemafenix.com.br/) and CDI Brasil
(www.cdibrasil.com.br/) for sponsoring this development.
Signed-off-by: Gilberto <gilberto@sistemafenix.com.br>
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds sound support for NEC Versa S9100
With it, we get sound on the internal speaker and headphone (with
automute working) while there is no sound by default.
External mic also works fine but I don't know if there is an internal
one (if there is an internal mic it does not work currently), and I
had to send back the hardware.
Signed-off-by: Pascal Terjan <pterjan@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
After the transition from cs4321_lib to wss_lib, azt2320 probe visits
snd_ad1848_probe during detection. It expects register 0 (LEFT_INPUT)
to be able to retain the value 0xaa during detection but AZT2320 does
not support MIC gain meaning it reads back as 0x8a instead.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Acked-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Remove initializations of spinlock and mutexes which
are done earlier in snd_wss_new(). The snd_wss_new()
is called from snd_wss_create().
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
snd-opti92x-ad1848 mistakingly passes WSS_HW_OPTI93X currently. This
fixes it as tested with a OPTi 82C929A/AD1848 card.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Acked-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Limit opti93x cards capture formats to only linear ones.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss detection code and kill the ad1848 library.
The library is fully assimilated into the new wss library.
This required reworking of the AD1848 family code
so the code is changed to correctly detect chips from
the AD1848 and CS4231 families.
I have tested it on following cards:
Gallant SC-6600 (codec: AD1848, driver: snd-sc6600)
SoundScape VIVO/90 (codec: AD1845, driver: snd-sscape)
SG Waverider (codec: CS4231A, driver: Rene Herman's snd-galaxy)
Opti930 (codec: built-in - CS4231 compatible, driver: snd-opti93x)
Opti931 (codec: built-in - CS4231 compatible, driver: snd-opti93x)
Gallant SC-70P (chip/codec: CS4237B, driver: snd-cs4236)
Audio Plus 3D (chip/codec: CMI8330A, driver: snd-cmi8330)
Dell Latitude CP (chip/codec: cs4236, driver snd-cs4232)
Sound playback and recording works on all these cards.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss pcm code and kill the ad1848 pcm code.
The AD1848 chip is much slower than CS4231 chips
so the waiting loop was increased 100x (10x is not
enough).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss mixer code and kill the ad1848 mixer code.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use CS4231P instead of AD1848P (kill the AD1848P).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss macros instead of ad1848 ones.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use wss constants for mode.
Move ad1848 hardware constants to the wss.h.
Move mixer tlv macros into the ad1848_lib.c from the ad1848.h.
Drop the MODE_RUNNING spurious IRQ guard on AD1848 as it doesn not seem
to be needed.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The snd_wss is superset of the snd_ad1848 so kill
the latter and replace it with the snd_wss.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rename functions and structures from the former
cs4321_lib to names more corresponding with the
new name: wss_lib.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rename file include/sound/cs4231.h
into include/sound/wss.h
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Move the file sound/isa/cs423x/cs4231_lib.c
into sound/isa/cs423x/wss_lib.c
This is the first step toward merging all libraries
for Windows Sound System compatible chips
into a single library.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine
to have more than 8 PCM devices per card, except one place - the
SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate
devices > 7. This patch fixes the issue, changing the devices list
organisation.
Instead of adding new device to the tail, the list is now kept always
ordered (by card number, then device number). Thus, during enumeration,
it is easy to discover the fact that there is no more given card's
devices.
Additionally the device field of struct snd_pcm had to be changed to int,
as its "unsignednity" caused a lot of problems when comparing it to
potentially negative signed values. (-1 is 0xffffffff or even more then ;-)
Signed-off-by: Pawel Moll <pawel.moll@st.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a new US122L usb-audio driver. This driver works together with a
dedicated alsa-lib plugin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This reverts commit fb3d6f2b77bdec75d45aa9d4464287ed87927866.
New, updated patch with same subject replaces this commit.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Updated IEC958 consumer status channel definitions according
to the third edition of IEC60958-3 spec.
Signed-off-by: Pawel Moll <pawel.moll@st.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine
to have more than 8 PCM devices per card, except one place - the
SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate
devices > 7. This patch fixes the issue, changing the devices list
organisation.
Instead of adding new device to the tail, the list is now kept always
ordered (by card number, then device number). Thus, during enumeration,
it is easy to discover the fact that there is no more given card's
devices. The same limit was present in OSS emulation code. It has
been fixed as well.
Additionally the device field of struct snd_pcm is now int, instead of
unsigned int, as there is no obvious reason for keeping it unsigned.
This caused a lot of problems with comparing this value with other
(almost always signed) variables. There is just one more place where
device number is unsigned - in struct snd_pcm_info, which should be
also sorted out in future.
Signed-off-by: Pawel MOLL <pawel.moll@st.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASOC: convert use of uint to unsigned int
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Allow all the CODEC drivers to be built without a machine driver in order
to facilitate testing of subsystem-wide changes and gain better coverage
from automated testing efforts. This also helps things like the generic
OpenFirmware machine driver for PowerPC.
Currently AC97 CODECs are not included since the current setup relies
on having a controller driver available.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This fixes sparse warnings and allows non-OpenFirmware systems to attempt
to bind to the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASoC Codec driver for the TLV320AIC26 device. As it stands, this driver
doesn't support all the modes and clocking options of the AIC16, but it
is a start.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
