It turned out that there are different pin configurations for this
PCI SSID, including multi-channel modes. And more proper fix for
allowing line-out mutes will come up in 2.6.40 tree, so we won't need
this fixup any more there.
The PCI SSID is 1025:031c and the codec SSID is 1025:031d,
so the driver mistakes this for a SKU value, but looking at
the numbers, this is obviously wrong.
ALSA: hda - Add Auto-Mute Mode enum for two-output cases
The Auto-Mute Mode control is useful even when only two outputs
(e.g. HP and speaker) are available. Then user can enable/disable
the auto-mute behavior on the fly.
ALSA: hda - More line-out auto-mute support for Realtek
Not only supporting the line-out automute as additional feature
to the existing headphone automute, now the headphone jack can
mute the line-out alone even without the speaker outs.
ALSA: hda - Add support for Line-Out automute to Realtek auto-parser
By popular demands, I add the functionality to mute / unmute the
line-out jacks per the headphone plug / unplug. For achieving this
and keeping the compatibility with the old behavior, the new mixer
enum "Auto-Mute Mode" is added. With this, user can control the
auto-mute behavior either disabled, speaker-only or lineout+speaker.
ALSA: hda - Consolidate auto-mute with master-switch for Realtek
Yet another consolidation of auto-mute functions for the devices
controlling the output muts together with the master mixer switch,
typically found for ALC262 machines.
ALSA: hda - Add common automute support for mxier-amp on/off for Reatek
Some models do mute on/off the connected mixer widget for the automatic
muting, instead of controlling the pin widget itself. This patch adds
the implementation of such type of auto-mute in the common helper
function, and reduces the redundant codes for each model preset.
ALSA: hda - Consolidate default automute functions for Realtek
There are two entry points for the headphone automute functions for
Realtek, alc_automute_amp() and alc_automute_pin(). These call the
same function in the end, so we can basically consolidate these
with a flag in spec.
In notify_aa_path_ctls(), adds 'rear mic' item and confirms the A-A
path control existing before notifying card that the A-A path volume
is muted if smart5.1 is enabled.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some crappy USB-audio devices give broken dB ranges, e.g. both min and max
are 0dB. This confuses the volume control that prefers dB expression such
as alsactl or PulseAudio. In such a case, it's much better not to expose
the broken dB information.
Raymond Yau [Mon, 25 Apr 2011 04:05:45 +0000 (12:05 +0800)]
ALSA - au88x0 - Add buffer bytes constraints
This allow application such as gstreamer and wine which use
snd_pcm_hw_params_set_buffer_time_near() won't fail any more
since sound chips require special containt power 2 period bytes
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: hda - Add channel-mode support to Realtek auto-parser
This patch adds the support of "Channel Mode" enum control to Realtek
auto-parser. When line-in or mic-in jacks are capable to output and
free DACs are available, the driver allows to switch to multi-channel
mode via "Channel Mode" enum switch, as already implemented in some
preset cases.
Not implemented in all Realtek codecs. Currently, ALC880, 882, 861,
662 and the compatible codecs are supported.
ALSA: hda - Minor update for alc662-parser functions
Allow alc662_dac_to_mix() and alc662_look_for_dac() to parse
down the selector widget that is found in ALC880-type codecs,
and rename them to alc_auto_*() accordingly.
This is for the next coming multi-io extensions.
ALSA: hda - Enable sync_write workaround for AMD generically
The workaround for AMD chipset via sync_write flag seems needed for
machines with Realtek codecs. So, it's better to activate it
generically in hda_intel.c from the beginning.
ALSA: hda - Move EAPD power-down into shutup callback for AD codecs
EAPD power-down should be called also for normal shutup cases.
Let's move to there. This also fixes the compile warnings when
CONFIG_PM isn't set automatically.
ALSA: hda - Enable sync_write for AMD chipset with IDT 92HD8x codecs
The AMD chipset seems unstable in the normal operation mode, and it
seems requring more sensible access for each verb. Enabling sync_write
mode and allowing bus-reset is a sort of workaround for these chipset
stability issues.
"Front Playback Volume" and "Surround Playback Volume" have same dB range
since I2S DAC of SB Live! and SB Live! Platinum does not has any hardware
volume control.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: hda - Add a fix-up for Acer dmic with ALC271x codec
Acer laptops with ALC271x needs a magic initialization for digital-mic
to make it working with mono streams (and PulseAudio).
Added a fix-up applied to Acer with ALC271x generically.
Stephen Warren [Tue, 12 Apr 2011 17:40:39 +0000 (11:40 -0600)]
ASoC: Tegra: Support more boards
* Ventana is identical to Harmony.
* Seaboard, Kaen, and Aebl are all pretty similar, mainly with slightly
different sets of GPIOs, and slightly different WM8903 pin connectivity.
Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stephen Warren [Tue, 12 Apr 2011 17:40:38 +0000 (11:40 -0600)]
ASoC: Tegra: Don't store snd_soc_jack_gpio in an array
Storing the struct in an array makes the assignments to the GPIO member a
little non-obvious, and is pointless when there's only a single GPIO.
(I thought I fixed this during the review cycle when first submitting this
driver, but I guess I overlooked that)
Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stephen Warren [Tue, 12 Apr 2011 17:40:37 +0000 (11:40 -0600)]
ASoC: Tegra: Rename Kconfig SND_TEGRA_SOC_* to SND_SOC_TEGRA_*
The previous commit renames SND_TEGRA_SOC_HARMONY to SND_TEGRA_SOC_WM8903.
While we're breaking people's .config files, rename all Tegra/SOC-related
Kconfig variables to be more consistent with at least the core codec
variables. Note that there exist machines that name their variables both
ways.
Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stephen Warren [Tue, 12 Apr 2011 17:40:36 +0000 (11:40 -0600)]
ASoC: Tegra: Rename harmony.c to tegra_wm8903.c
Soon, this machine driver will be updated to handle a number of Tegra boards
using the WM8903 codec. Rename the file in advance to reflect this.
Fix the content of tegra_wm8903.c to match the rename; replace references
to Harmony board with something more generic.
* s/struct tegra_harmony/struct tegra_wm8903/
* s/harmony/machine/ # variable name
* Similar rename for some functions
* Similar comment fix
* Similar MODULE_DESCRIPTION fix
Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stephen Warren [Tue, 12 Apr 2011 17:29:01 +0000 (11:29 -0600)]
ARM: Tegra: Add to tegra_wm8903_platform_data
Seaboard derivate Kaen has a GPIO to mute the headphone output. Add a field
to tegra_wm8903_platform_data so the board files can pass the GPIO number
for that to the ASoC machine driver.
Also, initialize this new field to a "not present" value for Harmony.
Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stephen Warren [Tue, 12 Apr 2011 17:29:00 +0000 (11:29 -0600)]
ASoC: Tegra: Rename pdev tegra-snd-harmony to tegra-snd-wm8903
Soon, this machine driver will be updated to handle a number of Tegra boards
using the WM8903 codec. Rename the platform device in advance to reflect this.
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>
ASoC: JZ4740: qi_lb60: Use gpio_request_array to request and setup gpios
This patch changes the qi_lb60 setup code to use gpio_request_array instead of
manually calling gpio_request and gpio_direction_output for each gpio.
Doing so makes the code a bit more compact.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: Make struct snd_soc_card's dapm_widgets and dapm_routes const
Those should not be modified (and are not) by the core code, so make them const.
This also makes them consistent with the same members of snd_soc_codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit ce6120cc(ASoC: Decouple DAPM from CODECs) changed the signature of
snd_soc_dapm_widgets_new to take an pointer to a snd_soc_dapm_context instead of
a snd_soc_codec. The call to snd_soc_dapm_widgets_new in jz4740_codec_dev_probe
was not updated to reflect this change, which results in a compiletime warning
and a runtime OOPS.
Since the core code calls snd_soc_dapm_widgets_new after the codec has been
registered it can be dropped here.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Mark Brown [Wed, 13 Apr 2011 00:37:52 +0000 (17:37 -0700)]
ASoC: Mark Speyside widgets as ignoring suspend
Allow audio paths through the Speyside system to be kept active while the
system is suspended (for example, when on a voice call) by marking all the
external widgets and the DAI link to the WM1250-EV1 baseband module as
ignoring suspend.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 12 Apr 2011 21:15:10 +0000 (14:15 -0700)]
ASoC: Add stub baseband link on Speyside
Demonstrate the connection of a baseband to the system. We add a DAI for
the link to the baseband. This will become visible to the application
layer - audio should be started from the application layer using an
application such as this:
which starts up audio as for CPU based playback and record up to the point
where data is streamed.
Due to non-availability of baseband simulation hardware we reuse the
configuration for the CPU link with the CODEC acting as clock master,
allowing signals to be observed with a scope. A more standard system
would have separate configuration for the baseband with its own ops
structure and operations. Normally the baseband would be clock master
as the baseband audio will be synchronised to the external telephony
network.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 12 Apr 2011 07:09:53 +0000 (00:09 -0700)]
ASoC: Add pin switches for fixed analogue inputs and outputs on Speyside
Pin switches enable direct control of the DAPM state from userspace,
enabling simple enabling and disabling of the path. This is especially
useful for outputs such as the speaker which are composed of several
physical devices as it allows them to be controlled as a group.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 12 Apr 2011 07:00:36 +0000 (00:00 -0700)]
ASoC: Add Speyside headset jack detection support
Speyside makes use of support the WM8915 has for detecting the polarity
of the microphone and ground connections on headsets, using a GPIO to
control the polarity of the ground connection and switching between the
two microphone bias supplies available on the device in order to do so.
As a result of this the detection support is more involved than for most
other CODECs, using a callback to configure the current polarity of the
jack and translate this into the board-specific connections required for
the current scenario.
On Android some additional work is required to hook this up to the
application layer as the Android HeadsetObserver monitors a custom
drivers/switch API rather than the standard Linux APIs. This can be
done by either updating HeadsetObserver or modifying the ALSA core to
report via drivers/switch as well.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 12 Apr 2011 06:42:25 +0000 (23:42 -0700)]
ASoC: Support the sub speaker driver on Speyside
Speyside includes a WM9081 configured as an external speaker driver taking
an analogue input from HPOUT2 on the WM8915 on the system. Add support for
this to the driver, using a prefix of "Sub" for the WM9081 controls to
ensure we avoid collisions with controls on the WM8915.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 12 Apr 2011 06:32:03 +0000 (23:32 -0700)]
ASoC: Optimise clock management for WM8915 Speyside
Dynamically enable and disable the FLL on the WM8915, configuring the
system clock to 256fs for 48kHz when the device is active but reverting
to using the input 32.768kHz clock directly at other times to support
features such as jack detection with minimal power consumption.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 12 Apr 2011 06:09:15 +0000 (23:09 -0700)]
ASoC: Add basic widgets for WM8915 Speyside
Provide widgets for the basic widgets connected directly to the WM8915
on Speyside - the headphones, speaker, digital and analogue microphones.
For the outputs this is just documentation, for the inputs this ensures
that the relevant microphone biases are enabled when they are in use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Wed, 13 Apr 2011 00:24:39 +0000 (17:24 -0700)]
ASoC: Initial audio support for Speyside on Cragganmore 6410
This is minimal code required to get audio out of the Speyside audio
subsystem on the Wolfson Cragganmore 6410 reference platform. It sets
up the link between the CPU and AIF1 of the WM8915 on the system,
enabling audio playback via the headphone and speaker outputs of the
device (which require no further configuration except runtime). It
allows verification of basic functionality of the system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jassi Brar <jassisinghbrar@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com>
projects / karo-tx-linux.git / log