Age | Commit message (Collapse) | Author |
|
Make sure we set the bias level for all DAPM contexts when
changing level.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Use dev_ style logging throughout soc_new_pcm()
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Codec shutdown performs a DAPM power sequence that might cause conflicts
and/or race conditions if another stream power event is running simultaneously.
Use card's dapm mutex to protect any potential race condition between them.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
|
|
In the new PCM streaming logic, the interface number is assigned to
usb stream instance (subs->interface) after the format and rate setups
are succeeded, but some codes are still passing subs->interface as the
reference to helper functions. This leads to initializing with an
invalid iface number (-1).
This patch replaces the wrong references with the ones from the target
fmt correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Not previously noticed due to normal usage being with CODEC<->CODEC links.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
This enables common telephony use cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
If an audio interface is configured to use ASYNCCLK then update the
asynchronous sample rate rather than one of our primary sample rates.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
mx27pdk board also has a mc13783 codec.
Add support for it and do a run-time machine type check to perform the correct
audiomux settings.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
If we don't set IMX_AUDMUX_V2_PTCR_TCLKDIR in the AUDMUX PTCR
register (means Tx clock pin is input), we don't need to set
IMX_AUDMUX_V2_PTCR_TCSEL as well. Since both i.MX35, i.MX51 and
i.MX6 datasheet says "When Tx clock pin set as an input, the TCSEL
settings are ignored".
Signed-off-by: Hui Wang <jason77.wang@gmail.com>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
They aren't modified by the core so the drivers can declare them const.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
With the model parsers out of the way, we have no custom unsol
events to worry about, we can therefore simplify the code.
In addition, this fixes a bug on the ASUS TC710, which has only
a headphone jack and a mic jack, but no internal mic or speakers.
Therefore the unsol_event pointer was not set, and as a result,
the jack kcontrols were not correctly updated.
BugLink: https://bugs.launchpad.net/bugs/1021192
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Allow any BCLK which can be divided down to generate LRCLK, not just the
lowest possible BCLK to clock out the samples.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Allow the user to select which of the system clocks each AIF is referenced
to and constran the DAI to the set of frequencies which can be generated
from that clock.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
They're the rates for the BCLK, not for the sample rate, so rename so that
we don't confuse ourselves.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
This patch implements the spdif out driver for ST peripheral. This
peripheral implements IEC60958 standard
Signed-off-by: Vipin Kumar <vipin.kumar@st.com>
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The STA529 is a digital stereo class-D audio amplifier. It includes an
audio DSP, an ST proprietary high-efficiency class-D driver and CMOS
power output stage. It is intended for high-efficiency
digital-to-power-audio conversion for portable applications.
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Adds missing register default values to cache.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Adds register and bit shift definitions in header file.
Changes are for TLV320AIC310x based on data sheet.
Signed-off-by: Jiri Prchal <jiri.prchal@aksignal.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
|
|
Leave it up to the machine driver to disable accessory detection if
desired, the common pattern is to have accessory detection be a wake
source.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Since most new drivers are expected to use regmap and since frequently the
only thing we need to do in the CODEC probe function is configure the I/O
try to initialise the register I/O using regmap if the driver hasn't done
so after probe().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
|
|
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
dapm_set_path_status() sets connect incorrectly in the case max > 1 with invert.
In that case, the raw disconnect value should be max, which corresponds to the
userspace value 0.
This use case currently does not appear upstream, but it could break
SOC_DAPM_SINGLE() or SOC_DAPM_SINGLE_TLV() elsewhere or in the future.
This patch completes commit 3a9abe8.
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: <alsa-devel@alsa-project.org>
Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau@advansee.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Use the dmaengine based Tegra APB DMA driver for
data transfer between SPI fifo and memory in
place of legacy Tegra APB DMA.
Because generic soc-dmaengine-pcm uses the DMAs API
based on dmaengine, using the exported APIs provided
by this generic driver.
The new driver is selected if legacy driver is not
selected and new dma driver is enabled through config
file.
Signed-off-by: Laxman Dewangan <ldewangan@nvidia.com>
Acked-by: Stephen Warren <swarren@wwwdotorg.org>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Stereo is not yet supported by dapm widgets, so remove stereo code from
snd_soc_dapm_get_volsw(), and warn if stereo controls are detected.
Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau@advansee.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Acked-by: Sangbeom Kim <sbkim73@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Call cancel_work_sync() when turning off the beep switch so that the
mute call is executed in a proper order.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
It's no longer necessary since beep_mode=2 option was dropped.
It can be checked simply via codec->beep != NULL.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Move snd_hda_mixer_amp_switch_put_beep() to hda_beep.c as a clean up
to remove one more ifdef.
Also add the corresponding get callback to return consistently the
digital beep state independently from the mixer amp value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The beep_mode=2 option was introduced to make the beep mixer
controlling the beep input allocation/deallocation dynamically, so
that a user can switch between HD-audio codec digital beep and the
system beep only via mixer API. This was necessary because the
keyboard driver took only the first input beep instance at that time.
However, the recent keyboard driver already processes the multiple
input instances, thus there is no point to keep this mode.
Let's remove it.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This is a bit clean up of public sound header directory.
Some header files in include/sound aren't really necessary to be
located there but can be moved to their local directories gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.
Since we change vx222, vx_core and vxpocket have to be converted,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The mioa701 board code is converted to the snd_soc_register_card()
and snd_soc_unregister_card() APIs.
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Windows use hidden register to control EAPD.
Linux use verb to control EAPD.
If windows reboot to Linux, it must change the EAPD control to verb
control.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
In sound/soc/codecs/tlv320aic3x.c
data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG);
snd_soc_write(codec, AIC3X_PLL_PROGA_REG,
data | (pll_p << PLLP_SHIFT));
In the above code, pll-p value is OR'ed with previous value without
clearing it. Bug is not seen if pll-p value doesn't change across
Sampling frequency.
However on some platforms (like AM335x EVM-SK), pll-p may have different
values across different sampling frequencies. In such case, above code
configures the pll with a wrong value.
Because of this bug, when a audio stream is played with pll value
different from previous stream, audio is heard as differently(like its
stretched).
Signed-off-by: Hebbar, Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
|
|
Without this very high BCLKs will be configured incorrectly.
Reported-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
|
|
snd_soc_dapm_put_volsw() sets connect incorrectly in the case max > 1 with
invert. In that case, the raw disconnect value should be max, which corresponds
to the userspace value 0.
This use case currently does not appear upstream, but it could break
SOC_DAPM_SINGLE() or SOC_DAPM_SINGLE_TLV() elsewhere or in the future.
Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau@advansee.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Otherwise we fail to link when building as modules due to multiple
init/exit functions.
Reported-by: Fengguang Wu <wfg@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
This patch adds the support for spdif in audio codec.
Signed-off-by: vipin Kumar <vipin.kumar@st.com>
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Otherwise sparse gets very upset with us.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here you find quite a few changes for HD-audio and a copule of quirk
additions for USB-audio. All reasonably small and/or trivial."
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix power-map regression for HP dv6 & co
ALSA: hda - Initialize caches at codec reconfiguration
ALSA: hda - Fix memory leaks at module unload
ALSA: hda - Fix memory leaks in Realtek & Conexant codec parsers
ALSA: hda - Add Realtek ALC280 codec support
ALSA: hda - Remove obsoleted CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
ASoC: wm8994: remove duplicate code
ALSA: usb-audio: add BOSS GT-100 support
ALSA: HDA: Add inverted internal mic quirk for Lenovo S205
ALSA: hda - Fix ALC272X codec detection
ALSA: snd_usb_audio: ignore ctrl errors on QuickCam Pro for Notebooks
ALSA: snd_usb_audio: ignore ctrl errors on QuickCam E3500
|
|
This 3-pin jack was labeled "Headphone Jack", but it could also be
used as a mic jack just by switching "Input Source". Therefore we need
to call the jack something else, to make sure PulseAudio can use the
speaker together with the external mic. (PulseAudio might mute the
speaker if it detects a headphone being plugged in.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Some ASUS device has a single 3-pin jack that can either be a mic or
a headphone, but the pin does not have VREF capabilities. We've been
told by Realtek to instead enable VREF on pin 0x18 in that case.
BugLink: https://bugs.launchpad.net/bugs/1018262
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
Fixes the following warning:
CC [M] sound/usb/caiaq/device.o
sound/usb/caiaq/device.c: In function ‘snd_probe’:
sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used
uninitialized in this function [-Wmaybe-uninitialized]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|