Now we have a complete list of loopback paths, thus we can initialize
the paths more completely based on it, instead of assuming a direct
connection from pin to mixer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't forget to take boost volumes into account in the managed path
list. Since it's an additional volume, we need to extend the ctls[]
array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The paths used for multi-io haven't been initialized properly, so
far. It's usually no big matter because the pins are set to input as
default, but it's still cleaner to initialize the paths properly.
Now with the path active/inactive check, we can do it easily.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pin widget has only a single amp value for the input even if it
has multiple "sources". Handle the situation in activate_path().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc_auto_is_dac_reachable() can be replaced fully with
is_reachable_path(). The only difference is the order of arguments.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... and rewrite the initialization of output paths as a generic
function that is applicable for both i/o directions.
The new flag, active, is introduced to each nid_path entry. This
indicates whether the given path is active, and it's used for checking
whether a certain widget can be turned off or changed when a path is
no longer used or newly enabled.
It's still used only in the output paths. More wider adaption for
input and loopback paths will be achieved in the later patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We are using only AUTOMUTE_MODE_PIN in patch_realtek.c and all others
have been already dropped. Let's remove the old superfluous codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new function snd_hda_codec_amp_init() (and the stereo variant)
initializes the amp value only once at the first access. If the amp
was already initialized or updated, this won't do anything more.
It's useful for initializing the input amps that are in the part of
the path but never used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For optimizing the verb executions, a new mechanism to cache the verbs
and amp update commands is introduced. With the new "write to cache
and flush" way, you can reduce the same verbs that have been written
multiple times.
When codec->cached_write flag is set, the further
snd_hda_codec_write_cache() and snd_hda_codec_amp_stereo() calls will
be performed only on the command or amp cache table, but not sent to
the hardware yet. Once after you call all commands and update amps,
call snd_hda_codec_resume_amp() and snd_hda_codec_resume_cache().
Then all cached writes and amp updates will be written to the
hardware, and the dirty flags are cleared.
In this implementation, the existing cache table is reused, so
actually no big code change is seen here. Each cache entry has a new
dirty flag now (so the cache key is now reduced to 31bit).
As a good side-effect by this change, snd_hda_codec_resume_*() will no
longer execute verbs that have been already issued during the resume
phase by checking the dirty flags.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The generated defines in the header are pre-shifted.
Reported-by: Heather Lomond <Heather.Lomond@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable bypass when the regulator is idle, not when it is in use. This is
consistent with what the few existing users actually want.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I've removed several unreachable returns.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With commit f2818d0 (ASoC: fsl: fix miscompilation of snd-soc-imx-pcm),
we will see the following build error when building modules with
CONFIG_SND_IMX_SOC=m in imx_v6_v7_defconfig.
CC [M] sound/soc/fsl/phycore-ac97.o
LD [M] sound/soc/fsl/snd-soc-fsl-ssi.o
LD [M] sound/soc/fsl/snd-soc-fsl-utils.o
LD [M] sound/soc/fsl/snd-soc-imx-ssi.o
LD [M] sound/soc/fsl/snd-soc-imx-audmux.o
LD [M] sound/soc/fsl/snd-soc-imx-pcm.o
sound/soc/fsl/imx-pcm-dma.o: In function `init_module':
imx-pcm-dma.c:(.init.text+0x0): multiple definition of `init_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.init.text+0x0): first defined here
sound/soc/fsl/imx-pcm-dma.o: In function `cleanup_module':
imx-pcm-dma.c:(.exit.text+0x0): multiple definition of `cleanup_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.exit.text+0x0): first defined here
make[4]: *** [sound/soc/fsl/snd-soc-imx-pcm.o] Error 1
Instead of using bool for SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA
to fix the original issue, we should completely remove SND_SOC_IMX_PCM
and have imx-pcm.o statically linked with imx-pcm-fiq.o or imx-pcm-dma.o.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As the IEC-60958 channel status word is set by ANDing and ORing with
the appropriate definitions, the word bytes need to be initialized
to zero to avoid misconfiguration due to previous hw_params calls.
Signed-off-by: Ricardo Neri <rneri@dextratech.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code to do the ramp has been removed a long time ago. Remove the
remaining code as well since this is not needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There's no need to create a queue for this anymore
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In this way we can clean up the probe and remove paths
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No need to set the bias_level twice to _STANDBY - since this is the only
state the device could be at suspend time. The driver do not support
idle_bias_off yet.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With idle_bias_off these are no longer needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert headset PLUGINT interrupt to NO_SUSPEND type in order to
allow handling of insertion/removal events while device is suspended.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for
audioformat mismatch] introduced the correction of parameters to be
set for sync EP. But since the new code assumes that the sync EP is
always paired with the data EP of another direction, it triggers Oops
when a device only with a single direction is used.
This patch adds a proper check of sync EP type and the presence of the
paired substream for avoiding the crash.
Reported-and-tested-by: Jens Axboe <axboe@kernel.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The header can be local in mach-s3c24xx/ and sort out inclusions.
Accordingly, the GTA02_ macro in driver can be replaced.
Cc: Sangbeom Kim <sbkim73@samsung.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
Nothing terribly exciting here except for the DOUBLE_RANGE fix which
just hadn't worked before, nobody noticed due to lack of use.
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Merge tag 'asoc-fix-3.8-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.8
Nothing terribly exciting here except for the DOUBLE_RANGE fix which
just hadn't worked before, nobody noticed due to lack of use.
Support for loading the ak4642 codec module via devicetree.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When initializing the output paths, we assumed the input amps have
almost two inputs blindly. It's not only generic but even incorrect
for some codecs like ALC268 & co. Also, the same assumption (two
sources) exists for the bind input-amp controls.
This patch changes the codes in these places to handle the input
connections in a more generic way.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For handling the analog-loopback paths more generically, check the amp
capabilities of the aa-mixer widget, and create only the appropriate
mixer elements.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve the parser of analog loopback paths and handle in a more
generic way. The following changes are included in this patch:
- Instead of assuming direct connections between pins and
the mixer widget, track the whole path between them. This fixes
some missing connections like ALC660.
- Introduce the path list for loopback paths like input and output
path lists. Currently it's not used for any real purposes, yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like the output paths, parse the whole paths for inputs as well
and store in a path list. For that purpose, rewrite the output parser
code to be generically usable.
The input path list is not referred at all in this patch. It'll be
used to replace the fixed adc/capsrc array in later patches for more
flexible input path selections.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, idx[i] and multi[i] indicate the attribute of the widget
path[i - 1]. This was just for simplifying the code in
__parse_output_path(), but this is rather confusing for later use.
It's more natural if both idx[i] and multi[i] point to the same widget
of path[i]. This patch changes to that way.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the output path initialization using the existing path
information instead of assuming the topology specific to Realtek
codecs. This is also implicitly a fix for some amp values on output
pins where the old parser missed (e.g. ALC260 output pins).
The same function alc_auto_set_output_and_unmute() can be used now for
the multi-io activation, since the output selection means nothing but
activating the given output path.
And, finally at this stage, we can get rid of alc_go_down_to_selector()
and other functions that are codec really specifically to Realtek
codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, Realtek codec driver evaluates the NIDs for volume and mute
controls twice, once while parsing the DACs and evaluating the
assignment, and another while creating the mixer elements. This is
utterly redundant and even fragile, as it's assuming that the ctl
element evaluation is identical between both parsing DACs and creating
mixer elements.
This patch simplifies the code flow by doing the volume / mute
controls evaluation only once while parsing the DACs. The patch ended
up in larger changes than expected because of some cleanups became
mandatory.
As a gratis bonus, this patch also fixes some cases where the stereo
channels are used wrongly for mono amps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mono widget is always connected to the left channel, thus the left
channel amp value also should be referred for mono widgets instead of
the right channel.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
