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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added snd_hda_get_input_pin_label() helper function to return the
string that can be used for control or capture-source ids.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added the new fields to contain all input-pins to struct auto_pin_cfg.
Unlike the existing input_pins[], this array contains all input pins
even if the multiple pins are assigned for a single role (i.e. two
front mics). The former input_pins[] still remains for a while, but
will be removed in near future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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patch_via.c has redundant codes for parsing the input-pins. Although
they are pretty similar, but all implemented in different functions
just because of hard-coded ids and slight incompatibilities.
This patch refactors the codes to use the common helper function,
resulting in the reduction of many lines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Instead of defining each content as a separate struct, put all into the
definition of struct alc_fixup arrays so that reader doesn't go back to
see the definition again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There were some new formats added in commit 15c0cee6c809 "ALSA: pcm:
Define G723 3-bit and 5-bit formats". That commit increased
SNDRV_PCM_FORMAT_LAST as well. My concern is that there are a couple
places which do:
for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
if (dummy->pcm_hw.formats & (1ULL << i))
snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
}
I haven't tested these but it looks like if "i" were equal to
SNDRV_PCM_FORMAT_G723_24 or higher then we might read past the end of
the array.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently output controls are not uniform. Some routes are adjusted by
mono controls that don't match to associated mixer switch, many routes are
not covered at all and stereo controls have following variants:
- L-to-L & R-to-R
- R-to-L & R-to-R
- L-to-L & R-to-L
This patch attempts to fix these issues. First, for the convenience, only
direct L-to-L, R-to-R and [L | R]-to-Mono routes are controlled by the
stereo controls. This logic is also used with the output pin mute controls
so all of them except mono output are controlled by stereo switches.
Then rest of the swapped L-to-R and R-to-L routes are controlled by the
mono controls that map to mixer switches with a same name. Mixers can then
associate these switches and volumes together.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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It turned out that the output mixers and their routes were misdefined: They
are not mixing output pins to internal signals but opposite. This has worked
for direct left-to-left and right-to-right routes since for those there are
complete routes. For swapped left-to-right and right-to-left routes this is
not working since there are no routes defined between them.
Another consequence is that those misdefined mixers are incorrectly routed
to several output pins leading unnecessary pin powerings even if there is no
route active to them.
Fix these by reimplementing the output mixers and routes as they are in
hardware. For completeness add also a few missing links between internal
signals and outputs.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Each output pin has 7 consecutive control registers in tlv320aic3x register
map. First 6 of them control the signal mixing and one is for output level
and power control.
Sort these registers as they are sorted clearly in hardware, it makes also
definitions more readable and easier to pinpoint missing register
definitions.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Bit 3 in output pin_CTRL register mutes the whole output pin not just the
route from DAC so remove misleading DAC from control name. Currently only
"Line[L | R] Playback Switch" were correct.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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The spinlock lock in sound_timer.c is used without initialization.
Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If hw error is ignored, status is updated with invalid info.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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I think this is a typo, debugfs_pop_time should not be executable.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimloogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The attached patch enables playback on a Sony VAIO machine.
BugLink: http://launchpad.net/bugs/618271
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The "priv" allocated in pxa_ssp_probe() should be kfreed in pxa_ssp_remove().
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In synchronous mode the SSI_SRCCR values are ignored. Instead
SSI_STCCR must be used for both receiving and transmitting.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The Makefile and Kconfig updates for WL1273 appear to have been mising
from the patch posted, add them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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This makes it that little bit easier to spot the diagnostics in the
logs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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Speaker amplifier is controlled by TWL4030 GPIO which may sleep. Therefore
use gpio_set_value_cansleep to get rid of runtime warning that is introduced
after the commit 9c4ba94 and to get a stack trace if ever executing this
code in atomic context.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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aic3x_init does a soft reset first and thus TLV320AIC3x GPIO setup must be
done after doing the basic init. Before multi-component the init was done
at i2c probe time and GPIO setup at soc probe time.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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This patch adds quirk for the Lenovo S10-3t so the headphone &
microphone jacks will now work.
Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This device is similar to the M-Audio Delta 1010LT in that it uses the
AK4524VF ADC/DAC, but it does not use the CS8427 for SPDIF.
The SPDIF appears to be set up correctly, but I am not able to test it
as I do not have any devices that use it.
This patch makes the ADC/DAC's and the hardware mixer visible to apps
such as alsamixer and envy24control.
Signed-off-by: Garnet MacPhee <dhubsith@comcast.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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'struct of_device' no longer exists, and its functionality has been merged
into platform_device. Update the MPC8610 HPCD audio drivers (fsl_ssi, fsl_dma,
and mpc8610_hpcd) accordingly.
Also add a #include for slab.h, which is now needed for kmalloc and kfree.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into for-2.6.37
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Otherwise we generate worrying (but benign) warnings for amps.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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This is an ALSA codec for the Texas Instruments WL1273 FM Radio.
Signed-off-by: Matti J. Aaltonen <matti.j.aaltonen@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
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git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6
Conflicts:
arch/sh/kernel/process_32.c
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
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The Freescale P1022 is a dual-core e500-based SOC with multimedia capabilities,
specifically the same SSI audio controller on the MPC8610. The P1022 DS
reference board includes a P1022 and a Wolfson Microelectronics WM8776
codec.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Fix reference to moved header file, which was unused anyway.
This change fixes below build error:
CC sound/soc/pxa/pxa2xx-ac97.o
sound/soc/pxa/pxa2xx-ac97.c:27:24: error: pxa2xx-pcm.h: No such file or directory
make[3]: *** [sound/soc/pxa/pxa2xx-ac97.o] Error 1
make[2]: *** [sound/soc/pxa] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Haojian Zhuang <haojian.zhuang@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This patch adds support for the tlv320aic3007 codec to the tlv320aic3x
driver.
The tlv320aic3007 is similar to the aic31, but has an additional class-D
speaker amp. The speaker amp control register overlaps with the mono
output register of other codecs in this family, so we add logic to
identify the actual codec being registered to set things up accordingly.
Signed-off-by: Randolph Chung <tausq@parisc-linux.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Some codecs have separate DAIs for playback and capture, so the DMA driver
should allocate a DMA buffer only for the streams that are valid when the
driver is opened.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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In e740_init(), we call gpio_request() for
GPIO_E740_MIC_ON, GPIO_E740_AMP_ON and GPIO_E740_WM9705_nAVDD2.
We should free the these gpio accordingly in e740_exit().
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The new sticky PCM parameter introduced the delayed clean-ups of
stream- and channel-id tags. In the current implementation, this check
(adding dirty flag) and actual clean-ups are done only for the codec
chip. However, with HD-audio architecture, multiple codecs can be
on a single bus, and the controller assign stream- and channel-ids in
the bus-wide.
In this patch, the stream-id and channel-id are checked over all codecs
connected to the corresponding bus. Together with it, the mutex is
moved to struct hda_bus, as this becomes also bus-wide.
Reported-and-tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Intel and Nvidia HDMI codec drivers have own implementations of
sticky PCM parameters. Now HD-audio core part already has it,
thus both setups conflict. The fix is simply remove the part in
patch_intelhdmi.c and patch_nvhdmi.c and simply call
snd_hda_codec_setup_stream() as usual.
Reported-and-tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In the process of unification of codec DAI names while implementing
multi-component, the CX20442 codec DAI has been renamed to "cx20442-hifi".
This new name seems not adequate for a 8kHz voice codec.
Use a better name, "cx20442-voice", as suggested by Liam Girdwood.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The tlv320aic3x codec driver only supports symmetric rates for capture/
playback. Set the flag in the DAI accordingly.
Signed-off-by: Randolph Chung <tausq@parisc-linux.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The current code in pcm_lib.c do all checks using only the position
in the ring buffer. Unfortunately, where the interrupts gets delayed or
merged into one, we need another timing source to check when the
buffer size boundary overlaps to avoid the wrong updating of the
ring buffer pointers.
This code uses jiffies to check the right time window without any
performance impact.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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