The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_be_hw_params_fixup() between in these
2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_dai_init() between in these 2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_hw_param() between in these 2 drivers.
One note is that only simple-card supports simple_set_clk_rate()
at hw_param from commit e9be4ffd4f ("ASoC: simple-card: set cpu
dai clk in hw_params").
By this patch, audio-graph has same feature.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_shutdown() between in these 2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The difference between simple-card / audio-graph are just using
OF graph style, or not. In other words, other things should be same.
This means, simple-card/audio-graph common functions should be
implemented at simple-card-utils, and its own functions should be
implemented at each files.
Current simple-card / audio-graph have almost same functions.
This patch shares asoc_simple_startup() between in these 2 drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Historically, simple-card/simple-scu-card/audio-graph/audio-graph-scu
are similar but different generic sound card.
simple-scu-card which was for DPCM was merged into simple-card, and
audio-graph-scu which was for DPCM was merged into audio-graph.
simple-card is for non OF graph sound card, and
audio-graph is for OF graph sound card.
And, small detail difference (= function parameter, naming, etc)
between simple-card/audio-graph has been unified.
So today, the difference between simple-card/audio-graph are
just using OF graph style, or not.
In other words, there should no difference other than OF graph sytle.
simple-card/audio-graph are using own priv today , but we can merge it.
This patch merge it at simple_card_utils.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils has dev_dbg(), but people want to
add #define DEBUG at simple-card/audio-graph, not simple-card-utils.
And, people want to get all information.
This patch adds new asoc_simple_debug_info() to indicates information.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As the DAI clocks for DA7219 have now been split into BCLK and WCLK,
the clock lookup name needs to be udpated here to select BCLK to
achieve the same functionality as before with regards to DAI clock
gating.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For the purposes of platforms which use the codec as DAI clock
master for the CPU and other codec devices, there is the need to
not only expose the clock gating of BCLK and WCLK but also the
ability to set those rates without going through the ASoC APIs.
To make this possible, the previous CCF implementation in the
driver has been extended to separate BCLK and WCLK out. WCLK is
the parent clock to BCLK, and is also the clock gate for both.
BCLK in HW is a factor/multiplier of WCLK so derives from whatever
SR is chosen for WCLK, hence the need to make it a child of WCLK
for the purposes of CCF. Enabling/disabling either BCLK or WCLK
will result in clocks being ungated/gated accordingly. To simplify
matters, these clocks can only be configured if the codec is set
as master, otherwise CCF control is disallowed.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is very low possibility ( < 0.1% ) that channel swap happened
in beginning when multi output/input pin is enabled. The issue is
that hardware can't send data to correct pin in the beginning with
the normal enable flow.
This is hardware issue, but there is no errata, the workaround flow
is that: Each time playback/recording, firstly clear the xSMA/xSMB,
then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled
before xSMA). Which is to use the xSMA as the trigger start register,
previously the xCR_TE or xCR_RE is the bit for starting.
Fixes commit 43d24e76b6 ("ASoC: fsl_esai: Add ESAI CPU DAI driver")
Cc: <stable@vger.kernel.org>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a constraint for the channel number setting on the
asrc of older version (e.g. imx35), the channel number should
be even, odd number isn't valid.
So add this constraint when the asrc of older version is used.
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The CS4270 does not by default increment the register address on
consecutive writes. During normal operation it doesn't matter as all
register accesses are done individually. At resume time after suspend,
however, the regcache code gathers the biggest possible block of
registers to sync and sends them one on one go.
To fix this, set the INCR bit in all cases.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Acer laptop Aspire E5-523G and ES1-432 with ALC255 can't detect
the headset microphone until ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk
applied.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer Aspire Z24-890 cannot detect the headset MIC until
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dev is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/core/seq/oss/seq_oss_synth.c:626 snd_seq_oss_synth_make_info() warn: potential spectre issue 'dp->synths' [w] (local cap)
Fix this by sanitizing dev before using it to index dp->synths.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
info->stream is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/core/rawmidi.c:604 __snd_rawmidi_info_select() warn: potential spectre issue 'rmidi->streams' [r] (local cap)
Fix this by sanitizing info->stream before using it to index
rmidi->streams.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot
record sound from headset MIC. This patch adds the
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue.
Fixes: 9f8aefed96 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660G")
Fixes: b72f936f6b ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860G")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Register platform component with a prefix, to avoid warnings
on debugfs entries creation, due to component name
redundancy.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DFSDM must be stopped when a new setting is applied.
restart systematically DFSDM on multiple prepare calls,
to apply changes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The wm_adsp_ops structures should be static and correct two printf
specifiers.
Fixes: 170b1e123f ("ASoC: wm_adsp: Add support for new Halo core DSPs")
Fixes: 4e08d50d1f ("ASoC: wm_adsp: Factor out DSP specific operations")
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can simplify the code by caching the CPU DAI master/slave
information rather than reading previously set register bit.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Break the clock setting logic out from the main hw_params. It's
rather large and unweildy and makes for a large function. This
also better enables some of the following changes to the clock
tree access in the driver.
Signed-off-by: Annaliese McDermond <nh6z@nh6z.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Depending on MACH_JZ4740 prevent us from creating a generic kernel that
works on more than one MIPS board. Instead, we just depend on MIPS being
set.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A Halo Core DSP has a memory protection unit that can trap and signal
memory access faults. This patch adds a function that dumps the fault
information.
The interrupt reaches the host via the parent codec interrupt controller
so this fault function is exported to be called by the codec driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Halo core is a new generation of audio DSP architecture from
Cirrus Logic. A new iteration of the WMFW file format (v3) is also
added, for this new architecture. Currently this format is not
supported on the old ADSP2 architecture however support may be
added for it in the future.
Signed-off-by: Wen Shi <wenshi@opensource.cirrus.com>
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change the signature of mtk_regmap_update_bits to also take a shift, and
warn when reg >= 0 but shift < 0. This reduce the code repetition
on the calling side, and prevent future UBSAN warning when some of the
xxx_shift and xxx_reg are both set to -1.
Signed-off-by: Pi-Hsun Shih <pihsun@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In sound/soc/mediatek/common/mtk-afe-fe-dai.c, when xxx_reg is -1, it's
a no-op to call mtk_regmap_update_bits, but since both xxx_reg and
xxx_shift are set to -1, the (1 << xxx_shift) in the argument would
trigger a UBSAN warning.
Fix the warning by setting those xxx_shift to 0 instead.
Note that since the code explicitly checks .mono_shift >= 0 and
.fs_shift >= 0 before using them in '<<' operator, those two members are
not set to 0.
Signed-off-by: Pi-Hsun Shih <pihsun@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for the addition of more types of DSP core refactor the
handling of DSP specific operations such as starting the memory or
enabling the core into a set of callbacks. This should make it easier to
add new core types and allow for more code reuse between them.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to duplicate this code for both ADSP1 and 2 as the
handling is exactly the same.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for further additions refactor the reading of the
firmware status.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The original wm_adsp2_early_event took an additional frequency
argument for clocking control so could not be used directly as a
DAPM callback. But this setup could equally be done by the codec
driver function wrapping wm_adsp2_early event. In preparation
for adding support for new core types wm_adsp2_set_dspclk has
been exported, and the freq argument removed so that it can
be used directly as a DAPM callback.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This function is not presently called from outside the adsp code and nor
should it be, as such stop exporting it.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a watchdog timeout is received from the DSP it is safe to assume the
DSP is not functioning anymore and as such any active compressed streams
should be put into an error state.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Best to lock across handling the bus error to ensure the DSP doesn't
change power state as we are reading the status registers.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During recent logging improvements it seems two error messages lost
their updates during patch application/rebasing. Add these back in.
Fixes: 0d3fba3e7a ("ASoC: wm_adsp: Improve logging messages")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously support was added to allow streams to be stopped and
started again without the DSP being power cycled and this was done
by clearing the buffer state in trigger start. Another supported
use-case is using the DSP for a trigger event then opening the
compressed stream later to receive the audio, unfortunately clearing
the buffer state in trigger start destroys the data received
from such a trigger. Correct this issue by moving the call to
wm_adsp_buffer_clear to be in trigger stop instead.
Fixes: 61fc060c40 ("ASoC: wm_adsp: Support streams which can start/stop with DSP active")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some strings are allocated by kstrdup, but not freed when error
happened.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The stream_name is allocated by kstrdup. We have to free it when the
dai is removed or return from error.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Recently we found the audio jack detection stop working after suspend
on many machines with Realtek codec. Sometimes the audio selection
dialogue didn't show up after users plugged headhphone/headset into
the headset jack, sometimes after uses plugged headphone/headset, then
click the sound icon on the upper-right corner of gnome-desktop, it
also showed the speaker rather than the headphone.
The root cause is that before suspend, the codec already call the
runtime_suspend since this codec is not used by any apps, then in
resume, it will not call runtime_resume for this codec. But for some
realtek codec (so far, alc236, alc255 and alc891) with the specific
BIOS, if it doesn't run runtime_resume after suspend, all codec
functions including jack detection stop working anymore.
This problem existed for a long time, but it was not exposed, that is
because when problem happens, if users play sound or open
sound-setting to check audio device, this will trigger calling to
runtime_resume (via snd_hda_power_up), then the codec starts working
again before users notice this problem.
Since we don't know how many codec and BIOS combinations have this
problem, to fix it, let the driver call runtime_resume for all codecs
in pm_resume, maybe for some codecs, this is not needed, but it is
harmless. After a codec is runtime resumed, if it is not used by any
apps, it will be runtime suspended soon and furthermore we don't run
suspend frequently, this change will not add much power consumption.
Fixes: cc72da7d4d ("ALSA: hda - Use standard runtime PM for codec power-save control")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 3baffc4a84 (ALSA: hda/intel: Refactoring PM code) changed
the behaviour of azx_resume(), it triggers the jackpoll_work after
applying this commit.
This change introduced a new issue, all codecs are runtime active
after S3, and will not call runtime_suspend() automatically.
The root cause is the jackpoll_work calls snd_hda_power_up/down_pm,
and it calls up_pm before snd_hdac_enter_pm is called, while calls
the down_pm in the middle of enter_pm and leave_pm is called. This
makes the dev->power.usage_count unbalanced after S3.
To fix it, let azx_resume() don't trigger jackpoll_work as before
it did.
Fixes: 3baffc4a84 ("ALSA: hda/intel: Refactoring PM code")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is parsing mclk_fs at many places, but it should be
same operation. This patch adds graph_parse_mclk_fs()
and parse it.
This patch also renames similar function graph_get_conversion()
to graph_parse_convert().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
use same naming rule, and this patch add missing of_node_put() on it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is parsing mclk_fs at many places, but it should be
same operation. This patch adds simple_parse_mclk_fs()
and parse it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If ASRC turns on, HW will use clk_dac as the reference clock
whether recording or playback.
Both of clk_dac and clk_adc should set proper clock while using ASRC.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The jack type detection needs the main bias power of analog.
The modification makes sure the main bias power on/off while jack plug/unplug.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The IRQ function may not work when system suspend.
We remove snd_soc_dapm_force_enable_pin function call to
make sure the bias off when idle and run into suspend/resume function.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skip for i2s5 in mck_disable which is also bypassed in mck_enable.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In kernel API of Linux FireWire subsystem, handlers of isochronous
receive (IR) context can get context headers as an argument of
callback. When 4 byte header is used, the context header includes
isochronous packet header for each packet. When 8 byte header is
used, it includes isochronous cycle as well.
ALSA IEC 61883-1/6 engine uses 4 byte header, and computes isochronous
cycle from the cycle of interrupt. The usage of 8 byte header can
obsolete the computation.
Furthermore, this change works well for a case that a series of
packet in one interrupt includes skipped isochronous cycle,
This commit uses 8 byte header to handle isochronous cycle.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds support for MOTU 8pre FireWire, which was shipped 2007
and nowadays already discontinued. Userspace applications can transmit
and receive PCM frames and MIDI messages for this model via ALSA PCM
interface and RawMidi/Sequencer interfaces.
Like the other models of MOTU FireWire series, this model has many
quirks in its CIP.
At first, data channels for two pairs of optical interfaces. At lower
sampling transmission frequency, i.e. 44.1 and 48.0 kHz, one pair is
available for ADAT data, thus 8 data chunks are transferred by CIP.
At middle sampling transmission frequency, i.e. 88.2 and 96.0 kHz,
two pairs are available to keep 8 chunks for ADAT data, thus CIP
still includes 8 data chunks.
Apart from data chunks for optical interface, CIP includes fixed number
of data chunks. In tx stream, two chunks for status message, eight
chunks for samples from analog 1-8 input, two chunks for mix-return.
In rx stream, two chunks for control message, two chunks for main 1-2
output, two chunks for phone 1-2 output, two chunks for dummy 1-2.
CIP header in tx stream includes quirks for its dbs and dbc fields.
The value of dbs field is fixed to 0x13, against its actual size.
The value of dbc field is firstly updated to 0x07 from zero, then
it's incremented continuously according to actual number of data h
blocks.
Finally, the model has own bits to disable frame fetch.
This commit uses several options to absorb the above quirks.
$ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 0410b57d bus_info_length 4, crc_length 16, crc 46461
404 31333934 bus_name "1394"
408 20001000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 1 (4)
40c 0001f200 company_id 0001f2 |
410 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb
root directory
-----------------------------------------------------------------
414 0004c65c directory_length 4, crc 50780
418 030001f2 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 8d000006 --> eui-64 leaf at 438
424 d1000001 --> unit directory at 428
unit directory at 428
-----------------------------------------------------------------
428 0003991c directory_length 3, crc 39196
42c 120001f2 specifier id
430 1300000f version
434 17103800 model
eui-64 leaf at 438
-----------------------------------------------------------------
438 00022681 leaf_length 2, crc 9857
43c 0001f200 company_id 0001f2 |
440 00083dfb device_id 0000083dfb | EUI-64 0001f20000083dfb
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function snd_opl3_drum_switch declaration in the header file
has the order of the two arguments on_off and vel swapped when
compared to the definition arguments of vel and on_off. Fix this
by swapping them around to match the definition.
This error predates the git history, so no idea when this error
was introduced.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another machine which does not like the power saving (noise):
https://bugzilla.redhat.com/show_bug.cgi?id=1689623
Also, reorder the Lenovo C50 entry to keep the table sorted.
Reported-by: hs.guimaraes@outlook.com
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add BYT_RT5651_JD_NOT_INV quirk for devices with an inverted
(active-high instead of the normal active-low) jack-detect switch.
And add a quirk for the Complet Electro Serv MY8307 tablet which has
an inverted jack-detect switch (and a mono-speaker).
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some boards use a jack-receptacle with a switch which reports the
jack-inserted status as active-high, rather then the standard active-low
reporting most jacks use.
This commit adds support for it. This is activated by a boolean
"realtek,jack-detect-not-inverted" device-property. The not-inverted
in the device-property name, rather then active-high, was chosen to keep
the device-property naming consistent with the rt5640 codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suspend and resume sleep callbacks to STM32 SPDIFRX driver,
to support system low power modes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dpcm get from fe_clients/be_clients
may be free before use
Add a spin lock at snd_soc_card level,
to protect the dpcm instance.
The lock may be used in atomic context, so use spin lock.
possible race condition between
void dpcm_be_disconnect(
...
list_del(&dpcm->list_be);
list_del(&dpcm->list_fe);
kfree(dpcm);
...
and
for_each_dpcm_fe()
for_each_dpcm_be*()
race condition example
Thread 1:
snd_soc_dapm_mixer_update_power()
-> soc_dpcm_runtime_update()
-> dpcm_be_disconnect()
-> kfree(dpcm);
Thread 2:
dpcm_fe_dai_trigger()
-> dpcm_be_dai_trigger()
-> snd_soc_dpcm_can_be_free_stop()
-> if (dpcm->fe == fe)
Excpetion Scenario:
two FE link to same BE
FE1 -> BE
FE2 ->
Thread 1: switch of mixer between FE2 -> BE
Thread 2: pcm_stop FE1
Exception:
Unable to handle kernel paging request at virtual address dead0000000000e0
pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
sound/soc/soc-pcm.c:3226
if (dpcm->fe == fe)
lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
Backtrace:
[<ffffff89602dba80>] notify_die+0x68/0xb8
[<ffffff896028c7dc>] die+0x118/0x2a8
[<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c
[<ffffff89602a27f4>] do_translation_fault+0x64/0xa0
[<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0
[<ffffff8960282ad0>] el1_da+0x24/0x40
[<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
[<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
[<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44
[<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c
[<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c
[<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128
[<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0
[<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14
[<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244
[<ffffff8960283740>] el0_svc_naked+0x34/0x38
[<ffffffffffffffff>] 0xffffffffffffffff
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some amplifier may not have a GPIO to control the power, but instead simply
rely on the regulator to power up and down the amplifier.
In order to support those setups, let's make the GPIO optional.
Signed-off-by: Mylène Josserand <mylene.josserand@bootlin.com>
Signed-off-by: Maxime Ripard <maxime.ripard@bootlin.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ALSA firewire-motu driver uses the value of 'model' field
of unit directory in configuration ROM for modalias for MOTU
FireWire models. However, as long as I checked, Pre8 and
828mk3(Hybrid) have the same value for the field (=0x100800).
unit | version | model
--------------- | --------- | ----------
828mkII | 0x000003 | 0x101800
Traveler | 0x000009 | 0x107800
Pre8 | 0x00000f | 0x100800 <-
828mk3(FW) | 0x000015 | 0x106800
AudioExpress | 0x000033 | 0x104800
828mk3(Hybrid) | 0x000035 | 0x100800 <-
When updating firmware for MOTU 8pre FireWire from v1.0.0 to v1.0.3,
I got change of the value from 0x100800 to 0x103800. On the other
hand, the value of 'version' field is fixed to 0x00000f. As a quick
glance, the higher 12 bits of the value of 'version' field represent
firmware version, while the lower 12 bits is unknown.
By induction, the value of 'version' field represents actual model.
This commit changes modalias to match the value of 'version' field,
instead of 'model' field. For degug, long name of added sound card
includes hexadecimal value of 'model' field.
Fixes: 6c5e1ac0e1 ("ALSA: firewire-motu: add support for Motu Traveler")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # v4.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case request_region fails, the fix returns an error code to
avoid NULL pointer dereference.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case ioremap_nocache fails, the fix releases chip and returns
an error code upstream to avoid NULL pointer dereference.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some gleaning after the first batch; mostly about HD-audio quirks but
also some NULL dereference fixes in corner cases and a random build
error fix, too.
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Merge tag 'sound-fix-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Some cleaning after the first batch; mostly about HD-audio quirks but
also some NULL dereference fixes in corner cases and a random build
error fix, too"
* tag 'sound-fix-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Add support headset mode for New DELL WYSE NB
ALSA: hda/realtek - Add support headset mode for DELL WYSE AIO
ALSA: hda/realtek: merge alc_fixup_headset_jack to alc295_fixup_chromebook
ALSA: pcm: Fix function name in kernel-doc comment
ALSA: hda: hdmi - add Icelake support
ALSA: hda - add more quirks for HP Z2 G4 and HP Z240
ALSA: hda/realtek - Fixed Headset Mic JD not stable
ALSA: hda/realtek: Enable headset MIC of Acer TravelMate X514-51T with ALC255
ALSA: hda/tegra: avoid build error without CONFIG_PM
ALSA: usx2y: Fix potential NULL pointer dereference
ALSA: hda: Avoid NULL pointer dereference at snd_hdac_stream_start()
Component driver may want to use tlv data. Create tlv before
soc_tplg_init_kcontrol so component driver can use the tlv data
in the control_load ops.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ALC225_FIXUP_HEADSET_JACK fixup can be merged to alc295_fixup_chromebook.
There are no other users for ALC225_FIXUP_HEADSET_JACK other than
the chromebook hardware.
Fixes: 10f5b1b85e ("ALSA: hda/realtek - Fixed Headset Mic JD not stable")
Cc: Kailang Yang <kailang@realtek.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is just a port of the ASoC Icelake HDMI codec code to the legacy
HDA driver with some cleanups.
ASoC commit 019033c854a20e10f691f6cc0e897df8817d9521:
"ASoC: Intel: hdac_hdmi: add Icelake support"
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: Bard liao <bard.liao@intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver will select correct BCLK automatically according to
BCLK and FS information in I2S master mode.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After commit fbeec965b8 ("ASoC: samsung: odroid: Fix 32000 sample rate
handling") the audio root clock frequency is configured improperly for
44100 sample rate. Due to clock rate rounding it's 20070401 Hz instead
of 22579000 Hz. This results in a too low value of the PSR clock divider
in the CPU DAI driver and too fast actual sample rate for fs=44100. E.g.
1 kHz tone has actual 1780 Hz frequency (1 kHz * 20070401/22579000 * 2).
Fix this by increasing the correction passed to clk_set_rate() to take
into account inaccuracy of the EPLL frequency properly.
Fixes: fbeec965b8 ("ASoC: samsung: odroid: Fix 32000 sample rate handling")
Reported-by: JaeChul Lee <jcsing.lee@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver changes the stream name of DAC and ADC to avoid the issue of
widget with prefixed name. When the machine adds prefixed name for codec,
the stream name of DAI may not find the widgets.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Apply the HP_MIC_NO_PRESENCE fixups for the more HP Z2 G4 and
HP Z240 models.
Reported-by: Jeff Burrell <jeff.burrell@hp.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It will be lose Mic JD state when Chrome OS boot and headset was plugged.
Implement of reset combo jack JD. It will show normally.
Fixes: e854747d75 ("ALSA: hda/realtek - Enable headset button support for new codec")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer TravelMate X514-51T with ALC255 cannot detect the headset MIC
until ALC255_FIXUP_ACER_HEADSET_MIC quirk applied. Although, the
internal DMIC uses another module - snd_soc_skl as the driver. We still
need the NID 0x1a in the quirk to enable the headset MIC.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The #ifdef protection around the PM functions is wrong, leading to
a failed reference in some configurations:
sound/pci/hda/hda_tegra.c: In function 'hda_tegra_runtime_suspend':
sound/pci/hda/hda_tegra.c:273:2: error: implicit declaration of function 'hda_tegra_disable_clocks'; did you mean 'hda_tegra_enable_clocks'? [-Werror=implicit-function-declaration]
Better remove the #ifdefs entirely and rely on the compiler silently
dropping unused functions marked __maybe_unused.
Fixes: 707e0759f2 ("ALSA: hda/tegra: implement runtime suspend/resume")
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
usb_alloc_urb() can fail due to kmalloc failure and push the error
upstream. Further this can cause a NULL pointer dereference in
init_pipe_urbs(). This patch avoids such a scenario.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both the capture and playback channels are optional in the axi_i2s IP
block. Reflect this in the driver by enabling only the channel(s) that
have a DMA.
Signed-off-by: Luca Ceresoli <luca@lucaceresoli.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
compiler complains about following declarations
sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern1[] = {
^~~~~
sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern2[] = {
^~~~~
sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsisr_table[] = {
^~~~~
sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan288888[] = {
^~~~~
sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan244888[] = {
^~~~~
sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan222222[] = {
^~~~~
This patch moves the 'static' keyword to the front of the
declaration to fix the compiler warnings
Fixes: linux-next commit 7674bec4fc ("ASoC: rsnd: update BSDSR/BSDISR handling")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
lockdep warns us that priv->lock and k->k_lock can cause a
deadlock when after acquire of k->k_lock, process is interrupted
by src, while in another routine of src .init, k->k_lock is
acquired with priv->lock held.
This patch avoids a potential deadlock by not calling soc_device_match()
in SRC .init callback, instead it adds new soc fields in priv->flags to
differentiate SoCs.
Fixes: linux-next commit 7674bec4fc ("ASoC: rsnd: update BSDSR/BSDISR handling")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
- Declare SR as volatile, as it is changed by hardware.
- Remove TXDR from readable and volatile register list,
as it is intended for write accesses only.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver has two issues when machine add prefix name for codec.
(1)The stream name of DAI can't find the AIF widgets.
(2)The drivr can enable/disalbe the MICBIAS and SAR widgets.
The patch will fix these issues caused by prefixed name added.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dpcm get from fe_clients/be_clients
may be free before use
Add a spin lock at snd_soc_card level,
to protect the dpcm instance.
The lock may be used in atomic context, so use spin lock.
Use irq spin lock version,
since the lock may be used in interrupts.
possible race condition between
void dpcm_be_disconnect(
...
list_del(&dpcm->list_be);
list_del(&dpcm->list_fe);
kfree(dpcm);
...
and
for_each_dpcm_fe()
for_each_dpcm_be*()
race condition example
Thread 1:
snd_soc_dapm_mixer_update_power()
-> soc_dpcm_runtime_update()
-> dpcm_be_disconnect()
-> kfree(dpcm);
Thread 2:
dpcm_fe_dai_trigger()
-> dpcm_be_dai_trigger()
-> snd_soc_dpcm_can_be_free_stop()
-> if (dpcm->fe == fe)
Excpetion Scenario:
two FE link to same BE
FE1 -> BE
FE2 ->
Thread 1: switch of mixer between FE2 -> BE
Thread 2: pcm_stop FE1
Exception:
Unable to handle kernel paging request at virtual address dead0000000000e0
pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
sound/soc/soc-pcm.c:3226
if (dpcm->fe == fe)
lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
Backtrace:
[<ffffff89602dba80>] notify_die+0x68/0xb8
[<ffffff896028c7dc>] die+0x118/0x2a8
[<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c
[<ffffff89602a27f4>] do_translation_fault+0x64/0xa0
[<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0
[<ffffff8960282ad0>] el1_da+0x24/0x40
[<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c
[<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c
[<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44
[<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c
[<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c
[<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128
[<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0
[<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14
[<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244
[<ffffff8960283740>] el0_svc_naked+0x34/0x38
[<ffffffffffffffff>] 0xffffffffffffffff
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If playback and capture are enabled concurrently, when the capture stops
the output becomes inaudile. The playback application will become stuck
and underrun after a timeout.
This is caused by mistaken use of the stream_id, which should only be
set for playback and not for capture
Tested on Apollolake and Kabylake with SST driver.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current implementation of the hdac_hda codec results in zero-valued
samples on capture and noise with headset playback when SOF is used on
platforms with an on-board HDaudio codec. This is root-caused to SOF
using be_hw_params_fixup, and the prepare() call using invalid runtime
fields to determine the format.
This patch moves the format handling to the hw_params() callback, as
done already for hdac_hdmi, to make sure the fixed-up information is
taken into account but keeps the codec initialization in prepare() as
the stream_tag is only available at that time. Moving everything in the
prepare() callback is possible but the code is less elegant so this
two-step solution was chosen.
The solution was tested with the SST driver with no regressions, and all
the issues with SOF playback and capture are solved.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On HDaudio platforms, if playback is started when capture is working,
there is no audible output.
This can be root-caused to the use of the rx|tx_mask to store an HDaudio
stream tag.
If capture is stared before playback, rx_mask would be non-zero on HDaudio
platform, then the channel number of playback, which is in the same codec
dai with the capture, would be changed by soc_pcm_codec_params_fixup based
on the tx_mask at first, then overwritten by this function based on rx_mask
at last.
According to the author of tx|rx_mask, tx_mask is for playback and rx_mask
is for capture. And stream direction is checked at all other references of
tx|rx_mask in ASoC, so here should be an error. This patch checks stream
direction for tx|rx_mask for fixup function.
This issue would affect not only HDaudio+ASoC, but also I2S codecs if the
channel number based on rx_mask is not equal to the one for tx_mask. It could
be rarely reproduecd because most drivers in kernel set the same channel number
to tx|rx_mask or rx_mask is zero.
Tested on all platforms using stream_tag & HDaudio and intel I2S platforms.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SND_SOC_DAVINCI_MCASP driver can use either edma or sdma as
a back-end, and it takes the presence of the respective dma engine
drivers in the configuration as an indication to which ones should be
built. However, this is flawed in multiple ways:
- With CONFIG_TI_EDMA=m and CONFIG_SND_SOC_DAVINCI_MCASP=y,
is enabled as =m, and we get a link error:
sound/soc/ti/davinci-mcasp.o: In function `davinci_mcasp_probe':
davinci-mcasp.c:(.text+0x930): undefined reference to `edma_pcm_platform_register'
- When CONFIG_SND_SOC_DAVINCI_MCASP=m has already been selected by
another driver, the same link error appears even if CONFIG_TI_EDMA
is disabled
There are possibly other issues here, but it seems that the only reasonable
solution is to always build both SND_SOC_TI_EDMA_PCM and
SND_SOC_TI_SDMA_PCM as a dependency here. Both are fairly small and
do not have any other compile-time dependencies, so the cost is
very small, and makes the configuration stage much more consistent.
Fixes: f2055e145f ("ASoC: ti: Merge davinci and omap directories")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
clang points out that SOC_ENUM_SINGLE_EXT_DECL() contains a 'const'
modifier already, so adding another one does not make it more const:
sound/soc/ti/ams-delta.c:203:14: error: duplicate 'const' declaration specifier [-Werror,-Wduplicate-decl-specifier]
static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum,
^
include/sound/soc.h:351:2: note: expanded from macro 'SOC_ENUM_SINGLE_EXT_DECL'
const struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts)
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After running into a link error:
sound/soc/ti/edma-pcm.o:(.rodata+0x18): undefined reference to `edma_filter_fn'
I checked all users of this, and they have new-style 'dma_slave_map' tables,
so none of them should still need it. Removing the associated lines
simplifies the code and avoids the build-time dependency on the
respective dmaengine drivers.
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use alsa snd_pcm_hw_constraint_single service to manage
channels restriction. This provides better status on driver
limitations, to the application.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Update traces to log capture/playback stream start/stop.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Building with clang shows a variable that is only used by the
suspend/resume functions but defined outside of their #ifdef block:
sound/soc/ti/davinci-mcasp.c:48:12: error: variable 'context_regs' is not needed and will not be emitted
We commonly fix these by marking the PM functions as __maybe_unused,
but here that would grow the davinci_mcasp structure, so instead
add another #ifdef here.
Fixes: 1cc0c054f3 ("ASoC: davinci-mcasp: Convert the context save/restore to use array")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
compiler complains about following declarations
sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern1[] = {
^~~~~
sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsdsr_table_pattern2[] = {
^~~~~
sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 bsisr_table[] = {
^~~~~
sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan288888[] = {
^~~~~
sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan244888[] = {
^~~~~
sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration]
const static u32 chan222222[] = {
^~~~~
This patch moves the 'static' keyword to the front of the
declaration to fix the compiler warnings
Fixes: linux-next commit 7674bec4fc ("ASoC: rsnd: update BSDSR/BSDISR handling")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch sets missing stream_name of capture part of the DAI driver
so we can define DAPM routing properly also for the capture stream.
While at it "Playback" suffix is added to the playback stream names
to clearly identify playback/capture.
Together with related dts patch this fixes NULL pointer dereference
when opening ALSA device for recording on Odroid XU3.
Fixes: 64aba9bca5 ("ASoC: samsung: i2s: Add widgets and routes for DPCM support")
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Here is the big char/misc driver patch pull request for 5.1-rc1.
The largest thing by far is the new habanalabs driver for their AI
accelerator chip. For now it is in the drivers/misc directory but will
probably move to a new directory soon along with other drivers of this
type.
Other than that, just the usual set of individual driver updates and
fixes. There's an "odd" merge in here from the DRM tree that they asked
me to do as the MEI driver is starting to interact with the i915 driver,
and it needed some coordination. All of those patches have been
properly acked by the relevant subsystem maintainers.
All of these have been in linux-next with no reported issues, most for
quite some time.
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Merge tag 'char-misc-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/char-misc
Pull char/misc driver updates from Greg KH:
"Here is the big char/misc driver patch pull request for 5.1-rc1.
The largest thing by far is the new habanalabs driver for their AI
accelerator chip. For now it is in the drivers/misc directory but will
probably move to a new directory soon along with other drivers of this
type.
Other than that, just the usual set of individual driver updates and
fixes. There's an "odd" merge in here from the DRM tree that they
asked me to do as the MEI driver is starting to interact with the i915
driver, and it needed some coordination. All of those patches have
been properly acked by the relevant subsystem maintainers.
All of these have been in linux-next with no reported issues, most for
quite some time"
* tag 'char-misc-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/char-misc: (219 commits)
habanalabs: adjust Kconfig to fix build errors
habanalabs: use %px instead of %p in error print
habanalabs: use do_div for 64-bit divisions
intel_th: gth: Fix an off-by-one in output unassigning
habanalabs: fix little-endian<->cpu conversion warnings
habanalabs: use NULL to initialize array of pointers
habanalabs: fix little-endian<->cpu conversion warnings
habanalabs: soft-reset device if context-switch fails
habanalabs: print pointer using %p
habanalabs: fix memory leak with CBs with unaligned size
habanalabs: return correct error code on MMU mapping failure
habanalabs: add comments in uapi/misc/habanalabs.h
habanalabs: extend QMAN0 job timeout
habanalabs: set DMA0 completion to SOB 1007
habanalabs: fix validation of WREG32 to DMA completion
habanalabs: fix mmu cache registers init
habanalabs: disable CPU access on timeouts
habanalabs: add MMU DRAM default page mapping
habanalabs: Dissociate RAZWI info from event types
misc/habanalabs: adjust Kconfig to fix build errors
...
Commit 78a24e10cd ("ASoC: soc-core: clear platform pointers on error")
re-worked the clean-up of any platform pointers that may have been
initialised by the function snd_soc_init_platform(). This commit missed
one error path where if any of the prelinks for a soundcard failed to
initialise, then these platform pointers would not be cleaned-up. This
then prevents the soundcard from being initialised following a probe
deferral when any of the soundcard prelinks cannot be found.
Fix this by ensuring that soc_cleanup_platform() is called when
initialising the soundcard prelinks fails.
Fixes: 78a24e10cd ("ASoC: soc-core: clear platform pointers on error")
Signed-off-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Limiting the value of the passed in params->msbits in the hw_params()
callback is redundant on three counts:
1. We already specify in the DAI driver that we can only handle up to
24 bits. This means msbits will be limited to 24 via the ALSA
constraints imposed by the ASoC core, unless we have multiple codecs
that can handle more bits.
2. Nothing in our hw_params() implementation uses this value.
3. The copy of the params that we are passed by the ASoC core never
reads back the msbits value.
Consequently, this code is unnecessary and does nothing useful. Remove
it.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add error check on set_sync function return.
Add of_node_put() as of_get_parent() takes a reference
which has to be released.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_pcm_stop_xrun() is called in interrupt routine,
substream context may have already been released.
Add protection on substream context.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change capabilities exposed in SAI S/PDIF mode, to match
actually supported formats.
In S/PDIF mode only 32 bits stereo is supported.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow indexation of sai iec958 controls according
to device id.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to enabling -Wimplicit-fallthrough, mark switch
cases where we are expecting to fall through.
This patch fixes the following warning:
In file included from sound/soc/codecs/ab8500-codec.c:24:
sound/soc/codecs/ab8500-codec.c: In function ‘ab8500_codec_set_dai_fmt’:
./include/linux/device.h:1485:2: warning: this statement may fall through [-Wimplicit-fallthrough=]
_dev_err(dev, dev_fmt(fmt), ##__VA_ARGS__)
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/ab8500-codec.c:2129:3: note: in expansion of macro ‘dev_err’
dev_err(dai->component->dev,
^~~~~~~
sound/soc/codecs/ab8500-codec.c:2132:2: note: here
default:
^~~~~~~
Warning level 3 was used: -Wimplicit-fallthrough=3
This patch is part of the ongoing efforts to enable
-Wimplicit-fallthrough.
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When using the S/PDIF DAI, there is no requirement to call
snd_soc_dai_set_fmt() as there is no DAI format definition that defines
S/PDIF. In any case, S/PDIF does not have separate clocks, this is
embedded into the data stream.
Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt
to configure TDA998x via the hw_params callback fails as the
hdmi_codec_daifmt is left initialised to zero.
Since the S/PDIF DAI will only be used by S/PDIF, prepare the
hdmi_codec_daifmt structure for this format.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add an entry to the quirks-table to for usb-audio to recognize the
Microbook II (although it only exposes vendor interfaces). A simple boot
quirk is also implemented to set up the sample rate and make sure that
no audio urbs are sent before the device is ready.
This patch only provides audio playback and capture at 96kHz sample
rate. Notice the following shortcomings:
- The sample rate is currently hardcoded to 96k although the device also
supports 48k and 44.1k.
- The various mixer controls of the MicroBook are not made available.
- The keep-iface control should be on by default because the device
shuts down whenever the altsetting is reset which is usually unwanted.
(I don't know the best way to do this)
- The communication format used by the MicroBook for sample rate setting
and also other setup has been reverse engineered by looking at the
usbmon output while running the windows driver through virtualbox. In
this patch the first byte of every message is set to \0 while in the
observed communications the first byte acts as a "message-counter"
increasing its value with every message sent. Leaving it at \0 does
not seem to affect the device.
Signed-off-by: Manuel Reinhardt <manuel.rhdt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes a bug that prevents freeing the reset gpio on unloading
the module.
aic3x_i2c_probe is called when loading the module and it calls list_add
with a probably uninitialized list entry aic3x->list (next = prev = NULL)).
So even if list_del is called it does nothing and in the end the gpio_reset
is not freed. Then a repeated module probing fails silently because
gpio_request fails.
When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move
list_del to aic3x_i2c_remove because aic3x_remove may be called
multiple times without aic3x_i2c_remove being called which leads to
a NULL pointer dereference.
Signed-off-by: Philipp Puschmann <philipp.puschmann@emlix.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Another batch of changes for ASoC, no big core changes - it's mainly
small fixes and improvements for individual drivers.
- A big refresh and cleanup of the Samsung drivers, fixing a number of
issues which allow the driver to be used with a wider range of
userspaces.
- Fixes for the Intel drivers to make them more standard so less likely
to get bitten by core issues.
- New driver for Cirrus Logic CS35L26.
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Merge tag 'asoc-v5.1-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More changes for v5.1
Another batch of changes for ASoC, no big core changes - it's mainly
small fixes and improvements for individual drivers.
- A big refresh and cleanup of the Samsung drivers, fixing a number of
issues which allow the driver to be used with a wider range of
userspaces.
- Fixes for the Intel drivers to make them more standard so less likely
to get bitten by core issues.
- New driver for Cirrus Logic CS35L26.
Dummy write in capture master mode is used to gate
bus clocks. This write is useless in slave mode
as the clocks are not managed by slave.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_pcm_stop_xrun() is called in interrupt routine,
substream context may have already been released.
Add protection on substream context.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Clocks do not need to be released on driver removal,
as this is already managed before.
Remove useless remove callback.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DMA configuration is not balanced on start/stop.
Move DMA configuration to trigger callback.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move counter handling to trigger start section
to manage multiple start/stop events.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
I2S supports 16 bits data in 32 channel length.
However the expected driver behavior, is to
set channel length to 16 bits when data format is 16 bits.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because of regmap cache, interrupts may not be cleared
as expected.
Declare IFCR register as write only and make writings
to IFCR register unconditional.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_CROS_EC_CODEC depends on MFD_CROS_EC.
Add that dependency to SND_SOC_SDM845 to fix unmet direct dependencies
warning.
Fixes: 74c6ecf419 (ASoC: qcom: Kconfig: select dmic for sdm845)
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Tested-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Tested-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch enables the reuse of kbl_da7219_max98927 machine driver to
support max98373. The same machine driver is modified for cases where one
amplifier is swapped out with another. Most of the changes are about
renaming the codec and codec_dai names, with minor differences due to
support for 24 bits in one case and 16 in the other.
Signed-off-by: Jenny TC <jenny.tc@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently each SSI unit 's busif mode/adinr/dalign address is
registered by: (in busif4 case)
RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80)
RSND_GEN_M_REG(SSI_BUSIF4_ADINR,0x504, 0x80)
RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80)
But according to user manual 41.1.4 Register Configuration
ssi9 4/5/6/7 busif mode/adinr/dalign register address
( SSI9-[4/5/6/7]_BUSIF_[MODE/ADINR/DALIGN] )
are out of this rule.
This patch registers ssi9 4/5/6/7 mode/adinr/dalign register
as single register, and access these registers in case of
SSI9 BUSIF 4/5/6/7.
Fixes: commit 8c9d750333 ("ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In data blocks of common isochronous packet for MOTU devices, PCM
frames are multiplexed in a shape of '24 bit * 4 Audio Pack', described
in IEC 61883-6. The frames are not aligned to quadlet.
For capture PCM substream, ALSA firewire-motu driver constructs PCM
frames by reading data blocks byte-by-byte. However this operation
includes bug for lower byte of the PCM sample. This brings invalid
content of the PCM samples.
This commit fixes the bug.
Reported-by: Peter Sjöberg <autopeter@gmail.com>
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 4641c93940 ("ALSA: firewire-motu: add MOTU specific protocol layer")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I set 10 seconds for the timeout of the i915 audio component binding
with a hope that recent machines are fast enough to handle all probe
tasks in that period, but I was too optimistic. The binding may take
longer than that, and this caused a problem on the machine with both
audio and graphics driver modules loaded in parallel, as Paul Menzel
experienced. This problem haven't hit so often just because the KMS
driver is loaded in initrd on most machines.
As a simple workaround, extend the timeout to 60 seconds.
Fixes: f9b54e1961 ("ALSA: hda/i915: Allow delayed i915 audio component binding")
Reported-by: Paul Menzel <pmenzel+alsa-devel@molgen.mpg.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the compressed stream implementation has acquired support for
multiple DAI links and compressed streams it has become harder to
interpret messages in the kernel log. Add additional macros to include
the compressed DAI name in the log messages, allowing different streams
to be easily disambiguated.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, only a single compressed stream is supported per firmware.
Add support for multiple compressed streams on a single firmware, this
allows additional features like completely independent trigger words or
separate debug capture streams to be implemented.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make the code slightly clearer and prepare things for the addition of
multiple compressed streams on a single DSP core.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for more refactoring add a helper function to strip the
padding from ADSP data.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The irq_get_irq_data() function doesn't return error pointers, it
returns NULL.
Fixes: 6ba9dd6c89 ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A platform can have multiple sound cards for different audio paths.
Following is the print seen duirng device boot for jetson-xavier,
ALSA device list:
#0: nvidia,p2972-0000 at 0x3518000 irq 17
By looking at above, it is not very clear if the sound card is for
HDA. It becomes confusing when platform has registered multiple cards,
and platform model name is used for card.
This patch uses "nvidia,model" property mentioned in hda device tree
to get the card name. Since property is optional, legacy boards will
continue to use "tegra-hda". Custom name can be passed wherever needed.
This naming convention is conistent with the way sound cards are named
in general.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS UX362FA with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone. This issue can be fixed
by the quirk in the commit 4e0511067 ALSA: hda/realtek: Enable audio
jacks of ASUS UX533FD with ALC294.
Besides, ASUS UX362FA and UX533FD have the same audio initial pin config
values. So, this patch replaces SND_PCI_QUIRK of UX533FD with a new
SND_HDA_PIN_QUIRK which benefits both UX362FA and UX533FD.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Ming Shuo Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This addresses an issue pointed out by compiler warning:
sound/soc/samsung/odroid.c: In function ‘odroid_audio_probe’:
sound/soc/samsung/odroid.c:298:22: warning: ‘cpu_dai’ may be used
uninitialized in this function [-Wmaybe-uninitialized]
priv->clk_i2s_bus = of_clk_get_by_name(cpu_dai, "iis");
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need 32ea33a044 ("mei: bus: export to_mei_cl_device for mei
client devices drivers") for the mei-hdcp patches.
References: https://lkml.org/lkml/2019/2/19/356
Signed-off-by: Daniel Vetter <daniel.vetter@intel.com>
Here are a few last-minute fixes for 5.0. The most significant one
is the OF-node refcount fix for ASoC simple-card, which could be
triggered on many boards. Another fix for ASoC core is for the
error handling in topology, while others are device-specific fixes
for Samsung and HD-audio.
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Merge tag 'sound-5.0' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are a few last-minute fixes for 5.0.
The most significant one is the OF-node refcount fix for ASoC
simple-card, which could be triggered on many boards. Another fix for
ASoC core is for the error handling in topology, while others are
device-specific fixes for Samsung and HD-audio"
* tag 'sound-5.0' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: simple-card: fixup refcount_t underflow
ASoC: topology: free created components in tplg load error
ALSA: hda/realtek: Disable PC beep in passthrough on alc285
ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5
ASoC: samsung: i2s: Fix prescaler setting for the secondary DAI
Although qcom_snd_parse_of() tries to manage the of-node refcount,
there are still a few places that lead to the unblanced refcount in
the error code path. Namely,
- for_each_child_of_node() needs to unreference the iterator node if
aborting the loop in the middle,
- cpu, codec and platform node objects have to be unreferenced at each
iteration,
- platform and codec node objects have to be referred before jumping
to the error handling code that unreference them unconditionally.
This patch tries to address these by moving the assignment of platform
and codec node objects to the beginning of the loop and adding the
of_node_put() calls adequately.
Fixes: c25e295cd7 ("ASoC: qcom: Add support to parse common audio device nodes")
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The apq8016 driver leaves the of-node refcount at aborting from the
loop of for_each_child_of_node() in the error path. Not only the
iterator node of for_each_child_of_node(), the children nodes referred
from it for codec and cpu have to be properly unreferenced.
Fixes: bdb052e81f ("ASoC: qcom: add apq8016 sound card support")
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
In odroid_audio_probe() some OF nodes are left without reference count
decrease after use. Fix it by ensuring required of_node_calls() are done
before exiting probe.
Reported-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Dell Precision 5820 with ALC3234 codec (which is equivalent with
ALC255) shows click noises at (runtime) PM resume on the headphone.
The biggest source of the noise comes from the cleared headphone pin
control at resume, which is done via the standard shutup procedure.
Although we have an override of the standard shutup callback to
replace with NOP, this would skip other needed stuff (e.g. the pull
down of headset power). So, instead, this "fixes" the behavior of
alc_fixup_no_shutup() by introducing spec->no_shutup_pins flag.
When this flag is set, Realtek codec won't call the standard
snd_hda_shutup_pins() & co. Now alc_fixup_no_shutup() just sets this
flag instead of overriding spec->shutup callback itself. This allows
us to apply the similar fix for other entries easily if needed in
future.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function simple_for_each_link() has a few missing places that
forgot unrefereing of-nodes after the use. The main do-while loop
may abort when loop=0, and this leaves the node object still
referenced. A similar leak is found in the error handling of NULL
codec that aborts the loop as well. Last but not least, the inner
for_each_child_of_node() loop may abort in the middle, and this leaks
the refcount of the iterator node.
This patch addresses these missing refcount issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We forgot to unreference the platform node object obtained from
of_get_child_by_name(). This leads to the unbalance of node
refcount.
Fixes: e0ae225b7e ("ASoC: simple-card: support platform in dts parse")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The node obtained from of_find_node_by_path() has to be unreferenced
after the use, but we forgot it for the root node.
Fixes: f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component Support")
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, compressed buffers can only be specified in the XM memory
region. There is no reason to have such a restriction with the newer
meta-data based way of specifying the buffers, so remove it.
Signed-off-by: Andrew Ford <aford@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a compressed stream is restarted after getting an error, the cached
error value will still be used on the next pointer request, preventing
the stream from starting. Resolve this by ensuring the error status is
updated on trigger start.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Also contains the prep work in the component helpers plus adjustements
for the snd-hda/i915 component interface.
Plus one small static inline in the drm_hdcp.h header that both i915
and mei_hdcp will need.
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Merge tag 'topic/mei-hdcp-2019-02-19' of git://anongit.freedesktop.org/drm/drm-intel into drm-intel-next-queued
Prep patches + headers for the mei-hdcp/i915 component interfaces
Also contains the prep work in the component helpers plus adjustements
for the snd-hda/i915 component interface.
Plus one small static inline in the drm_hdcp.h header that both i915
and mei_hdcp will need.
Signed-off-by: Joonas Lahtinen <joonas.lahtinen@linux.intel.com>
From: Daniel Vetter <daniel.vetter@ffwll.ch>
Link: https://patchwork.freedesktop.org/patch/msgid/20190219071619.GA11016@phenom.ffwll.local
We forgot to unreference the node when aborting from the loop of
for_each_child_of_node() in snd_pmac_tumbler_init(). This leads to
unbalanced node refcount. Fix it by adding the missing of_node_put()
call.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We forgot to unreference a node obtained via of_find_node_by_name()
after its usage.
Reviewed-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ac97_of_get_child_device() take the refcount of the node explicitly
via of_node_get(), but this leads to an unbalance. The
for_each_child_of_node() loop itself takes the refcount for each
iteration node, hence you don't need to take the extra refcount
again.
Fixes: 2225a3e6af ("ALSA: ac97: add codecs devicetree binding")
Reviewed-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ADCs are sleeping when the SLEEP bit is set and running when it's
cleared, so the bit should be inverted.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
On some SoCs (e.g. Exynos5433) there are multiple "IIS multi audio
interfaces" and the driver will try to register there multiple times
same platform device for the secondary FIFO, which of course fails
miserably. To fix this we derive the secondary platform device name
from the primary device name. The secondary device name will now
be <primary_dev_name>-sec instead of fixed "samsung-i2s-sec".
The fixed platform_device_id table entry is removed as the secondary
device name is now dynamic and device/driver matching is done through
driver_override.
Reported-by: Marek Szyprowski <m.szyprowski@samsung.com>
Suggested-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This fixes unregistration of the secondary platform device so all
resources are properly released. Additionally the removal sequence
is corrected so it is in reverse order comparing to probe sequence.
The test against NULL priv->pdev_sec is removed as it is not necessary.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add FE DAI link to support parallel playback on 2 ports
simultaneously.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch enables support for GeminiLake with the DA7219 codec and
MAX98357A amplifier. To avoid duplicating code, the existing machine
driver for ApolloLake is reused with only changes in hardware
connectivity (SSP2 for DA7219 and SSP1 for MAX98357A).
The dailinks are directly modified in this patch. Using a helper would
be nicer, but it'll be done in a follow-up step with validation done
across multiple machine drivers.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A64 datasheet lists the supply rail for the headphone amp's charge
pump as "CPVDD". cpvdd-supply is the name of the property for this power
rail specified in the device tree bindings. "HPVCC" was the name used in
the A33 datasheet for the same function.
Rename the supply so it matches the datasheet, bindings, and the subject
from the original commit.
Fixes: ca0412a057 ("ASoC: sunxi: sun50i-codec-analog: Add support for cpvdd regulator supply")
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
If platform_data is NULL add reading of optional adi,micbias
property from DT. If adi,micbias is not set keep the default
value for micbias.
Signed-off-by: Bogdan Togorean <bogdan.togorean@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should free "w" on the error path.
Fixes: 199ed3e81c ("ASoC: dapm: fix use-after-free issue with dailink sname")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With old DTS there will be missing DAPM routes linking BE with CODECs.
Add those routes in the card driver so sound works properly on Odroid
XU3/4 also without DTS updates enabling the secondary PCM.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following sparse warning:
sound/soc/codecs/wm8741.c:371:5: warning:
symbol 'wm8741_mute' was not declared. Should it be static?
Fixes: 36b1599340 ("ASoC: wm8741: Add digital mute callback")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to
wrong register, correct it.
Fixes 43d24e76b6 ("ASoC: fsl_esai: Add ESAI CPU DAI driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Ackedy-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A few small fixes, a driver fix for Samsung, a fix for refcounting of
of_nodes in the simple-card driver that triggered on a lot of systems
and a fix for topology error handling.
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Merge tag 'asoc-fix-v5.0-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
A few small fixes, a driver fix for Samsung, a fix for refcounting of
of_nodes in the simple-card driver that triggered on a lot of systems
and a fix for topology error handling.
The of_find_device_by_node() takes a reference to the underlying device
structure, we should release that reference.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device;
call of_find_device_by_node on line 105, but without a corresponding
object release within this function.
./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device;
call of_find_device_by_node on line 105, but without a corresponding
object release within this function.
Signed-off-by: Wen Yang <yellowriver2010@hotmail.com>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Shawn Guo <shawnguo@kernel.org>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Pengutronix Kernel Team <kernel@pengutronix.de>
Cc: NXP Linux Team <linux-imx@nxp.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-arm-kernel@lists.infradead.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
commit da215354eb ("ASoC: simple-card: merge simple-scu-card")
merged simple-card and simple-scu-card. Then it had refcount
underflow bug. This patch fixup it.
We will get below error without this patch.
OF: ERROR: Bad of_node_put() on /sound
CPU: 3 PID: 237 Comm: kworker/3:1 Not tainted 5.0.0-rc6+ #1514
Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT)
Workqueue: events deferred_probe_work_func
Call trace:
dump_backtrace+0x0/0x150
show_stack+0x24/0x30
dump_stack+0xb0/0xec
of_node_release+0xd0/0xd8
kobject_put+0x74/0xe8
of_node_put+0x24/0x30
__of_get_next_child+0x50/0x70
of_get_next_child+0x40/0x68
asoc_simple_card_probe+0x604/0x730
platform_drv_probe+0x58/0xa8
...
Reported-by: Vicente Bergas <vicencb@gmail.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_get_child_by_name() takes a reference we'll need to drop
later so when we substitute in top we need to take a reference
as well as just assigning.
Without this patch we hit the following error:
[ 1.246852] OF: ERROR: Bad of_node_put() on /sound-wm8524
[ 1.262261] Hardware name: NXP i.MX8MQ EVK (DT)
[ 1.266807] Workqueue: events deferred_probe_work_func
[ 1.271950] Call trace:
[ 1.274406] dump_backtrace+0x0/0x158
[ 1.278074] show_stack+0x14/0x20
[ 1.281396] dump_stack+0xa8/0xcc
[ 1.284717] of_node_release+0xb0/0xc8
[ 1.288474] kobject_put+0x74/0xf0
[ 1.291879] of_node_put+0x14/0x28
[ 1.295286] __of_get_next_child+0x44/0x70
[ 1.299387] of_get_next_child+0x3c/0x60
[ 1.303315] simple_for_each_link+0x1dc/0x230
[ 1.307676] simple_probe+0x80/0x540
[ 1.311256] platform_drv_probe+0x50/0xa0
This patch is based on an earlier version posted by Kuninori Morimoto
and commit message includes explanations from Mark Brown.
https://patchwork.kernel.org/patch/10814255/
Reported-by: Vicente Bergas <vicencb@gmail.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when playing sound with different sample rates actual
sample rate will be determined by audio stream which starts first
on either primary or secondary PCM. The audio root clock will be
configured appropriately only for the first stream. As the hardware
is limited to same sample rate on both interfaces we need to disallow
streams with different sample rates. It is done by this patch by
returning error in FE hw_params if there is already active stream
running with different sample rate.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_configure_clock':
sound/soc/stm/stm32_sai_sub.c:902:11: warning:
variable 'mask' set but not used [-Wunused-but-set-variable]
sound/soc/stm/stm32_sai_sub.c:902:6: warning:
variable 'cr1' set but not used [-Wunused-but-set-variable]
It's not used any more after 8307b2afd3 ("ASoC: stm32: sai: set sai as
mclk clock provider")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology resources are no longer needed if any element failed to load.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following sparse warnings:
sound/soc/codecs/cs35l36.c:135:20: warning:
symbol 'cs35l36_reg' was not declared. Should it be static?
sound/soc/codecs/cs35l36.c:248:6: warning:
symbol 'cs35l36_readable_reg' was not declared. Should it be static?
sound/soc/codecs/cs35l36.c:398:6: warning:
symbol 'cs35l36_precious_reg' was not declared. Should it be static?
sound/soc/codecs/cs35l36.c:410:6: warning:
symbol 'cs35l36_volatile_reg' was not declared. Should it be static?
Fixes: 6ba9dd6c89 ("ASoC: cs35l36: Add support for Cirrus CS35L36 Amplifier")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Acked-by: James Schulman <james.schulman@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sdm845 uses dmic on EC so it should select CROS_EC_CODEC.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to RM SPDIF STC SYSCLK_DF field is 9-bit wide, values
being in 0..511 range. Use a proper type to handle sysclk_df.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The on-chip PLL can be disabled if on the MCLKI pin we have an external
clock at 512 x fs. This clock can be used as direct internal clock for
ADCs or DACs.
To support this, we add an extra clock id that can be configured
using the set_sysclk() callback.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver only supports DPS_A for DAC, which is configured at probe.
This patch adds support for DSP_A and I2S modes by using the set_fmt()
callback.
A trivial break is also removed from a case's default branch.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
By default, the codec starts to interpret the left (first) channel on
the falling edge (low polarity) of LRCLK. However, for DSP_A, the left
channel needs to start on the rising edge of LRCLK. This patch fixes
this channel swap by toggling the bit which selects the LRCLK polarity.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DACs and ADCs on ad193x codecs require a 32 bit slot size. We should
assure that no other size is used.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some ad193x codecs don't have ADCs, so they have no capture capabilities.
This way, we can use this driver in multicodec cards.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in a dev_err message. Fix it.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
At least some USB devices use (MSB-aligned) audio format larger
than the actual resolution of the device. In order to expose the
actual device resolution (bBitResolution), add extra field to the
procfs stream info interface.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
drm/i915 is tracking all wakeref owners with a cookie in order to
identify leaks. To that end, each rpm acquisition ops->get_power is
assigned a cookie which should be passed to ops->put_power to signify
its release (and removal from the list of wakeref owners). As snd/hda is
already using a bool to track current status of display_power extending
that to an unsigned long to hold the boolean cookie is a trivial
extension, and will quell all doubt that snd/hda is the cause of the
device runtime pm leaks.
v2: Keep using the power abstraction for local wakeref tracking.
v3: BUILD_BUG_ON impedance mismatch
Signed-off-by: Chris Wilson <chris@chris-wilson.co.uk>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jani Nikula <jani.nikula@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Mika Kuoppala <mika.kuoppala@linux.intel.com>
Link: https://patchwork.freedesktop.org/patch/msgid/20190213152109.16997-1-chris@chris-wilson.co.uk
When np is NULL i2s_pdata could also be NULL but i2s_pdata is now being
dereferenced without proper check. Fix this and shorten the error message
so we don't exceed 80 characters limit.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is now no users of this flag so remove it together with
related code. The chan_name field of snd_dmaengine_dai_dma_data
data structure is not removed as it is still in use by the PXA
platform.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous implementatation was restrictive with regards to
BCLK rates for slave mode where the driver would not allow rates
the codec couldn't provide itself as clock master. The codec
is able to automatically determine and handle whatever rate is
provided so this restriction isn't necessary for slave mode. The
code was also flawed with regards to setting of the frame offset
as using rx_mask to explicitly set the offset has the knock on
effect of impacting the min and max channels for the codec, in
soc_pcm_hw_params() through the call to
soc_pcm_codec_params_fixup().
With this update, the driver now only limits frame size if codec
is clock master, and dynamically determines the BCLK offset
relating to WCLK using the tx_mask for slot offset along with the
slot width provided.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Previously the driver would default the BCLK periods per WCLK to
64, to cover all possible non-TDM scenarios when the codec was
DAI clock master. However some devices require a lower BCLK rate
to operate correctly so with this in mind, this commit updates
the code to be more dynamic, with BCLK rate now based on SR and
word length provided to hw_params().
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Spelling error fixes, upper/lower case letter changes.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change indentation so this macro definition spans 2 rows and looks
more consistent with surrounding code.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the probe call is on the primary DAI we can use 'other' in place of
i2s->sec_dai, if the probe call is on the secondary DAI we can use 'i2s'
in place of other->sec_dai.
While at it fix one whitespace issue.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The of_find_device_by_node() takes a reference to the underlying device
structure, we should release that reference.
Fixes: 7dd0d83558 ("ASoC: stm32: sai: simplify sync modes management")
Signed-off-by: Wen Yang <yellowriver2010@hotmail.com>
Acked-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is a part of conversion of Samsung platforms to use the custom DMA
config for specifying DMA channel names, in addition to passing custom
DMA device for the secondary CPU DAI's "PCM" component for some variants
of the I2S controller.
We also don't set the SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME any more
as setting it wouldn't allow to specify DMA channels through the custom
DMA config.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds DPCM links in order to support the secondary I2S interface.
For the secondary PCM interface to be actually available one more entry
should be added to the sound-dai property in sound/cpu node in DT.
The changes in driver are done in a way so we are backwards compatible
with existing DTS/DTB, i.e. if the cpu sound-dai property contains only
one entry only one PCM will be registered.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch makes the spinlock serializing access to the primary/secondary
PCM a per I2S controller lock, rather than a global one. There is no need
to have a global lock across multiple I2S controllers in the SoC.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The quirk flags are common for the primary and the secondary DAI
so move respective field from struct i2s_dai to common driver data
structure.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The IP variant data is another thing common for both DAIs, move it
to the driver's common data structure.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
As we now have the 'priv' pointer in most of the places we can use
priv->lock directly, dropping extra indirection in the SFR region
spinlock access.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SFR region is common for both DAIs so move related data structure
field from struct i2s_dai to the common driver data structure.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/isa/es1688/es1688_lib.c: In function 'snd_es1688_probe':
sound/isa/es1688/es1688_lib.c:124:31: warning:
variable 'hw' set but not used [-Wunused-but-set-variable]
unsigned short major, minor, hw;
^
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is reported that there's a constant background "hum/whitenoise"
in the headset on the Lenovo X1 machines with the codec alc285, and it
is confirmed that if we run the command below, the noise will stop.
sudo hda-verb /dev/snd/hwC0D0 0x1d SET_PIN_WIDGET_CONTROL 0x0
Then I consulted this issue with Kailang, he told me the pin 0x1d on
this codec is used for PC beep in, the noise probably comes from this
pin and we can also disable the PC beep in passthrough, then the PC
beep in will not affect other sound playback.
Fixes: c4cfcf6f42 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops")
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1660581
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the System76 Oryx Pro (oryp5), there is a headset microphone input
attached to 0x19 that does not have a jack detect. In order to get it
working, the pin configuration needs to be set correctly, and the
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied. This is
similar to the MIC_NO_PRESENCE fixups for some Dell laptops, except we
have a separate microphone jack that is already configured correctly.
Since the ALC1220 does not have a fixup similar to
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, I have exposed the fixup from the
ALC269 in a way that it can be accessed from the
alc1220_fixup_system76_oryp5 function. In addition, the
alc1220_fixup_clevo_p950 needs to be applied to gain speaker output.
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes the following sparse warning:
sound/soc/codecs/cros_ec_codec.c:209:27: warning:
symbol 'cros_ec_dai' was not declared. Should it be static?
Fixes: b291f42a37 ("ASoC: cros_ec_codec: Add codec driver for Cros EC")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following sparse warnings:
sound/soc/mediatek/mt8183/mt8183-dai-i2s.c:966:5: warning:
symbol 'mt8183_dai_i2s_get_share' was not declared. Should it be static?
sound/soc/mediatek/mt8183/mt8183-dai-i2s.c:986:5: warning:
symbol 'mt8183_dai_i2s_set_priv' was not declared. Should it be static?
Fixes: a94aec035a ("ASoC: mediatek: mt8183: add platform driver")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove including <linux/version.h> that don't need it.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to keep the PM suspend/resume register cache separate
for each DAI as those registers are common, move related i2s_dai data
structure to the driver's common data structure. This will allow us
to simplify the code a little eventually and to make it easier to follow.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The clock for generating I2S signals is also common for both CPU DAIs
so move it to the driver's common data structure.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The core clock is also common for both CPU DAIs so move it to
the driver's private data structure.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds DAPM widgets required to model the internal mixer
of the I2S controller merging audio streams from the primary and
from the secondary PCM interface.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Having the clocks provider data in struct samsung_i2s_priv, i.e. per the I2S
controller instance, rather than per CPU DAI better models the hardware and
simplifies the code a little. The clock provider is common for both DAIs.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch introduces again registration of additional platform device as
we still need it for registering the secondary dmaengine PCM component.
This patch in most part is a revert of changes done in commit
be2c92eb64 ("ASoC: samsung: i2s: Remove virtual device for secondary DAI")
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This avoids bringing back the problem introduced by
62ba568f7a ("ALSA: pcm: Return 0 when size <
start_threshold in capture") and fixed in 00a399cad1
("ALSA: pcm: Revert capture stream behavior change in
blocking mode"), which prevented the user from starting
capture from another thread.
Signed-off-by: Ricardo Biehl Pasquali <pasqualirb@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's a bit of surprising that we've got more changes than hoped
at this late stage, but the all don't look too scaring but small
fixes.
One change in ALSA core side is again the PCM regression fix that
was partially addressed for OSS, but now the all relevant change
is reverted instead. Also, a few ASoC core fixes for UAF and OOB
are included, while the rest are usual random device-specific
fixes.
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Merge tag 'sound-5.0-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"It's a bit of surprising that we've got more changes than hoped at
this late stage, but they all don't look too scary but small fixes.
One change in ALSA core side is again the PCM regression fix that was
partially addressed for OSS, but now the all relevant change is
reverted instead. Also, a few ASoC core fixes for UAF and OOB are
included, while the rest are usual random device-specific fixes"
* tag 'sound-5.0-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: pcm: Revert capture stream behavior change in blocking mode
ALSA: usb-audio: Fix implicit fb endpoint setup by quirk
ALSA: hda - Add quirk for HP EliteBook 840 G5
ASoC: samsung: Prevent clk_get_rate() calls in atomic context
ASoC: rsnd: ssiu: correct shift bit for ssiu9
ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check
ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables
ASoC: topology: fix oops/use-after-free case with dai driver
ASoC: rsnd: fixup MIX kctrl registration
ASoC: core: Allow soc_find_component lookups to match parent of_node
ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter
ASoC: MAINTAINERS: fsl: Change Fabio's email address
ASoC: hdmi-codec: fix oops on re-probe
This field was never used, let's remove it
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This field was never used, let's remove it
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This field was never used, let's remove it
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This field was never used, let's remove it
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This field was never used, let's remove it
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This field was never used, let's remove it
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This field was never used, let's remove it
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prefix is now handled in the code. This allows for default and
alternate paths, and more flexibility for OEMs and distros
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prefix is now handled in the code. This allows for default and
alternate paths, and more flexibility for OEMs and distros
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prefix is now handled in the code. This allows for default and
alternate paths, and more flexibility for OEMs and distros
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prefix is now handled in the code. This allows for default and
alternate paths, and more flexibility for OEMs and distros
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prefix is now handled in the code. This allows for default and
alternate paths, and more flexibility for OEMs and distros
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prefix is now handled in the code. This allows for default and
alternate paths, and more flexibility for OEMs and distros
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prefix is now handled in the code. This allows for default and
alternate paths, and more flexibility for OEMs and distros
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prefix is now handled in the code. This allows for default and
alternate paths, and more flexibility for OEMs and distros
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch includes minimal changes as a prerequisite for adding support
for the Exynos secondary I2S interface as second DAI of the I2S component.
Doing it that way allows to avoid problems as indicated in commmit
6b01e0365b1689 ("ASoC: samsung: i2s: disable secondary DAI until it gets fixed")
The samsung_i2s_get_pri_dai() helper added in this patch is temporary and
will be removed in one of subsequent patches.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The additional function argument will allow to select proper DMA device
for requesting DMA channel for the secondary CPU DAI.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are currently two ways to specify custom DMA channel names:
- through the SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME flag and
snd_dmaengine_dai_dma_data data structure,
- through chan_names field of struct snd_dmaengine_pcm_config.
In order to replace the DAI DMA data method with the custom DMA config
one on non-DT platforms the dmaengine_pcm_new() function is extended
to also consider channel names specified in the custom DMA config.
If both config->chan_names and dma_data->chan_name are provided
the former will be used.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when of_node of the "PCM" device is null
dmaengine_pcm_request_chan_of() function will bail out, including cases
when custom DMA device is intended to be used. To have the channels
properly requested when custom DMA device is provided extend the of_node
test to also consider dma_dev->of_node.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make sure i2s->rclk_srcrate is properly initialized also during
playback through the secondary DAI.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Silence warnings (triggered at W=1) by adding relevant __printf
attributes.
sound/soc/soc-dapm.c:149:2: warning: function 'pop_dbg' might be a candidate for 'gnu_printf' format attribute [-Wsuggest-attribute=format]
Signed-off-by: Mathieu Malaterre <malat@debian.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
As stated in 'TLV320AIC3254 Application Reference Guide' ([1]):
3.2 Device Startup Lockout Times
After the TLV320AIC3254 initializes through hardware reset at power-up
or software reset, the internal registers initialize to default values.
This initialization takes place within 1ms after pulling the RESET
signal high. During this initialization phase, no register-read or
register-write operation should be performed on ADC or DAC coefficient
buffers. Also, no block within the codec should be powered up during
the initialization phase.
[1] http://www.ti.com/lit/an/slaa408a/slaa408a.pdf
Signed-off-by: Peter Seiderer <ps.report@gmx.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
For correct operation of the digital filtering and other processing on the
WM8741, the user must ensure the correct value of OSR[1:0] is set at all
times.[1] Hence, depending the selected sampling rate, set the OSR (over-
sampling rate) mode in hw_params().
References:
[1] "WM8741 Data Sheet"
Signed-off-by: Sergej Sawazki <sergej@taudac.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sergej Sawazki <sergej@taudac.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
BE dai links only have internal PCM's and their substream ops may
not be set. Suspending these PCM's will result in their
ops->trigger() being invoked and cause a kernel oops.
So skip suspending PCM's if their ops are NULL.
[ NOTE: this change is required now for following the recent PCM core
change to get rid of snd_pcm_suspend() call. Since DPCM BE takes
the runtime carried from FE while keeping NULL ops, it can hit this
bug. See details at:
https://github.com/thesofproject/linux/pull/582
-- tiwai ]
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Loading/unloading modules exposes issues with memory allocation, which
is a mix of devm_kzalloc and manual kzalloc. Move to devm_k routines
everywhere to simplify all this.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.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: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ADC and DAC can be clocked from separate or same sources CLK1 and CLK2.
By default, ADC is clocked from CLK1, and DAC - from CLK2.
This commits allows sound cards to selest a proper clock source during
`hw_params()` via `snd_soc_dai_set_sysclk()`. It makes possible to have a
single clock source for both ADC and DAC.
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
Softly reset registers values on module probe
Signed-off-by: Kirill Marinushkin <kmarinushkin@birdec.tech>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no risk of the module being removed while the platform
components are in use. This solves the problem of the snd_soc_skl
module not being removable with rmmod
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ASoC core has for the longest time increased the module reference
counts, even before the transition to the component model. This is
probably fine on most platforms, but it introduces a deadlock case on
Intel devices with the Skylake and SOF drivers which cannot be removed
due to their reference counts being modified by the core.
In these 2 cases, the PCI or ACPI driver .probe creates a platform
device to let the machine driver .probe register the audio
card. Conversely the PCI or ACPI driver .remove will unregister the
platform device which results in the card being removed by the machine
driver .remove.
With ascii art, this can be represented as
modprobe
snd_soc_skl/
soc-pci-dev/sof-acpci-dev ----------> pci/acpi probe
^ |
| ---------------|
| | |
| V V
increase register register machine
refcount component platform_device
^ |
| |
| V
component <---- register card <---- probe
probe
The issue is that by playing with the component's module reference
counts during the card registration, it's no longer possible to remove
the module which controls the component. This can be shown, e.g. with
the following error:
root@plb-XPS-13-9350:~# lsmod | grep snd_soc_skl
snd_soc_skl 110592 1
root@plb-XPS-13-9350:~# rmmod snd_soc_skl
rmmod: ERROR: Module snd_soc_skl is in use
Increasing the reference count during the component probe is not
useful. If the PCI/ACPI module is removed, the card will be removed
anyway.
To avoid breaking existing platforms and allowing Intel platforms to
safely deal with module load/unload cases, this patch introduces a
flag which needs to be set during the component initialization. This
is a strictly opt-in capability that should only be used when the
handling of the component module does not require a reference count
increase to prevent removal during use.
Note that this solution is not directly applicable to the legacy
Atom/SST driver, which uses a different device hierarchy. There are
however additional refcount issues which prevent the ACPI driver from
being removed. This is a different issue which would need a different
patch.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
New quirk enforces search for GPIO based on its type,
i.e. iterate over GpioIo resources only.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Mika Westerberg <mika.westerberg@linux.intel.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the missing license and copyright information which never made it
into the analog driver when the original driver was split in two as part
of the review process.
Link: https://lkml.kernel.org/r/1465582725-30183-3-git-send-email-srinivas.kandagatla@linaro.org
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Johan Hovold <johan@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Regulator notifiers, that were registered during codec driver probing,
must be unregistered during driver release, or device managed versions
have to be used. This patch fixes codec drivers, that weren't explicitly
unregistering notifiers and simplifies those, that did that manually.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since we need multiple components for I915 for different purposes
(Audio & Mei_hdcp), we adopt the subcomponents methodology introduced
by the previous patch (mentioned below).
Author: Daniel Vetter <daniel.vetter@ffwll.ch>
Date: Mon Jan 28 17:08:20 2019 +0530
components: multiple components for a device
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by-by: Ramalingam C <ramalinagm.c@intel.com> (commit message)
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch> (code)
cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
cc: Russell King <rmk+kernel@arm.linux.org.uk>
cc: Rafael J. Wysocki <rafael@kernel.org>
cc: Jaroslav Kysela <perex@perex.cz>
cc: Takashi Iwai <tiwai@suse.com>
cc: Rodrigo Vivi <rodrigo.vivi@intel.com>
cc: Jani Nikula <jani.nikula@linux.intel.com>
Link: https://patchwork.freedesktop.org/patch/msgid/20190207232759.14553-4-daniel.vetter@ffwll.ch
In the commit 62ba568f7a ("ALSA: pcm: Return 0 when size <
start_threshold in capture"), we changed the behavior of
__snd_pcm_lib_xfer() to return immediately with 0 when a capture
stream has a high start_threshold. This was intended to be a
correction of the behavior consistency and looked harmless, but this
was the culprit of the recent breakage reported by syzkaller, which
was fixed by the commit e190161f96 ("ALSA: pcm: Fix tight loop of
OSS capture stream").
At the time for the OSS fix, I didn't touch the behavior for ALSA
native API, as assuming that this behavior actually is good. But this
turned out to be also broken actually for a similar deployment,
e.g. one thread goes to a write loop in blocking mode while another
thread controls the start/stop of the stream manually.
Overall, the original commit is harmful, and it brings less merit to
keep that behavior. Let's revert it.
Fixes: 62ba568f7a ("ALSA: pcm: Return 0 when size < start_threshold in capture")
Fixes: e190161f96 ("ALSA: pcm: Fix tight loop of OSS capture stream")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now all callers no longer check the return value from
snd_pcm_lib_preallocate_pages() and co, let's make them to return
void, so that any new code won't fall into the same pitfall.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots and lots of new drivers so far, a highlight being the MediaTek
BTCVSD which is a driver for a Bluetooth radio chip - the first such
driver we've had upstream. Hopefully we will soon also see a baseband
with an upstream driver!
- Support for only powering up channels that are actively being used.
- Quite a few improvements to simplify the generic card drivers,
especially the merge of the SCU cards into the main generic drivers.
- Lots of fixes for probing on Intel systems, trying to rationalize
things to look more standard from a framework point of view.
- New drivers for Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341,
Google ChromeOS embedded controllers, Ingenic JZ4725B, MediaTek
BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328, Spreadtrum
DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM formatters.
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Merge tag 'asoc-v5.1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v5.1
Lots and lots of new drivers so far, a highlight being the MediaTek
BTCVSD which is a driver for a Bluetooth radio chip - the first such
driver we've had upstream. Hopefully we will soon also see a baseband
with an upstream driver!
- Support for only powering up channels that are actively being used.
- Quite a few improvements to simplify the generic card drivers,
especially the merge of the SCU cards into the main generic drivers.
- Lots of fixes for probing on Intel systems, trying to rationalize
things to look more standard from a framework point of view.
- New drivers for Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341,
Google ChromeOS embedded controllers, Ingenic JZ4725B, MediaTek
BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328, Spreadtrum
DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM formatters.
A selection of driver specific fixes here, along with a few core fixes:
- A fixup for some MFD devices that were broken by the previous fixes
for deferred probe.
- A fix for potential out of bounds array accesses when ordering DAPM
power/up down sequences.
- Avoid use after free issue when unloading and reloading drivers using
topologies.
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Merge tag 'asoc-fix-v5.0-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
A selection of driver specific fixes here, along with a few core fixes:
- A fixup for some MFD devices that were broken by the previous fixes
for deferred probe.
- A fix for potential out of bounds array accesses when ordering DAPM
power/up down sequences.
- Avoid use after free issue when unloading and reloading drivers using
topologies.
Add suspend and resume sleep callbacks,
to support system low power modes.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add driver support for Cirrus Logic CS35L36 boosted
speaker amplifier
Signed-off-by: James Schulman <james.schulman@cirrus.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit a60945fd08 ("ALSA: usb-audio: move implicit fb quirks to
separate function") introduced an error in the handling of quirks for
implicit feedback endpoints. This commit fixes this.
If a quirk successfully sets up an implicit feedback endpoint, usb-audio
no longer tries to find the implicit fb endpoint itself.
Fixes: a60945fd08 ("ALSA: usb-audio: move implicit fb quirks to separate function")
Signed-off-by: Manuel Reinhardt <manuel.rhdt@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This enables mute LED support and fixes switching jacks when the laptop
is docked.
Signed-off-by: Jurica Vukadin <jurica.vukadin@rt-rk.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the header comment to use C++ style, so that it looks more
consistent with the rest of ASoC.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Show the knob to enable or disable the jz4740-codec driver, add a
proper description, and add a dependency on MIPS || COMPILE_TEST, as
this driver is only useful on MIPS.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add license information as a standard SPDX license notifier instead of
custom text.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Line Playback Volume for Allwinner A10 and Allwinner A20.
Add Line Boost Volume for Allwinner A10 and Allwinner A20.
Add Line Right, Line Left, Line Playback Switch for Allwinner A10 and
Allwinner A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add FM Playback Volume for Allwinner A10 and Allwinner A20.
Add FM Left, FM Right, FM Playback Switch for Allwinner A10 and
Allwinner A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Mic1 Playback Switch and Mic2 Playback Switch for Allwinner A10 and
Allwinner A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since it's now possible to have a DAPM mixer control with multiple
channels, use it to cut down the total number of controls.
Keep "Left Mixer Left DAC Playback Switch" and "Right Mixer Right DAC
Playback Switch" name & layout the same as before for compatibility.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Mic1 Boost Volume and Mic2 Boost Volume for Allwinner A10 and for
Allwinner A20.
Those controls are in different registers per chip model, so put the
Allwinner A10 controls and the Allwinner A20 controls into the newly
split sun4i_codec_controls and sun7i_codec_controls, respectively.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Introduce sun7i_codec_controls because some of the controls are different
on Allwinner A20 compared to Allwinner A10.
Also introduce sun7i_codec_codec in order to use sun7i_codec_controls and
make sun7i_codec_quirks use sun7i_codec_codec.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a control "Mic Playback Volume" that allows the user to control the
MIC gain stage (common for Mic1 and Mic2) leading to the output mixer.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add MIC2 Pre-Amplifier, Mic2 input for Allwinner A10 and Allwinner A20.
Previously, there only the Mic1 input and MIC1 Pre-Amplifier was exposed.
This exposes the Mic2 input and MIC2 Pre-Amplifier.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in the SOC_SINGLE control name. Fix this.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently "0xf << 36" is used to
clear SSIU-9 internal buffer state, which overflows 32-bit value
according to user reference manual, it is always bit4 ~ bit7
of SSI_SYS_STATUS[1,3,5,7] registers indicate
SSIU-9's buffer state, so "0xf << 4" should be used.
This patch fix incorrect shifting issue in SSIU-9 case
Fixes: commit b7169ddea2 ("ASoC: rsnd: remove RSND_REG_ from rsnd_reg")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A collection of a few small fixes. The most significant one is the
fix for the possible race at loading HD-audio drivers. This has been
present for long time and surfaced only in a rare occasion, but
finally spotted out.
The rest are usual device-specific fixes for HD-audio and USB-audio.
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Merge tag 'sound-5.0-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of a few small fixes.
The most significant one is the fix for the possible race at loading
HD-audio drivers. This has been present for long time and surfaced
only in a rare occasion, but finally spotted out"
* tag 'sound-5.0-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/ca0132 - Fix build error without CONFIG_PCI
ALSA: compress: Fix stop handling on compressed capture streams
ALSA: usb-audio: Add support for new T+A USB DAC
ALSA: hda - Serialize codec registrations
ALSA: hda/realtek - Use a common helper for hp pin reference
ALSA: hda/realtek - Fix lose hp_pins for disable auto mute
ALSA: hda/realtek - Headset microphone support for System76 darp5
Add jz4725b-codec driver to support the internal CODEC found in the
JZ4725B SoC from Ingenic.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 4d230d1271 ("ASoC: rsnd: fixup not to call clk_get/set
under non-atomic") added new rsnd_ssi_prepare() and moved
rsnd_ssi_master_clk_start() to .prepare.
But, ssi user count (= ssi->usrcnt) is incremented at .init
(= rsnd_ssi_init()).
Because of these timing exchange, ssi->usrcnt check at
rsnd_ssi_master_clk_start() should be adjusted.
Otherwise, 2nd master clock setup will be no check.
This patch fixup this issue.
Fixes: commit 4d230d1271 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic")
Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Reported-by: Valentine Barshak <valentine.barshak@cogentembedded.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>
To detect potential errors, let's add:
a) build-time warnings when the table size isn't aligned with the enum
list
b) run-time warnings when the values are not initialized. This
requires an increase by one of all values to avoid the default 0.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
KASAN reports and additional traces point to out-of-bounds accesses to
the dapm_up_seq and dapm_down_seq lookup tables. The indices used are
larger than the array definition.
Fix by adding missing entries for the new widget types in these two
lookup tables, and align them with PGA values.
Also the sequences for the following widgets were not defined. Since
their values defaulted to zero, assign them explicitly
snd_soc_dapm_input
snd_soc_dapm_output
snd_soc_dapm_vmid
snd_soc_dapm_siggen
snd_soc_dapm_sink
Fixes: 8a70b4544e ('ASoC: dapm: Add new widget type for constructing DAPM graphs on DSPs.').
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Again no functional changes, but only code clean up.
Use a standard macro for initializing the procfs entries, also drop
the info entries stored in dsp_spos_instance, as they are removed
recursively by a single snd_info_free_entry() calls.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The creation of card's id proc file can be moved gracefully into
info.c. Also, the assignment of card->proc_id is superfluous and can
be dropped. So let's do it.
Basically this is no functional change but code refactoring, but one
potential behavior change is that now it returns properly the error
code from snd_info_card_register(), which is a good thing (tm).
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just a minor code optimization to reduce the source code size
slightly. No functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's referred only in snd_card_id_read() which can receive the card
object via private_data.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the proc fs creation code with new helper functions,
snd_card_ro_proc_new() and snd_card_rw_proc_new().
Just a code refactoring and no functional changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two new helper functions are added here for cleaning up the existing
lengthy calls.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The calls of snd_info_register() are superfluous and should be avoided
at the procfs creation time. They are called at the end of the whole
initialization via snd_card_register(). This patch drops such
superfluous calls, as well as dropping the superfluous setup of
SNDRV_INFO_CONTENT_TEXT.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The calls of snd_info_register() are superfluous and should be avoided
at the procfs creation time. They are called at the end of the whole
initialization via snd_card_register(). This patch drops such
superfluous calls.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The calls of snd_info_register() are superfluous and should be avoided
at the procfs creation time. They are called at the end of the whole
initialization via snd_card_register(). This patch drops such
superfluous calls, as well as cleaning up the calls of substream proc
entries with a common helper.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The calls of snd_info_register() are superfluous and should be avoided
at the procfs creation time. They are called at the end of the whole
initialization via snd_card_register(). This patch drops such
superfluous calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The calls of snd_info_register() are superfluous and should be avoided
at the procfs creation time. They are called at the end of the whole
initialization via snd_card_register(). This patch drops such
superfluous calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The calls of snd_info_register() are superfluous and should be avoided
at the procfs creation time. They are called at the end of the whole
initialization via snd_card_register(). This patch drops such
superfluous calls.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop old license header and switch to SPDX-License-Identifier.
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The trigger and set_params callbacks are called from 3 and 2 separate
loops respectively, tidy up the code a little by factoring these out
into helper functions.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For callbacks configuring the state of the components (trigger,
set_params, ack and set_metadata) simplify the code a little and make
intention clearer by aborting as soon as an error is encountered. The
operation has already failed and there is nothing to be gained from
processing the callbacks on additional components. The operations
currently abort after the callbacks, so this simply shortens the
error path.
For callbacks returning information from the driver (copy,
get_metadata, pointer, get_codec_caps, get_caps and get_params)
only look for the first callback provided, currently the code will
call every callback only returning the information provided by the
last. Since we can only return one set of data, it makes no sense to
request the data from every component. Again this just makes the
currently supported feature set a little more clear.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
These are arrays, not pointers, and they can't be NULL.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the ssm2603 data sheet (control register sequencing), the
digital core should be activated only after all necessary bits in the
power register are enabled, and a delay determined by the decoupling
capacitor on the VMID pin has passed. If the digital core is activated
too early, or even before the ADC is powered up, audible artifacts
appear at the beginning of the recorded signal.
The digital core is also needed for playback, so when recording starts
it may already be enabled. This means we cannot get the power sequence
correct when we want to be able to start recording after playback.
As a workaround put the MIC mute switch into the DAPM routes. This
way we can keep the recording disabled until the MIC Bias has settled
and thus get rid of audible artifacts.
Signed-off-by: Philipp Zabel <p.zabel@pengutronix.de>
m.felsch@pengutronix.de: adapt commit message
m.felsch@pengutronix.de: drop of configuration as mentioned by Mark:
https://patchwork.kernel.org/patch/10407449/
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAIs linked to the dummy will not have an associated playback/capture
widget, so we need to skip the update in that case.
Fixes: 078a85f280 ("ASoC: dapm: Only power up active channels from a DAI")
Reported-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Tested-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A call of pci_iounmap() call without CONFIG_PCI leads to a build error
on some architectures. We tried to address this and add a check of
IS_ENABLED(CONFIG_PCI), but this still doesn't seem enough for sh.
Ideally we should fix it globally, it's really a corner case, so let's
paper over it with a simpler ifdef.
Fixes: 1e73359a24 ("ALSA: hda/ca0132 - make pci_iounmap() call conditional")
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the T+A VID to the generic check in order to enable
native DSD support for T+A devices. This works with the new T+A USB
DAC model SD3100HV and will also work with future devices which
support the XMOS/Thesycon style DSD format.
Signed-off-by: Udo Eberhardt <udo.eberhardt@thesycon.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make sure that all children entries are registered by a single call of
snd_info_register(). OTOH, don't register if a parent isn't
registered yet.
This allows us to create the whole procfs tree in a shot at the last
stage of card registration phase in a later patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we covered all callers with NULL device pointer, let's catch the
remaining calls with NULL and warn explicitly.
Acked-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should pass a proper non-NULL device object to memory allocators
although it was accepted in the past. The card->dev points to the
most appropriate device object in such a case, so let's put it.
Acked-by: Christoph Hellwig <hch@lst.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should pass a proper non-NULL device object to memory allocators
although it was accepted in the past. The card->dev points to the
most appropriate device object in such a case, so let's put it.
Acked-by: Christoph Hellwig <hch@lst.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This driver use the gpio consumer interface.
Add the header as it's needed.
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in a dev_warn message. Fix this.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should pass a proper non-NULL device object to memory allocators
although it was accepted in the past. The card->dev points to the
most appropriate device object in such a case, so let's put it.
Acked-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should pass a proper non-NULL device object to memory allocators
although it was accepted in the past. The card->dev points to the
most appropriate device object in such a case, so let's put it.
Acked-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
soc_tplg_link_config() will find the physical dai link and call
soc_tplg_dai_link_load() to load the BE dai link. Currently remove_link()
is only used to remove the FE dai link which is created by the topology.
The BE dai link cannot however be unloaded in snd_soc_tplg_component
_remove(), which is problematic if anything needs to be released or
reinitialized.
This patch aligns the definitions of dynamic types with the existing
UAPI and adds a new remove_backend_link() routine to unload the the BE
dai link when snd_soc_tplg_component_remove() is invoked.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Originally snd_soc_init_platform was not cleaning up its pointers, this
was fixed to always reallocate dynamic memory but created a memory leak
when snd_soc_init_platform was called multiple times during the same
probe attempt and also threw away any changes made to the struct between
calls. In order to avoid reallocating memory that is still valid, the
behaviour will be changed to clear the dynamically set pointers on a
probe error and a unregister event and snd_soc_init_platform will go
back to its original behaviour of only allocating null pointers so it will
stop throwing away valid changes.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The of_find_device_by_node() takes a reference to the underlying device
structure, we should release that reference.
Signed-off-by: Wen Yang <yellowriver2010@hotmil.com>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, in some complex cases, more than one widgets have same
name and registed from differnt dapm context, and route add from
another context too. When snd_soc_dapm_add_route, the previous
registered widget will overwritten by the latest same name widget,
will cause unexpect error. For Asoc framework we cant avoid this
situation and we cant decide which widget that wanted with route.
At least we can give users a notice.
Signed-off-by: Zhiwei Jiang <qq282012236@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently all widgets attached to a DAI link will be powered
up when the DAI is active, however this may include routes
that are not actually in use if there are unused channels
available on the DAI.
The macros for creating AIF widgets already include an entry for
slot, it is proposed to change that to channel. The effective
difference here being respresenting the logical channel index
rather than the physical slot index. The CODECs currently
using the slot entry on the DAPM_AIF macros are using it in
a manner consistent with this, the CODECs not using it just
have the field set to zero.
A variable is added to snd_soc_dapm_widget to represent
this channel index and then for each AIF widget attached to
a DAI this is compared against the number of channels on
the stream. Enabling the links for those which will be in
use. This has the nice property that the CODECs which haven't
used the slot/channel entry in the macro will function exactly
as before due to all the AIF widgets having a channel of zero
and a stream by definition having at least one channel.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rmmod/modprobe tests expose a kernel oops when accessing the dai
driver pointer. This comes from the topology design which operates in
multiple passes. Each object removal happens at a specific iteration,
and the code checks for the iteration (order) number after the memory
containing the order was freed.
Fix this be clearing a reference to the dai driver and check its
validity to avoid dereferences.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 7620fe9161 ("ASoC: topology: fix memory leak in
soc_tplg_dapm_widget_create") fixed a memory leak issue, but
additional tests and KASAN reports show a use-after-free in soc-dapm.
The widgets are created with a kmemdup operating on a template. The
"name" string is also duplicated, but the "sname" string is not. As a
result, when the template is freed after widget creation, its sname
string is still used.
Fix by explicitly duplicating the "sname" string, and freeing it when
required.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver function for transferring/receiving
BT encoded data to/from BT firmware.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set the channel number on each AIF widget to allow unused channels not
to be powered up across AIFs.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound card need to judge that whether it is using
"TDM Split mode". To judge it and for other purpose, it has
rsnd_parse_connect_simple() and rsnd_parse_connect_graph(),
but these are using different judgement policy for
TDM Split mode.
It is pointless and confusable.
This patch add new rsnd_parse_tdm_split_mode() and use common
judgement policy for simple-card/audio-graph.
Without this patch, CTU will be judged as TDM Split mode
on audio-graph card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver has below function to check connection
rsnd_parse_connect_simple()
rsnd_parse_connect_graph()
But these have different parameters. This patch synchronize these
for cleanup.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound device has many IPs and many situations.
If platform/board uses MIXer, situation will be more complex.
To avoid duplicate DVC kctrl registration when MIXer was used,
it had original flags.
But it was issue when sound card was re-binded, because
no one can't cleanup this flags then.
To solve this issue, commit 9c698e8481 ("ASoC: rsnd: tidyup
registering method for rsnd_kctrl_new()") checks registered
card->controls, because if card was re-binded, these were cleanuped
automatically. This patch could solve re-binding issue.
But, it start to avoid MIX kctrl.
To solve these issues, we need below.
To avoid card re-binding issue: check registered card->controls
To avoid duplicate DVC registration: check registered rsnd_kctrl_cfg
To allow multiple MIX registration: check registered rsnd_kctrl_cfg
This patch do it.
Fixes: 9c698e8481 ("ASoC: rsnd: tidyup registering method for rsnd_kctrl_new()")
Reported-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-By: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We used to pass NULL to memory allocators for ISA devices due to
historical reasons. But we prefer rather a proper device object to be
assigned, so let's fix it by replacing snd_dma_isa_data() call with
card->dev reference, and kill snd_dma_isa_data() definition.
Reviewed-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DMA API generally relies on a struct device to work properly, and
only barely works without one for legacy reasons. Pass the easily
available struct device from the platform_device to remedy this.
Also use GFP_KERNEL instead of GFP_USER as the gfp_t for the memory
allocation, as we should treat this allocation as a normal kernel one.
Signed-off-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DMA API generally relies on a struct device to work properly, and
only barely works without one for legacy reasons. Pass the easily
available struct device from the platform_device to remedy this.
Signed-off-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current code, the codec registration may happen both at the
codec bind time and the end of the controller probe time. In a rare
occasion, they race with each other, leading to Oops due to the still
uninitialized card device.
This patch introduces a simple flag to prevent the codec registration
at the codec bind time as long as the controller probe is going on.
The controller probe invokes snd_card_register() that does the whole
registration task, and we don't need to register each piece
beforehand.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the open-codes in many places with a new common helper for
performing the same thing: referring to the primary headphone pin.
This eventually fixes the potentially missing headphone pin on some
weird devices, too.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When auto_mute = no or spec->suppress_auto_mute = 1, cfg->hp_pins will
lose value.
Add this patch to find hp_pins value.
I add fixed for ALC282 ALC225 ALC256 ALC294 and alc_default_init()
alc_default_shutup().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only three fixes: a fix for Realtek HD-audio looks lengthy, but it's
just a code shuffling, and the actual changes are fairly small. The
rest are a PCM core fix for a long-standing bug that was recently
scratched by syzkaller, and a trivial USB-audio quirk for DSD
support.
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Merge tag 'sound-5.0-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Only three fixes.
The fix for Realtek HD-audio looks lengthy, but it's just a code
shuffling, and the actual changes are fairly small.
The rest are a PCM core fix for a long-standing bug that was recently
scratched by syzkaller, and a trivial USB-audio quirk for DSD support"
* tag 'sound-5.0-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fixed hp_pin no value
ALSA: pcm: Fix tight loop of OSS capture stream
ALSA: usb-audio: Add Opus #3 to quirks for native DSD support
On the System76 Darter Pro (darp5), there is a headset microphone
input attached to 0x1a that does not have a jack detect. In order to
get it working, the pin configuration needs to be set correctly, and
the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied.
This is similar to the MIC_NO_PRESENCE fixups for some Dell laptops,
except we have a separate microphone jack that is already configured
correctly.
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declaration of snd_pcm_drop() in sound/core/pcm_native.c is superfluous
since the function isn't called before being defined. Remove the
declaration.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Spreadtrum DMA engine uses the link-list mode to support audio playback
or capture, thus this patch adds audio DMA platform support for CPU DAI to
trigger DMA link-list transfer.
Signed-off-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
struct snd_soc_dapm_route has been modified to be a dynamic
object so that it can be used to save driver specific
data while parsing topology and clean up
driver-specific data during driver unloading.
This patch makes the following changes to accomplish the above:
1. Set the dobj member of snd_soc_dapm_route during the
SOC_TPLG_PASS_GRAPH pass of topology parsing.
2. Add the remove_route() routine that will be called while
removing all dynamic objects from the component driver.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
template.sname and template.name are only freed when an error occur.
They should be freed in the success return case, too.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
dtexts is two dimensional array, so we also need to free it after
freeing its fields.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when we unload and reload machine driver few times we end with
corrupted list and try to cleanup no longer existing objects. Fix this
by removing dobj from the list.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We already have passed dobj, there is no reason to access it through
containing structs.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the registration and free of beep input device was done
manually from the register and the disconnect callbacks of the
assigned codec object. This seems working in most cases, but this may
be a cause of some races at probe. Moreover, due to these manual
calls, the total code became unnecessarily lengthy.
This patch rewrites the beep registration code to follow the standard
sound device object style. This allows us reducing the code, in
addition to avoiding the nested device registration calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The init sequence for ALC294 headphone stuff is needed not only for
the boot up time but also for the resume from hibernation, where the
device is switched from the boot kernel without sound driver to the
suspended image. Since we record the PM event in the device
power_state field, we can now recognize the call pattern and apply the
sequence conditionally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we deal with single codec and suspend codec callbacks for
all S3, S4 and runtime PM handling. But it turned out that we want
distinguish the call patterns sometimes, e.g. for applying some init
sequence only at probing and restoring from hibernate.
This patch slightly modifies the common PM callbacks for HD-audio
codec and stores the currently processed PM event in power_state of
the codec's device.power field, which is currently unused. The codec
callback can take a look at this event value and judges which purpose
it's being called.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For some reason we test if the machine is passed as a parameter before
fixing up the codec name. This is unnecessary, generates false
positives in static analysis tools and done only in this machine
driver, remove and adjust indentation.
Reported-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix hp_pin always no value.
[More notes on the changes:
The hp_pin value that is referred in alc294_hp_init() is always zero
at the moment the function gets called, hence this is actually
useless as in the current code.
And, this kind of init sequence should be called from the codec init
callback, instead of the parser function. So, the first fix in this
patch to move the call call into its own init_hook.
OTOH, this function is needed to be called only once after the boot,
and it'd take too long for invoking at each resume (where the init
callback gets called). So we add a new flag and invoke this only
once as an additional fix.
The one case is still not covered, though: S4 resume. But this
change itself won't lead to any regression in that regard, so we
leave S4 issue as is for now and fix it later. -- tiwai ]
Fixes: bde1a74596 ("ALSA: hda/realtek - Fixed headphone issue for ALC700")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a codec driver to control ChromeOS EC codec.
Use EC Host command to enable/disable I2S recording and control other
configurations.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Reviewed-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
XMOS/Thesycon family of USB Audio Class firmware flags DSD altsetting
separate from the PCM ones. Thus the DSD altsetting can be auto-detected
based on the flag and doesn't need maintaining specific altsetting
whitelist.
In addition, static VID:PID-to-altsetting whitelisting causes problems
when firmware update changes the altsetting, or same VID:PID is reused
for another device that has different kind of firmware.
This patch removes existing explicit whitelist mappings for XMOS VID
(0x20b1) and Thesycon VID (0x152a).
Also corrects placement of Hegel HD12 and NuPrime DAC-10 to keep list
sorted based on VID.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds audio routing for both playback and capture.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds required dapm widgets for playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds basic controls found in wcd9335 codec.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
CLASS-H controller/Amplifier is common accorss Qualcomm WCD codec series.
This patchset adds basic CLASS-H controller apis for WCD codecs after
wcd9335 to use.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC,
It supports both I2S/I2C and SLIMbus audio interfaces.
On slimbus interface it supports two data lanes; 16 Tx ports
and 8 Rx ports. It has Seven DACs and nine dedicated interpolators,
Seven (six audio ADCs, and one VBAT ADC), Multibutton headset
control (MBHC), Active noise cancellation and Sidetone paths
and processing.
This patchset adds very basic support for playback and capture
via the 9 interpolators and ADC respectively.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Duende Classic was produced by Solid State Logic in 2006, as a
first model of Duende DSP series. The following model, Duende Mini
was produced in 2008. They are designed to receive isochronous
packets for PCM frames via IEEE 1394 bus, perform signal processing by
downloaded program, then transfer isochronous packets for converted
PCM frames.
These two models includes the same embedded board, consists of several
ICs below:
- Texus Instruments Inc, TSB41AB3 for physical layer of IEEE 1394 bus
- WaveFront semiconductor, DICE II STD ASIC for link/protocol layer
- Altera MAX 3000A CPLD for programs
- Analog devices, SHARC ADSP-21363 for signal processing (4 chips)
This commit adds support for the two models to ALSA dice driver. Like
support for the other devices, packet streaming is just available.
Userspace applications should be developed if full features became
available; e.g. program uploader and parameter controller.
$ ./hinawa-config-rom-printer /dev/fw1
{ 'bus-info': { 'adj': False,
'bmc': False,
'chip_ID': 349771402425,
'cmc': True,
'cyc_clk_acc': 255,
'generation': 1,
'imc': True,
'isc': True,
'link_spd': 2,
'max_ROM': 1,
'max_rec': 512,
'name': '1394',
'node_vendor_ID': 20674,
'pmc': False},
'root-directory': [ ['VENDOR', 20674],
['DESCRIPTOR', 'Solid State Logic'],
['MODEL', 112],
['DESCRIPTOR', 'Duende board'],
[ 'NODE_CAPABILITIES',
{ 'addressing': {'64': True, 'fix': True, 'prv': True},
'misc': {'int': False, 'ms': False, 'spt': True},
'state': { 'atn': False,
'ded': False,
'drq': True,
'elo': False,
'init': False,
'lst': True,
'off': False},
'testing': {'bas': False, 'ext': False}}],
[ 'UNIT',
[ ['SPECIFIER_ID', 20674],
['VERSION', 1],
['MODEL', 112],
['DESCRIPTOR', 'Duende board']]]]}
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In function rt5651_i2c_probe(), local variable "ret" could
be uninitialized if function regmap_read() returns -EINVAL.
However, this value is used in if statement. This is
potentially unsafe.
Signed-off-by: Yizhuo <yzhai003@ucr.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rationale behind the current calculation is somewhat obscure [1]
and can yield slightly wrong dividers in certain cases, which the
machine drivers for some boards (like the HiFiBerry DAC+ Pro)
seemingly try to circumvent, by updating the rate fraction so as to
suit this calculation.
The updated calculation should correctly yield the smallest bit clock
rate that would fit the frame.
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2019-January/144219.html
Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some boards, such as the HiFiBerry DAC+ Pro, use a pair of external
oscillators, to generate 44.1 or 48kHz multiples and are forced to
resort to hacks [1] in order to support 24-bit data without ending up
with fractional dividers. This patch allows the machine driver to use
32-bit frames for 24-bit data to avoid such issues.
Although the datasheet (p. 15) seems to suggest that only a handful
of ratios are supported, it's not very explicit about it, so we allow
the full range of values supported by the underlying register in the
callback, to avoid needlessly rejecting potentially usable
configurations.
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143442.html
Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can use for_each_link_codecs() without waiting
for_each_rtd_codec_dai() on soc_bind_dai_link().
Let's use for_each macro.
Fixes: 50acc7e49 ("ASoC: core: Fix multi-CODEC setups")
Fixes: 10dff9b0d ("ASoC: soc-core: use for_each_link_codecs() for dai_link codecs")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the trigger=off is passed for a PCM OSS stream, it sets the
start_threshold of the given substream to the boundary size, so that
it won't be automatically started. This can be problematic for a
capture stream, unfortunately, as detected by syzkaller. The scenario
is like the following:
- In __snd_pcm_lib_xfer() that is invoked from snd_pcm_oss_read()
loop, we have a check whether the stream was already started or the
stream can be auto-started.
- The function at this check returns 0 with trigger=off since we
explicitly disable the auto-start.
- The loop continues and repeats calling __snd_pcm_lib_xfer() tightly,
which may lead to an RCU stall.
This patch fixes the bug by simply allowing the wait for non-started
stream in the case of OSS capture. For native usages, it's supposed
to be done by the caller side (which is user-space), hence it returns
zero like before.
(In theory, __snd_pcm_lib_xfer() could wait even for the native API
usage cases, too; but I'd like to stay in a safer side for not
breaking the existing stuff for now.)
Reported-by: syzbot+fbe0496f92a0ce7b786c@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk VID/PID IDs for the Opus #3 DAP (made by 'The Bit')
in order to enable Native DSD support.
[ NOTE: this could be handled in the generic way with fp->dvd_raw if
we add 0x10cb to the vendor whitelist, but since 0x10cb shows a
different vendor name (Erantech), put to the individual entry at
this time -- tiwai ]
Signed-off-by: Olek Poplavsky <woodenbits@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For devices implemented as a MFD it is common to only have a single node
in devicetree representing the whole device. As such when looking up
components in soc_find_components we should match against both the devices
of_node and the devices parent's of_node, as is already done in the rest
of the ASoC core.
This causes regressions for some DAI links at the moment as
soc_find_component was recently added as a check in soc_init_dai_link.
Fixes: 8780cf1142 ("ASoC: soc-core: defer card probe until all component is added to list")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
An open-coded error path in __snd_pcm_lib_xfer() can be replaced with
the simple goto to the common error path. This also makes the error
handling more consistent, i.e. when some samples have been already
processed, return that size instead of the error code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The > should be >= or otherwise we potentially read one element beyond
the end of the ff->tx_midi_substreams[] array.
Fixes: 73f5537fb2 ("ALSA: fireface: support tx MIDI functionality of Fireface UCX")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't populate the const arrays on the stack but instead make
it static. Makes the object code smaller, for example:
Before:
text data bss dec hex filename
14107 8832 224 23163 5a7b bytcht_es8316.o
After:
text data bss dec hex filename
14015 8896 224 23135 5a5f bytcht_es8316.o
(gcc version 8.2.0 x86_64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A significant amount of fixes at this time, mostly for covering the
recent ASoC issues.
- Fixes for the missing ASoC driver initialization with non-deferred
probes; these triggered other problems in chain, which resulted in
yet more fix commits
- DaVinci runtime PM fix; the diff looks large but it's just a code
shuffling
- Various fixes for ASoC Intel drivers: a regression in HD-A HDMI,
Kconfig dependency, machine driver adjustments, PLL fix.
- Other ASoC driver-specific stuff including the trivial fixes
caught by static analysis
- Usual HD-audio quirks
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Merge tag 'sound-5.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A significant amount of fixes at this time, mostly for covering the
recent ASoC issues.
- Fixes for the missing ASoC driver initialization with non-deferred
probes; these triggered other problems in chain, which resulted in
yet more fix commits
- DaVinci runtime PM fix; the diff looks large but it's just a code
shuffling
- Various fixes for ASoC Intel drivers: a regression in HD-A HDMI,
Kconfig dependency, machine driver adjustments, PLL fix.
- Other ASoC driver-specific stuff including the trivial fixes caught
by static analysis
- Usual HD-audio quirks"
* tag 'sound-5.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (30 commits)
ALSA: hda - Add mute LED support for HP ProBook 470 G5
ASoC: amd: Fix potential NULL pointer dereference
ASoC: imx-audmux: change snprintf to scnprintf for possible overflow
ASoC: rt5514-spi: Fix potential NULL pointer dereference
ASoC: dapm: change snprintf to scnprintf for possible overflow
ASoC: rt5682: Fix PLL source register definitions
ASoC: core: Don't defer probe on optional, NULL components
ASoC: core: Make snd_soc_find_component() more robust
ASoC: soc-core: fix init platform memory handling
ASoC: intel: skl: Fix display power regression
ALSA: hda/realtek - Fix typo for ALC225 model
ASoC: soc-core: Hold client_mutex around soc_init_dai_link()
ASoC: Intel: Boards: move the codec PLL configuration to _init
ASoC: soc-core: defer card probe until all component is added to list
ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages
ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode
ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks
ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized
ASoC: rt5682: Fix recording no sound issue
ASoC: Intel: atom: Make PCI dependency explicit
...
This patch changes the parent pointer assignment of snd_info_entry
object to be always non-NULL. More specifically,check the parent
argument in snd_info_create_module_entry() & co, and assign
snd_proc_root if NULL is passed there.
This assures that the proc object is always freed when the root is
freed, so avoid possible memory leaks. For example, some error paths
(e.g. snd_info_register() error at snd_minor_info_init()) may leave
snd_info_entry object although the proc file itself is freed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The proc files are recursively freed by calling with the root
snd_info_entry object, so we don't have to keep each object for
releasing one by one. Move the release of the PCM stream proc root at
the beginning, so that we can remove the redundant code and resource.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In Fireface series, registration of higher 4 bytes of destination
address for asynchronous transaction of MIDI messages is done by
a write transaction to model-specific register.
On the other hand, registration of lower 4 bytes of the address is
selectable from 4 options. A register for this registration includes
the other purpose options such as input attenuation. Thus this
driver expects userspace applications to configure the register.
Actual behaviour for the asynchronous transaction is different
depending on protocols. In former protocol, destination offset
of each transaction is the same as the registered address even if
it is block request. In latter models, destination offset of each
transaction is the offset of previous transaction plus 4 byte
and the transaction is quadlet request.
This commit cleanups comments about the above mechanism.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the previous code refactoring, the PCM stream locking code
became nothing but the PCM group lock with self_group object. Use the
existing helper function for simplifying the code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the hackish down_write_nonfifo() that was introduced as a
workaround of rwsem deadlock.
It used to be a problem for non-atomic PCM streams that take the rwsem
for the locking and hit the high lock contention. Since the current
PCM locking refactoring, we'll no longer hit it as the hot code-paths
don't take global locks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have currently two global locks, a rwlock and a rwsem, that are
used for managing linking the PCM streams. Due to these global locks,
once when a linked stream is used, the lock granularity suffers a
lot.
This patch attempts to eliminate the former global lock for atomic
ops. The latter rwsem needs remaining because of the loosy way of the
loop calls in snd_pcm_action_nonatomic(), as well as for avoiding the
deadlock at linking. However, these are used far rarely, actually
only by two actions (prepare and reset), where both are no timing
critical ones. So this can be still seen as a good improvement.
The basic strategy to eliminate the rwlock is to assure group->lock at
adding or removing a stream to / from the group. Since we already
takes the group lock whenever taking the all substream locks under the
group, this shouldn't be a big problem. The reference to group
pointer in snd_pcm_substream object is protected by the stream lock
itself.
However, there are still pitfalls: a race window at re-locking and the
lifecycle of group object. The former is a small race window for
dereferencing the substream group object opened while snd_pcm_action()
performs re-locking to avoid ABBA deadlocks. This includes the unlink
of group during that window, too. And the latter is the kfree
performed after all streams are removed from the group while it's
still dereferenced.
For addressing these corner cases, two new tricks are introduced:
- After re-locking, the group assigned to the stream is checked again;
if the group is changed, we retry the whole procedure.
- Introduce a refcount to snd_pcm_group object, so that it's freed
only when it's empty and really no one refers to it.
(Some readers might wonder why not RCU for the latter. RCU in this
case would cost more than refcounting, unfortunately. We take the
group lock sooner or later, hence the performance improvement by RCU
would be negligible. Meanwhile, because we need to deal with
schedulable context depending on the pcm->nonatomic flag, it'll become
dynamic RCU/SRCU switch, and the grace period may become too long.)
Along with these changes, there are a significant amount of code
refactoring. The complex group re-lock & ref code is factored out to
snd_pcm_stream_group_ref() function, for example.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert 10dff9b0d (ASoC: soc-core: use for_each_link_codecs() for
dai_link codecs) for now as Sylwester Nawrocki reports that it causes
oopses on at least Odroid boards.
Reported-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In latter model of Fireface series, asynchronous transaction includes
a prefix byte to indicate the way to decode included MIDI bytes.
Upper 4 bits of the prefix byte indicates port number, and the rest 4
bits indicate the way to decode rest of bytes for MIDI messages.
Basically the rest bits indicates the number of bytes for MIDI message.
However, if the last byte of each MIDi message is included, the rest
bits are 0xf. For example:
message: f0 00 00 66 14 20 00 00 f7
offset: content (big endian, port 0)
'0030: 0x02f00000
'0030: 0x03006614
'0030: 0x03200000
'0030: 0x0ff70000
This commit supports encoding scheme for the above and allows
applications to transfer MIDI messages via ALSA rawmidi interface.
An unused member (running_status) is reused to keep state of
transmission of system exclusive messages.
For your information, this is a dump of config rom.
$ sudo ./hinawa-config-rom-printer /dev/fw1
{ 'bus-info': { 'bmc': False,
'chip_ID': 13225063715,
'cmc': False,
'cyc_clk_acc': 0,
'imc': False,
'isc': True,
'max_rec': 512,
'name': '1394',
'node_vendor_ID': 2613},
'root-directory': [ [ 'NODE_CAPABILITIES',
{ 'addressing': {'64': True, 'fix': True, 'prv': False},
'misc': {'int': False, 'ms': False, 'spt': True},
'state': { 'atn': False,
'ded': False,
'drq': True,
'elo': False,
'init': False,
'lst': True,
'off': False},
'testing': {'bas': False, 'ext': False}}],
['VENDOR', 2613],
['DESCRIPTOR', 'RME!'],
['EUI_64', 2873037108442403],
[ 'UNIT',
[ ['SPECIFIER_ID', 2613],
['VERSION', 4],
['MODEL', 1054720],
['DESCRIPTOR', 'Fireface UCX']]]]}
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Between former and latter models, content of asynchronous transaction
for MIDI messages from driver to device is different.
This commit is a preparation to support latter models. A protocol-specific
operation is added to encode MIDI messages to the transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Between former and latter models, destination address to receive
asynchronous transactions for MIDI messages is different.
This commit adds model-dependent parameter for the addresses.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface UCX transfers asynchronous transactions for MIDI messages.
One transaction includes quadlet data therefore it can transfer 3
message bytes as maximum. Base address of the destination is
configured by two settings; a register for higher 8 byte of the
address, and a bitflag to option register indicates lower 8byte.
The register for higher address is 0x'ffff'0000'0034. Unfortunately,
firmware v24 includes a bug to ignore registered value for the
destination address and transfers to 0x0001xxxxxxxx always. This
driver doesn't work well if the bug exists, therefore users should
install the latest firmware (v27).
The bitflag is a part of value to be written to option register
(0x'ffff'0000'0014).
lower addr: bitflag (little endian)
'0000'0000: 0x00002000
'0000'0080: 0x00004000
'0000'0100: 0x00008000
'0000'0180: 0x00010000
This register includes more options but they are not relevant to
packet streaming or MIDI functionality. This driver don't touch it.
Furthermore, the transaction is sent to address offset incremented
by 4 byte to the offset in previous time. When it reaches base address
plus 0x7c, next offset is the base address.
Content of the transaction includes a prefix byte. Upper 4 bits of
the byte indicates port number, and the rest 4 bits indicate the way
to decode rest of bytes for MIDI message.
Except for system exclusive messages, the rest bits are the same as
status bits of the message without channel bits. For system exclusive
messages, the rest bits are encoded according to included message bytes.
For example:
message: f0 7e 7f 09 01 f7
offset: content (little endian, port 0)
'0000: 0x04f07e7f
'0004: 0x070901f7
message: f0 00 00 66 14 20 00 00 00 f7
offset: content (little endian, port 1)
'0014: 0x14f00000
'0018: 0x14661420
'001c: 0x14000000
'0020: 0x15f70000
message: f0 00 00 66 14 20 00 00 f7
offset: content (little endian, port 0)
'0078: 0x04f00000
'007c: 0x04661420
'0000: 0x070000f7
This commit supports decoding scheme for the above and allows
applications to receive MIDI messages via ALSA rawmidi interface.
The lower 8 bytes of destination address is fixed to 0x'0000'0000,
thus this driver expects userspace applications to configure option
register with bitflag 0x00002000 in advance.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In Fireface series, drivers can register destination address for
asynchronous transaction which transfers MIDI messages from device.
In former models, all of the transactions arrive at the registered
address without any offset. In latter models, each of the transaction
arrives at the registered address with sequential offset within 0x00
to 0x7f. This seems to be for discontinuity detection.
This commit adds model-dependent member for the address range.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a series of Fireface, devices transfer asynchronous transaction with
MIDI messages. In the transaction, content is different depending on
models. ALSA fireface driver has protocol-dependent handler to pick up
MIDI messages from the content.
In latter models of the series, the transaction is transferred to range
of address sequentially. This seems to check continuity of transferred
messages.
This commit changes prototype of the handler to receive offset of
address for received transactions.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD/DA ASRC function control two ASRC clock sources separately.
Whether AD/DA filter select which clock source, we enable AD/DA ASRC
function for all cases.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AZX_DCAPS_PM_RUNTIME flag is added to indicate support for runtime PM.
azx_has_pm_runtime() is used to check if above is enabled and thus
forbid runtime PM calls if needed.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch moves clock enable/disable from system resume/suspend to
runtime resume/suspend respectively. Along with this hda controller
chip init or stop is also moved. System resume/suspend can invoke
runtime callbacks and do necessary setup.
chip->running can be used to check for probe completion and device
access during runtime_resume or runtime_suspend can be avoided if
probe is not yet finished. This helps to avoid kernel panic during
boot where runtime PM callbacks can happen from system PM.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Explicit clock enable is not required during probe, as this would be
managed by runtime PM calls. Clock can be enabled/disabled in runtime
resume/suspend. This way it is easier to balance clock enable/disable
counts.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moved devm_clk_get() API calls to a separate function and the same
can be called early in the probe. This is done before runtime PM
for the device is enabled. The runtime resume/suspend callbacks can
later enable/disable clocks respectively(the support would be added
in subsequent patches). Clock handles should be available by the
time runtime suspend/resume calls can happen.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables runtime power management(runtime PM) support for
hda. pm_runtime_enable() and pm_runtime_disable() are added during
device probe and remove respectively. The runtime PM callbacks will
be forbidden if hda controller does not have support for runtime PM.
pm_runtime_get_sync() and pm_runtime_put() are added for hda register
access. The callbacks for above will be added in subsequent patches.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current simple-card is using asoc_simple_card_canonicalize_dailink().
Its naming is "dailink", but is for "platform".
We already have asoc_simple_card_canonicalize_cpu() for "cpu",
let's follow same naming rule.
It never return error, so, void function is better idea.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can use for_each_link_codecs() without waiting
for_each_rtd_codec_dai() on soc_bind_dai_link().
Let's use for_each macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to cleanup component when soc_probe_component() was
failed, or when soc_remove_component() was called.
But they are cleanuping component on each way.
(And soc_probe_component() doesn't call snd_soc_dapm_free(),
but it should).
Same code in many places makes code un-understandable.
This patch adds new soc_cleanup_component() and call it from
snd_probe_component() and snd_remove_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto