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Julia Lawall [Thu, 28 Jul 2011 12:46:05 +0000 (14:46 +0200)]
ALSA: sound/core/pcm_compat.c: adjust array index
Convert array index from the loop bound to the loop index.
A simplified version of the semantic patch that fixes this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e1,e2,ar;
@@
for(e1 = 0; e1 < e2; e1++) { <...
ar[
- e2
+ e1
]
...> }
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Andy Whitcroft [Wed, 27 Jul 2011 16:48:41 +0000 (17:48 +0100)]
sound: oss: rename local change_bits to avoid powerpc bitsops.h definition
This collides with powerpc exported functions from bitops.h. Rename the
local copy in the oss soundblaster mixer and ad1848 driver.
Signed-off-by: Andy Whitcroft <apw@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 27 Jul 2011 14:41:57 +0000 (16:41 +0200)]
ALSA: hda - Fix duplicated DAC assignments for Realtek
Copying hp_pins and speaker_pins from line_out_pins may confuse the
parser, and it can lead to duplicated initializations for the same pin
with a wrong DAC assignment. The problem appears in 3.0 kernel code.
Cc: <stable@kernel.org> (for 3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dan Carpenter [Wed, 27 Jul 2011 12:02:26 +0000 (15:02 +0300)]
ALSA: asihpi - off by one in asihpi_hpi_ioctl()
"adapter" is used as an array index in the adapters[] array so
the off by one would make us read past the end.
1c073b67979 "ALSA: asihpi - Remove spurious adapter index check"
reverted Dan Rosenberg's check that would have prevented the
overflow here.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 27 Jul 2011 12:01:24 +0000 (14:01 +0200)]
ALSA: hda - Fix Oops with Realtek quirks with NULL adc_nids
Somce quirk models don't set adc_nids but let the parser filling it.
But the recent code has unnecessary NULL-checks of spec->input_mux,
and it resulted in NULL dereferences.
This patch fixes that regression.
Reported-and-tested-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Wed, 27 Jul 2011 08:03:51 +0000 (20:03 +1200)]
ALSA: asihpi - bug fix pa use before init.
Fixes bug introduced by
1c073b67.
Also declare pa local to block in which it is used.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vitaliy Kulikov [Tue, 26 Jul 2011 21:56:20 +0000 (16:56 -0500)]
ALSA: hda - Add support for vref-out based mute LED control on IDT codecs
This patch also registers all necessary callbacks to support mute LED
only when such control is enabled. And it keeps codec AFG in D0 or D1
state all the time when aggressive power managemnt is enabled for vref-out
control (and mute LED) work correctly.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 26 Jul 2011 15:47:05 +0000 (17:47 +0200)]
Merge branch 'fix/asoc' into for-linus
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>
Mark Brown [Thu, 21 Jul 2011 13:58:05 +0000 (14:58 +0100)]
ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver"
This reverts commit
d7c3e9525ac8e898f1156a1f3a7c5038f6560186 as it does
not currently build due to missing dependencies in the Samsung tree.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
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>
Liam Girdwood [Wed, 20 Jul 2011 11:23:33 +0000 (12:23 +0100)]
ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
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 <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sangbeom Kim [Wed, 20 Jul 2011 08:07:13 +0000 (17:07 +0900)]
ASoC: SAMSUNG: Add I2S0 internal dma driver
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 <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sangbeom Kim [Wed, 20 Jul 2011 08:07:12 +0000 (17:07 +0900)]
ASoC: SAMSUNG: Modify I2S driver to support idma
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 <sbkim73@samsung.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rajashekhara, Sudhakar [Wed, 20 Jul 2011 12:07:18 +0000 (17:37 +0530)]
ASoC: davinci: add missing break statement
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 <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Rajashekhara, Sudhakar [Wed, 20 Jul 2011 12:06:04 +0000 (17:36 +0530)]
ASoC: davinci: fix codec start and stop functions
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 <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Liam Girdwood [Wed, 20 Jul 2011 18:42:20 +0000 (19:42 +0100)]
ASoC: dapm - add DAPM macro for external enum widgets
Add a convenience macro for external enumerated widgets.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 20 Jul 2011 12:50:10 +0000 (13:50 +0100)]
ASoC: Acknowledge WM8962 interrupts before acting on them
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 <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Wolfram Sang [Mon, 18 Jul 2011 15:53:04 +0000 (17:53 +0200)]
ASoC: sgtl5000: guide user when regulator support is needed
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 <w.sang@pengutronix.de>
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Wolfram Sang [Mon, 18 Jul 2011 15:53:03 +0000 (17:53 +0200)]
ASoC: sgtl5000: refactor registering internal ldo
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 <w.sang@pengutronix.de>
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Wolfram Sang [Sun, 17 Jul 2011 16:00:26 +0000 (18:00 +0200)]
ASoC: core: make comments fit the code
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 <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Mon, 18 Jul 2011 04:17:13 +0000 (13:17 +0900)]
ASoC: Mark cache as dirty when suspending
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 <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
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>