change id to elem_id as it is used to initialize each mixer element
sound/pci/maestro3.c:2071:25: warning: symbol 'id' shadows an earlier one
sound/pci/maestro3.c:67:13: originally declared here
index is used in each of these places to count over the dsp's memory,
change to the name dsp_index
sound/pci/maestro3.c:2572:9: warning: symbol 'index' shadows an earlier one
sound/pci/maestro3.c:66:12: originally declared here
sound/pci/maestro3.c:2604:9: warning: symbol 'index' shadows an earlier one
sound/pci/maestro3.c:66:12: originally declared here
[tiwai - fixed coding style issues as well]
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
id was only used as a counter in a for loop, move the declaration
to where it is used and change it to i.
sound/pci/fm801.c:1288:6: warning: symbol 'id' shadows an earlier one
sound/pci/fm801.c:51:13: originally declared here
[tiwai - fixed a coding style issue as well]
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
id is used when initializing the mixer elements, use elem_id here
instead.
sound/pci/es1968.c:1963:25: warning: symbol 'id' shadows an earlier one
sound/pci/es1968.c:129:13: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
index is incremented only when AC97_EI_SPDIF and then assigned to
the index field. Change the temporary name to is_spdif.
sound/pci/ens1370.c:1638:10: warning: symbol 'index' shadows an earlier one
sound/pci/ens1370.c:84:12: originally declared here
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A temporary variable for each mixer element is used in an initialization
loop, use the name elem_id.
sound/pci/cmipci.c:2747:26: warning: symbol 'id' shadows an earlier one
sound/pci/cmipci.c:56:13: originally declared here
[tiwai - fixed a coding style issue as well]
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bits indicating SPDIF frequency in the status register are not the same for
the 9632 than for the other cards, because it also supports 192kHz. A specific
bitmask has thus been added (used in hdsp_spdif_sample_rate()).
The 9632 does not seem to report external sample rates greater than 96kHz. In
this case, the best seems to report spdif rate when autosync reference is
spdif. This also required to move function hdsp_spdif_sample_rate().
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added logic to check if AUTO_PIN_FRONT_MIC is available for output
switch, if AUTO_PIN_MIC isn't.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some laptops have a internal analog microphone that is not setup by the BIOS.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added PCI_QUIRKS for laptop that have the 92HDxxx family of codecs.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC_DELL_BIOS quirks were setting the association value wrong
for port 0x0f, which prevented it from being included in hp_outs[].
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix issue on STAC927x codecs that first DAC was getting powered down
even if was being used.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize the capture source properly for auto model.
It's especially important for cases that only mic is detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the beep volume control to ALC268 codec support code.
Since the codec doesn't return the correct AMP caps, we need to override
the value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a private array for TLV entries of virtual master controls instead
of (supposed) static array. This cleans up the existing codes.
Also, now vmaster assumes the simple dB-range TLV that is the only type
it can handle.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the codes for virtual master controls to sound core part so that
not only hda-intel drivers can use it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of 8 channel sound for codecs that are known to work.
So far, only ALC850 is marked as a 8ch-support codec.
This fix is a modified version of the patch on ALSA BTS#2097 by
Martin Ellis:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2097
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of new AD codecs: AD1883, AD1884A, AD1984A and AD1984B.
These are almost compatible except for additional digital pins, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC262 must have capsrc_nids defined as well as in ALC882.
Also, add a NULL check in alc882_mux_enum_put to avoid Oops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last patch for fixing the auto-config pin setting breaks the resume
due to a wrong use of snd_hda_codec_amp_stereo(). The code in the init
hook shouldn't touch the amp cache.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new mixer switch to enable/disable the sharing of the default
PCM stream with analog and SPDIF outputs. When "IEC958 Default PCM"
switch is on, the PCM stream is routed both to analog and SPDIF outputs.
This is the behavior in the earlier version.
Turning this switch off has a merit for some codecs, though. Some codec
chips don't support 24bit formats for SPDIF but only for analog outputs.
In this case, you can use 24bit format by disabling this switch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes some bugs in the auto-configurator of Realtek codecs:
- add missing pin set-up for speaker pins
- fix the speaker auto-mute function not to conflict with the existing
"Speaker" mixer switch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some cases, the BIOS sets up only the HP pins with different assoc
and sequence numbers, e.g. on FSC Esprimo with ALC262.
This patch adds a fix-up for such a case. When multiple HPs are defined
and no line-outs is found, the configurator tries to re-assign some pins
from HP list to line-out, judging from the sequence number.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implemented the auto-mic jack sensing for Samsung laptops with AD1986A
codec chip (model=laptop-eapd).
The hardware uses pin 0x1d and 0x1f for the internal and external
mics, respectively.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the codes of the capture source selection for Realtek codecs.
Now using common helper functions with the new capsrc_nids field.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reprogram the speaker-pin setting at each HP pin plug to make sure
the spekaer auto-muting on AD1981HD hp model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I've just noticed that there are a handful of duplicate controls in the
ALC268 test model mixer. This patch (against alsa-driver 1.0.16) removes
them.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
See ALSA bug#3327 for more details. Experimental.
Also fix support for M-Audio Delta 1010E - subdevice check.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The HD-audio hardware usually supports 64bit address for DMA and other
buffers. The patch enables the feature if supported.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On several laptops that have STAC9228 codecs have unused DACs,
this powers them down to a D3 state.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes the problems with Midi In on Hoontech/STA dsp24 cards, for example with
DSP2000 box, without restricting the box configurations available. Also adds
mpu_401 name strings.
Signed-off-by: Alan Horstmann <gineera@aspect135.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current scheme, PCM device numbers are assigned incrementally
in the order of codecs. This causes problems when the codec number
is irregular, e.g. codec #0 for HDMI and codec #1 for analog. Then
the HDMI becomes the first PCM, which is picked up as the default
output device. Unfortuantely this doesn't work well with normal
setups.
This patch introduced the fixed device numbers for the PCM types,
namely, analog, SPDIF, HDMI and modem. The PCM devices are assigned
according to the corresponding PCM type. After this patch, HDMI will
be always assigned to PCM #3, SPDIF to PCM #1, and the first analog
to PCM #0, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
