The rate controls are codec-specific, it's not possible to
generically say what the range or the meaning of each control
is (or even if they exist at all) - that depends on the
particular codec.
This is currently being handled for Arizona codecs by putting
an Arizona-specific table of controls inside the wm_adsp driver.
This creates a dependency between wm_adsp and arizona.c, and is an
awkward solution if the ADSP is used in another family of codecs
Fix this by moving the Arizona-specific rate controls into the
Arizona codec drivers.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial typo.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A typical io ops use simple io accessors which can be common for most
drivers, so provide a default ops which will be used if driver doesn't
provide one
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In HDA extended bus the HDA link objects are created when multilink
capabilities are parsed. We need a routine which free up these link objects
for a bus. So add snd_hdac_link_free_all routine
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDAC extended core should create streams for an extended bus and also free
up those on cleanup. So introduce snd_hdac_ext_stream_init_all and
snd_hdac_stream_free_all routines
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'047000278da3a17f8("ASoC: rsrc-card: cleanup for DPCM")'
cleanuped rsrc-card driver, but then, unused ret was left.
Below warning happen without this patch
${LINUX}/sound/soc/sh/rcar/rsrc-card.c: In function 'rsrc_card_startup':
${LINUX}/sound/soc/sh/rcar/rsrc-card.c:78:6: warning: unused variable \
'ret' [-Wunused-variable]
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since this is common option for HDA driver to specfiy pre-allocated
buffer, we should make this option availble to all HDA driver by
moving this to HDA core
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the DPCM based machine driver with rt5650 and rt5676
Signed-off-by: Nicolas Boichat <drinkcat@chromium.org>
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the DPCM based platform driver of AFE (Audio Front End) unit.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many path
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. rsnd_mod_to_io() is no longer needed. Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship.
This patch checks module working status via io instead of mod
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from snd_kcontrol
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_src_xxx()
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_ssi_xxx()
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_dma_xxx()
and related function
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_get_adinr()
and its related function
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. Then, interrupt handler can't use rsnd_mod_to_io().
This patch adds SSI/SRC/DMA common interrupt handler frame
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This means we can't call rsnd_mod_to_io() any more.
This patch adds struct rsnd_dai_stream to each function as parameter.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This means we can't use rsnd_mod_to_io() in SSI/SRC/DMA
interrupt handler. In such case, we need to check all io in interrupt
handler, and then, "priv" is needed.
This patch adds rsnd_priv pointer in rsnd_mod for prepare it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. Then, we don't need to re-call each mod function
that had been called. This patch count each mod status.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsrc-card driver is based on simple-card driver which is caring about
CPU / Codec connection. OTOH, rsrc-card is used for DPCM system.
FE portion is constituted by CPU and dummy Codec, and BE is constituted
by dummy CPU and Codec in DPCM system.
Because of this, current rsrc-card is doing pointless method. It works well
if FE/BE was 1:1, but not good for multi FE/BE.
This patch cleanups rsrc-card driver for DPCM. and this is prepare for
MIX support for Renesas sound driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is prepare for DPCM cleanup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsrc-card is assuming 1 FE (= CPU), 1 BE (= codec) on card.
But, it will support multi FE/BE card. This is prepare for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current dai_link name is using "cpu_dai_name + codec_dai_name",
but one of them is always "snd-soc-dummy-dai" when DPCM.
This patch uses "fe.xxx" for cpu, "be.xxx" for codec.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
'a9e1ac1a9e4585b5("ASoC: rsnd: spin lock for interrupt handler")'
added spin lock under interrupt handler to solve HW restart issue.
OTOH, current rsnd driver calls snd_pcm_period_elapsed() from
rsnd_dai_pointer_update(). but, it will be called under spin lock
if SSI was PIO mode.
If it was called under spin lock, it will call
snd_pcm_update_state() -> snd_pcm_drain_done().
Then, it calls rsnd_soc_dai_trigger() and will be dead-lock.
This patch doesn't call rsnd_dai_pointer_update() under spin lock
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PIO is used only for checking data path / codec settings. And underrun
is very normal when PIO mode. Let's don't care about under/over run
error when PIO case. Otherwise, 1) too many HW restart happens, 2) some
sounds which need much data transfer can't play since it falls into
error detection method which was created for DMA transfer
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When CONFIG_SND_HDA_I915=n, we get a compile warning:
sound/pci/hda/hda_intel.c: In function ‘azx_probe_continue’:
sound/pci/hda/hda_intel.c:1882:2: warning: label ‘skip_i915’ defined but not used [-Wunused-label]
Fix it by putting again ifdef to it. Sigh.
Fixes: bf06848bdb ('ALSA: hda - Continue probing even if i915 binding fails')
Reported-by: Borislav Petkov <bp@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Developing a driver for an Asus X205TA laptop I get these dmesg
errors:
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC1 Swap Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC2 Swap Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC3 Swap Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 L Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 R Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 L Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 R Mux has no paths
so, move these muxes to the rt5650_specific_dapm_widgets[] list.
Signed-off-by: Michele Curti <michele.curti@gmail.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The new Dell XPS13 also requires the similar quirk for fixing the
noisy outputs. (But, as the codec was changed, now the fixup for
Latitude is used instead.)
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99851
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Behringer FCA610 transmits packets with periodic noisy PCM samples
when receiving no streams, and generates a bit noisy sound.
ALSA BeBoB driver is programmed to establish both in/out connections
when starting streaming, then transfers packets as userspace applications
requested. This means that there's a case that one of incoming/outgoing
streams is running, to save CPU and bandwidth usage. Although, it's natural
to start transferring packets in both direction.
This commit makes this driver to keeps duplex streams always.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Behringer FCA610 and UFX1604 is confirmed to require more time till
transmitting packets after establishing connections. This seems to
be a quirk of DM1500 ASIC which ArchWave produced.
For this quirk, this commit extends the time to wait up to 2 seconds.
As a result, in worst cases, below userspace functions require 2 seconds
to return.
- snd_pcm_prepare()
- snd_pcm_hw_params()
- snd_pcm_recover()
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BeBoB installed devices have BeBoB register area. This area stores
basic information about its firmware. A register has its protocol
version.
This commit adds 'version' member and store the device's protocol
version to handle v3 quirks in following commits.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commits, this driver can detect the source of clock as mush
as possible. SYT-Match mode is also available.
This commit purge the restriction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The old string literals were completely replaced by new normalized
representation.
This commit obsoletes it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit changes function prototype and its processing. As a result,
function caller can execute additional processing according to detected
clock source.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previous commit adds a enumerator as a normalized representation of
clock source, while model-dependent structures still use string literals
for this purpose.
This commit is a preparation for replacement.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previous commit allows this driver to detect several types of clock
source, while there's no normalized expression for it.
This commit adds a new enumerator for this purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With BeBoB version 3, current ALSA BeBoB driver detects the type of
current clock signal source wrongly. This is due to a lack of proper
implementation to parse the information.
This commit renews the parser. As a result, this driver detects
SYT-Match clock signal, thus it can start streams with two modes;
SYT-Match mode and the others. SYT-Match mode will be supported in future
commits.
There's a constrain about detected internal/external clock source.
When detecting external clock source, this driver allows userspace
applications to use current sampling rate only. This is due to consider
abour synchronization to external clock sources such as S/PDIF, ADAT or
word-clock.
According to several information from some devices, I guesss that the
internal clock of most devices synchronize to IEEE 1394 cycle start
packet. In this case, by a usual way, it's detect as 'Sync type
of output Music Sub-Unit' connected to 'Sync type of PCR output Unit
(oPCR)', and this driver judges it as internal clock. Therefore,
userspace applications is allowed to request arbitrary supported sampling
rates.
On the other hand, several devices based on BeBoB version 3 have
additional internal clock. In this case, by a usual way, it's detect as
'Sync/Additional type of External input Unit'. Unfortunately, there's no
way to distinguish this sync type from the other external clock sources
such as word-clock. In this case, this driver handles it as external and
userspace applications is forced to use current sampling rate.
I note that when the source of clock is detected as 'Isochronous stream
type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the
synchronization clock is generated according to SYT-series in received
packets. In this case, this driver generates the series by myself. I
experienced this mode often make the device silent suddenly during
playbacking. This means that the mode is easy to lost synchronization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the missing dependency on PCM stuff.
[Add the same fix for HAL2, too -- tiwai]
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Constify the ACPI device ID array, it doesn't need to be writable at
runtime.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime. Also drop the unneeded RT5670_INIT_REG_LEN
define.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime. Also drop the unneeded RT5640_INIT_REG_LEN
define.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current code has duplicate code for 16000, 32000 and 48000 sample rates.
get_srate() returns negative error code for unsupported rate, so we can
remove the duplicate code in the swith cases by calling get_srate() first.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We still got a report that the audio crackles and noises occur with
the recent 4.1 kernels on Dell machines. These machines seem to need
similar workarounds that have been applied to the recent Dell XPS 13
models. Since the codec of these machines (Dell Latitute E7240 and
E7440) is different from XPS 13's one, we need a new fixup entry.
Also, it was confirmed that the previous workaround to disable the
widget power-save (commit [219f47e4f9: ALSA: hda - Disable widget
power-saving for ALC292 & co]) is no longer needed after this fix.
So, this patch includes the partial revert of the commit, too.
Reported-and-tested-by: Mihai Donțu <mihai.dontu@gmail.com>
Tested-by: Jonathan McDowell <noodles@earth.li>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to make TI button interrupt working max98090 codec
Need provide mic bias all the time as long as mic is present
so SHDN and micbias pin are forced on.we also need set max98090
codec bias close or lower than TI bias.We set them in bios/coreboot
kernel reads them from device property
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some tiny improvements, cutting 180 bytes off the generated code.
- use strchr() for single-character needle
- compute index using pointer subtraction instead of two strlen()
calls
- factor out the common check for whether the initial part of
kctl->id.name (before the space) is identical to w->name.
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On a HP Envy TouchSmart laptop, there are 2 speakers (main speaker
and subwoofer speaker), 1 headphone and 2 DACs, without this fixup,
the headphone will be assigned to a DAC and the 2 speakers will be
assigned to another DAC, this assignment makes the surround-2.1
channels invalid.
To fix it, here using a DAC/pin preference map to bind the main
speaker to 1 DAC and the subwoofer speaker will be assigned to another
DAC.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use kernel.h macro definition.
Thanks to Julia Lawall for Coccinelle scripting support.
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use kernel.h macro definition.
Thanks to Julia Lawall for Coccinelle scripting support.
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dev_attrs field of struct bus_type is going away, use dev_groups instead.
This converts the soundbus code to use the correct field.
These modifications were made using Coccinelle.
Signed-off-by: Quentin Lambert <lambert.quentin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When running in master mode the bclk divider must be configured to generate a
sane bitclock. Pick the smallest fs multiplicator, which can hold all
transmitted bits.
Signed-off-by: Julian Scheel <julian@jusst.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the enum of "DAC Polarity" should be wm8960_enum[1].
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Add mclk-fs ratio property per dai-link sub node. This will
allow to manage several codecs with different ratio.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixed below error/warnings
sound/built-in.o: In function `rt5645_i2c_probe':
>> rt5645.c:(.text+0xe38f5): undefined reference to
>> `devm_regmap_init_i2c'
sound/built-in.o: In function `rt5645_i2c_driver_exit':
>> rt5645.c:(.exit.text+0x60e): undefined reference to `i2c_del_driver'
sound/built-in.o: In function `rt5645_i2c_driver_init':
>> rt5645.c:(.init.text+0x1a90): undefined reference to
>> `i2c_register_driver'
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since hp-detect is optional, use devm_gpiod_get_optional instead.
In additional, it should return error if devm_gpiod_get_optional fails.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This was spotted by Fernando Lopez-Lezcano <nando@ccrma.Stanford.EDU>
while he tried to compile a -RT kernel with this driver enabled.
"make C=2" would also warn about this. This is is based on his patch.
Reported-by: Fernando Lopez-Lezcano <nando@ccrma.Stanford.EDU>
Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is currently possible to have CONFIG_SND_ATMEL_SOC_SSC=y with either
CONFIG_SND_ATMEL_SOC_PDC=m or CONFIG_SND_ATMEL_SOC_DMA=m. This results in a
driver that compiles but does not link with this kind of error:
sound/built-in.o: In function `atmel_ssc_set_audio':
(.text+0x87d90): undefined reference to `atmel_pcm_pdc_platform_register'
sound/built-in.o: In function `atmel_ssc_put_audio':
(.text+0x8879a): undefined reference to `atmel_pcm_pdc_platform_unregister'
Introduce new config options SND_ATMEL_SOC_SSC_PDC and
SND_ATMEL_SOC_SSC_DMA which should be used by the board drivers and the
correct logic to properly select the SND_ATMEL_SOC_PDC and
SND_ATMEL_SOC_DMA states.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After discussing with the Kconfig maintainer, we found a better fiw
allowing to keep each driver as modules.
This reverts commit 0ef9dc139d.
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Modify the RT5645 driver to parse platform data from device tree. This is
missing from previous patch in sound/soc/codecs/rt5645.c, that was present
in v3.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch corrected pcm_delay calculation in BSW sst driver
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch separates the IO function from the rt286. It is prepared to share
for new chips that support the same IO function.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous patch tried to continue the probe if i915 binding fails.
For for simplicity reason, we haven't implemented abort even for
controller chips that are dedicated for HDMI/DP on HSW and BDW.
However, Mengdong suggested that this can be dangerous; BIOS may
disable gfx power well although the PCI entry for HD-audio is left,
and this may result in the unexpected behavior, kernel errors, etc.
For avoiding this situation, abort the probe at i915 binding failure
only for HSW/BDW chips selectively. For other chips, it still
continues.
Fixes: bf06848bdb ('ALSA: hda - Continue probing even if i915 binding fails')
Reported-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
commit ea178d1456 ("ASoC: tas2552: Make the enable-gpio really optional")
makes enable-gpio optional. devm_gpiod_get_optional() is the better
function for optional gpio, so let's switch to use it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds some debugfs nodes to get information
about the currently running firmware.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that we have a codec_probe stage initialization in the wm_adsp
driver, we can make the wm_adsp driver create its own ALSA controls
instead of having that responsibility pushed to every codec driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the only init function in wm_adsp is called by the
codec driver early in its probe before the codec has been
registered with SOC.
This patch adds stubs for the codec_probe and codec_remove stages
and calls them from WM5102 and WM5110 codec drivers. This allows us
to hang anything that needs setup during the codec probe stage off
these functions without further modification of the codec drivers.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have the bus and controller code added to find and initialize
the extended capabilities. Now we need to use them in stream code to
decouple stream, manage links etc
So this patch adds the stream handling code for extended capabilities
introduced in preceding patches
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The controller needs to support the new capabilities and allow
reading, parsing and initializing of these capabilities, so this patch
does it
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new HDA controllers from Intel support new capabilities like
multilink, pipe processing, SPIB, GTS etc In order to use them we
create an extended HDA bus which embed the hdac bus and contains the
fields for extended configurations
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed ctl type for Input Gain Control and Input Gain Pad Control to
USB_MIXER_S16 as per section 5.2.5.7.11-12 in the USB Audio Class 2.0
definition.
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another regression by the transition to regmap cache; for better
usability, we had the fake mute control using the zero amp value for
Conexant codecs, and this was forgotten in the new hda core code.
Since the bits 4-7 are unused for the amp registers (as we follow the
syntax of AMP_GET verb), the bit 4 is now used to indicate the fake
mute. For setting this flag, snd_hda_codec_amp_update() becomes a
function from a simple macro. The bonus is that it gained a proper
function description.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This merges and resolves the non-trivial conflicts with the recent fix
for hda-i915 binding fallback.
Conflicts:
sound/pci/hda/hda_intel.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently snd-hda-intel driver aborts the probing of Intel HD-audio
controller with i915 power well management when binding with i915
driver via hda_i915_init() fails. This is no big problem for Haswell
and Broadwell where the HD-audio controllers are dedicated to
HDMI/DP, thus i915 link is mandatory. However, Skylake, Baytrail and
Braswell have only one controller and both HDMI/DP and analog codecs
share the same bus. Thus, even if HDMI/DP isn't usable, we should
keep the controller working for other codecs.
For fixing this, this patch simply allows continuing the probing even
if hda_i915_init() call fails. This may leave stale sound components
for HDMI/DP devices that are unbound with graphics. We could abort
the probing selectively, but from the code simplicity POV, it's better
to continue in all cases.
Reported-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My static checker complains that:
sound/soc/fsl/imx-wm8962.c:196 imx_wm8962_probe() warn:
we tested 'ret' before and it was 'false'
The intent was that we use "ret" to check imx_audmux_v2_configure_port().
Fixes: 8de2ae2a7f ('ASoC: fsl: add imx-wm8962 machine driver')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Otherwise, Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move gpio to gpio_desc and use gpiod APIs in codec driver.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
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Merge tag 'asoc-v4.2' into asoc-rt5645
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
Register RT5650_TDM_CTRL_4(0x7A) is readable and used for mixer
setting. It should be added in rt5645_readable_register function.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5645_enable_push_button_irq uses snd_soc_dapm_*_unlocked
functions, so it needs to lock the required dapm mutex.
Signed-off-by: Nicolas Boichat <drinkcat@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds apq8016 machine driver support. This patch is tested on
DB410c and msm8916-mtp board for both hdmi and analog audio
features.
Acked-by: Kenneth Westfield <kwestfie@codeaurora.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the missing NULL checks so that snd_hdac_i915*() can be called
even after the binding with i915 failed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Along with the transition to regmap for managing the cached parameter
reads, the caps overwrite was also moved to regmap cache. The cache
change itself works, but it still tries to write the non-existing verb
(the HDA parameter is read-only) wrongly. It's harmless in most
cases, but some chips are picky and may result in the codec
communication stall.
This patch avoids it just by adding the missing flag check in
reg_write ops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Find the configured DMA controller by asking for a DMA channel in the
probe phase and releasing it right after. The controller device can be
found via the dma_chan struct and the controller is recognized from
the compatible property of its device node. The patch assumes EDMA if
there is no device node.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the schematics the external speaker is differential, so the
HPLCOM and HPRCOM pins of the CODEC should be connected to it.
Otherwise the routing looks complete, so add the missing routes and set the
fully_routed flag of the card instead of manually marking the unused inputs
and outputs as not connected. This makes the code a bit shorter and
cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes special casing the EPROBE_DEFER error handling in the
driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch moves static allocation of snd_soc_card to dynamic allocation,
the reason to do this is to avoid holding up any dangling pointers
in this static structures. And I see no use for having this struct as static
given that the card->name is also populated dynamically from dt.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Enable runtime PM of the HDMI audio codec on the latest Intel platforms.
So the HD-A controller or HDMI codec can suspend when idle timeout by
default and release the GFX power well.
The patch influences HSW/BDW/BYT/BSW/SKL. Eariler platforms and third
party analog codecs will not be influenced.
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new regmap code seems to cache this, which isn't helpful
for the hotplug dock situation where this gets updated.
Use the uncached query for this.
Signed-off-by: Dave Airlie <airlied@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
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Merge tag 'asoc-v4.2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
The set of supported sample rates depends on the master clock supplied
to the codec. Allow the machine driver to set the required master clock
in hw_params().
Signed-off-by: Sergej Sawazki <ce3a@gmx.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In tas2552_sw_shutdown() tas_data is used while the rest of the driver uses
tas2552 when dealing with the 'struct tas2552_data'
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Double semicolon was added by the following commit:
ea178d1456 ASoC: tas2552: Make the enable-gpio really optional
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The module can not be loaded again after it has been removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Align the numbers in the header file to the same column.
At the same time change the wrapping of CFG_2 register write in the probe
function to be uniform with the other calls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Correct the bit definition so the code will change the bits what it
supposed to change. Also rename the register define to
TAS2552_BOOST_APT_CTRL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Do not write to DOUT Tristate register at probe time, specially not write
data which is defined to be used in Output Data Register.
Fix the defines for the Output Data Register and correct the register write
at probe time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
'DIN source' enum can be used to select the DIN Source (muted, left, right
or average of left and right channels).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Do not restrict the sampling rate to 44.1/48KHz. The pll_clk clock should
be (sampling rate * 512) in all cases.
Correct the J.D calculation (the D part was incorrectly calculated).
Restore PLL enable status after we are done with the configuration.
Implement hardware constraint handling towards the pll_clkin:
if D != 0 (in J.D) then 1.1MHz <= pll_clkin <= 9.2MHz needs to be checked.
If the PLL setup does not met with this constraint, fall back to BCLK as
reference clock, if BCLK fails, use the internal 1.8MHz clock.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The default state when serializers are in inactive slots is Hi-Z.
In some cases, there are no additional components driving the data
lines to a safe state so they might have noise.
While in inactive slots, the McASP AXR pins configured as outputs
can be driven low through the serializer pin drive mode setting
(DISMOD) to prevent such noise.
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@kernel.org>
This patch adds native DSD support for the XMOS based JLsounds I2SoverUSB board
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of hard wiring the WCLK frequency at probe time do it runtime.
The hard wired 88_96KHz was not even setting the correct bits since it was
defined as (1 << 6) which will change the I2S_OUT_SEL bit and will leave
the amplifier configured for 8KHz.
At the same time clean up and fix the CFG3 register bits.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Configure the word length based on the params_width of the stream.
Also configure the clock per frame value which is used when tas2552 is bus
master.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Certain sequence need to be followed in order to have smooth power up and
power down performance.
Execute this sequence via DAPM_POST widget.
Remove patching the RESERVED_0D register at probe time since it has to be
handled every time when we stop or start the amplifier.
In order to be able to execute the sequence at the correct time, the driver
need to request to ignore the pmdown time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The last parameter for DECLARE_TLV_DB_SCALE() is to tell if the gain will
be muted or not when it is set to raw 0. IN this case it is not muted.
The PGA_GAIN is in 0-4 bits in the register. Fix the offset in the
SOC_SINGLE_TLV() for this.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The strings should be: 'static const char * const tas2552_input_texts[]'
SOC_DAPM_ENUM should have "Route" in place of xname and no need to have it
as an array.
Also align the parameters.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
TDM support is achieved using DSP transfer mode and setting a programmable
offset which specifies where data begins with respect to the frame sync.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use names from the datasheet for the definitions.
Correct the data format definitions since they were not correct.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DSP_A mode require one bit delay from the FS, DSP_B is without data delay.
When checking the requested format, also match the bit and fs inversion
flag along with the format since it is not possible to change inversion.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of hard wiring the PLL_CLKIN and PDM_CLK to be sourced from BCLK add
proper clock configuration via the set_dai_sysclk callback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prevents frequent panic on boot, if the irq handler rt5645_irq
gets called before the workqueue rt5645_jack_detect_work is
initialized.
Signed-off-by: Nicolas Boichat <drinkcat@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
sound/soc/zte/zx296702-i2s.c:428:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
MCLK is one of the possible source for the pll_clkin frequency. Make this
clear by renaming the variable.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The function name and parameters of:
tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown)
implies that if sw_shutdown is 1 we should be entering to the software
shutdown mode.
The code can be simplified as well within the function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Initialize the cfg1_reg to 0 and set the mute bit only when it is needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the _MASK postfix of the bit definitions, collect the CFG1 bit
definition in one place and correct the bit shifts at the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The PDM clock can be selected via bit0-1.
PDM_DATA_ES bit is at bit2.
The code were trying to select BCLK as PDM reference clock but instead
it was selecting PLL and set the DATA_ES bit to 1.
Selecting the PLL output as reference clock as default does make sense,
but the driver should not change the PDM data edge.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SOC_DAPM_SINGLE("Playback AMP", ..) should not be under kcontrols. It
causes kernel crash (NULL pointer) when the mixers are listed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
If the card is not part of any card the tas_data->codec is NULL since it is
set only during snd_soc_codec_driver.probe, which is not yet called.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Do not fail the probe if the enable-gpio is not specifiedbut handle
deferred probe case.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/zte/zx296702-spdif.c:191:1-4: WARNING: end returns can be simpified
Simplify a trivial if-return sequence. Possibly combine with a
preceding function call.
Generated by: scripts/coccinelle/misc/simple_return.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/zte/zx296702-spdif.c:361:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the following error:-
All error/warnings (new ones prefixed by >>):
>
> sound/built-in.o: In function `soc_tplg_dapm_widget_create':
> >> :(.text+0x25a90): undefined reference to `snd_soc_dapm_new_control'
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The topology core parses the FW topology file for known block types and
instanciates any common ALSA/ASoC objects that it discovers. The core
also passes any block that is does not understand to client component
drivers for enumeration.
The core exports some APIs to client drivers in order to load and unload
firmware topology data as use case require.
Currently the core deals with the following object types :-
o kcontrols. This includes TLV, enumerated and bytes controls.
o DAPM widgets. All types with any associated kcontrol.
o DAPM graph.
o FE PCM. FE PCM capabilities and configuration can be defined.
o BE DAI Link. BE DAI link capabilities and configuration can be defined.
o Codec <-> codec style links capabilities and configuration.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DM_INH = 1 (stereo downmix prohibited) and CA = 0x00 (Channel
Allocation: FR, FL) is an invalid combination according to the
HDMI Compliance Test 7.31 "Audio InfoFrame".
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ti.com>
Acked-by: Mark Brown <broonie@kernel.org>
There is a constraint in the OMAP4 HDMI IP that requires to use
the 8-channel code when transmitting more than two channels.
The constraint doesn't apply for OMAP5 so don't force the channel
allocation in the sound driver as it can be done specifically for
OMAP4 later in the hdmi4 core.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ti.com>
Acked-by: Mark Brown <broonie@kernel.org>
Nothing in <asm/io.h> uses anything from <linux/vmalloc.h>, so
remove it from there and fix up the resulting build problems
triggered on x86 {64|32}-bit {def|allmod|allno}configs.
The breakages were triggering in places where x86 builds relied
on vmalloc() facilities but did not include <linux/vmalloc.h>
explicitly and relied on the implicit inclusion via <asm/io.h>.
Also add:
- <linux/init.h> to <linux/io.h>
- <asm/pgtable_types> to <asm/io.h>
... which were two other implicit header file dependencies.
Suggested-by: David Miller <davem@davemloft.net>
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
[ Tidied up the changelog. ]
Acked-by: David Miller <davem@davemloft.net>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Cc: Andrew Morton <akpm@linux-foundation.org>
Cc: Anton Vorontsov <anton@enomsg.org>
Cc: Boris Ostrovsky <boris.ostrovsky@oracle.com>
Cc: Colin Cross <ccross@android.com>
Cc: David Vrabel <david.vrabel@citrix.com>
Cc: H. Peter Anvin <hpa@zytor.com>
Cc: Haiyang Zhang <haiyangz@microsoft.com>
Cc: James E.J. Bottomley <JBottomley@odin.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: K. Y. Srinivasan <kys@microsoft.com>
Cc: Kees Cook <keescook@chromium.org>
Cc: Konrad Rzeszutek Wilk <konrad.wilk@oracle.com>
Cc: Kristen Carlson Accardi <kristen@linux.intel.com>
Cc: Len Brown <lenb@kernel.org>
Cc: Linus Torvalds <torvalds@linux-foundation.org>
Cc: Peter Zijlstra <peterz@infradead.org>
Cc: Rafael J. Wysocki <rjw@rjwysocki.net>
Cc: Suma Ramars <sramars@cisco.com>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: Tony Luck <tony.luck@intel.com>
Signed-off-by: Ingo Molnar <mingo@kernel.org>
HDA codec drivers can be matched using vendor id and revision id typically.
So provide a match function which does this and is loaded when driver hasn't
provided one (default behaviour)
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver worked around an error in the MAYA44 USB(+)'s mixer unit
descriptor by aborting before parsing the missing field. However,
aborting parsing too early prevented parsing of the other units
connected to this unit, so the capture mixer controls would be missing.
Fix this by moving the check for this descriptor error after the parsing
of the unit's input pins.
Reported-by: nightmixes <nightmixes@gmail.com>
Tested-by: nightmixes <nightmixes@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer control names for the ESI Maya44 USB+ (which appears to be
identical width the AudioTrak Maya44 USB).
Reported-by: nightmixes <nightmixes@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For SKL, only the HDMI codec is in the display power well while the
HD-A controller isn't. So the controller flag 'need_i915_power' is
not set to release the display power after probe, and the codec flag
'link_power_control' is set to request/release the display power via
bus link_power ops.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If an ASoC component device does not have a device tree node, use its
parent's node instead, when looking for a matching DAI based on a
device tree reference.
This allows video device drivers to register a separate child device
for their ASoC side audio functionality. [And MFDs in general --
broonie]
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The WM8997 and WM5102 codecs need to boost DVFS for higher sample rates.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In theory the ADSP driver should not need to know anything
about the codec it is part of. But the WM5102 needs DVFS
control based on ADSP clocking speed. This was being handled
by bundling part of the knowledge of this into the ADSP driver.
This change moves this handling out of the ADSP driver and
into the WM5102 driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The WM5102 and WM8997 codecs have an internal dynamic clock booster.
When this booster is active, the DCVDD voltage must be increased.
If all the currently active audio paths can run with the root SYSCLK
we can disable the booster, allowing us to turn down DCVDD voltage
to save power.
Previously this was being done by having the booster enable bit set
as a side-effect of the LDO1 regulator driver, which is unexpected
behaviour of a regulator and not compatible with using an external
regulator. [Originally this was documented as a feature of the internal
LDO -- broonie]
This patch exports functions to handle the booster enable and
DCVDD voltage, with each relevant subsystem flagging whether it can
currently run without the booster. Note that these subsystems are
stateless and none of them are nestable, so there's no need for
reference counting, we only need a simple boolean for each subsystem
of whether their current condition could require the booster or will
allow us to turn the codec down to lower operating power.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch is a fixup to correct dependencies in patch 9bae4880ac
("ASoC: qcom: move ipq806x specific bits out of lpass driver.")
Originally this change-set was suggested by Arnd on mailing list.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Acer Aspire 9420 with ALC883 (1025:0107) needs the fixup for EAPD to
make the sound working like other Aspire models.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94111
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The change fixes the following compilation problem:
sound/soc/soc-dapm.c: In function 'dapm_kcontrol_data_alloc':
sound/soc/soc-dapm.c:388:4: error: implicit declaration of function
'snd_soc_dapm_new_control' [-Werror=implicit-function-declaration]
data->widget = snd_soc_dapm_new_control(widget->dapm,
^
sound/soc/soc-dapm.c:387:17: warning: assignment makes pointer
from integer without a cast [enabled by default]
data->widget = snd_soc_dapm_new_control(widget->dapm,
^
sound/soc/soc-dapm.c: At top level:
sound/soc/soc-dapm.c:3269:1: error: conflicting types for
'snd_soc_dapm_new_control'
snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
^
In addition to the fix add static qualifier to
snd_soc_dapm_new_control() function to silence checkpatch.
Fixes: 02aa78abec ("ASoC: DAPM: Add APIs to create individual DAPM controls.")
Signed-off-by: Vladimir Zapolskiy <vz@mleia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
replace of_property_read_u32 with device_property_read_u32
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It needs free pcm runtime modules before unloading firmware, here
add hsw_pcm_suspend() to handle this procedure:
suspends firmware ==> frees runtime modules ==> unloads firmware.
This fixes the broadwell module unload failed issue.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add check runtime module pointers before freeing them, and clear
them to NULL after freed.
With this implemented, we can avoid NULL pointer dereference or
double free errors.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 01f202c7b4.
We shouldn't leave the device as suspended state after module freed,
it is not good to do runtime suspend at driver free, here revert
this fixing, and replace it with the procedure:
suspends firmware ==> frees runtime modules ==> unloads firmware.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 506c148ee5.
We still need this hsw_pcm_free_modules(), we plan to remove the
runtime modules at both fw_unload(D0->D3) and snd_soc_sst_haswell_pcm
module removing.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't need compress offload feature for broadwell broadwell machine,
here remove the non exist dependency.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
