This provides an ASoC DAI interface for Tegra's I2S controller.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This provides an ASoC platform driver that manages Tegra's APB DMA
controller.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DAS (Digital Audio Switch) is a mux/crossbar which sits between
the DACs (Digital Audio Controllers) and the DAPs (Digital Audio
Ports). Audio data may be routed between DACs and DAPs in various
combinations as required by board design/application.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A couple Tegra ASoC drivers will create debugfs entries. Mark requested
these by under debugfs/asoc/ not just debugfs/. To enable this, export
the dentry representing debugfs/asoc/.
Also, rename debugfs_root -> asoc_debugfs_root now it's exported to
prevent potential symbol name clashes.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The infrastructure for handling NULL default register maps is already
included in the soc-cache code, just ensure that we don't try to dereference
a NULL pointer while accessing a NULL register cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of the generic Oxygen, use the actual card name, if known.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apparently, the revision is 2 on all sold sound cards, so this
information is not actually useful.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to select between the on-board and extension board
S/PDIF inputs for the X-Meridian (2G).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to prevent capturing S/PDIF samples that are not
marked as valid (non-audio or corrupted samples).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the 24-bit audio I/Os of the Edirol SD-90 interface.
Reported-any-tested-by: Jim Grusendorf <alsa-user@grusendorf.ca>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify info callbacks by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the info callback by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify info callbacks by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the info callback by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the helper function snd_ctl_enum_info() to fill out the
elem_info fields for an enumerated control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add experimental support for the Asus Xonar HDAV1.3 Slim sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add experimental support for the Asus Xonar DG sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the AuzenTech X-Meridian 7.1 2G sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the CSxxxx and AKxxxx DAC/ADC chips, the MCLK factor in double rate
modes (64-96 kHz) can be reduced to 128x without reducing sound quality.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the get_i2s_mclk callback with tables of MCLK values. This
simplifies the MCLK-handling code in both the framework and the model-
specific drivers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not apply the headphone gain offset to any but the front DAC. These
DACs would not be used in headphone mode, so this saves a few register
writes.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the DAC Oversampling mixer control because this setting does not
make much sense.
For cards with the H6 daughterboard, 128x oversampling was disabled
anyway because these high MCLK frequency would not be compatible with
the connector cable.
For cards without the H6 daughterboard, 128x gives a slightly higher
output quality; there is no reason to reduce it to 64x except for saving
power, but then these cards have not been designed to be power efficient
anyway (the D2's blinkenlights cannot be disabled).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because of the unshielded connector cable, it is important to use as low
a master clock frequency as possible with the H6.
For double rate modes (64-96 kHz), the MCLK rate is unconditionally
lowered from 512x to 256x because the higher rate would not improve
anything.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The clock output of the CS2000, which is used as master clock for the
DACs, was using half the actual master clock frequency for some reason.
Using the theoretically correct frequency seems also to work in practice.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Xonar Essence ST Deluxe, remove all mixer controls that would
require I2C communication with the third DAC, which does not work
because of an addressing conflict with the CS2000 chip.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the PCM format used for the PCM1796 from left-justified to I2S to
ensure that the correct format is used even for the Essence ST Deluxe's
center/LFE DAC, where I2C does not work because of an address conflict
with the CS2000 chip.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM1796 needs the master clock for I2C communication to work, so
add delays after clock changes to ensure that the clock is stable when
we try to write the DACs' registers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To make the I2C communication reliable when using the H6 daughterboard,
reduce the I2C clock frequency.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix wrong register bits for SPI clock cycle times longer than 160 ns,
and adjust the polling loop timeout for these speeds.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of DACs can now be deduced from the dac_channels_mixer field,
so the private_data field is no longer needed.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For cards like the Xonar HDAV1.3, differentiate between the number of
PCM channels that can be played and the number of channels whose volume
can be adjusted.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, Realtek auto-parser assumed that the multiple pins are only for
line-outs, and assigned the channel names like Front, Surround, etc for
the multiple outputs. But, there are devices that have multiple
headphones, and these can be better controlled with the corresponding
control-name like "Headphone" with indicies.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple headphone pins are defined without line-out pins, the
driver takes them as primary outputs. But it forgot to set line_out_type
to HP by assuming there is some rest of HP pins. This results in some
mis-handling of these pins for Realtek codec parser. It takes as if
these are pure line-out jacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Enable Mic Jack during glue driver init, otherwise capture will not work.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since direct register writes may confuse the drivers and are supposed
to be used only in diagnostic situations discourage their use in
production by tainting the kernel when we do a write.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch moves PCM specific definitions, they doesn't need to be
shared, into 'pcm.c' from 'pcm.h'.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch moves DMA specific definitions, they doesn't need to be
shared, into 'dma.c' from 'dma.h'. And remove unnecessery definitions
on 'dma.h'.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch moves AC97 specific definitions, they doesn't need to be shared,
into 'ac97.c' from 'ac97.h'. This patch also remove header file, 'ac97.h',
and remove includes on machine drivers.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
previous commit e62255f2ad introduced DDEBUG flag in Makefile
This causes all debug statemenets to be ON.
This patch removes this flag
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
this patch fixes the style inconsistency by removing empty space in MODULE_ALIAS
also removes a redundant return statement
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
this patch fixes inconsistencies commented by Mark.
This also fixes few other style things in audio_map & header file
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_pcm_hw_param_near() will leak the memory allocated to 'save' if the
call to snd_pcm_hw_param_max() returns less than zero.
This patch makes sure we never leak.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5184
A user reported on the alsa-devel mailing list that he needs to use
the vostro model quirk to have audible playback, so apply it for his
PCI SSID.
Reported-and-tested-by: Fernando Lemos <fernandotcl@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/689036
Many new Lenovos need the ideapad quirk. Also, since the
auto parser for this chip is far from optimal, the regression
risk is low (although not zero).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If more than one mic is present with different locations,
e g "Front Mic" and "Rear Mic", they can use the same index (0),
since their names are different.
Previous behavior was to have "Front Mic" as index 1, causing it
to be ignored by e g PulseAudio.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/697240
If the "Volume" suffix is not given, alsa-lib gets confused and
loses the dB information at the simple element level.
Boosts generally affects both playback and capture, as they are
applied early in the chain. Hence no "Playback" or "Capture" in
the suffix.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/696493
According to datasheet (and real-world testing), IDT 92HD88B can
have internal mics at NID 0x11 and 0x20, so enable them accordingly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the makefile and kconfig changes for mid asoc drivers:
platform and machine driver which are introduced in 2 preceeding patches
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the medfield machine driver
Machine driver glues the sn95031 codec driver and sst_platform driver to form the asoc sound driver
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stephen Warren