Tim Howe [Fri, 22 Jul 2011 21:41:00 +0000 (16:41 -0500)]
ALSA: hda - Cirrus Logic CS421x support
This update includes the changes necessary for supporting the
CS421x family of codecs. Previously this file only supported
the CS420x family of codecs.
This file also contains init verbs to correct several issues in
the CS421x hardware.
Behavior between the CS421x and CS420x codec families is similar,
so several functions have been reused with "if" statements to
determine which codec family (CS421x or CS420x) is present.
Also, this file will be updated sometime in the near future in
order to add support for a system using CS421x that requires
mono mix on the speaker output only.
[Fix const usages and adaption for new APIs by tiwai]
Signed-off-by: Tim Howe <tim.howe@cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 26 Jul 2011 10:02:56 +0000 (12:02 +0200)]
ALSA: Make pcm.h self-contained
Move the macros depending on snd_mask_min() and co out of pcm.h into
pcm_params.h. Otherwise using some params_*() macros will give comiple
errors without inclusion of pcm_params.h.
Also use hw_param_interval_c() and hw_param_mask_c() for const pointer.
Reported-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 26 Jul 2011 08:33:10 +0000 (10:33 +0200)]
ALSA: hda - Allow codec-specific set_power_state ops
The procedure for codec D-state change may have exceptional cases
depending on the codec chip, such as a longer delay or suppressing D3.
This patch adds a new codec ops, set_power_state() to override the system
default function. For ease of porting, snd_hda_codec_set_power_to_all()
helper function is extracted from the default set_power_state() function.
As an example, the Conexant codec-specific delay is removed from the
default routine but moved to patch_conexant.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 26 Jul 2011 08:19:20 +0000 (10:19 +0200)]
ALSA: hda - Add post_suspend patch ops
Add a new ops, post_suspend(), which is called after suspend() ops is
performed. This is called only in the case of the real PM suspend, and
the codec driver can use this for further changing of D-state or
clearing the LED, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 26 Jul 2011 07:52:50 +0000 (09:52 +0200)]
ALSA: hda - Make CONFIG_SND_HDA_POWER_SAVE depending on CONFIG_PM
It makes little sense to enable power-saving without PM.
This removes SND_HDA_NEEDS_RESUME define so that we can use CONFIG_PM
in all places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vitaliy Kulikov [Mon, 25 Jul 2011 22:52:57 +0000 (17:52 -0500)]
ALSA: hda - Make sure mute led reflects master mute state
This patch adds checking of mute state on all outputs besides just
speakers to calculate the master mute state for mute led support.
It also renames and splits the function that does it for better code
clarity.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vitaliy Kulikov [Fri, 22 Jul 2011 23:18:15 +0000 (18:18 -0500)]
ALSA: hda - Fix invalid mute led state on resume of IDT codecs
Codec state is not restored immediately on resume but on the first
access when power-save is enabled. That leads to an invalid mute led
state after resume until either sound is played or some control is
changed. This patch adds a possibility for a vendor specific patch to
restore codec state immediately after resume if required. And it adds
code to restore IDT codecs state immediately on resume on HP systems
with mute led support.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vitaliy Kulikov [Fri, 22 Jul 2011 22:50:37 +0000 (17:50 -0500)]
ALSA: hda - Add support of the 4 internal speakers on certain HP laptops
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Sat, 23 Jul 2011 00:36:25 +0000 (12:36 +1200)]
ALSA: Make snd_pcm_debug_name usable outside pcm_lib
Formatting a PCM name is useful for module debug too.
Add snd_prefix when making function public.
[minor coding-style fixes by tiwai]
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Sat, 23 Jul 2011 16:57:11 +0000 (18:57 +0200)]
ALSA: hda - Fix DAC filling for multi-connection pins in Realtek parser
Fix a regression in the DAC filling code in patch_realtek.c. The already
filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0,
thus always pointed to the first DAC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 22 Jul 2011 06:43:27 +0000 (08:43 +0200)]
Merge branch 'topic/hda' into for-linus
Takashi Iwai [Fri, 22 Jul 2011 06:43:24 +0000 (08:43 +0200)]
Merge branch 'topic/misc' into for-linus
Takashi Iwai [Fri, 22 Jul 2011 06:43:19 +0000 (08:43 +0200)]
Merge branch 'topic/asoc' into for-linus
Takashi Iwai [Fri, 22 Jul 2011 05:57:44 +0000 (07:57 +0200)]
ALSA: asihpi - Replace with snd_ctl_boolean_mono_info()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:53:04 +0000 (15:53 +1200)]
ALSA: asihpi - HPI version 4.08
HPI Version is used to check for firmware compatibility.
This version will accept 4.08.xx released firmware,
and will also accept 4.09.xx beta firmware
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:53:03 +0000 (15:53 +1200)]
ALSA: asihpi - Add volume mute controls
Mute functionality was recently added to the DSP firmware
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:53:00 +0000 (15:53 +1200)]
ALSA: asihpi - Control name updates
Add names corresponding to new HPI node types.
Shorten some names so that constructed names don't overflow the
maximum name length.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:58 +0000 (15:52 +1200)]
ALSA: asihpi - Use size_t for sizeof result
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:56 +0000 (15:52 +1200)]
ALSA: asihpi - Explicitly include mutex.h
Because mutex is used in adapter struct defined here.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:55 +0000 (15:52 +1200)]
ALSA: asihpi - Add new node and message defines
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:52 +0000 (15:52 +1200)]
ALSA: asihpi - Make local function static
Fixes a sparse warning.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:50 +0000 (15:52 +1200)]
ALSA: asihpi - Fix minor typos and spelling
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:48 +0000 (15:52 +1200)]
ALSA: asihpi - Remove unused structures, macros and functions
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:46 +0000 (15:52 +1200)]
ALSA: asihpi - Remove spurious adapter index check
Subsystem requests don't have or need a valid adapter index.
The adapter index is already checked further on, before it is used to index
the adapters array. (Reverts
4a122c10f)
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:44 +0000 (15:52 +1200)]
ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro
Work towards moving the function into alsa common header.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:42 +0000 (15:52 +1200)]
ALSA: asihpi - DSP code loader API now independent of OS
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 <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:40 +0000 (15:52 +1200)]
ALSA: asihpi - Remove controlex structs and associated special data transfer code
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 <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:38 +0000 (15:52 +1200)]
ALSA: asihpi - Increase request and response buffer sizes
Allow for up to 256 bytes of extra data on top of standard hpi
request and response sizes.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 22 Jul 2011 03:52:36 +0000 (15:52 +1200)]
ALSA: asihpi - Give more meaningful name to hpi request message type
Having a 'request message' makes more sense than a 'message message'
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David G Turner [Thu, 21 Jul 2011 17:00:57 +0000 (19:00 +0200)]
ALSA: usb-audio - Add quirk for Roland / BOSS BR-800
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 <dgturner@iee.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 21 Jul 2011 12:23:35 +0000 (14:23 +0200)]
ALSA: hda - Remove a superfluous argument of via_auto_init_output()
"force" argument is always true, so let's strip it off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 21 Jul 2011 11:45:56 +0000 (13:45 +0200)]
ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
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 <tiwai@suse.de>
Takashi Iwai [Tue, 19 Jul 2011 07:34:10 +0000 (09:34 +0200)]
ALSA: hda - Add documentation for codec-specific mixer controls
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 18 Jul 2011 14:54:40 +0000 (16:54 +0200)]
ALSA: hda - Switch HP DAC dynamically with indep-HP switch for VIA
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 <tiwai@suse.de>
Takashi Iwai [Mon, 18 Jul 2011 10:49:25 +0000 (12:49 +0200)]
ALSA: hda - Implement dynamic loopback control for VIA codecs
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 <tiwai@suse.de>
Clemens Ladisch [Sun, 17 Jul 2011 20:18:05 +0000 (22:18 +0200)]
ALSA: virtuoso: fix silent analog output on Xonar Essence ST Deluxe
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 <clemens@ladisch.de>
Cc: 2.6.38+ <stable@kernel.org>
Mark Brown [Sun, 17 Jul 2011 09:25:58 +0000 (18:25 +0900)]
Merge branch 'for-3.0' into for-3.1
Mark Brown [Thu, 14 Jul 2011 09:21:37 +0000 (18:21 +0900)]
ASoC: Correct WM8994 MICBIAS supply widget hookup
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 <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Mark Brown [Sat, 16 Jul 2011 01:55:08 +0000 (10:55 +0900)]
ASoC: Don't use -1 to boostrap subseq so it can be used by drivers
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 <broonie@opensource.wolfsonmicro.com>
Mark Brown [Thu, 14 Jul 2011 08:11:38 +0000 (17:11 +0900)]
ASoC: Reduce power consumption for idle DAIs in WM8994
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 <broonie@opensource.wolfsonmicro.com>
Mark Brown [Sat, 16 Jul 2011 02:34:58 +0000 (11:34 +0900)]
ASoC: Report an error for unknown adav80x formats
Not only fixes error handling but also some uninitialized variable
warnings.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Mark Brown [Fri, 15 Jul 2011 18:12:18 +0000 (03:12 +0900)]
ASoC: Handle failed WM8994 FLL lock waits
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 <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 15 Jul 2011 08:33:26 +0000 (17:33 +0900)]
ASoC: Handle spurious wm_hubs DC servo done interrupts
Don't assume the first fire indicates that we're done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Dimitris Papastamos [Fri, 15 Jul 2011 12:51:30 +0000 (13:51 +0100)]
ASoC: WM8983: Initial driver
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 <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 15 Jul 2011 13:43:07 +0000 (22:43 +0900)]
Merge branch 'for-3.0' into for-3.1
Mark Brown [Fri, 15 Jul 2011 13:28:32 +0000 (22:28 +0900)]
ASoC: Fix shift in WM8958 accessory detection default implementation
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Daniel T Chen [Fri, 15 Jul 2011 02:06:06 +0000 (22:06 -0400)]
ALSA: intel8x0: Apply headphones+mute LED quirk for Dell Inspiron 9300
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 <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 14 Jul 2011 13:57:27 +0000 (15:57 +0200)]
ALSA: hda - Fix krealloc() replacement in hda_codec.c
It was obviously wrong, grr....
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 14 Jul 2011 13:31:21 +0000 (15:31 +0200)]
ALSA: hda - Re-add need_dac_fix check for multi-io jacks of Realtek codecs
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 <tiwai@suse.de>
Axel Lin [Thu, 14 Jul 2011 10:14:46 +0000 (18:14 +0800)]
ASoC: wm8900: fix a memory leak if wm8900_set_fll fails
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Thu, 14 Jul 2011 03:38:18 +0000 (12:38 +0900)]
ASoC: Log WM8994 FIFO errors from the interrupt
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 <broonie@opensource.wolfsonmicro.com>
Giridhar Maruthy [Wed, 13 Jul 2011 11:22:06 +0000 (16:52 +0530)]
ASoC: SAMSUNG: 24-bit audio playback on Exynos4210
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 <giridhar.maruthy@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 13 Jul 2011 06:52:13 +0000 (15:52 +0900)]
ASoC: Don't warn on low WM8994/58 AIFnCLKs
We can have valid but very low clocks in accessory detection modes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 12 Jul 2011 10:47:59 +0000 (19:47 +0900)]
ASoC: Use WM8994 FLL lock interrupt
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 <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 12 Jul 2011 06:47:17 +0000 (15:47 +0900)]
ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 12 Jul 2011 06:25:03 +0000 (15:25 +0900)]
ASoC: Implement DC servo completion IRQ handling for wm_hubs devices
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 <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 29 Jun 2011 07:21:09 +0000 (00:21 -0700)]
ASoC: Use late enable handling for direct voice, speaker and headphone
This ensures appropriate clocking for bypass paths to speaker and
headphone and direct voice paths on affected revisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Johannes Stezenbach [Mon, 11 Jul 2011 15:01:24 +0000 (17:01 +0200)]
ASoC: STA32x: Preserve reserved register bits
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 <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Johannes Stezenbach [Mon, 11 Jul 2011 15:01:23 +0000 (17:01 +0200)]
ASoC: STA32x: Add mixer controls for biquad coefficients
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 <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Paul Menzel [Tue, 12 Jul 2011 17:53:56 +0000 (19:53 +0200)]
ALSA: hda - fix up typos in Kconfig help for default buffer size introduced in
acfa634f
This commit is a fix up for commit
acfa634f.
commit
acfa634f7e199193ec28282e82a5a6dd8edebcb7
Author: Takashi Iwai <tiwai@suse.de>
Date: Tue Jul 12 17:27:46 2011 +0200
ALSA: hda - Add Kconfig for the default buffer size
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Guillaume Pellerin [Tue, 12 Jul 2011 16:13:46 +0000 (18:13 +0200)]
ALSA: usb-audio - Add quirks for M-Audio Fast Track Pro and Quattro
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 <yomguy@parisson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 12 Jul 2011 15:27:46 +0000 (17:27 +0200)]
ALSA: hda - Add Kconfig for the default buffer size
Add a Kconfig entry to specify the default buffer size.
Distros using PulseAudio can choose a larger value here.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 12 Jul 2011 06:05:16 +0000 (08:05 +0200)]
ALSA: Use krealloc() in possible places
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 11 Jul 2011 15:05:04 +0000 (17:05 +0200)]
ALSA: hda - Expose secret DAC-AA connection of some VIA codecs
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 <tiwai@suse.de>
Takashi Iwai [Mon, 11 Jul 2011 13:42:52 +0000 (15:42 +0200)]
ALSA: hda - Always read raw connections for proc output
In the codec proc outputs, read the raw connections instead of the
cached connection list, i.e. proc files contain only raw values.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 4 Jul 2011 14:23:26 +0000 (16:23 +0200)]
ALSA: hda - Add snd_hda_override_conn_list() helper function
Add a function to add/modify the connection-list cache entry.
It'll be useful to fix a buggy hardware result.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 11 Jul 2011 12:46:44 +0000 (14:46 +0200)]
ALSA: hda - Turn on extra EAPDs on Conexant codecs
Some machines seem to use EAPD control of the unused pin for controlling
the overall EAPD. Since the driver currently doesn't check the EAPD of
unused pins, the EAPD isn't enabled. For avoiding such a problem, turn
all extra EAPDs on as default.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 11 Jul 2011 09:36:44 +0000 (11:36 +0200)]
ALSA: hda - Preserve input pin-ctl bits in HP-automute for VIA codec
For smart51 pins, we need to preserve the input pin-control bits at
auto-mute controls instead of overwriting zero or pin-out-only.
Otherwise the VREF won't be set properly when smart51 is disabled
again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 11 Jul 2011 09:28:13 +0000 (11:28 +0200)]
ALSA: hda - Set line-out pin-ctls properly when indep-HP mode changes
When Independent-HP mode is changed for VIA, the driver needs to
re-issue the auto-mute check so that the line-out pins are set properly
without influence of HP pin state.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 11 Jul 2011 08:33:47 +0000 (10:33 +0200)]
ALSA: hda - Via Fix speaker-mute checks in VIA driver
When the line-jack is plugged/unplugged, the driver must check also
the headphone jack state in addition to the line-out jack. Currently
it checks only the line-out state and ignores the headphone.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Wed, 6 Jul 2011 07:08:43 +0000 (00:08 -0700)]
ASoC: Conditionalize the enable of WM8994 ADC TDM mode
Future devices will not benefit from this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Sat, 9 Jul 2011 14:16:12 +0000 (23:16 +0900)]
Merge branch 'topic/asoc' of git://git./linux/kernel/git/tiwai/sound-2.6 into for-3.1
Takashi Iwai [Sat, 9 Jul 2011 12:42:25 +0000 (14:42 +0200)]
ALSA: hda - Implement 44kHz workaround for IdeadPad as fixup
Instead of checking the model quirk, use a fixup table for workaround
of 44kHz-fixed PCM for Lenovo IdeaPad with ALC269.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Sat, 9 Jul 2011 10:06:33 +0000 (19:06 +0900)]
Merge branch 'for-3.0' into for-3.1
Kuninori Morimoto [Tue, 5 Jul 2011 07:16:17 +0000 (00:16 -0700)]
ASoC: sh: fsi-hdmi: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kuninori Morimoto [Tue, 5 Jul 2011 07:16:03 +0000 (00:16 -0700)]
ASoC: sh: fsi-da7210: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Kuninori Morimoto [Tue, 5 Jul 2011 07:15:04 +0000 (00:15 -0700)]
ASoC: sh: fsi-ak4642: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Sat, 9 Jul 2011 09:58:06 +0000 (11:58 +0200)]
Merge branch 'fix/hda' into topic/hda
Conflicts:
sound/pci/hda/patch_realtek.c
Takashi Iwai [Sat, 9 Jul 2011 09:56:43 +0000 (11:56 +0200)]
Merge branch 'fix/asoc' into for-linus
Takashi Iwai [Sat, 9 Jul 2011 09:55:28 +0000 (11:55 +0200)]
ALSA: hda - Fix a copmile warning
It's harmless but annyoing.
sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Sat, 9 Jul 2011 09:43:04 +0000 (11:43 +0200)]
Merge branch 'for-3.1' of git://git./linux/kernel/git/broonie/sound-2.6 into topic/asoc
Mark Brown [Sat, 9 Jul 2011 09:20:36 +0000 (18:20 +0900)]
Merge branch 'for-3.0' into for-3.1
Takashi Iwai [Sat, 9 Jul 2011 07:44:09 +0000 (09:44 +0200)]
Merge branch 'for-3.0' of git://git./linux/kernel/git/broonie/sound-2.6 into fix/asoc
Takashi Iwai [Fri, 8 Jul 2011 14:55:13 +0000 (16:55 +0200)]
ALSA: hda - Merge alc*_parse_auto_config() functions in patch_realtek.c
Now all alc*_parse_auto_config() do almost same thing except for the
NID list to ignore and the PINs for SSID-check, we can merge all these
to a single function. A good amount of code reduction.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 14:19:48 +0000 (16:19 +0200)]
ALSA: hda - Merge ALC260 auto-parser code
Finally the last one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 14:12:05 +0000 (16:12 +0200)]
ALSA: hda - Merge ALC269 parser code
One more code reduction. This codec has less DACs, thus the wiring
to DAC can't be filled uniquely for all output pins, i.e. some outputs
share the same volume control.
Except for that, all seems working fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 14:01:47 +0000 (16:01 +0200)]
ALSA: hda - Merge ALC268/269 auto-parser codes
Now coming to ALC268/269 parser codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 13:16:55 +0000 (15:16 +0200)]
ALSA: hda - Merge ALC861 auto-parser code
Merge more auto-parser code in patch_realtek.c, now for ALC861.
The topology of this codec is pretty simple, and can be parsed well
by the current starndard parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 13:14:19 +0000 (15:14 +0200)]
ALSA: hda - Fix amp-cap checks in patch_realtek.c
query_amp_caps() may return non-zero if the amp cap isn't supported
by the codec. Thus one needs to check widget-caps first, then check
the corresponding amp-caps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 12:39:03 +0000 (14:39 +0200)]
ALSA: hda - Merge ALC861-VD auto-parse to the standard parser
The existing standard auto-parser can work well with this codec, too.
Let's merge.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 12:37:35 +0000 (14:37 +0200)]
ALSA: hda - Fix auto-mic detection in Realtek codec-parser
A regression fix from commit
21268961d3d1bbdd22a19b68adb80119e8c72dcd
ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs
The auto-mic wasn't detected properly when no ADC-switch is needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Fri, 8 Jul 2011 10:28:47 +0000 (18:28 +0800)]
ALSA: hda - Fix output-path of VT1812 codec
For VT1812, add dac_mixer_idx for initialization.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 09:35:11 +0000 (11:35 +0200)]
ALSA: hda - Fix Oops in smart51 parsing in VIA codec
Typical off-by-one thinko.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 09:11:35 +0000 (11:11 +0200)]
ALSA: hda - Provide the standard auto_init for Realtek codecs
Remove redundant definitions. Ideally, all init functions should be
identical in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 8 Jul 2011 09:07:59 +0000 (11:07 +0200)]
ALSA: hda - Merge ALC680 auto-parser to the standard parser
Improved the standard Realtek auto-parser to support the codec topology
like ALC680.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 29 Jun 2011 15:21:00 +0000 (17:21 +0200)]
ALSA: hda - Add a fix-up for HP RP5800
The BIOS provides bogus pin configs, and also invalid SSID.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 14 Jun 2011 13:57:08 +0000 (15:57 +0200)]
ALSA: pcmcia - Use pcmcia_request_irq()
The drivers don't require the exclusive irqs. Let's fix the deprecated
warnings.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pavel Roskin [Wed, 6 Jul 2011 15:20:13 +0000 (11:20 -0400)]
ALSA: usb-audio: replace "void *" with more specific pointers
Signed-off-by: Pavel Roskin <proski@gnu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Fri, 8 Jul 2011 06:04:33 +0000 (14:04 +0800)]
ALSA: hda - Fix Independent-HP detection on VT2002P/1802/1812 codecs
For VT2002P, VT1802 and VT1812 codecs, to create Independent HP
control.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Fri, 8 Jul 2011 06:03:43 +0000 (14:03 +0800)]
ALSA: hda - Fix DAC checks for VT2002P/1802/1812 codecs
For VT2002P, VT1802 and VT1812 codecs, there're only two DACs. So smart51
control shouldn't be created.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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