Current rsnd driver is using "%s[%d]" for mod name and ID,
but, this ID portion might confusable.
For example currently, CTU ID is 0 to 7, but using 00 to 13
(= 00, 01, 02, 03, 10, 11, 12, 13) is very best matching to datasheet.
In the future, we will support BUSIFn, but it will be more complicated
numbering. To avoid future confusable code, this patch modify
rsnd_mod_name() to return understandable name.
To avoid using pointless memory, it uses static char and snprintf,
thus, rsnd_mod_name() user should use it immediately, and shouldn't keep
its pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ID for CTU and SSIU are confusable.
1 CTU has 4 sub nodes. This means, CTU0 has CTU01 - CTU03, CTU1 has
CTU10 - CTU13. SSIU is more confusable. Gen2 SSIU has BUSIF0-3, Gen3
SSIU has BUSIF0-7, but not for all SSIU.
In rsnd driver, each mod drivers are assuming rsnd_mod_id() returns
main device ID (In CTU case CTU0-1, SSIU case SSIU0-9), not serial
number.
This patch adds new .id/.id_sub to handling more detail ID.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each mod needs to have .get_status, but current driver is handling it
under rsnd_mod, instead of rsnd_mod_ops.
It is not any make sence. This patch moves it to rsnd_mod_ops, and
tidyup its parameter order to align to other callback functions.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
hw constraint for LEFT_J/I2S are same, but had duplicated code.
This code cleanup these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For samples sizes of 20/24/32 bit the DRR2/DXR2 registers must be used to
receive or transmit the most significant part of the 32bit sample.
We can not trick the system by using the DRR2/DXR2 register offset and use
32bit element size since these (and other McBSP) registers are 16bit ones.
We would need support for port_window in cyclic mode from the sDMA driver,
but it is not straight forward as we would need to have the sDMA frame to
cover the 32bit (DRR2+DRR1/DXR2+DXR1), but the frames must cover the ALSA
period to be able to receive periodic interrupts.
Since the 32bit samples are not working (DMA timeout), just remove it in
case we have McBSP with reg_size of 2.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The regulator_desc structure can be const as it is only passed as the
second argument of devm_regulator_register and the corresponding
parameter is declared as const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_dai_ops structures are only stored in the ops field of a
snd_soc_dai_driver structure, so make the snd_soc_dai_ops structures
const as well.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_dai_ops structures are only stored in the ops field of a
snd_soc_dai_driver structure, so make the snd_soc_dai_ops structures
const as well.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
WARN_ON() already contains an unlikely(), so it's not necessary to use
unlikely.
Signed-off-by: Yangtao Li <tiny.windzz@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Randconfig testing revealed a very old bug, with gcc-8:
sound/soc/intel/atom/sst/sst_loader.c: In function 'sst_load_fw':
sound/soc/intel/atom/sst/sst_loader.c:357:5: error: 'fw' may be used uninitialized in this function [-Werror=maybe-uninitialized]
if (fw == NULL) {
^
sound/soc/intel/atom/sst/sst_loader.c:354:25: note: 'fw' was declared here
const struct firmware *fw;
We must check the return code of request_firmware() before we look at the
pointer result that may be uninitialized when the function fails.
Fixes: 9012c9544e ("ASoC: Intel: mrfld - Add DSP load and management")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
gcc notices that without either the ac97 bus or the pdata, we never
initialize the regmap pointer, which leads to an uninitialized variable
access:
sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe':
sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized]
Since that configuration is invalid, it's better to return an error
here. I tried to avoid adding complexity to the conditions, and turned
the #ifdef into a regular if(IS_ENABLED()) check for readability.
This in turn requires moving some header file declarations out of
an #ifdef.
The same code is used in three drivers, all of which I'm changing
the same way.
Fixes: 2ed1a8e0ce ("ASoC: wm9712: add ac97 new bus support")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_ops structure can be const as it is only stored in the
ops field of a snd_soc_dai_link structure and this field is const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, the comparison of div < 0 is always false because div is
an unsigned int. Fix this by making div an int.
Detected by CoverityScan, CID#1475309 ("Unsigned compared against 0")
Fixes: 8307b2afd3 "(ASoC: stm32: sai: set sai as mclk clock provider")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SUN50I_CODEC_ANALOG selects SND_SUNXI_ADDA_PR_REGMAP which is leftover
of renaming SND_SUNXI_ADDA_PR_REGMAP to SND_SUN8I_ADDA_PR_REGMAP. Replace
it with SND_SUN8I_ADDA_PR_REGMAP to fix possible link errors for some
configurations:
sound/soc/sunxi/sun50i-codec-analog.o: In function `sun50i_codec_analog_probe':
sun50i-codec-analog.c:(.text+0x62): undefined reference to `sun8i_adda_pr_regmap_init'
Fixes: 42371f327d ("ASoC: sunxi: Add new driver for Allwinner A64 codec's analog path controls")
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Reviewed-by: Andre Przywara <andre.przywara@arm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A few device-specific fixes: a fix for SPDIF on old Creative PCI
board, and two additional fixes for the recent changes in FireWire
audio stack.
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Merge tag 'sound-fix-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few device-specific fixes: a fix for SPDIF on old Creative PCI
board, and two additional fixes for the recent changes in FireWire
audio stack"
* tag 'sound-fix-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size
ALSA: ca0106: Disable IZD on SB0570 DAC to fix audio pops
ALSA: dice: fix to wait for releases of all ALSA character devices
Ever since commit 46e831abe8 ("drm/i915/lpe: Mark LPE audio runtime
pm as "no callbacks"") the runtime suspend/resume hooks are no longer
used. Inline them into the system suspend hooks.
Cc: Chris Wilson <chris@chris-wilson.co.uk>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Ville Syrjälä <ville.syrjala@linux.intel.com>
Link: https://patchwork.freedesktop.org/patch/msgid/20181024154825.18185-2-ville.syrjala@linux.intel.com
Reviewed-by: Chris Wilson <chris@chris-wilson.co.uk>
Acked-by: Takashi Iwai <tiwai@suse.de>
Commit 46e831abe8 ("drm/i915/lpe: Mark LPE audio runtime pm as
"no callbacks"") broke runtime PM with lpe audio. We can no longer
runtime suspend the GPU since the sysfs power/control for the
lpe-audio device no longer exists and the device is considered
always active. We can fix this by not marking the device as
active.
Cc: Chris Wilson <chris@chris-wilson.co.uk>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Fixes: 46e831abe8 ("drm/i915/lpe: Mark LPE audio runtime pm as "no callbacks"")
Signed-off-by: Ville Syrjälä <ville.syrjala@linux.intel.com>
Link: https://patchwork.freedesktop.org/patch/msgid/20181024154825.18185-1-ville.syrjala@linux.intel.com
Reviewed-by: Chris Wilson <chris@chris-wilson.co.uk>
Acked-by: Takashi Iwai <tiwai@suse.de>
commit 4d230d1271 ("ASoC: rsnd: fixup not to call clk_get/set under
non-atomic") fixuped clock start timing. But it exchanged clock start
checker from ssi->usrcnt to ssi->rate.
Current rsnd_ssi_master_clk_start() is called from .prepare,
but some player (for example GStreamer) might calls it many times.
In such case, the checker might returns error even though it was not
error. It should check ssi->usrcnt instead of ssi->rate.
This patch fixup it. Without this patch, GStreamer can't switch
48kHz / 44.1kHz.
Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In a former commit, PCM constraint based on LCM of SYT_INTERVAL was
obsoleted with PCM rule. However, the new PCM rule brings -EINVAL in
some cases that max/min values of size of buffer/period is not
multiples of one of values of SYT_INTERVAL. For example, pulseaudio
always fail to configure PCM substream.
This commit changes strategy for the PCM rule. Although the buggy rules
had a single dependency (rate from period, period from rate, rate from
buffer, buffer from rate), a revised rule has double dependencies
(period from period/rate, buffer from buffer/rate). A step of value is
calculated with table of SYT_INTERVAL and list of available rates. This
prevents interval template which brings -EINVAL to a call of
snd_interval_refine().
Fixes: 5950229582bc('ALSA: firewire-lib: add PCM rules to obsolete PCM constraints based on LCM of SYT_INTERVAL')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Creative Audigy SE (SB0570) card currently exhibits an audible pop
whenever playback is stopped or resumed, or during silent periods of an
audio stream. Initialise the IZD bit to the 0 to eliminate these pops.
The Infinite Zero Detection (IZD) feature on the DAC causes the output
to be shunted to Vcap after 2048 samples of silence. This discharges the
AC coupling capacitor through the output and causes the aforementioned
pop/click noise.
The behaviour of the IZD bit is described on page 15 of the WM8768GEDS
datasheet: "With IZD=1, applying MUTE for 1024 consecutive input samples
will cause all outputs to be connected directly to VCAP. This also
happens if 2048 consecutive zero input samples are applied to all 6
channels, and IZD=0. It will be removed as soon as any channel receives
a non-zero input". I believe the second sentence might be referring to
IZD=1 instead of IZD=0 given the observed behaviour of the card.
This change should make the DAC initialisation consistent with
Creative's Windows driver, as this popping persists when initialising
the card in Linux and soft rebooting into Windows, but is not present on
a cold boot to Windows.
Signed-off-by: Alex Stanoev <alex@astanoev.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a development period for Linux kernel v4.20, drivers in ALSA firewire
stack were changed to wait for releases of all ALSA character devices at
.remove callback of bus driver. However, ALSA dice driver is partly out
of this change. This bug can bring fault to user process which holds
the last of character device in unplugging.
This commit fixes the driver to wait in the callback.
Fixes: 61ccc6f6b2 ('ALSA: firewire: block .remove callback of bus driver till all of ALSA character devices are released')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There have been little changes in ALSA core stuff, but ASoC core still
kept rolling for the continued restructuring. The rest are lots of
small driver-specific changes and some minor API updates.
Here are highlights:
General:
- Appropriate fall-through annotations everywhere
- Some code cleanup in memalloc code, handling non-cacahed pages more
commonly in the helper
- Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently
Drivers:
- More HD-audio CA0132 codec improvement for supporting other Creative
boards
- Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
give move support of existing HD-audio devices with DSP
- A few device-specific HD-audio quirks as usual
- New quirk for RME CC devices and correction for B&W PX for USB-audio
- FireWire: code refactoring including devres usages
ASoC Core:
- Continued componentization works; it's almost done!
- A bunch of new for_each_foo macros
- Cleanups and fixes in DAPM code
ASoC Drivers:
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060
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Merge tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There have been little changes in ALSA core stuff, but ASoC core still
kept rolling for the continued restructuring. The rest are lots of
small driver-specific changes and some minor API updates. Here are
highlights:
General:
- Appropriate fall-through annotations everywhere
- Some code cleanup in memalloc code, handling non-cacahed pages more
commonly in the helper
- Deployment of SNDRV_PCM_INFO_SYNC_APPLPTR flag consistently
Drivers:
- More HD-audio CA0132 codec improvement for supporting other Creative
boards
- Plumbing legacy HD-audio codecs as ASoC BE on Intel SST; this will
give move support of existing HD-audio devices with DSP
- A few device-specific HD-audio quirks as usual
- New quirk for RME CC devices and correction for B&W PX for USB-audio
- FireWire: code refactoring including devres usages
ASoC Core:
- Continued componentization works; it's almost done!
- A bunch of new for_each_foo macros
- Cleanups and fixes in DAPM code
ASoC Drivers:
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060"
* tag 'sound-4.20-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (299 commits)
ASoC: stm32: sai: fix master clock naming
ASoC: stm32: add clock dependency for sai
ALSA: hda/ca0132 - Actually fix microphone issue
ASoC: sun4i-i2s: move code from startup/shutdown hooks into pm_runtime hooks
ASoC: wm2000: Remove wm2000_read helper function
ASoC: cs42l51: fix mclk support
ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate
ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
ASoC: cs42l51: add mclk support
ASoC: stm32: sai: set sai as mclk clock provider
ASoC: dt-bindings: add mclk support to cs42l51
ASoC: dt-bindings: add mclk provider support to stm32 sai
ASoC: soc-core: fix trivial checkpatch issues
ASoC: dapm: Add support for hw_free on CODEC to CODEC links
ASoC: Intel: kbl_da7219_max98927: minor white space clean up
ALSA: i2c/cs8427: Fix int to char conversion
ALSA: doc: Brush up the old writing-an-alsa-driver
ASoC: rsnd: tidyup SSICR::SWSP for TDM
ASoC: rsnd: enable TDM settings for SSI parent
ASoC: pcm3168a: add hw constraint for capture channel
...
sound/soc/stm/stm32_sai_sub.c:393:26-32: ERROR: application of sizeof to pointer
sizeof when applied to a pointer typed expression gives the size of
the pointer
Generated by: scripts/coccinelle/misc/noderef.cocci
Fixes: 8307b2afd3 ("ASoC: stm32: sai: set sai as mclk clock provider")
CC: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A couple of fixes for build bot issues in the STM32 SAI driver.
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Merge tag 'asoc-v5.0-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Additional fixes for the next release
A couple of fixes for build bot issues in the STM32 SAI driver.
Pull perf updates from Ingo Molnar:
"The main updates in this cycle were:
- Lots of perf tooling changes too voluminous to list (big perf trace
and perf stat improvements, lots of libtraceevent reorganization,
etc.), so I'll list the authors and refer to the changelog for
details:
Benjamin Peterson, Jérémie Galarneau, Kim Phillips, Peter
Zijlstra, Ravi Bangoria, Sangwon Hong, Sean V Kelley, Steven
Rostedt, Thomas Gleixner, Ding Xiang, Eduardo Habkost, Thomas
Richter, Andi Kleen, Sanskriti Sharma, Adrian Hunter, Tzvetomir
Stoyanov, Arnaldo Carvalho de Melo, Jiri Olsa.
... with the bulk of the changes written by Jiri Olsa, Tzvetomir
Stoyanov and Arnaldo Carvalho de Melo.
- Continued intel_rdt work with a focus on playing well with perf
events. This also imported some non-perf RDT work due to
dependencies. (Reinette Chatre)
- Implement counter freezing for Arch Perfmon v4 (Skylake and newer).
This allows to speed up the PMI handler by avoiding unnecessary MSR
writes and make it more accurate. (Andi Kleen)
- kprobes cleanups and simplification (Masami Hiramatsu)
- Intel Goldmont PMU updates (Kan Liang)
- ... plus misc other fixes and updates"
* 'perf-core-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip: (155 commits)
kprobes/x86: Use preempt_enable() in optimized_callback()
x86/intel_rdt: Prevent pseudo-locking from using stale pointers
kprobes, x86/ptrace.h: Make regs_get_kernel_stack_nth() not fault on bad stack
perf/x86/intel: Export mem events only if there's PEBS support
x86/cpu: Drop pointless static qualifier in punit_dev_state_show()
x86/intel_rdt: Fix initial allocation to consider CDP
x86/intel_rdt: CBM overlap should also check for overlap with CDP peer
x86/intel_rdt: Introduce utility to obtain CDP peer
tools lib traceevent, perf tools: Move struct tep_handler definition in a local header file
tools lib traceevent: Separate out tep_strerror() for strerror_r() issues
perf python: More portable way to make CFLAGS work with clang
perf python: Make clang_has_option() work on Python 3
perf tools: Free temporary 'sys' string in read_event_files()
perf tools: Avoid double free in read_event_file()
perf tools: Free 'printk' string in parse_ftrace_printk()
perf tools: Cleanup trace-event-info 'tdata' leak
perf strbuf: Match va_{add,copy} with va_end
perf test: S390 does not support watchpoints in test 22
perf auxtrace: Include missing asm/bitsperlong.h to get BITS_PER_LONG
tools include: Adopt linux/bits.h
...
Fixes: 8307b2afd3 ("ASoC: stm32: sai: set sai as mclk clock provider")
Fix warning issued by strncat when bound equals to source length.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes: 8307b2afd3 ("ASoC: stm32: sai: set sai as mclk clock provider")
Add COMMON_CLK dependency for STM32 SAI,
as it is required by clock provider.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As ever there's a lot of small and driver specific changes going on
here, but we do also have some relatively large changes in the core
thanks to the hard work of Charles and Morimoto-san:
- More component transitions from Morimoto-san, I think we're about
finished with this. Thanks for all the hard work!
- Morimoto-san also added a bunch of for_each_foo macros
- A bunch of cleanups and fixes for DAPM from Charles.
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373.
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060.
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Merge tag 'asoc-v5.0' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.0/v4.20
As ever there's a lot of small and driver specific changes going on
here, but we do also have some relatively large changes in the core
thanks to the hard work of Charles and Morimoto-san:
- More component transitions from Morimoto-san, I think we're about
finished with this. Thanks for all the hard work!
- Morimoto-san also added a bunch of for_each_foo macros
- A bunch of cleanups and fixes for DAPM from Charles.
- MCLK support for several different devices, including CS42L51, STM32
SAI, and MAX98373.
- Support for Allwinner A64 CODEC analog, Intel boards with DA7219 and
MAX98927, Meson AXG PDM inputs, Nuvoton NAU8822, Renesas R8A7744 and
TI PCM3060.
This patch fixes the microphone issue for all cards. The previous fix
worked on the ZxR, but not on the AE-5 or Z. This patch has been tested
to work for all cards.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
startup() and shutdown() hooks are called for both substreams,
so stopping either substream when another is running breaks the
latter.
E.g. playback breaks if capture is stopped when playback is running.
Move code from startup() and shutdown() to resume() and suspend()
hooks respectively to fix this issue
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The return type "unsigned int" was used by the wm2000_read()
function despite of the aspect that it will eventually return
a negative error code.
The resulting function doesn't add much to the code, so replace
wm2000_read with regmap_read.
Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The MCLK clock is made optional for cs42l51 codec.
However, ASoC DAPM clock supply widget, expects the clock to be defined
unconditionally.
Register MCLK DAPM conditionally in codec driver,
depending on clock presence in DT.
Fixes: 5e8d63a726 ("ASoC: cs42l51: add mclk support")
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Increase the address value width in the debug log from 4 digits to
8 digits to allow for DSP cores with larger memory address ranges.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The wm_adsp_buffer struct is the control header of a circular
buffer used to transfer data from the firmware over the
control interface to an ALSA compressed stream.
The original names of the fields pointing to the data buffer
were based on ADSP2V2 memory layout where they correspond to
{XM, XM, YM}. But this circular buffer could be used on other
types of DSP core that have different memory region types.
Also the names and description of the size fields were not
very clear. The field names and descriptions have been changed
to be generic and not imply any particular memory types.
This patch updates the wm_adsp driver to the new field names.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add MCLK dapm to allow configuration of cirrus CS42l51
codec as a master clock consumer.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add master clock generation support in STM32 SAI.
The master clock provided by SAI can be used to feed a codec.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix a few trivial aka cosmetic only checkpatch issues like long lines,
wrong indentations, spurious blanks and newlines, missing newlines,
multi-line comments etc.
Signed-off-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, on power down for a CODEC to CODEC DAI link we only call
digital_mute and shutdown. Provide a little more flexibility for drivers
by adding a call to hw_free as well.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Compiling with clang yields the following warning:
sound/i2c/cs8427.c:140:31: warning: implicit conversion from 'int'
to 'char' changes value from 160 to -96 [-Wconstant-conversion]
data[0] = CS8427_REG_AUTOINC | CS8427_REG_CORU_DATABUF;
~ ~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~
Because CS8427_REG_AUTOINC is defined as 128, it is too big for a
char field.
So change data from char to unsigned char, that it can hold the value.
This patch does not change the generated code.
Signed-off-by: Philipp Klocke <philipp97kl@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
R-Car datasheet is indicating that WS output settings of SSICR::SWSP
is inverted on TDM mode from non TDM mode settings.
But, it is meaning that TDM should use 0 here.
Without this patch, sound input/output 1ch will be 2ch, 2ch will be 3ch
..., be jumbled on I2S + TDM settings. This patch fixup it.
This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 TDM sound.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some SSIs are sharing each pins (= WS/CLK pin for playback/capture).
Then, SSI parent needs control WS/CLK setting for SSI slave.
In such case, SSI parent needs TDM settings if SSI slave is working as
TDM mode. But it is not cared in current driver.
It can't capture TDM sound without this patch if SSIs were pin sharing.
This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
LEFT_J / I2S only can use TDM.
commit 594680ea4a ("ASoC: pcm3168a: add hw constraint for channel")
commit 3809688980 ("ASoC: pcm3168a: add HW constraint for non
RIGHT_J") added channel constraint for it, but, it was only for playback.
This patch adds constraint for capture.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The STA32x chips feature an XTI clock input that needs to be stable before
the reset signal is released. Therefore, the chip driver needs to get a
handle to the clock. Instead of relying on other parts of the system to
enable the clock, let the codec driver grab a handle itself.
In order to keep existing boards working, clock support is made optional.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Similar to the following:
commit 4321723648 ("ASoC: tegra_alc5632: fix device_node refcounting")
commit 7c5dfd5496 ("ASoC: tegra: fix device_node refcounting")
Signed-off-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Acked-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allwinner A64 uses the same digital codec part as in A33, so we need
to build this driver on ARM64 as well.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The internal codec on A64 is split into 2 parts. The analog path controls
are routed through an embedded custom register bus accessed through
the PRCM block.
Add an ASoC component driver for it. This should be tied to the codec
audio card as an auxiliary device.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It will be reused by sun50i-codec-analog later.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
BCLK / LRCK ratio should be sample size * channels, but it was
hardcoded to 32 (0x1 is 32 as per A33 and A64 datasheets).
Calculate it basing on sample size and number of channels.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The I2S block used for the audio codec in the A64 differs from other 3
I2S modules in A64 and isn't compatible with H3. But it is very similar
to what is found in A10(sun4i). However, its TX FIFO is
located at a different address.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DMA API does its own zone decisions based on the coherent_dma_mask.
Signed-off-by: Christoph Hellwig <hch@lst.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Power-saving is causing plops on audio start/stop on Dell Precision T3600
laptops and Intel DZ77BH boards, add these to the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds da7219_max98927 machine driver entry into
machine table
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds Kabylake I2S machine driver with:
DA7219 audio codec(SSP1) and MAXIM98927(SSP0) speaker amplifier.
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some 3 pole connectors report impedance greater than threshold of
1000Ohm. Thus, da7219 reports them as LINEOUT.
Adding the SND_JACK_LINEOUT type so that we don't fail to detect
any 3 pole jack type.
Also, changing
SND_JACK_HEADPHONE | SND_JACK_MICROPHONE -> SND_JACK_HEADSET
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DMA API does its own zone decisions based on the coherent_dma_mask.
[ Note: as the driver doesn't set the DMA coherent mask, we can assume
the default 32bit DMA, hence it should be safe to drop the flag here
-- tiwai ]
Signed-off-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pcm3168a is supporting TDM on I2S/Left_J, but there is no
settings for it. This patch add it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
LEFT_J / I2S only can use TDM.
This patch adds channel constraint for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
RIGHT_J only can handle 16bit data bits.
Current driver just errored if user requests non RIGHT_J
+ 16bit combination. But it is not useful for user.
This patch adds HW constraint for it, and avoid
error on such situation.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ESD watchdog code in sta32x_watchdog() dereferences the pointer
which is never assigned.
This is a regression from a1be4cead9 ("ASoC: sta32x: Convert to direct
regmap API usage.") which went unnoticed since nobody seems to use that ESD
workaround.
Fixes: a1be4cead9 ("ASoC: sta32x: Convert to direct regmap API usage.")
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, add the "fall through" annotation in au88xx driver.
Although the usages in both both functions vortex_adbdma_setbuffers()
and vortex_wtdma_setbuffers() look a bit suspicious, we keep the
behavior as before, just to be safer. If it were broken, we should
have already received bug reports.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, replace with the standard "fall through" annotation
at the right places. They have to be put right before the next
labels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, replace with the standard "fall through" annotation.
Unfortunately gcc doesn't understand a chattier text.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, add the "fall through" annotation in caiaq driver.
Note that this seems necessary to be put exactly before the next
label, so it's outside the ifdef block.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, add the "fall through" annotation in
snd_dma_alloc_pages(). Note that this seems necessary to be put
exactly before the next label, so it's outside the ifdef block.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some error paths, reference count of firewire unit is not decreased.
This commit fixes the bug.
Fixes: 5b14ec25a79b('ALSA: firewire: release reference count of firewire unit in .remove callback of bus driver')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo G50-30, like other G50 models, has a Conexant codec that
requires a quirk for its inverted stereo dmic.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1249364
Reported-by: Alexander Ploumistos <alex.ploumistos@gmail.com>
Tested-by: Alexander Ploumistos <alex.ploumistos@gmail.com>
Cc: stable@vger.kernel.org
Signed-off-by: Jeremy Cline <jcline@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current audio-graph-card is supporting mclk-fs on CPU node
side only. But having Codec node also is good idea.
It will be just ignored if not defined.
"rcpu_ep" is same as "cpu_ep", This patch tidyup it, too.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit fb2815f44a ("ASoC: rsnd: add support for 16/24 bit slot widths")
added TDM width check, and return error if it was not 16/24/32 bit.
But it is too strict. This patch uses 32bit same as default.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
max98988 only builds with I2C support enabled, otherwise we get a build error:
sound/soc/codecs/max98088.c:1789:1: error: data definition has no type or storage class [-Werror]
module_i2c_driver(max98088_i2c_driver);
^~~~~~~~~~~~~~~~~
sound/soc/codecs/max98088.c:1789:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int]
sound/soc/codecs/max98088.c:1789:1: error: parameter names (without types) in function declaration [-Werror]
sound/soc/codecs/max98088.c:1780:26: error: 'max98088_i2c_driver' defined but not used [-Werror=unused-variable]
Fixes: 24ae67c582 ("ASoC: max98988: make it selectable")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
In former commits, .private_free callback releases resources just for
data transmission. This release function can be called without the
resources are actually allocated in error paths.
This commit applies a small refactoring to clean up codes in error
paths.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In drivers of ALSA firewire stack, bebob and fireworks drivers have
local device entry table. At present, critical section to operate the
table is from the beginning/end of 'do_registration' call. This can be
more narrow and simplify codes.
This commit applies small refactoring for the above purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a previous commit, drivers in ALSA firewire stack blocks .remove
callback of bus driver. This enables to release members of private
data in the callback after releasing device of sound card.
This commit simplifies codes to release the members.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At present, in .remove callback of bus driver just decrease reference
count of device for ALSA card instance. This delegates release of the
device to a process in which the last of ALSA character device is
released.
On the other hand, the other drivers such as for devices on PCIe are
programmed to block .remove callback of bus driver till all of ALSA
character devices are released.
For consistency of behaviour for whole drivers, this probably confuses
users. This commit takes drivers in ALSA firewire stack to imitate the
above behaviour.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The front MIC on the Lenovo M715 can't record sound, after applying
the ALC294_FIXUP_LENOVO_MIC_LOCATION, the problem is fixed. So add
the pin configuration of this machine to the pin quirk table.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on ASUS G751 doesn't seem to map the headphone pin (NID 0x16)
correctly. Add a quirk to address it, as well as chaining to the
previous fix for the microphone.
Reported-by: Håvard <hovardslill@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes the echo cancellation control for desktop cards, and
makes use of the special 0x47 SCP command for noise reduction.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch cleans up the patch_ca0132() function with suggestions from
Takashi Sakamoto.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes microphone inconsistency issues by adding a delay to
each setup_defaults function. Without this, the microphone only works
intermittently.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the driver will build only if SND_SOC_ALL_CODECS is set.
Adding a Kconfig menu description to build the driver standalone.
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Cc: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Andreas Färber <afaerber@suse.de>
Acked-by: Tushar Behera <trblinux@gmail.com>
Reviewed-by: Javier Martinez Canillas <javier.martinez@collabora.co.uk>
[m.felsch@pengutronix.de: move mclk request to i2c_probe]
[m.felsch@pengutronix.de: make use of snd_soc_component_get_bias_level()]
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Lookup regulators for Vdd and Vdda during probe, and enable them when the
component is linked.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a corresponding model list entry for ASUS G751 so that user can
test the quirk for another compatible machines more easily.
Reported-and-tested-by: Håvard <hovardslill@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS G751 requires the extra COEF initialization to make it microphone
working properly.
Reported-and-tested-by: Håvard <hovardslill@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, replace with the standard "fall through" annotation.
gcc can't understand the mixed texts, unfortunately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, add the "fall through" annotation in Intel SST
skylake driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, add the "fall through" annotations in rt274 driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, replace with the standard "fall through" annotation.
Unfortunately gcc doesn't understand the mixed comment lines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
As a preparatory patch for the upcoming -Wimplicit-fallthrough
compiler checks, replace with the standard "fall through" annotation
at the right place. It has to be put right before the next label.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds several vendor specific mixer quirks for RME's Class Compliant
USB devices. These provide extra status information from the device
otherwise not available.
These include AES/SPDIF rate and status information, current system
sampling rate and measured frequency. This information is especially
useful in cases where device's clock is slaved to external clock
source.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A quirk in snd-usb-audio was added to automate setting sample rate to
4800k and remove the previously exposed nonfunctional microphone for
the Bowers & Wilkins PX:
commit 240a8af929https://lore.kernel.org/patchwork/patch/919689/
However the headphones where updated shortly after that to remove the
unintentional microphone functionality. I guess because of this the
headphones now crash when connecting them via USB while the quirk is
active. Dmesg:
snd-usb-audio: probe of 2-3:1.0 failed with error -22
usb 2-3: 2:1: cannot get min/max values for control 2 (id 2)
This patch removes the microfone and allows the headphones to connect
and work out of the box. It is based on the current mainline kernel
and successfully applied an tested on my machine (4.18.10.arch1-1).
Fixes: 240a8af929 ("ALSA: usb-audio: Add a quirck for B&W PX headphones")
Signed-off-by: Nicolas Huaman <nicolas@herochao.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Elo VuPoint 15MX has two headphone jacks of which neither work by
default. Disabling automute allows ALSA to work normally with the
speakers & left headphone jack.
Future pin configuration changes may be required in the future to get
the right headphone jack working in tandem.
Signed-off-by: Michael Pobega <mpobega@neverware.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For discarding the pending bytes on rawmidi, we process with a loop of
snd_rawmidi_transmit() which is just a waste of CPU power.
Implement a lightweight API function to discard the pending bytes and
the proceed the ring buffer instantly, and use it instead of open
codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver already has support for setting the FDRB bit in the CONFA
register through platform data, but there was no property to set it
in the device-tree bindings.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a "extclk" clock is given, enable and disable it when appropriate.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The q6asm_audio_client_alloc() doesn't return NULL, it returns error
pointers.
Fixes: 2a9e92d371 ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
skl_tplg_get_token() misses a break in the big switch() block for
SKL_TKN_U8_CORE_ID entry.
Spotted nicely by -Wimplicit-fallthrough compiler option.
Fixes: 6277e83292 ("ASoC: Intel: Skylake: Parse vendor tokens to build module data")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA oxfw driver allocates memory objects for cache of stream formats.
The objects are used to maintain packet streaming by components for
ALSA rawMIDI/PCM interface. They can be released as managed-resource
of 'struct snd_card.card_dev'.
This commit uses managed-resource of the sound card device for this
purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA oxfw driver allocates memory objects for data specific to some
models. These objects are used to maintain functionalities specific
to the models for ALSA rawMIDI/control interfaces. They can be
released as managed-resource of 'struct snd_card.card_dev'.
This commit uses managed-resource of the sound card device for this
purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA fireworks driver allocates memory object to handle response from
target unit. The object is used to initiate transaction unique to
Fireworks board module. This can be released as managed-resource
of 'struct snd_card.card_dev'.
This commit uses managed-resource of the sound card device for this
purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA bebob driver allocates memory object for data specific to M-Audio
FW-1884/ProjectMix I/O. The object is to maintain format of isochronous
packet payload for packet streaming by components for ALSA rawMIDI/PCM
interfaces. The object can be released as managed-resource of
'struct snd_card.card_dev'.
This commit uses managed-resource of the sound card device for this
purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At present, private data of each driver in ALSA firewire stack is
allocated/freed by kernel slab allocator for corresponding unit on
IEEE 1394 bus. In this case, resource-managed slab allocator is
available to release memory object automatically just before releasing
device structure for the unit. This idea can prevent runtime from
memory leak due to programming mistakes.
This commit uses the allocator for the private data. These drivers
already use reference counter to maintain lifetime of device structure
for the unit by a pair of fw_unit_get()/fw_unit_put(). The private data
is safely released in a callback of 'struct snd_card.private_free().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The issue is the same as commit dd9aa335c8 ("ALSA: hda/realtek - Can't
adjust speaker's volume on a Dell AIO"), the output requires to connect
to a node with Amp-out capability.
Applying the same fixup ALC298_FIXUP_SPK_VOLUME can fix the issue.
BugLink: https://bugs.launchpad.net/bugs/1775068
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace "fallthru" with a proper "fall through" annotation.
This fix is part of the ongoing efforts to enabling
-Wimplicit-fallthrough
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace "fallthru" with a proper "fall through" annotation.
This fix is part of the ongoing efforts to enabling
-Wimplicit-fallthrough
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In blocking mode of IEC 61883-1/6, when one isochronous packet includes
data for events, the data is for the same number of events as
SYT_INTERVAL decided according to sampling transmission frequency (SFC).
IEC 61883-1/6 engine of ALSA firewire stack applies constraints of
period and buffer size of PCM intermediate buffer of PCM substream.
At present, this constraint is designed to round the size up/down to
32 frames. This value comes from the least common multiple (LCM) of
SYT_INTERVAL. Although this looks to work well, in lower sampling
rate, applications are not allowed to set size of period quite near
period time constraint (at present 5 msec per period).
This commit adds PCM rules for period/buffer size and rate to obsoletes
the constraints based on LCM.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a control for 600 ohm gain on the Sound Blaster ZxR.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes the input select control for the ZxR, as it only has
one input option, rear microphone.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds commands for selecting input and output on the Sound
Blaster ZxR. The ZxR has no front panel header, and has line-in on the
separate daughter board, so it only does rear-mic.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds commands for setting up the ZxR after the DSP is
downloaded. The ZxR already shares most of the post-download commands
from the regular Sound Blaster Z.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds separate hda_codec_ops for the DBPro daughter board, as
it behaves more like a generic HDA codec than the other ca0132 cards,
despite having a ca0132 on board.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a pincfg for the ZxR, and defines which pins are used
for both.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk ID's for the ZxR and it's daughter board, the
DBPro. It also adds a function for determining the quirk for each board
through HDA subsytem ID's instead of PCI subsystem ID's.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes an issue where if surround sound was the selected
output and output effects were enabled, the sound wasn't sent to all
channels correctly.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch corrects the control type of the additional AE-5 controls
added in a previous patch from HDA_INPUT to HDA_OUTPUT.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
E.g. for snd_hdac_ext_bus_link_power_up(), we should set mask to be
AZX_MLCTL_SPA(it was 0), and AZX_MLCTL_SPA as value to power up it,
here correct it and several similar mismatches.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Going primarily by:
https://en.wikipedia.org/wiki/List_of_Intel_Atom_microprocessors
with additional information gleaned from other related pages; notably:
- Bonnell shrink was called Saltwell
- Moorefield is the Merriefield refresh which makes it Airmont
The general naming scheme is: FAM6_ATOM_UARCH_SOCTYPE
for i in `git grep -l FAM6_ATOM` ; do
sed -i -e 's/ATOM_PINEVIEW/ATOM_BONNELL/g' \
-e 's/ATOM_LINCROFT/ATOM_BONNELL_MID/' \
-e 's/ATOM_PENWELL/ATOM_SALTWELL_MID/g' \
-e 's/ATOM_CLOVERVIEW/ATOM_SALTWELL_TABLET/g' \
-e 's/ATOM_CEDARVIEW/ATOM_SALTWELL/g' \
-e 's/ATOM_SILVERMONT1/ATOM_SILVERMONT/g' \
-e 's/ATOM_SILVERMONT2/ATOM_SILVERMONT_X/g' \
-e 's/ATOM_MERRIFIELD/ATOM_SILVERMONT_MID/g' \
-e 's/ATOM_MOOREFIELD/ATOM_AIRMONT_MID/g' \
-e 's/ATOM_DENVERTON/ATOM_GOLDMONT_X/g' \
-e 's/ATOM_GEMINI_LAKE/ATOM_GOLDMONT_PLUS/g' ${i}
done
Signed-off-by: Peter Zijlstra (Intel) <peterz@infradead.org>
Cc: Alexander Shishkin <alexander.shishkin@linux.intel.com>
Cc: Arnaldo Carvalho de Melo <acme@redhat.com>
Cc: Jiri Olsa <jolsa@redhat.com>
Cc: Linus Torvalds <torvalds@linux-foundation.org>
Cc: Peter Zijlstra <peterz@infradead.org>
Cc: Stephane Eranian <eranian@google.com>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: Vince Weaver <vincent.weaver@maine.edu>
Cc: dave.hansen@linux.intel.com
Cc: len.brown@intel.com
Signed-off-by: Ingo Molnar <mingo@kernel.org>
Remove duplicated includes linux/of_platform.h
and linux/wait.h
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The 'ret' variable is now only used in an #ifdef, and causes a
warning if it is declared outside of that block:
sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe':
sound/soc/codecs/wm9712.c:641:6: error: unused variable 'ret' [-Werror=unused-variable]
Fixes: 2ed1a8e0ce ("ASoC: wm9712: add ac97 new bus support")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Selecting AC97_BUS_NEW from SND_PXA2XX_SOC_AC97 leads to a Kconfig
warning if any other driver selects AC97_BUS:
WARNING: unmet direct dependencies detected for AC97_BUS_COMPAT
Depends on [n]: SOUND [=y] && !UML && SND [=y] && AC97_BUS_NEW [=y] && !AC97_BUS [=y]
Selected by [y]:
- SND_SOC_WM9713 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && AC97_BUS_NEW [=y]
I don't know if that combination is supposed to work.
Assuming it is not, this adds a dependency on all users
for PXA to avoids the combination.
Fixes: 1c8bc7b3de ("ASoC: pxa: switch to new ac97 bus support")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The symbols 'dummy_dma_ops' is declared with different data types by
sound/soc/soc-utils.c and arch/arm64/include/asm/dma-mapping.h. This
leads to conflicts when soc-utils.c (indirectly) includes dma-mapping.h:
sound/soc/soc-utils.c:282:33: error: conflicting types for 'dummy_dma_ops'
static const struct snd_pcm_ops dummy_dma_ops = {
^
...
arch/arm64/include/asm/dma-mapping.h:27:33: note: previous declaration of 'dummy_dma_ops' was here
extern const struct dma_map_ops dummy_dma_ops;
^
Rename the symbol in soc-utils.c to 'snd_dummy_dma_ops' to avoid the
conflict.
Signed-off-by: Matthias Kaehlcke <mka@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
MAX98088 is an older version of the MAX98089 device.
Signed-off-by: Andreas Färber <afaerber@suse.de>
[m.felsch@pengutronix.de: add CONFIG_OF compile switch]
[m.felsch@pengutronix.de: adapt commit message]
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The cs4265 driver is missing a microphone preamp enable.
This patch enables/disables the microphone preamp when mic
selection is made using the kcontrol.
Signed-off-by: Matt Flax <flatmax@flatmax.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_component_dais() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_component() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prepare <-> Cleanup functions pair has balanced calls.
But in case of suspend mode no call to rsnd_soc_dai_shutdown()
function, so cleanup isn't called. OTOH during resume mode
function rsnd_soc_dai_prepare() is called, but calling
rsnd_ssi_prepare() is skipped (rsnd_status_update() returns zero,
bacause was not cleanup before).
We need to call rsnd_ssi_prepare(), because it enables SSI clocks
by calling rsnd_ssi_master_clk_start().
This patch allows to call prepare/cleanup functions always.
Signed-off-by: Dmytro Prokopchuk <dmytro.prokopchuk@globallogic.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
[kuninori: adjusted to upstream]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We include wait.h twice in q6adm.c. it is unnecessary. hence remove
it. Further, order the include files as alphabet.
Signed-off-by: zhong jiang <zhongjiang@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 7fe072b4df ("ASoC: add for_each_card_prelinks() macro")
added new for_each_card_prelinks() macro, but it had typo.
This patch fixup it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch removes the boost volume in the beginning of playback while the
DAC volume set to lower.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_dpcm_be() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_dpcm_fe() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_comp_order() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_card_components() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_card_rtds() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_card_links() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To be more readable code, this patch adds
new for_each_card_prelinks() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 0b7990e389 ("ASoC: add for_each_rtd_codec_dai() macro")
added for_each_rtd_codec_dai_reverse(). but _rollback() is better
naming than _reverse(). This patch rename it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 0b7990e389 ("ASoC: add for_each_rtd_codec_dai() macro")
added for_each_rtd_codec_dai(), but it didn't convert few loop
which is not using "rtd". This patch fixup it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds controls for the AE-5's headphone gain setting, and the
DAC's interpolation filter setting.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the input selection commands for the Sound BlasterX
AE-5.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch cleans up ca0132_alt_out_select by moving the card specific
output commands into a separate function. As more cards are added, the
function ca0132_alt_out_select is going to get more bloated with these,
so moving into a separate function tries to keep that from happening.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch cleans up some of the formatting of the post-dsp load setup
functions, and also merges some of the sub functions into individual
ones.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Recon3D, AE-5, Z and ZxR all share the same firmware file. Rename
this from the specific "ctefx-sbz.bin" to "ctefx-desktop.bin" and set
the AE-5 and Recon3D to use it as well.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds AE-5 specific stuff to the ca0132_alt_init function.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds AE-5 pre-init functions that happen before the main
ca0132_alt_init, and gives functions related to the ca0113 a ca0113
prefix instead of ca0132. It also adds functions to write to the 8051's
SFRs, and to write the special ca0113 commands for the AE-5.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the unique writes for the AE-5 on startup to
ca0132_mmio_init. The other cards share some addresses written to, but
use different values.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch sets the bool values for the AE-5, as well as selects the
mixer it will use.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the pincfg for the Sound BlasterX AE-5, and cleans up
the function it's assigned in.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the PCI subsys ID quirk for the Sound BlasterX AE-5.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch defines some new verbs found from reverse engineering of the
onboard 8051 CPU, and a control param found there as well. This clears
up usage of these verbs in other parts of the driver, and removes their
usage where they're now known to be unnecessary.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_ops are not supposed to change. So mark the
non-const structs as const. Also, refine indentation
to ncrease readability.
Signed-off-by: Nick Simonov <nicksimonovv@gmail.com>
Reviewed-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The timeout of audio component binding was incorrectly specified in
msec, not in jiffies, which results in way too shorter timeout than
expected.
Along with fixing it, add the information print about the binding
failure to show the unexpected situation more clearly.
Fixes: a57942bfdd ("ALSA: hda: Make audio component support more generic")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This SPI driver does not use the legacy GPIO header so
just delete it.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Minor code changes are:
- improve the readability in patch list
- add i2c remove function
- regmap_register_patch changes to regmap_multi_reg_write
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
alsa_conformance_test -C hw:0,4 -p 1024 --debug
would sometime show:
TIME_DIFF(s) HW_LEVEL READ RATE
0.000095970 1024 1024 10670001.041992
0.042609555 1024 2048 24032.168372
0.021330364 1024 3072 48006.681930
0.021339559 1024 4096 47985.996337
The issue is that in dma pointer function we can have stale value
of the register for current descriptor of channel.
The register retains the number of the last descriptor that
was transferred.
Fix ensures that we report position, 0, till the one period worth of
data is transferred. After one period of data, in handler of period
completion interrupt we update the config and correct value of descriptor
starts reflecting.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixed the boost volume at the begining of playback
while DAC volume set to lower level.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ADC/DAC path should open while calibration process.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removed Headphone Playback Volume control.
Due to codec settings, we don't want the user to change HP analog gain.
The user could use DAC1 Playback Volume control to
change playback volume.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We change the settings while HP power-up for better performance.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
module.h already contained moduleparam.h, so it is safe to remove
the redundant include.
The issue is detected with the help of Coccinelle.
Signed-off-by: zhong jiang <zhongjiang@huawei.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The value of period_bytes will get the zero before the hw_params() is not
run completely. Move the function snd_pcm_lib_period_bytes() to copy work,
and make sure that is not zero.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To find (CPU/)Codec/Platform, we need to find component first
(= on CPU/Codec/Platform), and find DAI from it (= CPU/Codec).
These are similar operation but difficult to be simple,
and has many duplicate code to finding component.
This patch adds new snd_soc_is_matching_component(),
and reduce duplicate codes.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now "platform" is controlled by snd_soc_dai_link_component,
thus its "name" can be initialized in snd_soc_init_platform(),
instead of soc_bind_dai_link() local.
This patch do it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace "fallthru" with a proper "fall through" annotation.
This fix is part of the ongoing efforts to enabling
-Wimplicit-fallthrough
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the usual set of small fixes scatterd around various drivers,
plus one fix for DAPM and a UAPI build fix. There's not a huge amount
that stands out here relative to anything else.
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Merge tag 'asoc-v4.19-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.19
This is the usual set of small fixes scatterd around various drivers,
plus one fix for DAPM and a UAPI build fix. There's not a huge amount
that stands out here relative to anything else.
Replace "fallthru" with a proper "fall through" annotation.
This fix is part of the ongoing efforts to enabling
-Wimplicit-fallthrough
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After allocating memory object for response buffer, ALSA fireworks
driver has leak of the memory object at error path.
This commit releases the object at the error path.
Fixes: 7d3c1d5901aa('ALSA: fireworks: delayed registration of sound card')
Cc: <stable@vger.kernel.org> # v4.7+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After finishing discover of stream formats, ALSA OXFW driver has memory
leak of allocated memory object at error path.
This commit releases the memory object at the error path.
Fixes: 6c29230e2a ('ALSA: oxfw: delayed registration of sound card')
Cc: <stable@vger.kernel.org> # v4.7+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After allocating model-dependent data, ALSA OXFW driver has memory leak
of the data at error path.
This commit releases the data at the error path.
Fixes: 6c29230e2a ('ALSA: oxfw: delayed registration of sound card')
Cc: <stable@vger.kernel.org> # v4.7+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After allocating model-dependent data for M-Audio FW1814 and ProjectMix
I/O, ALSA bebob driver has memory leak at error path.
This commit releases the allocated data at the error path.
Fixes: 04a2c73c97eb('ALSA: bebob: delayed registration of sound card')
Cc: <stable@vger.kernel.org> # v4.7+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace "fallthru" with a proper "fall through" annotation.
This fix is part of the ongoing efforts to enabling
-Wimplicit-fallthrough
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent change of vga_switcheroo allowed the runtime PM for
HD-audio on AMD GPUs, but this also resulted in a regression. When
the HD-audio controller driver gets runtime-suspended, HD-audio link
is turned off, and the hotplug notification is ignored. This leads to
the inconsistent audio state (the connection isn't notified and ELD is
ignored).
The best fix would be to implement the proper ELD notification via the
audio component, but it's still not ready. As a quick workaround,
this patch adds the check of runtime_idle and allows the runtime
suspend only when the vga_switcheroo is bound with discrete GPU.
That is, a system with a single GPU and APU would be again without
runtime PM to keep the HD-audio link for the hotplug notification and
ELD read out.
Also, the codec->auto_runtime_pm flag is set only for the discrete GPU
at the time GPU gets bound via vga_switcheroo (i.e. only dGPU is
forcibly runtime-PM enabled), so that APU can still get the ELD
notification.
For identifying which GPU is bound, a new vga_switcheroo client
callback, gpu_bound, is implemented. The vga_switcheroo simply calls
this when GPU is bound, and tells whether it's dGPU or APU.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200945
Fixes: 07f4f97d7b ("vga_switcheroo: Use device link for HDA controller")
Reported-by: Jian-Hong Pan <jian-hong@endlessm.com>
Tested-by: Jian-Hong Pan <jian-hong@endlessm.com>
Acked-by: Lukas Wunner <lukas@wunner.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although private data of sound card instance is usually allocated in the
tail of the instance, drivers in ALSA firewire stack allocate the private
data before allocating the instance. In this case, the private data
should be released explicitly at .private_free callback of the instance.
This commit fixes memory leak following to the above design.
Fixes: 6c29230e2a ('ALSA: oxfw: delayed registration of sound card')
Cc: <stable@vger.kernel.org> # v4.7+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although private data of sound card instance is usually allocated in the
tail of the instance, drivers in ALSA firewire stack allocate the private
data before allocating the instance. In this case, the private data
should be released explicitly at .private_free callback of the instance.
This commit fixes memory leak following to the above design.
Fixes: b610386c8a ('ALSA: firewire-tascam: deleyed registration of sound card')
Cc: <stable@vger.kernel.org> # v4.7+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although private data of sound card instance is usually allocated in the
tail of the instance, drivers in ALSA firewire stack allocate the private
data before allocating the instance. In this case, the private data
should be released explicitly at .private_free callback of the instance.
This commit fixes memory leak following to the above design.
Fixes: 86c8dd7f4d ('ALSA: firewire-digi00x: delayed registration of sound card')
Cc: <stable@vger.kernel.org> # v4.7+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interrupt handler has to be acquired after the other resource
initialization when allocated with IRQF_SHARED. Otherwise it's
triggered before the resource gets ready, and may lead to unpleasant
behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Include asm/io.h directly so we've got a definition of pci_iomap(), the
current set of includes do this implicitly on most architectures but not
on SH.
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Internally, skl_init_chip() calls snd_hdac_bus_init_chip() which
1) sets bus->chip_init to prevent multiple entrances before device
is stopped; 2) enables interrupt.
We shouldn't use it for the purpose of resetting device only because
1) when we really want to initialize device, we won't be able to do
so; 2) we are ready to handle interrupt yet, and kernel crashes when
interrupt comes in.
Rename azx_reset() to snd_hdac_bus_reset_link(), and use it to reset
device properly.
Fixes: 60767abcea ("ASoC: Intel: Skylake: Reset the controller in probe")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In snd_hdac_bus_init_chip(), we enable interrupt before
snd_hdac_bus_init_cmd_io() initializing dma buffers. If irq has
been acquired and irq handler uses the dma buffer, kernel may crash
when interrupt comes in.
Fix the problem by postponing enabling irq after dma buffer
initialization. And warn once on null dma buffer pointer during the
initialization.
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 12eeeb4f47.
The patch doesn't fix accessing memory with null pointer in
skl_interrupt().
There are two problems: 1) skl_init_chip() is called twice, before
and after dma buffer is allocate. The first call sets bus->chip_init
which prevents the second from initializing bus->corb.buf and
rirb.buf from bus->rb.area. 2) snd_hdac_bus_init_chip() enables
interrupt before snd_hdac_bus_init_cmd_io() initializing dma buffers.
There is a small window which skl_interrupt() can be called if irq
has been acquired. If so, it crashes when using null dma buffer
pointers.
Will fix the problems in the following patches. Also attaching the
crash for future reference.
[ 16.949148] general protection fault: 0000 [#1] PREEMPT SMP KASAN PTI
<snipped>
[ 16.950903] Call Trace:
[ 16.950906] <IRQ>
[ 16.950918] skl_interrupt+0x19e/0x2d6 [snd_soc_skl]
[ 16.950926] ? dma_supported+0xb5/0xb5 [snd_soc_skl]
[ 16.950933] __handle_irq_event_percpu+0x27a/0x6c8
[ 16.950937] ? __irq_wake_thread+0x1d1/0x1d1
[ 16.950942] ? __do_softirq+0x57a/0x69e
[ 16.950944] handle_irq_event_percpu+0x95/0x1ba
[ 16.950948] ? _raw_spin_unlock+0x65/0xdc
[ 16.950951] ? __handle_irq_event_percpu+0x6c8/0x6c8
[ 16.950953] ? _raw_spin_unlock+0x65/0xdc
[ 16.950957] ? time_cpufreq_notifier+0x483/0x483
[ 16.950959] handle_irq_event+0x89/0x123
[ 16.950962] handle_fasteoi_irq+0x16f/0x425
[ 16.950965] handle_irq+0x1fe/0x28e
[ 16.950969] do_IRQ+0x6e/0x12e
[ 16.950972] common_interrupt+0x7a/0x7a
[ 16.950974] </IRQ>
<snipped>
[ 16.951031] RIP: snd_hdac_bus_update_rirb+0x19b/0x4cf [snd_hda_core] RSP: ffff88015c807c08
[ 16.951036] ---[ end trace 58bf9ece1775bc92 ]---
Fixes: 2eeeb4f4733b ("ASoC: Intel: Skylake: Acquire irq after RIRB allocation")
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is strange if it has "dai" but doesn't have "dai->driver".
And more over "dai->driver->xxx" is used everywhere without
"dai->driver" pointer NULL checking.
It got Oops already if "dai->driver" was NULL.
Let's remove un-needed "dai->driver" NULL check.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current behaviour of ASoC core w.r.t to component removal is that it
unregisters dependent sound card totally. There is no support to
rebind the card if the component comes back.
Typical use case is DSP restart or kernel modules itself.
With this patch, core now maintains list of cards that are unbind due to
any of its depended components are removed and card not unregistered yet.
This list is cleared when the card is rebind successfully or when the
card is unregistered from machine driver.
This list of unbind cards are tried to bind once again after every new
component is successfully added, giving a fair chance for card bind
to be successful.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
platform_driver_register will set the .owner field. So it is safe
to remove the redundant assignment.
The issue is detected with the help of Coccinelle.
Signed-off-by: zhong jiang <zhongjiang@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds SNDRV_PCM_INFO_INTERLEAVED into PCM hardware info.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC is counting card->dai_link_list user,
but no-one is using it.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Nested code is not readable.
This patch avoid it on soc_remove_dai().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5668_i2c_remove() is empty, and no longer needed.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When CONFIG_X86=n function azx_snoop doesn't use the variable chip it
only returns true.
sound/pci/hda/hda_intel.c: In function ‘dma_alloc_pages’:
sound/pci/hda/hda_intel.c:2002:14: warning: unused variable ‘chip’ [-Wunused-variable]
struct azx *chip = bus_to_azx(bus);
^~~~
Create a inline function of azx_snoop.
Fixes: a41d122449 ("ALSA: hda - Embed bus into controller object")
Signed-off-by: Anders Roxell <anders.roxell@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In functions snd_soc_get_volsw_sx() or snd_soc_put_volsw_sx(),
if the result of (min + max) is negative, then fls() returns
signed integer with value as 32. This leads to signed integer
overflow as complete operation is considered as signed integer.
UBSAN: Undefined behaviour in sound/soc/soc-ops.c:382:50
signed integer overflow:
-2147483648 - 1 cannot be represented in type 'int'
Call trace:
[<ffffff852f746fe4>] __dump_stack lib/dump_stack.c:15 [inline]
[<ffffff852f746fe4>] dump_stack+0xec/0x158 lib/dump_stack.c:51
[<ffffff852f7b5f3c>] ubsan_epilogue+0x18/0x50 lib/ubsan.c:164
[<ffffff852f7b6840>] handle_overflow+0xf8/0x130 lib/ubsan.c:195
[<ffffff852f7b68f0>] __ubsan_handle_sub_overflow+0x34/0x44 lib/ubsan.c:211
[<ffffff85307971a0>] snd_soc_get_volsw_sx+0x1a8/0x1f8 sound/soc/soc-ops.c:382
Typecast the operation to unsigned int to fix the issue.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Switch to the new ac97 bus support in sound/ac97 instead of the legacy
snd_ac97 one.
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
If capture and playback are started on different channel (I2S/BT)
there is a possibilty that channel information passed from machine driver
is overwritten before the configuration is done in dma driver.
Example:
113.597588: cz_max_startup: ---playback sets BT channel
113.597694: cz_dmic1_startup: ---capture sets I2S channel
113.597979: acp_dma_hw_params: ---configures capture for I2S channel
113.598114: acp_dma_hw_params: ---configures playback for I2S channel
This is fixed by having 2 separate instance for playback and capture.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_emu10k1_fx8010_ioctl(SNDRV_EMU10K1_IOCTL_INFO) allocates
memory using kmalloc() and partially fills it by calling
snd_emu10k1_fx8010_info() before returning the resulting
structure to userspace, leaving uninitialized holes. Let's
just use kzalloc() here.
BugLink: http://blog.infosectcbr.com.au/2018/09/linux-kernel-infoleaks.html
Signed-off-by: Willy Tarreau <w@1wt.eu>
Cc: Jann Horn <jannh@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MMTLR bit is in the CS4265_SPDIF_CTL2 register at address 0x12 bit 0
and not at address 0x0 bit 1. Fix this.
Signed-off-by: Sébastien Szymanski <sebastien.szymanski@armadeus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
snd_soc_dapm_new_dai may return an error pointer and currently this
isn't checked for in dapm_connect_dai_link_widgets. Add code to check
the return value and not add routes in that case.
Fixes: 778ff5bb86 ("ASoC: dapm: Move connection of CODEC to CODEC DAIs")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
kmemdup has implemented the function that kzalloc() + memcpy() will
do. and we prefer to kmemdup rather than the open coded implementation.
Signed-off-by: zhong jiang <zhongjiang@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
kmemdup has implemented the function that kzalloc() + memcpy() will
do. and we prefer to use the kmemdup rather than the open coded implementation.
Signed-off-by: zhong jiang <zhongjiang@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have devm_snd_soc_register_component().
Let's use it instead of snd_soc_register_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have devm_snd_soc_register_component().
Let's use it instead of snd_soc_register_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have devm_snd_soc_register_component().
Let's use it instead of snd_soc_register_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have devm_snd_soc_register_component().
Let's use it instead of snd_soc_register_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have devm_snd_soc_register_component().
Let's use it instead of snd_soc_register_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have devm_snd_soc_register_component().
Let's use it instead of snd_soc_register_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have devm_snd_soc_register_component().
Let's use it instead of snd_soc_register_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have devm_snd_soc_register_component().
Let's use it instead of snd_soc_register_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we have devm_snd_soc_register_component().
Let's use it instead of snd_soc_register_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HW register descriptions says:
"DMA Channel Reset...Software must confirm that this bit is
cleared before reprogramming any of the channel configuration registers."
There could be cases where dma stop errored out leaving dma channel
in reset state. We need to ensure that before the start of another dma,
channel is out of the reset state.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Smatch complains that if both "psubstream" and "csubstream" are NULL
then "ret" is uninitialized. That probably can't happen, but it's
cleaner to just return zero anyway so let's do that.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Smatch complains that these variables could be uninitialized. The first
one in snd_soc_dai_link_event() is probably a false positive, because
probably we know the lists are not empty. I would normally ignore the
warning, but GCC complains here as well so I just silenced the warning.
The "ret" in snd_soc_dapm_new_dai() does need to be initialized or it
leads to a bogus dereference in the caller.
Fixes: 3bbf5d34fd ("ASoC: dapm: Move error handling to snd_soc_dapm_new_control_unlocked")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When executing 'fw_run_transaction()' with 'TCODE_WRITE_BLOCK_REQUEST',
an address of 'payload' argument is used for streaming DMA mapping by
'firewire_ohci' module if 'size' argument is larger than 8 byte.
Although in this case the address should not be on kernel stack, current
implementation of ALSA bebob driver uses data in kernel stack for a cue
to boot M-Audio devices. This often brings unexpected result, especially
for a case of CONFIG_VMAP_STACK=y.
This commit fixes the bug.
Reference: https://bugzilla.kernel.org/show_bug.cgi?id=201021
Reference: https://forum.manjaro.org/t/firewire-m-audio-410-driver-wont-load-firmware/51165
Fixes: a2b2a7798fb6('ALSA: bebob: Send a cue to load firmware for M-Audio Firewire series')
Cc: <stable@vger.kernel.org> # v3.16+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This ensures the transfer loop won't waste a run to read
the few frames (if any) between start and hw_ptr update.
It will wait for the next interrupt with wait_for_avail().
Signed-off-by: Ricardo Biehl Pasquali <pasqualirb@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit ca917f9fe1 added use of usleep_range() but not
the corresponding "include <linux/delay.h>". The result is
with Chrome OS won't build because warnings are forced
to be errors:
mnt/host/source/src/third_party/kernel/v4.4/sound/soc/codecs/max98373.c:734:2: error: implicit declaration of function 'usleep_range' [-Werror,-Wimplicit-function-declaration]
usleep_range(10000, 11000);
^
Including delay.h "fixes" this.
Signed-off-by: Grant Grundler <grundler@chromium.org>
Reviewed-by: Benson Leung <bleung@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Split regmap_config.use_single_rw into use_single_read and
use_single_write. This change enables drivers of devices which only
support bulk operations in one direction to use the regmap_bulk_*()
functions for both directions and have their bulk operation split into
single operations only when necessary.
Update all struct regmap_config instances where use_single_rw==true to
instead set both use_single_read and use_single_write. No attempt was
made to evaluate whether it is possible to set only one of
use_single_read or use_single_write.
Signed-off-by: David Frey <dpfrey@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Playback of 44.1Khz contents with HDMI plugged returns
"Invalid pipe config" because HDMI paths in the FW
topology are configured to operate at 48Khz.
This patch filters out sampling rates not supported
at hdac_hdmi_create_dais() to let user space SRC
to do the converting.
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 4a75aae17b ("ASoC: dapm: Add support for multi-CODEC
CODEC to CODEC links") adds loops that iterate over multiple
CODECs in snd_soc_dai_link_event. This also introduced a compiler
warning for a potentially uninitialised variable in the case
no CODECs are present. This should never be the case as the
DAI link must by definition contain at least 1 CODEC however
probably best to avoid the compiler warning by initialising ret
to zero.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In the case where lo_vag <= SGTL5000_LINE_OUT_GND_BASE, lo_vag
is set to zero and later vol_quot is computed by dividing by
lo_vag causing a division by zero error. Fix this by avoiding
a zero division and set vol_quot to zero in this specific case
so that the lowest setting for i is correctly set.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd driver sometimes want to know which address is used when debugging.
But it will indicate "(____ptrval____)" if it used "%p" on dev_dbg().
Let's use "%pa" or "%px" for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver will fallback to PIO mode if it can't get DMA
handler. But, DMA might return -EPROBE_DEFER when probe timing.
This driver always fallback to PIO mode especially from
commit ac6bbf0cdf ("iommu: Remove IOMMU_OF_DECLARE") because
of this reason.
The DMA driver will be probed later, but sound driver might be
probed as PIO mode in such case. This patch fixup this issue.
Then, -EPROBE_DEFER is not error. Thus, let's don't indicate error
message in such case.
And it needs to call rsnd_adg_remove() individually if probe failed,
because it registers clk which should be unregister.
Maybe PIO fallback feature itself is not needed,
but let's keep it so far.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ADG has buffer over flow bug if DT has more than 3 clock-frequency.
This patch fixup this issue, and uses first 2 values.
clock-frequency = <x y>; /* this is OK */
clock-frequency = <x y z>; /* this is NG */
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 8c9d750333 ("ASoC: rsnd: ssiu: Support BUSIF
other than BUSIF0") added new SSIU registers.
But it is using white-space for it.
This patch fixup it to use tab.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The !port check is redundant as it being performed in the following
check. Remove it.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Larger CODECs may contain many several hundred widgets and which set of
parameters is selected only needs to be recorded on a per DAI basis. As
such move the selected CODEC to CODEC link params to be stored in the
runtime rather than the DAPM widget, to save some memory.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently multi-CODEC is not supported on CODEC to CODEC links.
There are common applications where this would be useful, such
as connecting two mono amplifiers to an audio CODEC. Adding
support simply requires an update of snd_soc_dai_link_event
to loop over the attached CODEC DAIs.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, snd_soc_dapm_connect_dai_link_widgets connects up the routes
representing normal DAIs, however CODEC to CODEC links are hooked up
through separate infrastructure in soc_link_dai_widgets. Improve the
consistency of the code by using snd_soc_dapm_connect_dai_link for both
types of DAIs.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the function snd_soc_dapm_new_control to be next to
snd_soc_dapm_new_controls and add some kernel doc for it.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently DAPM has a lot of similar code to handle errors from
snd_soc_dapm_new_control_unlocked, and much of this code does
not really accurately reflect what the function returns.
Firstly, most places will check for a return value of
-EPROBE_DEFER and silence any error messages in that case. The
one notable exception here being dapm_kcontrol_data_alloc
which does currently print any error messages in the case
of snd_soc_dapm_new_control_unlocked returning NULL or an
error. Additionally the error prints being silenced in these
case are redundant as snd_soc_dapm_new_control_unlocked can
only return -EPROBE_DEFER or NULL when failing.
Secondly, most places will treat a return value of NULL as
an -ENOMEM. This is not correct either since any error except
EPROBE_DEFER will cause a return value of NULL from
snd_soc_dapm_new_control_unlocked.
Centralise this handling and the error messages within
snd_soc_dapm_new_control_unlocked and update the callers
to simply check IS_ERR and return. Note that this update is
slightly simpler in the case of dapm_kcontrol_data_alloc where
that is fairly close to the handling that was already in place.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC snd_soc_pcm_runtime has snd_soc_dai array for codec_dai.
To be more readable code, this patch adds
new for_each_rtd_codec_dai() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC snd_soc_dai_link has snd_soc_dai_link_component array
for codecs.
To be more readable code, this patch adds
new for_each_link_codecs() macro, and replace existing code to it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Similar to short circuit detection, when the ADC/DAC is saturated and
overflows poor audio quality can result and should be reported to the
user. This device support Automatic Dynamic Range Compression (DRC)
to reduce this but it is not enabled currently in this driver.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We shouldn't set up the indirect PCM parameters at trigger but they
should be set at prepare. Also, remove a useless debug message, too.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced PCM info flag assures the call of ack ops at
each applptr change, and this is mandatory for the indirect PCM
helpers.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced PCM info flag assures the call of ack ops at
each applptr change, and this is mandatory for the indirect PCM
helpers.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced PCM info flag assures the call of ack ops at
each applptr change, and this is mandatory for the indirect PCM
helpers.
Also, with the proper ack callback, we need no longer prefill at
trigger start. The relevant code can be killed.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced PCM info flag assures the call of ack ops at
each applptr change, and this is mandatory for the indirect PCM
helpers.
Also, with the proper ack callback, we need no longer prefill at
trigger start. The relevant code can be killed.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
E.g. for snd_hdac_ext_link_clear_stream_id(), we should set (1 << stream)
as mask, and 0 as value, here correct it and several similar mismatches.
And, here also remove unreadable register_mask usage for those mask value
updating.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation to remove the node name pointer from struct device_node,
convert printf users to use the %pOFn format specifier.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When dapm_power_widgets() is called, the dapm_pre_sequence_async() and
dapm_post_sequence_async() functions are scheduled for all DAPM contexts
(apart from the card DAPM context) regardless of whether the DAPM
context is already in the desired state. The overhead of this is not
insignificant and the more DAPM contexts there are the more overhead
there is.
For example, on the Tegra124 Jetson TK1, when profiling the time taken
to execute the dapm_power_widgets() the following times were observed.
Times for function dapm_power_widgets() are (us):
Min 23, Ave 190, Max 434, Count 39
Here 'Count' is the number of times that dapm_power_widgets() has been
called. Please note that the above time were measured using ktime_get()
to log the time on entry and exit from dapm_power_widgets(). So it
should be noted that these times may not be purely the time take to
execute this function if it is preempted. However, after applying this
patch and measuring the time taken to execute dapm_power_widgets() again
a significant improvement is seen as shown below.
Times for function dapm_power_widgets() are (us):
Min 4, Ave 16, Max 82, Count 39
Therefore, optimise the dapm_power_widgets() function by only scheduling
the dapm_pre/post_sequence_async() work if the DAPM context is not in
the desired state.
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the MikroElektronika PROTO audio codec board.
URL to the audio chip:
http://www.mikroe.com/add-on-boards/audio-voice/audio-codec-proto/
Signed-off-by: Florian Meier <florian.meier@koalo.de>
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Main purpose of .nolock_start is we need to call
some function without spinlock.
OTOH we have .prepare which main purpose is
called under atomic context.
Then, it is called without spinlock.
In summary, our main callback init/quit, and start/stop
are called under "atomic context and with spinlock".
And some function need to be called under
"non-atomic context or without spinlock".
Let's merge .nolock_start and prepare to be more clear code.
Then, let's rename nolock_stop to cleanup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently ssiu's .init is only called once during audio stream.
But SSIU with different BUSIF, shall be initialized each time,
even they are used in the same audio stream.
This patch introduces ssiu_status for BUSIF0 to BUSIF7 in rsnd_ssiu,
to make sure same .init for different BUSIF can always be executed.
To avoid the first stopped stream to stop the whole SSIU,
which may still has other BUSIF instance running, use usrcnt to count
the usage of SSIU, only the last user of SSIU can stop the whole SSIU.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently only BUSIF0 is supported by SSIU, all register setting
is done only for BUSIF.
Since BUSIF1 ~ BUSIF7 has been supported, so also support
these BUSIF from SSIU.
One note is that we can't support SSI9-4/5/6/7 so far,
because its address is out of calculation rule.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PDMACHCR setting for SSI only considers BUSIF0 so far.
But BUSIF1 ~ BUSIF7 also maybe used, in the future.
This patch updates table gen2_id_table_ssiu, to also consider
BUSIF number used by SSI.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[kuninori: adjust to upstreaming]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DMA address calculated by rsnd_dma_addr() only considers BUSIF0 so far.
But BUSIF1 ~ BUSIF7 also maybe used, in the future.
This patch updates DMA address calculations, to also consider
BUSIF number used by SSI.
One note is that we can't support SSI9-4/5/6/7 so far,
because its address is out of calculation rule.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[kuninori: adjust to upstreaming]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The number of channel handled by SSI maybe differs from the one set
in hw_params, currently SSI checks hw_params's channel number,
and constrains to use same channel number, when it is being
used by multiple clients.
This patch corrects to check runtime channel number rather
than channel number set in hw_params.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[kuninori: adjust to upstreaming]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Same SSI device may be used in different dai links,
by only having one dma struct in rsnd_ssi, after the first
instance's dma config be initilized, the following instances
can no longer configure dma, this causes issue, when their
dma data address are different from the first instance.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently rsnd_dai_stream *io is set to either &rdai->playback or
&rdai->capture based on whether it is a playback or capture stream,
in __rsnd_soc_hw_rule_* functions, but this is not necessary,
rsnd_dai_stream *io handler can be get from rule->private.
This patch removes 'is_play' parameter from hw_rule function.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for SNDRV_PCM_FMTBIT_S8 format.
Signed-off-by: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The slot width (system word length) was fixed at 32 bits.
This patch allows also setting it to 16 or 24 bits.
Signed-off-by: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add warning message to rsnd_kctrl_accept_runtime(), when kctrl
update is rejected due to corresponding dai-link is idle.
So that user can notice the reason of kctrl update failure.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[kuninori: adjust to upstream]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Clocking operations clk_get/set_rate, are non-atomic,
they shouldn't be called in soc_pcm_trigger() which is atomic.
Following issue was found due to execution of clk_get_rate() causes
sleep in soc_pcm_trigger(), which shouldn't be blocked.
We can reproduce this issue by following
> enable CONFIG_DEBUG_ATOMIC_SLEEP=y
> compile, and boot
> mount -t debugfs none /sys/kernel/debug
> while true; do cat /sys/kernel/debug/clk/clk_summary > /dev/null; done &
> while true; do aplay xxx; done
This patch adds support to .prepare callback, and moves non-atomic
clocking operations to it. As .prepare is non-atomic, it is always
called before trigger_start/trigger_stop.
BUG: sleeping function called from invalid context at kernel/locking/mutex.c:620
in_atomic(): 1, irqs_disabled(): 128, pid: 2242, name: aplay
INFO: lockdep is turned off.
irq event stamp: 5964
hardirqs last enabled at (5963): [<ffff200008e59e40>] mutex_lock_nested+0x6e8/0x6f0
hardirqs last disabled at (5964): [<ffff200008e623f0>] _raw_spin_lock_irqsave+0x24/0x68
softirqs last enabled at (5502): [<ffff200008081838>] __do_softirq+0x560/0x10c0
softirqs last disabled at (5495): [<ffff2000080c2e78>] irq_exit+0x160/0x25c
Preemption disabled at:[ 62.904063] [<ffff200008be4d48>] snd_pcm_stream_lock+0xb4/0xc0
CPU: 2 PID: 2242 Comm: aplay Tainted: G B C 4.9.54+ #186
Hardware name: Renesas Salvator-X board based on r8a7795 (DT)
Call trace:
[<ffff20000808fe48>] dump_backtrace+0x0/0x37c
[<ffff2000080901d8>] show_stack+0x14/0x1c
[<ffff2000086f4458>] dump_stack+0xfc/0x154
[<ffff2000081134a0>] ___might_sleep+0x57c/0x58c
[<ffff2000081136b8>] __might_sleep+0x208/0x21c
[<ffff200008e5980c>] mutex_lock_nested+0xb4/0x6f0
[<ffff2000087cac74>] clk_prepare_lock+0xb0/0x184
[<ffff2000087cb094>] clk_core_get_rate+0x14/0x54
[<ffff2000087cb0f4>] clk_get_rate+0x20/0x34
[<ffff20000113aa00>] rsnd_adg_ssi_clk_try_start+0x158/0x4f8 [snd_soc_rcar]
[<ffff20000113da00>] rsnd_ssi_init+0x668/0x7a0 [snd_soc_rcar]
[<ffff200001133ff4>] rsnd_soc_dai_trigger+0x4bc/0xcf8 [snd_soc_rcar]
[<ffff200008c1af24>] soc_pcm_trigger+0x2a4/0x2d4
Fixes: e7d850dd10 ("ASoC: rsnd: use mod base common method on SSI-parent")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[Kuninori: tidyup for upstream]
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
syzbot reported the uninitialized value exposure in certain situations
using virmidi loop. It's likely a very small race at writing and
reading, and the influence is almost negligible. But it's safer to
paper over this just by replacing the existing kvmalloc() with
kvzalloc().
Reported-by: syzbot+194dffdb8b22fc5d207a@syzkaller.appspotmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These devices support detecting and reporting short circuits across
the output stages. Add support for reporting these issue. Do this
by registering an interrupt if available and enabling this error
to trigger that interrupt in the device.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix kernel-doc warning:
../sound/soc/soc-core.c:2918: warning: Excess function parameter 'legacy_dai_naming' description in 'snd_soc_register_dais'
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some of the router data fields are left as default zeros which are
valid dai ids, so initialize these to invalid value of -1.
Without intializing these correctly get_session_from_id() can return
incorrect session resulting in not closing the opened copp and messing
up with the copp ref count.
Fixes: e3a33673e8 ("ASoC: qdsp6: q6routing: Add q6routing driver")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The TAS6426 has a register that reports channel faults such as
overcurrent and continuous DC output. Add reporting of this here.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When there is no fault bit set in a fault register we skip the fault
reporting section for that register. This also skips over saving that
registers value. We save the value so we will not double report an
error, but if an error clears then returns we will also not report it
as we did not save the all cleared register value. Fix this by saving
the fault register value in the all clear path.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The current short version of the register name may be
ambiguous when another fault register detection is added.
Use the full name.
While here fix comment about clearing faults, the CLEAR_FAULT
register actually only clears sticky bits, which are only
warnings, fault bits can only cleared by resolving the fault.
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now no simple/audio cards are using legacy dai_link style for platform.
Let's remove compatible code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for audio-graph-scu-card for platform.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for audio-graph-card for platform.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-scu-card for platform.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-card for platform.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-card-util for platform.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current struct snd_soc_dai_link is supporting multicodec,
and it is supporting legacy style of
codec_name
codec_of_node
code_dai_name
This is handled as single entry of multicodec.
We don't have multicpu support yet, but in the future we will.
In such case, we can use snd_soc_dai_link_component for both
cpu/codec. Then the code will be more simple and readble.
As next step, we want to use it for platform, too.
This patch adds snd_soc_dai_link_component style for platform.
We might have multiplatform support in the future, but we
don't know yet. To avoid un-known issue / complex code,
this patch supports just single-platform as 1st step.
If we could use snd_soc_dai_link_component for all CPU/Codec/Platform,
we will switch to new style, and remove legacy code.
This is prepare for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now no simple/audio cards are using legacy dai_link style for codec.
Let's remove compatible code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for audio-graph-scu-card for codec.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audi-graph-card and audio-graph-scu-card are very similar driver,
but using different feature. Thus we are keeping synchronization
on these 2 drivers style, because it is easy to confirm / check.
Current big difference between these 2 drivers are "dai_props" on
graph_card_data (= priv).
It will be difficult to keep synchronize if we will add new feature
on audio-graph-scu-card. Thus, this patch synchronize it.
[audio-graph]
struct graph_card_data {
...
struct graph_dai_props {
...
} *dai_props;
...
};
[audio-graph-scu]
struct graph_card_data {
...
struct asoc_simple_dai *dai_props;
...
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for audio-graph-card for codec.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-scu-card for codec.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card and simple-scu-card are very similar driver,
but using different feature. Thus we are keeping synchronization
on these 2 drivers style, because it is easy to confirm / check.
Current big difference between these 2 drivers are "dai_props" on
simple_card_data (= priv).
It will be difficult to keep synchronize if we will add new feature
on simple-scu-card. Thus, this patch synchronize it.
[simple]
struct simple_card_data {
...
struct simple_dai_props {
...
} *dai_props;
...
};
[simple scu]
struct simple_card_data {
...
struct asoc_simple_dai *dai_props;
...
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple-card for codec.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is supporting snd_soc_dai_link_component for binding,
it is more useful than current legacy style.
Currently only codec is supporting it as multicodec (= codecs).
CPU will support multi style in the future.
We want to have it on Platform too in the future.
If all Codec/CPU/Platform are replaced into snd_soc_dai_link_component
style, we can remove legacy complex style.
This patch supports snd_soc_dai_link_component style
for simple_card_utils for codec.
[current]
struct snd_soc_dai_link {
...
*cpu_name;
*cpu_of_node;
*cpu_dai_name;
*codec_name;
*codec_of_node;
*codec_dai_name;
*codecs;
num_codecs;
*platform_name;
*platform_of_node;
...
}
[in the future]
struct snd_soc_dai_link {
...
*cpus
num_cpus;
*codecs;
num_codecs;
*platform;
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds a S/PDIF enable switch as a SOC_SINGLE.
Signed-off-by: Matt Flax <flatmax@flatmax.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The cs4265 uses 32 bit transport on the I2S bus. This patch enables native
32 bit mode for machine drivers which use this sound card driver.
Signed-off-by: Matt Flax <flatmax@flatmax.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The cs4265 driver declares the "MMTLR Data Switch" register setting with
a 0 register value rather then the 0x12 register (CS4265_SPDIF_CTL2).
This incorrect value causes alsamixer to fault with the output :
cannot load mixer controls: Input/output error
This patch corrects the register value. alsamixer now runs.
Signed-off-by: Matt Flax <flatmax@flatmax.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Unlike the TAS5720, the TAS5722 can be configured to utilize 16-bit wide
slots in TDM mode. This can help easing audio clocking/frequency
requirements.
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The TAS5722 supports modifying volume in 0.25dB steps (as opposed to
0.5dB steps on the TAS5720). Introduce a custom mixer control that
allows taking advantage of this finer output volume granularity.
Also add custom getters/setters for access as the TAS5722 digital volume
controls are split over two registers.
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Signed-off-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
McASP have write and read FIFO, each 64 words deep.
From the WFIFOS/RFIFOS registers we can read the amount of data currently
in the FIFO which can be directly reported as delay.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For easier sharing with ASoC.
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Merge tag 'hda-codec-h-move' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ALSA: Move hda_codec.h to include/sound
For easier sharing with ASoC.
As suggested by Takashi, move this header file to make it easier
to include from e.g. the Intel Skylake driver in follow-up patches
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On AMD/ATI controllers, the HD-audio controller driver allows a bus
reset upon the error recovery, and its procedure includes the
cancellation of pending jack polling work as found in
snd_hda_bus_codec_reset(). This works usually fine, but it becomes a
problem when the reset happens from the jack poll work itself; then
calling cancel_work_sync() from the work being processed tries to wait
the finish endlessly.
As a workaround, this patch adds the check of current_work() and
applies the cancel_work_sync() only when it's not from the
jackpoll_work.
This doesn't fix the root cause of the reported error below, but at
least, it eases the unexpected stall of the whole system.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200937
Cc: <stable@vger.kernel.org>
Cc: Lukas Wunner <lukas@wunner.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the jackpoll_ms option value is passed indirectly by
referring to an array in chip->jackpoll_ms although each card needs to
see only the assigned value. Also, the sanity check is done at each
time in get_jackpoll_interval() although basically jackpoll_ms option
is a read-only, hence we need to evaluate only once at probe time.
This patch is the code simplification about the above points: the jack
polling interval is directly set to chip->jackpoll_interval so that it
can be simply copied to each codec.
No functional change but only code reduction.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add pdm input driver for the device found on the amlogic AXG SoC family
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Before this patch the only alias provided by the dmic module is:
alias: platform:dmic-codec
Device instantiated from DT will not probe automatically with this
After this patch, here is the new alias list:
alias: platform:dmic-codec
alias: of:N*T*Cdmic-codecC*
alias: of:N*T*Cdmic-codec
Now the dmic codec probes automatically when instantiated from DT.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Kconfig prompt for the generic digital mic to make it configurable
through menuconfig
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
adau17x1_setup_firmware and adau17x1_has_dsp is only used internally, so
making them static instead of exported.
Signed-off-by: Robert Rosengren <robertr@axis.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Modified some if-statements to make them more clear
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
Modified the complete file comments in C++ style, to make them look more
intentional
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
Convert to // comments in the leading comment, drop the boilerplate
license text and use the correct MODULE_LICENSE.
Signed-off-by: Peter Rosin <peda@axentia.se>
Signed-off-by: Mark Brown <broonie@kernel.org>
25Mhz MCLK which was earlier used was of spread type.
Thus, we were not getting accurate rate. The 48Mhz system
clk is of non-spread type and we are changing to it to get
accurate rate.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
We support dual channel, 48Khz. This constraint was set only for
da7219. It is being extended to DMIC and MAX98357a.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit adds support for TI PCM3060 CODEC.
The technical documentation is available at [1].
[1] http://ti.com/product/pcm3060
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Cc: Mark Brown <broonie@kernel.org>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: M R Swami Reddy <mr.swami.reddy@ti.com>
Cc: Vishwas A Deshpande <vishwas.a.deshpande@ti.com>
Cc: Kevin Cernekee <cernekee@chromium.org>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Having the headset buttons send BTN_0, BTN_1 and BTN_2 events is not
really useful. Add mappings to PLAYPAUSE VOLUME_UP and VOLUME_DOWN like
we do in other Intel machine drivers.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Thinkpad 8 tablet uses 10EC5640 as ACPI HID, but it has a rt5670 codec
add a quirk for this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Thinkpad 8 needs a quirk for jack-detect and the internal mic to
work correctly.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Safeload support has been implemented which is used
when updating for instance filter parameters using
alsa controls. Without safeload support audio can
become distorted during update.
Signed-off-by: Danny Smith <dannys@axis.com>
Signed-off-by: Robert Rosengren <robertr@axis.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixed range in safeload conditional to allow safeload to up to 20 bytes,
without a lower limit.
Signed-off-by: Danny Smith <dannys@axis.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds a kernel module which is used by the legacy HDA
codec drivers as library. This implements hdac_ext_bus_ops to enable
the reuse of legacy HDA codec drivers with ASoC platform drivers.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use hda_bus instead of hdac_bus in the SKL ASoC platform driver to enable
reuse of legacy HDA codec drivers.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for HDA BE DAIs in SKL platform driver.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When no I2S based codec entries are found in the BIOS, check if there are
any HDA codecs detected on the bus. Based on the number of codecs found
take appropriate action in machine driver. If there are two HDA codecs
i.e. iDisp + HDA found on the bus, register DAIs and DAI links for both.
If only one codec i.e. iDisp is found then load only iDisp machine driver.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add machine driver for Intel platforms (SKL/KBL/BXT/APL) with
HDA and iDisp codecs. This patch adds support for only iDisp (HDMI/DP)
codec. In the following patches support for HDA codecs will be added.
This should work for other Intel platforms as well e.g. GLK,CNL
however this series is not tested on all the platforms.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Expose a table containing machine driver information for HDAudio-based
platforms handled by ASoC on Intel hardware.
We only set constant values that are valid across multiple
platforms. The firmware name used by the DSP will be set dynamically
for each platform.
The table is made of a single entry for now, if we need more
complicated set-up where HDAudio is mixed with ACPI-enumerated devices
(I2C, SoundWire) then we'd expect the differentiation to be handled
through information provided by the BIOS (as done for KBL
Chromebooks).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As suggested by Takashi, move this header file to make it easier
to include from e.g. the Intel Skylake driver in follow-up patches
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HID made of either Wolfson/CirrusLogic PCI ID + 8804 identifier.
This helps enumerate the HifiBerry Digi+ HAT boards on the Up2 platform.
The scripts at https://github.com/thesofproject/acpi-scripts can be
used to add the ACPI initrd overlays.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
New calibration sequence allows rt5682 do calibration without
MCLK.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The step of DAC/ADC volume scale changes from 0.375dB to 0.75dB
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Restrict the formats possible on the TDM interface depending on the width
of the TDM slot and let dpcm merging do the rest.
Fixes: d60e4f1e4b ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Return value of request_irq() was irgnored. Fix this and report
the failure if any
Fixes: 6dc4fa179f ("ASoC: meson: add axg fifo base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add audio clock controller and ARB reset controller module
implication for the device using them
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the CPU DAI does not initialise rate_max, say if using
using KNOT or CONTINUOUS, then the rate_max field will be
initialised to 0. A value of zero in the rate_max field of
the hardware runtime will cause the sound card to support no
sample rates at all. Obviously this is not desired, just a
different mechanism is being used to apply the constraints. As
such update the setting of rate_max in dpcm_init_runtime_hw
to be consistent with the non-DPCM cases and set rate_max to
UINT_MAX if nothing is defined on the CPU DAI.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_pinctrl_get will only return NULL in the case that pinctrl
is not built into the kernel and all the pinctrl functions used
by the DAPM core are appropriately stubbed for that case. There
is no need to error out of snd_soc_dapm_new_control_unlocked
if pinctrl isn't built into the kernel, so change the
IS_ERR_OR_NULL to just an IS_ERR.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The clock code now has stub functions defined in its header files so
the ifdefs around clocking code should no longer be necessary.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to remove the node name pointer from struct device_node,
convert printf users to use the %pOFn format specifier.
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The HDMI LPE audio driver requires the non-cached page allocations for
its buffers. With the recent support of SNDRV_DMA_TYPE_DEV_UC type,
we can reduce lots of codes in the driver side and let the memalloc
core doing it properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
intel8x0 driver requires the non-cached pages for 440MX workaround,
and this can be implemented more easily with the new memalloc type,
SNDRV_DMA_TYPE_DEV_UC. This allows us to reduce lots of code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we dropped the memory page fiddling in the own allocators in
hda_intel.c, the substream allocation and free ops in both hda_intel.c
and hda_tegra.c became nothing but the simple calls of the standard
snd_pcm_lib helpers. As both are identical, there is no longer need
for indirect calls via ops; it's a good opportunity for removing ops
and simplifying the codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now the ALSA memory allocator helper supports the new non-cached
pages, let's use the new type, SNDRV_DMA_TYPE_DEV_UC_SG, for HD-audio
driver. This allows us to reduce lots of codes.
As another positive side-effect by this patch, the long-standing issue
with non-snoop mode playing in the non-mmap mode is fixed. The core
memalloc helper does the proper pgprot setup for non-cached pages for
vmap(), which was missing in the past.
Reported-and-tested-by: Hans Hu <HansHu@zhaoxin.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new flag, uc_buffer, to indicate that the controller
requires the non-cached pages for stream buffers, either as a
chip-specific requirement or specified via snoop=0 option.
This improves the code-readability.
Also, this patch fixes the incorrect behavior for C-Media chip where
the stream buffers were never handled as non-cached due to the check
of driver_type even if you pass snoop=0 option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some cases (mainly for x86), we need the DMA coherent buffer with
non-cached pages. Although this has been done in each driver side
like HD-audio and intel8x0, it can be done cleaner in the core memory
allocator.
This patch adds the new types, SNDRV_DMA_TYPE_DEV_UC and
SNDRV_DMA_TYPE_DEV_UC_SG, for allocating such non-cached buffer
pages. On non-x86 architectures, they work as same as the standard
SNDRV_DMA_TYPE_DEV and *_SG.
One additional change by this move is that we can assure to pass the
non-cached pgprot to the vmapped buffer, too. It eventually fixes the
case like non-snoop mode without mmap access on HD-audio.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_malloc_dev_pages() and snd_free_dev_pages() are local functions
and the parameters passed there are all contained in snd_dma_buffer
object. As a code-simplification, pass snd_dma_buffer object and
assign the address there like other allocators do (except for
snd_malloc_pages() which is called from outside, hence we can't change
easily).
Only code refactoring, no functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The size passed to dma_alloc_coherent() doesn't have to be aligned
with power-of-two, rather it should be the raw size. As a minor
optimization, remove the size adjustment in the current code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kuninori Morimoto