Don't take the codec mutex during machine probe until we have registered
with ALSA, fixing a lockdep warning reported by Dmitry Baryshkov.
Cc: Dmitry Baryshkov <dbaryshkov@gmail.com>
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>
The TempoTec HiFier has a somwhat different architecture; remove it from
the list of cards that are known to be supported.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The available sample rates on CS5535 depend on AC97 codec chip.
Set the additional hw params limit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add one more indirection to the lookup of the snd_pcm_hardware
structures so that we can save the space of the duplicate ones.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Change the oxygen_spdif_input_bits_changed() function so that clock
changes on the SPDIF input are correctly detected. This means that
sample rates greater than 96 kHz are now supported.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fix some wrong values for the definitions of the source masks for DACS 1
and 3.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Actually use the template that was maybe changed by the control filter
instead of the original one.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Instead of having model-specific fields in the common struct oxygen, put
them into a private structure that is allocated together with the card
structure.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
According to the datasheets, the SPI clock cycle must be at least 200 ns
for the AK4396 and the WM8785, so we cannot use the default 160 ns.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Allow to record with 32 kHz, 64 kHz, 88.2 kHz and 176.4 kHz with cards
that have a WM8785 ADC.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add symbol definitions for the various codecs and GPIO pins.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The MIDI bit in the MISC register is set by default and cannot be used
to detect the presence of a MIDI port. Instead, add a parameter to the
oxygen_pci_probe() function so that model drivers can specify this.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the DMA_PAUSE register for pausing instead of stopping DMA.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Reported by Al Viro:
In copy_tlv(), the size of kmalloc is wrongly calculated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
In azx_[rirb_]get_response(), the timeout is checked at the end of the loop.
It's better to be checked just after the check of the RIRB index to avoid
a bogus error with a too long msleep().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Clean up ice1712/ice1724 codes. The board-specific data is allocated
locally in each code instead of having an ungly union in struct ice1712.
Also, fix coding issues in prodigy_hifi.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds additional output to the /proc codec#X info.
The following pieces of information are added to the output:
- Balanced, L/R swap, trigger, impedance sense pin capabilities
- Vref pin capabilities
- Current Vref pin widget control setting
- Default configuration association, sequence, and misc bit test
- EAPD/BTL bits conveying balanced mode, EAPD, and L/R swap
- Power state modified to show state name as well as setting vs actual value
- GPIO parameter output on Audio Function Group, including enumeration of IO
pins which are indicated present (Any I and O pins are not output at this
time)
- Stripe and L/R swap widget capabilities
- All digital converter bits: enable, validity, validity config, preemphasis,
copyright, non-audio, professional, generation level, and content category
- Converter stream and channel values for in/out widgets
- SDI select value for in widgets
- Unsolicited response widget capability tag and enabled bit
- Delay widget capability value
- Processing widget capability benign bit and number of coefficients
- Realtek Define Registers: processing coefficient, coefficient index
[Also, fixed space/tab issues and make codes a bit more readable
-- Takashi]
Signed-off-by: Andrew Paprocki <andrew@ishiboo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The default playback routing must be 0xe4, not 0xe1; the front and
surround DACs were exchanged.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add more symbols for registers and register fields.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Enable mmap support in the MPC8610 ASoC driver. The driver can use ALSA's
default mmap functionality, it was just not enabled previously.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
For codec conexant 5045, I found that the name of 'Capture Source Items'
is different from the name of mixer.
The mixer is:
HDA_CODEC_VOLUME('Ext Mic Playback Volume', 0x17, 0x2, HDA_INPUT),
HDA_CODEC_MUTE('Ext Mic Playback Switch', 0x17, 0x2, HDA_INPUT),
But the capture source item is :
static struct hda_input_mux cxt5045_capture_source = {
.num_items = 2,
.items = {
{ 'IntMic', 0x1 },
{ 'LineIn', 0x2 },
}
};
I think that it's better to change the name of capture_source to avoid
misunderstanding.
Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The benq r55e laptop have 3 jacks on the front panel.
One for HP, one for Line In and one for Mic In.
This patch implemented a new model to support it.
Signed-off-by: Jiang Zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add proper register bit symbols for the I2S format field, and allow card
models to configure the I2S format to be used for the DACs and ADCs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
See ALSA bug#2384 for more details.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Konstantin Kletschke <konsti@ku-gbr.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Codec with id 11c1:1040 sitting on hda bus isn't si3054-compatible.
It should be removed from patch_si3054.c id list.
The detailed information
http://archives.linmodems.org/26457
From: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Some machines with STAC codecs seem to have problems (e.g. no audible
playback) when the delay in codec-read routine is too short.
I still don't figure out which command sequence causes this problem
(due to lack of test hardware), but it's known that increasing the
delay fixes. So, added a stupid workaround here temporarily...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This device has io_type == 1 (Multiface) and firmware_rev > 0xa
(fixes regression from changeset 5326)
Signed-off-by: Andreas Degert <ad@papyrus-gmbh.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch provides the snd-powermac sound driver with the ability to handle
dead DMA transfers. If a dead DMA transfer is detected, the driver now sets
up a new DMA transfer to continue with the sound output at the point where the
old transfer died.
This dead DMA transfer handling has become necessary with recent kernels on
certain G4 PowerMacs. Please refer to the following URLs for more information:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3126https://bugs.launchpad.net/ubuntu/+source/linux-source-2.6.20/+bug/87652http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=436723
The patch is based on the dead DMA transfer handling code from the old dmasound
driver which can be found in the file sound/oss/dmasound/dmasound_awacs.c in
the Linux source code.
Signed-off-by: T. H. Huth <th.huth@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fix section mismatch in caiaq: these __devinit functions can be
called at any time so they should not be __devinit.
WARNING: vmlinux.o(.text+0x10a8dae): Section mismatch: reference to .init.text:snd_usb_caiaq_audio_init (between 'setup_card' and 'create_card')
WARNING: vmlinux.o(.text+0x10a8dd6): Section mismatch: reference to .init.text:snd_usb_caiaq_midi_init (between 'setup_card' and 'create_card')
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fix section mismatch in hdsp: snd_hdsp_proc_init() can be called from
an ioctl at any time.
WARNING: vmlinux.o(.text+0x1089bc2): Section mismatch: reference to .init.text: (between 'snd_hdsp_create_alsa_devices' and 'snd_hdsp_free')
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fix section mismatch in atiixp by making some functions __devinit.
WARNING: vmlinux.o(.text+0xfd9304): Section mismatch: reference to .init.data:atiixp_quirks (between 'ac97_probing_bugs' and 'snd_atiixp_codec_detect')
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fix section mismatch in ad1889 by renaming the pci_driver variable to a
whitelisted variable name.
WARNING: vmlinux.o(.data+0x2e5ff0): Section mismatch: reference to .init.text:snd_ad1889_probe (between 'ad1889_pci' and 'index')
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fix section mismatches in mts64 by making a static variable __devinitdata.
WARNING: vmlinux.o(.data+0x2e33f0): Section mismatch: reference to .init.data:mts64_ctl_smpte_switch (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e33f8): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_hours (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e3400): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_minutes (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e3408): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_seconds (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e3410): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_frames (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e3418): Section mismatch: reference to .init.data:mts64_ctl_smpte_fps (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add a callback to the model structure to allow modification of the
hardware PCM limits.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Allow the models to modify mixer controls before they are added to the
card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Allow the card models to specify whether each of the hardware DMA
channels is used.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add a field to the model structure so that it is possible to have a card
where the SPI outputs 4 and 5 are used for an EEPROM.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Only initialize and create mixer controls for the first AC97 codec when
one has actually been detected.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Enable the SPDIF input-pin on VIA codecs when SPDIF-input is enabled
by BIOS. Also, including a bit code clean up.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added the support of Dell OEM laptops (Vostro 1200) with ALC268 codec.
The new model=dell is provided.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
It turned out that the PCBEEP element (0x1d) is disabled on some hardwares
although it's defined in the datasheet. Because of the error at info of
this element, the mixer gets totally unusable.
Since the PCBEEP isn't that important feature, it's safer to disable this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
There are two similar GPIO-enablement codes in patch_sigmatel.c.
Let's clean up.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch removes hardcoded values for the number of streams supported
by the southbridge in most chipsets, and reads these values from the
chipset directly. Most systems are hardwired for 4 streams in each
direction, but newer chipsets change that capability.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rename the 'PCM Playback Volume'/'Switch' mixer controls to 'Master'.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This busy_wait is not needed after latest changes
to the cs4231-lib
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>
The line input cannot be mixed with the other inputs, so we have to mute
the other input switches when it is selected.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use an array for the pointers to known controls so that it is easier to
add more.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Do not exchange the surround and back jacks except when in 7.1 mode
where the surround jack is not rear but side.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
'snd_shutdown_f_ops' is not a pointer so its address will never be NULL.
GCC will complain because 'fops_get' will do an unnecessary check because
'&snd_shutdown_f_ops' is always true.
Signed-off-by: Miguel Boton <mboton@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch fixes 5.1 surround output and headphone detection in the
Dell Inspiron 530 and possibly other Dell systems using the ALC888
codec (mode 6stack-dell).
Signed-off-by: Claudio Matsuoka <cmatsuoka@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fixes multiple device support by incrementing the static device counter
at the end of the azx_probe() call. Without this, subsequent probes would
always use the index specified for the first card.
Signed-off-by: Andrew Paprocki <andrew@ishiboo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add the ASoC drivers for the Freescale MPC8610 SoC and the MPC8610 HPCD
reference board.
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 ALC auto-probe creates mic boost controls automatically for the
probed pins, but it assumes that they are analog mics. The digital
mics have no boost controls and must be skipped.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Print the name of the defect controls in error messages in amp info
callback. This will make debugging easier.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a new STAC 9205 quirk for Vostro 1500.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Change semantics for SNDRV_PCM_TSTAMP_MMAP. Doing timestamping only in
the interrupt handler might cause that hw_ptr is not related to actual
timestamp. With this change, grab timestamp at every hw_ptr update to
have always valid timestamp + ring buffer position pair.
With this change, SNDRV_PCM_TSTAMP_MMAP was renamed to
SNDRV_PCM_TSTAMP_ENABLE. It's no regression (I think).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added support for advanced power management support for output ports on
92HD7xxx family of codecs. Inactive output ports are powered down when
the pin sense doesn't detect a connection, and powered back up when a
connection is sensed.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add master controls using vmaster to codecs that have no real hardware
master volume registers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added helper functions to implement virtual master volume controls.
The virtual master control is a control element that has multiple
slave controls. The value of master element is equally added to
slave elements.
The functions are written for general purpose, but it's put in the
HD-audio directory as now, since HD-audio driver is the only user.
It should be moved to the common place once after other drivers use
vmaster.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Currently only the AUX channel is used (touchscreen)
Signed-off-by: Ian Molton <spyro@f2s.com>
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>
Support for suspend/resume for the S3C2412 ASoC IIS
core driver.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
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>
S3C2412 SoC IIS support for ALSA/ASoC
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
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>
This one changes the DMA initialisation as it turns out the DMA driver
in s3c24xx doesnt store registers between suspend/resume so you have
to re-initialise the channels on every resume.
Signed-off-by: Graeme Gregory <graeme@openmoko.com>
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>
snd_soc_dapm_new_widgets() takes the codec lock when adding new widgets,
causing lockdep warnings when applications later call down through ALSA
to adjust controls. Since widgets are only added during probe this lock
should be unneeded so don't take it.
Thanks to Dmitry Baryshkov <dbaryshkov@gmail.com> for reporting this issue.
Cc: Dmitry Baryshkov <dbaryshkov@gmail.com>
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>
This fixes a bug whereby PCMs were not being suspended when the rest of the
audio subsystem was suspended.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a device level dapm event so that both the machine and codec are informed
when dapm events occur.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
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: Joe Sauer <jsauer@vernier.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
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>
Moved 2 systems PCI_QUIRK values to STAC_DELL_BIOS. Also the second
front HP jack is incorrect defined in the BIOS VT's for some laptops,
this patch corrects this.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Please refer to the [0003680] on ALSA bugtracking system.
The user found that 'model=mbp3' works and provided the ID.
From: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
1. Support HP rp5700
2. Fixed alc_subsystem_id function (Bug fixed and support Desktop)
3. Support ASUS EP20
Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
snd_rawmidi_kernel_read1/write1 weren't annotated but used
copy_to_user/copy_from_user when one of parameters (kernel) was equal to 0
remove it and add properly annotated parameter
Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a mono output mute mixer for the 92hd71bxx family of codecs, this
also removes the need for the mono out node to explicitly unmuted in the
core init.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This header file exists only for some hacks to adapt alsa-driver
tree. It's useless for building in the kernel. Let's move a few
lines in it to sound/core.h and remove it.
With this patch, sound/driver.h isn't removed but has just a single
compile warning to include it. This should be really killed in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The 'tick' in PCM is set (again) via sw_params. And, nobody uses
this feature at all except for a command line option of aplay.
(This is literally 'nobody', as I checked alsa-lib API calls in all
programs in major distros.)
Above all, if we need finer wake-ups for the position update, it's
basically an issue that the driver should solve, not tuned by each
application.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Introduce a common helper function for snd_pcm_lib_read and snd_pcm_lib_write
for cleaning up the code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The xfer_align sw_params parameter has never been used in a sane manner,
and no one understands what this does exactly. The current
implementation looks also buggy because it allows write of shorter size
than xfer_align. So, if you do partial writes, the write isn't actually
aligned at all.
Removing this parameter will make some pcm_lib_* code more readable
(and less buggy).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The snd_pcm_lib_write1() may block in some weird condition:
- the stream isn't started
- avail_min is big (e.g. period size)
- partial write up to buffer_size - avail_min
The patch fixes this invalid blocking problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch removes the indirect control access to the control elements.
The indirect access has never been used and is even broken on 32bit
ioctl wrapper. Let's clean it up.
The pointers still remain in snd_ctl_elem_* structs just to make sure
that the struct size won't change. Once after checking the size
consistency, we can get rid of them, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added the proper model=laptop for HP 350 laptop with Cxt5045 codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This implements a test model for the ALC268. It depends on the feature
added by alc260-test-eapd-0.2.diff. This patch also adds a mention of
the ALC260 test model to ALSA-Configuration.txt since this seems to have
been missed.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This implements a switch control for the EAPD signal output by the ALC26x
chips. Since some laptops may utilise this to activate useful things it
is handy to have a control for this in the ALC26x test models. The patch
includes the control in the ALC260 test model.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The condition caused that the returned ring buffer position does not match
with timestamp when SNDRV_PCM_TSTAMP_MMAP mode was enabled. Removing
condition makes unified behaviour and interrupt based timestamp can be
accessed via PCM_IOCTL_SYNC_PTR or mmaped status area.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Dynamically create digital gain mixers for dmics that have out-amp
support. Also some 92HD73xx's codecs don't have DMIC gains, so this also
prevents creating dead mixers.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
It turned out that there can be multiple HD-audio devices on a single
machine (e.g. on-board audio and HDMI on graphic cards), so we need to
support multiple devices with snd-hda-intel driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The functions time_before, time_before_eq, time_after, and time_after_eq
are more robust for comparing jiffies against other values.
A simplified version of the semantic patch making this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@ change_compare_np @
expression E;
@@
(
- jiffies <= E
+ time_before_eq(jiffies,E)
|
- jiffies >= E
+ time_after_eq(jiffies,E)
|
- jiffies < E
+ time_before(jiffies,E)
|
- jiffies > E
+ time_after(jiffies,E)
)
@ include depends on change_compare_np @
@@
#include <linux/jiffies.h>
@ no_include depends on !include && change_compare_np @
@@
#include <linux/...>
+ #include <linux/jiffies.h>
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch enables wavetable chips ES689/ES69X connected to
ESS ES18xx chips. The wavetable chip uses FM DAC if the clock signal
from the wavetable is active.
It has no effect if there is no ESS wavetable chip present.
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>
This patch adds IEC958 digital out support for the AD1984 sound card.
This card can be found in Lenovo Thinkapds T61/X61. The digital out is
not located on the Thinkpad, but optional docking station (it's coxial
digital out). I've add this support as it is done the exact same way
for the AD1983 & AD1884.
I have tested this patch with my Lenovo Thinkpad T61 hooked up to a
docking station (that has the digital coxial) and then run to my home
theater reciever. Works like a charm :-)
Signed-off-by: Jerone Young <jerone@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Some laptops incorrectly assume the front input jack as a line in
instead of a microphone in. Which in turn disables the voltage
reference, in which non-amplified input is not possible. This patch
enables VREF80 for the input jack.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When we store the DMA channel number in the substream's private_data
pointer, use uintptr_t as an intermediate step when casting from/to
unsigned int to prevent the compiler from whining when the pointer size
is different.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
sound/usb/usbaudio.c: In function 'usb_audio_suspend':
sound/usb/usbaudio.c:3674: error: implicit declaration of function 'snd_pcm_sus\pend_all'
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Sort the ad1986a config table by PCI SSID (the last toshiba entry was
added wrongly).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add the snd-virtuoso driver for the Asus Virtuoso 200 chip used on the
PCI and PCI-E models of the Xonar sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add the snd-oxygen driver for the C-Media CMI8788 (Oxygen) chip, used on
the Asound A-8788, AuzenTech X-Meridian, Bgears b-Enspirer,
Club3D Theatron DTS, HT-Omega Claro, Razer Barracuda AC-1,
Sondigo Inferno, and TempoTec HIFIER sound cards.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
According to the [0003659], the node 0x23,0x24 is a selector.
I checked the alc268 spec on the REALTEK website and it showed that they
were selectors indeed.
However, current code implement the alc268 input_mux in a mixer way.
Signed-off-by: Jiang Zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The model laptop-eapd get rid of the high-pitched noise.
(ALSA bug#3662)
Signed-off-by: Jiang Zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The commit that added support for ASUS P701 eeepc also changed the
mixers of other ALC662 models, duplicating entries for the Capture
items, making them to not work anymore. This fixes it by removing
duplicated entries using where possible the common alc662_capture_mixer.
Also alc662_capture_mixer should use alc662* functions and not alc882
(I checked /proc/asound/card0/codec* on an eepc model and it's ok).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add a quirk entry to handle Edirol UR-80 audio I/O.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to
AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added
AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK,
and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet
rev 1.3.
Signed-off-by: Andrew Paprocki <andrew@ishiboo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The digital mux on STAC codecs doesn't always exist although the
driver builds dmux enum mixer elements unconditionally.
Now the driver creates 'digital input source' mixer elements only
when dmux is available.
Also, the patch adds the missing dmux definition for STAC925x.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When CONFIG_SND_SOC_CS4270_VD33_ERRATA is set, there was a mismatch between
the mclk_ratios[] and cs4270_mode_ratios[] arrays. The two arrays have been
merged and code has been shuffled. One side effect is that the
cs4270_set_dai_sysclk() and cs4270_set_dai_fmt() functions are available only
if I2C has been enabled.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch fixes the locking bug in the at73c213 SPI sound driver. This bug was
triggered because spinlocks were wrapped around the spi_sync call which might
sleep. The fix was to add a mutex to the sound driver and replace the spinlocks
in the mixer functions with mutex lock/unlock.
Tested on STK1000/STK1002.
Signed-off-by: Hans-Christian Egtvedt <hcegtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Sort pci quirk list in the order of PCI SSID.
This makes easier to find out the buggy duplicated entries.
Thanks to Andy Shevchenko for providing the sort script.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added the missing link to struct device from the card instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The PCM volume of Cx5045 codec has overload that isn't useful but
rather harmful. Add a hack to override the amp info to set the max
level 0 dB.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cleaned up STAC927x and added several subsystem id's for more laptops.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
AC_KNBCAP_DELTA is incorrectly defined as (1<<8). According to the Intel
HDA spec, this is bit 7 after AC_KNBCAP_NUM_STEPS which is a 0x7f mask.
Signed-off-by: Andrew Paprocki <andrew@ishiboo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch implements suspend/resume support for USB audio devices.
It works with the microphone in my camera.
Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Basic audio support for the SiS 7019 Audio Accelerator as found in the
SiS 55x SoC. There is currently no synth support at the moment, but
audio playback and capture with two periods per buffer has seen
extensive use. Arbitrary period and buffer sizes (with multiple periods
per buffer) have seen light testing, but are believed to be production
ready.
Signed-off-by: David Dillow <dave@thedillows.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
info_oss: move prototype of snd_card_info_read_oss to info.h
Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
pcm_lib: fix sparse warning about different signedness
Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
pcm_lib: fix sparse warning about shadowing 'n' symbol
Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
via82xx: minor optimization in snd_via82xx_free
don't check X, when we just checked !X before goto
Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
pcm_native: fix sparse warning about shadowing 'state' symbol
Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
sound/isa/sb/sb16_csp.c: In function ‘snd_sb_csp_new’:
sound/isa/sb/sb16_csp.c:121: warning: ‘version’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
sound/core/control_compat.c: In function 'copy_ctl_value_from_user':
sound/core/control_compat.c:222: warning: 'count' may be used uninitialized in this function
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Replaced the old SNDRV_PCM_IOCTL_TSTAMP with the new one in
PCM compat_ioctl.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added more codecs to the 92HD71 family, as well as support for several
that don't have an analog mixer.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Changed hardware gain mixers for the digital mic's from HDA_OUTPUT to
HDA_INPUT.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Enable Analog CD Input from internal ATAPI connector on Asus M2N-SLI.
Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added support for new STAC92HD73 family of codecs. Additionally added
features for multiple analog loopbacks, and multiple dmux mixers.
Regression testing for the analog loopback changes for STAC9205 and
STAC9274D completed with any issues, as well for the dmux changes.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The attached patch adds initial support of the Mitac 8252D
(http://www.mitac-mtc.com.tw/English/products/8252Dspec.htm).
Working:
- Front speakers (volume + mute)
- Center/LFE speakers (volume + mute)
- HP out (with Front Volume)
- HP individual mute switch
- HP Jack sense
- Front Mic and its volume
Not tested:
- external mic and its volume
Not working while now:
- Mic Jack sense
Questionable:
- is Mic have Jack sense?
- one or two Mic volume controls?
- CD/Line-in: presense in the mixer
Signed-off-by: Andy Shevchenko <andy@smile.org.ua>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added the support of new Realtek codecs:
1. New ALC889 Support
2. New ALC267 Support
3. New ALC269 Support
Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds support for VIA new HD audio codec, VT1708B.
Signed-off-by: Josepch Chan <josephchan@via.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Convert alsa usx2y driver from nopage to fault.
Signed-off-by: Nick Piggin <npiggin@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Convert ALSA from nopage to fault.
Switch from OOM to SIGBUS if the resource is not available.
Signed-off-by: Nick Piggin <npiggin@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added the model targa-2ch-dig for MSL L745 (ALSA bug#3641).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
We need an accurate and continuous (monotonic) time sources to do
accurate synchronization among more timing sources. This patch allows
to enable monotonic timestamps for ALSA PCM devices and enables monotonic
timestamps for ALSA timer devices.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add a comment that explains why the 'Modem' control doesn't work with
newer chips.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On version 39 or newer chips, we better remove the 'Modem' control
because this register bit now mutes the front channels of the
multichannel stream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The attached patch should fix typo in auto initialization verbs for ALC883
codec.
Signed-off-by: Andy Shevchenko <andy@smile.org.ua>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* support for switching rate in STAC9460 - using set_rate_val of the akm
infrastructure
* listing all STAC9460 registers in proc
* disabling mpu401 device for Prodigy192 - otherwise the currently
flawed mpu401 code hangs kernel when opening the midi device
* removing old unused commented-out code
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
