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The loader API has been revised so that OS specific data is kept
local to hpidspcd.c, and the public API is unchanged across OSes.
Signed-off-by: Eliot Blennerhassett <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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Some cobranet control data would not fit in an original HPI message.
Now that HPI is able to transfer larger messages, this special handling
is no longer required.
Signed-off-by: Eliot Blennerhassett <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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Allow for up to 256 bytes of extra data on top of standard hpi
request and response sizes.
Signed-off-by: Eliot Blennerhassett <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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Having a 'request message' makes more sense than a 'message message'
Signed-off-by: Eliot Blennerhassett <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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Add support for Roland/BOSS BR-800 (0582:011e) to snd-usb-audio driver.
This allows playback and recording, which has been tested and found to
work. The third interface should be MIDI (MTC/SMPTE?) for DAW interface
and is set as per ME-25, but this has not been tested. SDHC card access
is already supported by usb-storage for Backup/Rhythm Editor/Wave
Convertor mode which should not conflict with this.
Signed-off-by: David G Turner <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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"force" argument is always true, so let's strip it off.
Signed-off-by: Takashi Iwai <[email protected]>
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This patch fixes non-working indep-HP control on VT1708* codecs.
The problems are that via_independent_hp_put() wasn't fixed to follow
the recent change of three HP paths, and hp_indep_path didn't contain
the amp nids of mixer elements.
Together with the fixes, a few code clean-ups are done.
Signed-off-by: Takashi Iwai <[email protected]>
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In preparation for ASoC Dynamic PCM (AKA DSP) support.
Provide convenience methods to retrieve the soc_card or snd_card from a
DAPM context.
Signed-off-by: Liam Girdwood <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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I2S in Exynos4 and S5PC110(S5PV210) has a internal dma.
It can be used low power audio mode and 2nd channel transfer.
This patch can support idma.
Signed-off-by: Sangbeom Kim <[email protected]>
Acked-by: Jassi Brar <[email protected]>
Acked-by: Liam Girdwood <[email protected]>
Acked-by: Jassi Brar <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Previously, I2S driver only can support system dma.
In this patch, i2s driver can support internal dma too.
IDMA h/w configuration is initialized on idma.c
Signed-off-by: Sangbeom Kim <[email protected]>
Acked-by: Liam Girdwood <[email protected]>
Acked-by: Jassi Brar <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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In davinci_vcif_trigger() function, a break() statement was missing
causing the davinci_vcif_stop() function to be called as a fallback
after calling davinci_vcif_start().
Signed-off-by: Rajashekhara, Sudhakar <[email protected]>
Acked-by: Liam Girdwood <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
Cc: [email protected]
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According to DM365 voice codec data sheet at [1], before starting
recording or playback, ADC/DAC modules should follow a reset and
enable cycle. Writing a 1 to the ADC/DAC bit in the register resets
the module and clearing the bit to 0 will enable the module. But the
driver seems to be doing the reverse of it.
[1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf
Signed-off-by: Rajashekhara, Sudhakar <[email protected]>
Acked-by: Liam Girdwood <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
Cc: [email protected]
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Add a convenience macro for external enumerated widgets.
Signed-off-by: Liam Girdwood <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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This closes the small race between a status being read in response to an
interrupt and clearing the interrupt, meaning that if the status changes
between those periods we might not get a reassertion of the interrupt.
Signed-off-by: Mark Brown <[email protected]>
Acked-by: Liam Girdwood <[email protected]>
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Print a hint when the user has a setup where CONFIG_REGULATOR is really
needed to make the driver work.
Signed-off-by: Wolfram Sang <[email protected]>
Tested-by: Dong Aisheng <[email protected]>
Tested-by: Shawn Guo <[email protected]>
Acked-by: Liam Girdwood <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The code for registering the internal ldo was present twice. Turn it
into a function instead. Also, inform the user if LDO is used now.
Signed-off-by: Wolfram Sang <[email protected]>
Tested-by: Dong Aisheng <[email protected]>
Tested-by: Shawn Guo <[email protected]>
Acked-by: Liam Girdwood <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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In one comment, cpu_dai was mentioned although codec_dai was used in the
code. Also, fix the name for the card dai list which has no seperation
into card_dai and codec_dai.
Signed-off-by: Wolfram Sang <[email protected]>
Acked-by: Liam Girdwood <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Since quite a few drivers are not managing to flag the cache as needing
to be resynced after suspend and it's a reasonable thing to do flag the
cache as needing sync automatically when suspending.
The expectation is that systems will mainly only keep the CODEC powered
when doing audio through the CODEC so we won't actually suspend the
device anyway; drivers which want to can override this behaviour when
they resume.
Signed-off-by: Mark Brown <[email protected]>
Acked-by: Liam Girdwood <[email protected]>
Cc: [email protected]
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Signed-off-by: Takashi Iwai <[email protected]>
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This patch changes the behavior of independent-HP enum switch. Now
instead of returning a busy error, the driver switches dynamically the
stream of the HP (and shared) DACs according to the current mode.
The logic is similar like the dual-mic ADC switch, but a bit more
complicated because of the presence of shared DAC.
Together with the change, a mutex is introduced to protect against the
possible races for the indep-HP mode setting.
Signed-off-by: Takashi Iwai <[email protected]>
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This patch adds the dynamic control of analog-loopback for VIA codecs.
When the loopback is enabled, the inputs from line-ins and mics are
mixed with the front DAC, and sent to the front outputs. The very same
input is routed to the headhpones and speakers in loopback mode.
However, since the loopback mix can't take other than the front DAC,
there is no longer individual volume controls for headphones and
speakers. Once when the loopback control is off, these volumes take
effect.
Since the individual volumes are more desired in general use caess, the
loopback mode is set to off as default for now.
Signed-off-by: Takashi Iwai <[email protected]>
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Commit dd203fa97bd5 (ALSA: virtuoso: remove non-working controls on
Essence ST Deluxe) made it impossible to adjust the volume after the
driver initialized it to muted.
Ensure that those DACs that can be accessed with I2C are initialized
to the same volume that is the reset default of the DAC without I2C.
Signed-off-by: Clemens Ladisch <[email protected]>
Cc: 2.6.38+ <[email protected]>
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The WM8994 and WM8958 series of devices have two MICBIAS supplies rather
than one, the current widget actually manages the microphone detection
control register bit (which is managed separately by the relevant API).
Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2
widgets.
Signed-off-by: Mark Brown <[email protected]>
Cc: [email protected]
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Makes life a little easier if you want to add subsequences to an existing
driver as you can use -1 to put things at the start of sequences.
Signed-off-by: Mark Brown <[email protected]>
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If DAIs are idle but their clocks are in use for some reason (eg, as
SYSCLK or for accessory detect) then set the clock dividers to the maximum
to reduce slightly the power consumption of the unclocked circuits.
Signed-off-by: Mark Brown <[email protected]>
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Not only fixes error handling but also some uninitialized variable
warnings.
Signed-off-by: Mark Brown <[email protected]>
Acked-by: Lars-Peter Clausen <[email protected]>
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Try the completion before we start the FLL so that if an interrupt was
delayed long enough for us to miss it we don't wait for the completion
it signalled.
Signed-off-by: Mark Brown <[email protected]>
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Don't assume the first fire indicates that we're done.
Signed-off-by: Mark Brown <[email protected]>
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The WM8983 is a low power, high quality stereo CODEC
designed for portable multimedia applications. Highly flexible
analogue mixing functions enable new application features,
combining hi-fi quality audio with voice communication.
Signed-off-by: Dimitris Papastamos <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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Signed-off-by: Mark Brown <[email protected]>
Cc: [email protected]
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BugLink: https://bugs.launchpad.net/bugs/774895
The original reporter states that his volume keys do not change the
desired Master and PCM mixer elements together, so apply the hp+mute led
quirk for his PCI SSID.
Reported-by: Jeffrey Finkelstein
Signed-off-by: Daniel T Chen <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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It was obviously wrong, grr....
Signed-off-by: Takashi Iwai <[email protected]>
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During the rewrite, the check of spec->need_dac_fix and the corresponding
num_dacs change was dropped from the channel-mode control.
This patch re-adds it, and also enables need_dac_fix for ALC880 as default,
as this feature was originally introduced to fix h/w bugs of this chip.
Signed-off-by: Takashi Iwai <[email protected]>
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Signed-off-by: Axel Lin <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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We should spot them anyway on state changes but logging them gives us
better time information about when the misconfiguration happened.
Signed-off-by: Mark Brown <[email protected]>
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Using 256fs or 512fs will result in distortion of 24-bit
audio samples. This is because the lrclk generated is not
proper. Using 384 fs generates proper output.
Signed-off-by: Giridhar Maruthy <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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We can have valid but very low clocks in accessory detection modes.
Signed-off-by: Mark Brown <[email protected]>
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If we have interrupts then wait for the FLL lock interrupt rather than
using dead reckoning when waiting for the FLL to start.
Signed-off-by: Mark Brown <[email protected]>
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Signed-off-by: Mark Brown <[email protected]>
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The individual devices should set the flag dcs_done_irq in the hubs
shared data structure to indicate that they will flag the interrupt
by calling wm_hubs_dcs_done().
Signed-off-by: Mark Brown <[email protected]>
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This ensures appropriate clocking for bypass paths to speaker and
headphone and direct voice paths on affected revisions.
Signed-off-by: Mark Brown <[email protected]>
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Chip documentation explicitly requires that the reset values
of reserved register bits are left untouched. It is possible
there are differences between STA326 and STA328 or future
chip revisions in these bits, and clobbering them might
cause malfunction.
Signed-off-by: Johannes Stezenbach <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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The STA32x has a number of preset EQ settings, but also
allows full user control of the biquad filter coeffcients
(when "Automode EQ" is set to "User").
Each biquad has five signed, 24bit, fixed-point coefficients
representing the range -1...1. The five biquad coefficients
can be uploaded in one atomic operation into on-chip
coefficient RAM.
There are also a few prescale, postscale and mixing
coefficients, in the same numeric format and range
(a negative coefficient inverts phase).
These coefficients are made available as SNDRV_CTL_ELEM_TYPE_BYTES
mixer controls.
Signed-off-by: Johannes Stezenbach <[email protected]>
Signed-off-by: Mark Brown <[email protected]>
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in acfa634f
This commit is a fix up for commit acfa634f.
commit acfa634f7e199193ec28282e82a5a6dd8edebcb7
Author: Takashi Iwai <[email protected]>
Date: Tue Jul 12 17:27:46 2011 +0200
ALSA: hda - Add Kconfig for the default buffer size
Signed-off-by: Paul Menzel <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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This patch gives M-Audio Fast Track Pro and M-Audio Quattro quirks and
endpoints to boot and setup those devices with special options (digital
inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are
just adapted to match the new global M-Audio parameters.
Special configurations can be then loaded through a modprobe conf file.
For example, to set the 24 bits mode on the Fast Track Pro add
/etc/modprobe.d/fast_track_pro.conf :
options snd_usb_audio vid=0x763 pid=0x2012 device_setup=0x08
Here is a list of the possibilities in this example :
http://files.parisson.com/debian/fast-track-pro.conf
Signed-off-by: Guillaume Pellerin <[email protected]>
Signed-off-by: Takashi Iwai <[email protected]>
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Add a Kconfig entry to specify the default buffer size.
Distros using PulseAudio can choose a larger value here.
Signed-off-by: Takashi Iwai <[email protected]>
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Signed-off-by: Takashi Iwai <[email protected]>
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VT1718S and co have a secret connection from DAC to AA-mix, which
doesn't appear in the connection list obtained from the h/w.
Currently the driver fixes the connection index locally at init, but
now we can expose it statically via snd_hda_override_connections()
so that this conection can be checked better by the parser in future.
Signed-off-by: Takashi Iwai <[email protected]>
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