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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>
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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>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Don't query connections for widgets have no connections
ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums
ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
ALSA: HDA: Fix dock mic for Lenovo X220-tablet
ASoC: format_register_str: Don't clip register values
ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare
ASoC: zylonite: set .codec_dai_name in initializer
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The DAPM pin operations currently require that the specific DAPM context
that the pin being operated in is contained in be specified. With multi
component and especially with the addition of a per-card DAPM context
this isn't ideal as it means that things like disabling unused pins on
CODECs require looking up the CODEC DAPM context.
Fix this by falling back to matching a widget in any context if there isn't
a match in the current context. The code isn't ideal currently but will do
the job.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
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Currently we allow all DAPM contexts to determine their own bias level.
While this should in general work in most situations and will deliver the
lowest possible power it causes problems for our integration with the
card bias level as we're calling the card bias level functions for each
DAPM context even though they're card wide but don't say which CODEC
we're calling them for. Mitigate against this by forcing everything to
be in the same state.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
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Since we recently explicitly set the register for registerless widgets
to no register there is no longer any need to special case power updates
for them, we can allow them to be handled with the register compression
code as other widgets are.
As this is the only remaining user of dapm_generic_apply_power() and
dapm_update_bits() also remove those functions.
Noticed-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The ssm2602 codec has a SPI interface as well as I2C, so add the simple
bit of glue to make it usable.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Allow CODEC and card drivers to point to an array of controls from their
driver structure rather than explicitly calling snd_soc_add_controls().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Update the headphone and line out mixers and PGAs use the same logical
set of register bits and sequencing as the speaker mixer/PGA.
This allows ALSA controls for mute and volume on headphone and line out
to operate correctly.
Per conversation on alsa-devel, earlier datasheets indicated that the
POWER_MANAGEMENT_* register bits 0 and 1 were aliases to ANALOG_* register
bits 0 and 4, and hence only one copy of those bits was programmed.
However, later datasheets corrected this.
From: Dilan Lee <dilee@nvidia.com>
[swarren: Applied same change to headphone widgets]
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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* 'for-linus2' of git://git.profusion.mobi/users/lucas/linux-2.6:
Fix common misspellings
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The connection lists are static and we can reuse the previous results
instead of querying via verb at each time. This will reduce the I/O
in the runtime especially for some codec auto-parsers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fixes the kernel warnings with IDT codecs like
hda_codec: connection list not available for 0x1e
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since we now set up the connections and mutes dynamically in the
auto-parser, all static initializations via alc662_init_verbs & co are
no longer needed. Let's drop them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Mute the ADC as default in the auto-parser dynamically instead of relying
on the static init verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Instead of static init array, better to determine the connection and
the mute status of the pin/mixer/DAC route dynamically. This fixes the
uninitialized mixer 0x0f on ALC892.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In cases where there is only one internal mic connected to ADC 0x11,
alc275_setup_dual_adc won't handle the case, so we need to add the
ADC node to the array of candidates.
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/752792
Reported-by: Vincenzo Pii
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The MCP7x hardware computes the audio infoframe channel count
automatically, but requires the audio driver to set the audio
infoframe checksum manually via the Nv_VERB_SET_Info_Frame_Checksum
control verb.
When audio starts playing, nvhdmi_8ch_7x_pcm_prepare sets the checksum
to (0x71 - chan - chanmask). For example, for 2ch audio, chan == 1
and chanmask == 0 so the checksum is set to 0x70. When audio playback
finishes and the device is closed, nvhdmi_8ch_7x_pcm_close resets the
channel formats, causing the channel count to revert to 8ch. Since
the checksum is not reset, the hardware starts generating audio
infoframes with invalid checksums. This causes some displays to blank
the video.
Fix this by updating the checksum and channel mask when the device is
closed and also when it is first initialized. In addition, make sure
that the channel mask is appropriate for an 8ch infoframe by setting
it to 0x13 (FL FR LFE FC RL RR RLC RRC).
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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alc269_suspend is just calling the shut-up, so we can use the new shutup
callback for the purpose.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add shutup callback to be called codec-specifically for avoiding pop
noises at suspend or shutdown. As a generic callback, just turn EAPD
off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Reduced the duplicated codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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So far, alc662_init_verbs[] is used for all ALC662-compatible chips,
but the EAPD controls for 0x15 in there is invalid for ALC892.
Also, since EAPDs should be set up in alc_auto_init_amp(), these static
elements aren't needed for auto-parser, too.
In this patch, the EAPD init verbs are split from alc662_init_verbs,
and applied only to static quirks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Ensures that we apply volume updates that don't affect the right channel.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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initialize ret to invalid value so that when we reach the config error path in
soc_pcm_open, it will return the correct error code. without this patch, though
config error path is executed, soc_pcm_open will return 0 in
snd_pcm_open_substream and then cause double release of substream.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Suppose we have:
cpu_dai
channels_min = 1
channels_max = 1
codec_dai
channels_min = 2
channels_max = 2
This is a mismatch that should not happen, however according to the current
code, the result of runtime->hw will be:
channels_min = 2
channels_max = 1
We better spot it early. This patch checks this mismatch.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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module_name will be checked in register_sst_card.
It will fail to register sst card if it's not initialized.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch adds the MAX98095 CODEC driver.
Signed-off-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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ASoC machine drivers that are their own platform_driver (as opposed to
those using the soc-audio platform_driver) need to explicitly set up
power-management operation callbacks.
To avoid cut/paste, snd_soc_pm_ops also needs to be exported.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The current alc662 parser doesn't set the DAC for the mixer 0x0f
properly for ALC892, which has 4 DACs while ALC662 has 3.
Fixed by implementing alc662_mix_to_dac() more genericly with the
dynamic widget list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some ALC272-quirks use alc662_dac_nids instead of alc272_dac_nids.
This patch fixes these entries. No functional change since the first
two elements are identical in both arrays.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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SB Live! Platinum CT4760P is just a 4 channels sound card with STAC9721 and
Philips UDA1334 DAC.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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alc662 series only have 3 DAC, so it can only support 5stack-dig
instead of 6stack-dig.
[updated HD-Audio-Models.txt as well by tiwai]
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There are many USB MIDI cables out there that have buggy
firmware that reports it can do more than 4 bytes in a
packet when they can only properly handle 4
This patch adds the ID of yet another one of those cables
Signed-off-by: Tarek Soliman <tarek@bashasoliman.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Remove some unneeded defintions
Use %pR to print resources
Make some data const
Consistent braces for else
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Replacing subsys_delete_adapter with adapter_delete
allows some special-case adapter lookup code to be removed.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Define and use pcm_debug_name if CONFIG_SND_DEBUG
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Allow older non DMA capable cards to use MMAP by
emulating the DMA using read and write functions,
and getting rid of copy & silence callbacks that
were used only by older cards.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the card drained status reporting for playback,
but allow it to persist for a few timer cycles before
signalling XRUN, to allow card to recover by itself.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Debug print full substream ID.
Other minor debug print updates.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Clock source is neither capture nor playback,
so change 'Capture Clock' to 'Clock'.
Add spaces to control name string for consistency,
always 'PCM 0' , never 'PCM0'
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Without the "thinkpad" quirk, the dock mic in
Lenovo X220 tablet edition won't work.
BugLink: http://bugs.launchpad.net/bugs/751033
Cc: stable@kernel.org
Tested-by: James Ferguson <james.ferguson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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