A small smattering of ASoC fixes for v5.4 - nothing too exciting
here, all small standalone things.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAl2JCfITHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0GP1B/wImLx7u5jHiiAAzeW9kig7R+m91/wm
+QMwcguVLsrTiI8qz2YnH8bMEbLU66bHsy7faKl5hG1r6vnSEct6MTw236eM7l5/
XqLRmnn6W2GtxnpShL/eL3F73i7HYdSUBNsdeV5V9loTMJKehDJ/AqK7k+eMZt0y
5xxEpkiTZilu1ue+GwWN2bhKHu4HnBmESkmbO2Adsws1ZKUNfuhye+IkorllJFeG
v9FmP8j37Z5YKMAUaKmdy4uaecvwJZVpWNuaNNS7A02+6OpeIdrYtNUbj4dnPqGF
U+mOs+Xsnsv09ykBEBdFBk50sKjZnIMeO+cxz/tjT8nL9QI+LCFonf72
=Ijmx
-----END PGP SIGNATURE-----
Merge tag 'asoc-fix-v5.4-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.4
A small smattering of ASoC fixes for v5.4 - nothing too exciting
here, all small standalone things.
The same fixup to enable laptop imic is needed for ASUS M9V with AD1986A
codec like another HP machine.
Signed-off-by: Shih-Yuan Lee (FourDollars) <fourdollars@debian.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190920134052.GA8035@localhost
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SND_SOC_DM365_VOICE_CODEC is a 'bool' option in a choice statement,
meaning it cannot be set to =m, but it selects two other drivers
that we may want to be loadable modules after all:
WARNING: unmet direct dependencies detected for SND_SOC_CQ0093VC
Depends on [m]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m]
Selected by [y]:
- SND_SOC_DM365_VOICE_CODEC [=y] && <choice>
Selected by [m]:
- SND_SOC_ALL_CODECS [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMPILE_TEST [=y]
Add an intermediate symbol that sets SND_SOC_CQ0093VC and
MFD_DAVINCI_VOICECODEC to =m if SND_SOC=m.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20190920075046.3210393-1-arnd@arndb.de
Signed-off-by: Mark Brown <broonie@kernel.org>
24 bits is supported in all modes and 16 bit only when the codec is slave
and the DAI is set to RIGHT_J.
Remove the unsupported sample format.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190919071652.31724-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
commit a0a4bf57a9 ("ASoC: core: delete component->card_list in soc_remove_component only")
was trying to fix a kernel oops when list_del was called twice without
re-init the list. Use list_del_init() can solve it, too. Besides, it
will be more readable if we cleanup all component related resource at
soc_cleanup_component().
Suggested-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20190918133131.15045-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The laptop has a combined jack to attach headsets on the right.
The BIOS encodes them as two different colored jacks at the front,
but otherwise it seems to be configured ok. But any adaption of
the pins config on its own doesn't fix the jack detection to work
in Linux. Still Windows works correct.
This is somehow fixed by chaining ALC256_FIXUP_ASUS_HEADSET_MODE,
which seems to register the microphone jack as a headset part and
also results in fixing jack sensing, visible in dmesg as:
-snd_hda_codec_realtek hdaudioC0D0: Mic=0x19
+snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19
[ Actually the essential change is the location of the jack; the
driver created "Front Mic Jack" without the matching volume / mute
control element due to its jack location, which confused PA.
-- tiwai ]
Signed-off-by: Jan-Marek Glogowski <glogow@fbihome.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/8f4f9b20-0aeb-f8f1-c02f-fd53c09679f1@fbihome.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's reported that the garbled sound on HP Envy x360 13z-ag000 (Ryzen
Laptop) is fixed by the same workaround applied to other AMD chips.
Update the driver_data entry for Raven (1022:15e3) to use the newly
introduced preset, AZX_DCAPS_PRESET_AMD_SB. Since it already contains
AZX_DCAPS_PM_RUNTIME, we can drop that bit, too.
Reported-and-tested-by: Dennis Padiernos <depadiernos@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190920073040.31764-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SSI need to use rsnd_channel_normalization() for TDM-split mode,
thus, channel check need to do after that.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874l1aw39d.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A potential bug was reported in the email "[BUG] atmel_ssc_dai: a
possible sleep-in-atomic bug in atmel_ssc_shutdown"[1]
Indeed in the function atmel_ssc_shutdown() free_irq() was called in a
critical section protected by spinlock.
However this spinlock is only used in atmel_ssc_shutdown() and
atmel_ssc_startup() functions. After further analysis, it occurred that
the call to these function are already protected by mutex used on the
calling functions.
Then we can remove the spinlock which will fix this bug as a side
effect. Thanks to this patch the following message disappears:
"BUG: sleeping function called from invalid context at
kernel/locking/mutex.c:909"
[1]: https://www.spinics.net/lists/alsa-devel/msg71286.html
Reviewed-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Signed-off-by: Gregory CLEMENT <gregory.clement@bootlin.com>
Link: https://lore.kernel.org/r/20190918100344.23629-1-gregory.clement@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We add component->card_list in the end of soc_probe_component(). In
other words, component->card_list will not be added if there is an
error in the soc_probe_component() function. So we can't delete
component->card_list in the error handling of soc_probe_component().
Fixes: 22d1423187 ("ASoC: soc-core: add soc_cleanup_component()")
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190916210353.6318-1-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
EDMA requires the period size to be multiple of maxburst. Otherwise the
remaining bytes are not transferred and thus noise is produced.
We can handle this issue by adding a constraint on
SNDRV_PCM_HW_PARAM_PERIOD_SIZE to be multiple of tx/rx maxburst value.
Signed-off-by: Mihai Serban <mihai.serban@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20190913192807.8423-2-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds quirk VID ID for Hiby portable players family with
native DSD playback support.
Signed-off-by: Ilya Pshonkin <sudokamikaze@protonmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190917074937.157802-1-ilya.pshonkin@netforce.ua
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At higher sampling rate (e.g. 192.0 kHz), Alesis iO26 transfers 4 data
channels per data block in CIP.
Both iO14 and iO26 have the same contents in their configuration ROM.
For this reason, ALSA Dice driver attempts to distinguish them according
to the value of TX0_AUDIO register at probe callback. Although the way is
valid at lower and middle sampling rate, it's lastly invalid at higher
sampling rate because because the two models returns the same value for
read transaction to the register.
In the most cases, users just plug-in the device and ALSA dice driver
detects it. In the case, the device runs at lower sampling rate and
the driver detects expectedly. For this reason, this commit leaves the
way to detect as is.
Fixes: 28b208f600 ("ALSA: dice: add parameters of stream formats for models produced by Alesis")
Cc: <stable@vger.kernel.org> # v4.18+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190916101851.30409-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few small fixes and one feature that came in since I sent you the
earlier pull request.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAl1+v7ETHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0N6DB/4kXPv5zew+Vm6uOz+4WguP+TqettIM
N47HXiWz66TKjM4m+7EWU+iZwOgtXGEisT0VvhVCNHvFiKRugViyURhgR2qNkBI8
KCion+EmehcRKLWesz3U06YXIaEctj54dUcdpi2wpymf1V1wye/wKJAWVDFpRd2z
g1/TcIp4KQaspEnfnTYEiN2jmeV4TGeJsWXMdeu2gbUf7zw0aEHMXt1QaiiNC0Vz
2T689MZBrBd/z0MT62EE0bkHoEqdgB+yxNSOWjoldjtx6yZ9wuz9f2Jc2WlPZT3i
97K2RJ3+TIwdeVtsN5ijZP61ALpz7Qif5q6UiSytyQcBdxYnwegUZlLT
=oi7v
-----END PGP SIGNATURE-----
Merge tag 'asoc-v5.4-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Final merge window fixes for v5.4
A few small fixes and one feature that came in since I sent you the
earlier pull request.
This is to allow machine drivers to set a certain bitclk rate
which might not be exactly rate * frame size.
Cc: NXP Linux Team <linux-imx@nxp.com>
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20190830215910.31590-1-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
While it is safe to use strncpy in this case, the advice is to move to
strscpy or strscpy_pad.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190911083331.16801-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
All these functions declares and initializes variable ret with
'0' and without modifying 'ret' variable, it is returned.
This patch removes this redundancy and returns '0' directly.
Signed-off-by: Saiyam Doshi <saiyamdoshi.in@gmail.com>
Link: https://lore.kernel.org/r/20190909174541.GA22718@SD
Signed-off-by: Mark Brown <broonie@kernel.org>
2 bytes in MSB of register for clock status is zero during intermediate
state after changing status of sampling clock in models of TASCAM FireWire
series. The duration of this state differs depending on cases. During the
state, it's better to retry reading the register for current status of
the clock.
In current implementation, the intermediate state is checked only when
getting current sampling transmission frequency, then retry reading.
This care is required for the other operations to read the register.
This commit moves the codes of check and retry into helper function
commonly used for operations to read the register.
Fixes: e453df44f0 ("ALSA: firewire-tascam: add PCM functionality")
Cc: <stable@vger.kernel.org> # v4.4+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190910135152.29800-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return value of snd_tscm_stream_get_clock() is ignored. This commit
checks the value and handle error.
Fixes: e453df44f0 ("ALSA: firewire-tascam: add PCM functionality")
Cc: <stable@vger.kernel.org> # v4.4+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190910135152.29800-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Quite a big update this time around, particularly in the core
where we've had a lot of cleanups from Morimoto-san - there's
not much functional change but quite a bit of modernization
going on. We've also seen a lot of driver work, a lot of it
cleanups but also some particular drivers.
- Lots and lots of cleanups from Morimoto-san and Yue Haibing.
- Lots of cleanups and enhancements to the Freescale, sunxi dnd
Intel rivers.
- Initial Sound Open Firmware suppot for i.MX8.
- Removal of w90x900 and nuc900 drivers as the platforms are
being removed.
- New support for Cirrus Logic CS47L15 and CS47L92, Freescale
i.MX 7ULP and 8MQ, Meson G12A and NXP UDA1334
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAl13cr4THGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0NKuB/9fvRIh6bJ4pUA26Bc7+shJQ1BtC/MN
jo1G4maN+hY5ZUwE5hvg04S6W6Unm1iNotQecKcF43Vh/4SZNiLtfSEM4b/6IBWw
IFUU6xDz8Q4HbF4HJMotpKQKMABpfds5flH2e1YrrNoMH+KlkC9kJOR26B2W36xW
TZclfquCDICxr8M7eYGM7N5hOqSrlugyWBZqTTnTDnsMrW4SAaH2HYwFhaeayd+I
ECyaXIoUHvo4FX5ueZv/mzBiMl0z4rgXn3tuqI6a8LoWJdRZTkcSQabtuIC+wmxb
P734RY6vjSUYZrv03cAtxHDrSVoC/RYedOzhT+iFF6y/NHzdu701lsJb
=aD0T
-----END PGP SIGNATURE-----
Merge tag 'asoc-v5.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v5.4
Quite a big update this time around, particularly in the core
where we've had a lot of cleanups from Morimoto-san - there's
not much functional change but quite a bit of modernization
going on. We've also seen a lot of driver work, a lot of it
cleanups but also some particular drivers.
- Lots and lots of cleanups from Morimoto-san and Yue Haibing.
- Lots of cleanups and enhancements to the Freescale, sunxi dnd
Intel rivers.
- Initial Sound Open Firmware suppot for i.MX8.
- Removal of w90x900 and nuc900 drivers as the platforms are
being removed.
- New support for Cirrus Logic CS47L15 and CS47L92, Freescale
i.MX 7ULP and 8MQ, Meson G12A and NXP UDA1334
Add an op in hdmi_codec_ops so codec driver can register callback
function to handle plug event.
Driver in DRM can use this callback function to report connector status.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20190717083327.47646-2-cychiang@chromium.org
Reviewed-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of clearing RT5677_PWR_ANLG2 (MX-64h) to 0 at SND_SOC_BIAS_OFF,
we only clear the RT5677_PWR_CORE bit which is set at SND_SOC_BIAS_PREPARE.
MICBIAS control bits are left unchanged.
This fixed the bug where if MICBIAS1 widget is forced on, MICBIAS
control bits will be cleared at suspend and never turned back on again,
since DAPM thinks the widget is always on.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20190906194636.217881-3-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
In order to simplify understanding what register values are being
written to the codec for debugging more advanced features (such as
hotwording) it is best to remove magic numbers
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20190906194636.217881-2-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc_unbind_aux_dev() implementation is very half,
thus it is very unreadable.
for_each_comp_order(order) {
for_each_card_auxs_safe(card, comp, _comp) {
(1) if (comp->driver->remove_order == order) {
...
=> soc_unbind_aux_dev(comp);
}
}
soc_unbind_aux_dev() itself is not related to remove_order (1).
And, it is called from soc_remove_aux_devices(), even though
its paired function soc_bind_aux_dev() is called from
snd_soc_instantiate_card().
It is very unbalance, and very difficult to understand.
This patch do
1) update soc_bind_aux_dev() to self contained
2) call it from soc_cleanup_card_resources() to make up balance
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r24wor0z.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.
soc-core.c has soc_bind_aux_dev(), but, there is no its paired
soc_unbind_aux_dev().
This patch adds soc_unbind_aux_dev().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgpcor14.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc_bind_aux_dev() implementation is very half,
thus it is very unreadable.
for_each_card_pre_auxs(xxx) {
=> ret = soc_bind_aux_dev(xxx);
...
}
This patch does all for_each_xxx() under soc_bind_aux_dev(),
and makes it to self contained.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tv9sor1b.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.
This patch moves soc_probe_link_dais() next to soc_remove_link_dais()
which is paired function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v9u8or1g.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc_probe_link_dais() implementation is very half,
thus it is very difficult to read.
for_each_comp_order(xxx) {
for_each_card_rtds(xxx)
=> soc_probe_link_dais(xxx);
}
This patch does all for_each_xxx() under soc_probe_link_dais(),
and makes it to self contained.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87woeoor1m.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc_probe_link_dais() (1) is called under probe_order (2),
and it will initialize dai_link related settings at *Last* turn (3)(B).
It is very complex code.
static int soc_probe_link_dais(..., order)
{
(A) /* probe DAIs here */
...
(3) if (order != SND_SOC_COMP_ORDER_LAST)
return 0;
(B) /* initialize dai_link related settings */
...
}
static int snd_soc_instantiate_card(...)
{
...
(2) for_each_comp_order(order) {
for_each_card_rtds(...) {
(1) ret = soc_probe_link_dais(..., order);
}
}
}
This patch separes soc_probe_link_dais() into "DAI probe" portion (A),
and dai_link settings portion (B).
The later is named as soc_link_init() by this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y2z4or1r.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.
This patch moves soc_probe_dai() next to soc_remove_dai() which is
paired function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zhjkor1x.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc_remove_link_dais() implementation is very half,
thus it is very difficult to read.
for_each_comp_order(xxx) {
for_each_card_rtds(xxx)
=> soc_remove_link_dais(xxx);
}
This patch does all for_each_xxx() under soc_remove_link_dais(),
and makes it to self contained.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871rwwq5mm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc_remove_link_components() implementation is very half,
thus it is very difficult to read.
for_each_comp_order(xxx) {
for_each_card_rtds(xxx)
=> soc_remove_link_components(xxx);
}
This patch does all for_each_xxx() under soc_remove_link_components(),
and makes it to self contained.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8736hcq5ms.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc_probe_link_components() implementation is very half,
thus it is very difficult to read.
for_each_comp_order(xxx) {
for_each_card_rtds(xxx) {
=> ret = soc_probe_link_components(xxx);
...
}
}
This patch does all for_each_xxx() under soc_probe_link_components(),
and makes it to self contained.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874l1sq5mx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't populate the array pd on the stack but instead make it
static const. Makes the object code smaller by 82 bytes.
Before:
text data bss dec hex filename
26548 7288 64 33900 846c sound/soc/codecs/rt1308.o
After:
text data bss dec hex filename
26370 7384 64 33818 841a sound/soc/codecs/rt1308.o
(gcc version 9.2.1, amd64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190907074634.22144-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't populate the array pd on the stack but instead make it
static const. Makes the object code smaller by 93 bytes.
Before:
text data bss dec hex filename
38961 9784 64 48809 bea9 sound/soc/codecs/rt1305.o
After:
text data bss dec hex filename
38804 9848 64 48716 be4c sound/soc/codecs/rt1305.o
(gcc version 9.2.1, amd64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190907074156.21907-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't populate the array pd on the stack but instead make it
static const. Makes the object code smaller by 100 bytes.
Before:
text data bss dec hex filename
51463 13016 128 64607 fc5f sound/soc/codecs/rt1011.o
After:
text data bss dec hex filename
51299 13080 128 64507 fbfb sound/soc/codecs/rt1011.o
(gcc version 9.2.1, amd64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190907073717.21632-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch set 0Hz to sysclk when shutdown the card.
Some codecs set rate constraints that derives from sysclk. This
mechanism works correctly if machine drivers give fixed frequency.
But simple-audio and audio-graph card set variable clock rate if
'mclk-fs' property exists. In this case, rate constraints will go
bad scenario. For example a codec accepts three limited rates
(mclk / 256, mclk / 384, mclk / 512).
Bad scenario as follows (mclk-fs = 256):
- Initialize sysclk by correct value (Ex. 12.288MHz)
- Codec set constraints of PCM rate by sysclk
48kHz (1/256), 32kHz (1/384), 24kHz (1/512)
- Play 48kHz sound, it's acceptable
- Sysclk is not changed
- Play 32kHz sound, it's acceptable
- Set sysclk to 8.192MHz (= fs * mclk-fs = 32k * 256)
- Codec set constraints of PCM rate by sysclk
32kHz (1/256), 21.33kHz (1/384), 16kHz (1/512)
- Play 48kHz again, but it's NOT acceptable because constraints
do not allow 48kHz
So codecs treat 0Hz sysclk as signal of applying no constraints to
avoid this problem.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907174501.19833-1-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch ignores sysclk setting if it is 0Hz.
Some codecs treat 0Hz sysclk as signal of applying no constraints.
This driver does not have such feature but current implementation
outputs 'Failed to set mclk' error message if machine driver sets
0Hz sysclk to this driver.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907174332.19586-1-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't populate the arrays on the stack but instead make them
static const. Makes the object code smaller by 37 bytes.
Before:
text data bss dec hex filename
16253 7200 0 23453 5b9d sound/soc/codecs/ad193x.o
After:
text data bss dec hex filename
16056 7360 0 23416 5b78 sound/soc/codecs/ad193x.o
(gcc version 9.2.1, amd64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190906161404.1440-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch supports some type of machine drivers that set 0 to mclk
when sound device goes to idle state. After applied this patch,
sysclk == 0 means there is no constraint of sound rate and other
values will set constraints which is derived by sysclk setting.
Original code refuses sysclk == 0 setting. But some boards and SoC
(such as RockPro64 and RockChip I2S) has connected SoC MCLK out to
ES8316 MCLK in. In this case, SoC side I2S will choose suitable
frequency of MCLK such as fs * mclk-fs when user starts playing or
capturing.
Bad scenario as follows (mclk-fs = 256):
- Initialize sysclk by correct value (Ex. 12.288MHz)
- ES8316 set constraints of PCM rate by sysclk
48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400),
24kHz (1/512), 16kHz (1/768), 12kHz (1/1024)
- Play 48kHz sound, it's acceptable
- Sysclk is not changed
- Play 32kHz sound, it's acceptable
- Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256)
- ES8316 set constraints of PCM rate by sysclk
32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400),
16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024)
- Play 48kHz again, but it's NOT acceptable because constraints
list does not allow 48kHz
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907163653.9382-2-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes redundant null checks for optional MCLK clock.
And fix DT binding document for changing clock property to optional
from required.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Link: https://lore.kernel.org/r/20190907163653.9382-1-katsuhiro@katsuster.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 1137ceee76 ("ARM: OMAP1: ams-delta: Don't request unused
GPIOs"), on-board audio has appeared muted. It has been discovered that
believed to be unused GPIO pins "hookflash1" and "hookflash2" need to be
set low for audible sound in handsfree and handset mode respectively.
According to Amstrad E3 wiki, the purpose of both pins hasn't been
clearly identified. Original Amstrad software used to produce a high
pulse on them when the phone was taken off hook or recall was pressed.
With the current findings, we can assume the pins provide a kind of
audio mute function, separately for handset and handsfree operation
modes.
Commit 2afdb4c41d ("ARM: OMAP1: ams-delta: Fix audio permanently
muted") attempted to fix the issue temporarily by hogging the GPIO pin
"hookflash1" renamed to "audio_mute", however the fix occurred
incomplete as it restored audible sound only for handsfree mode.
Stop hogging that pin, rename the pins to "handsfree_mute" and
"handset_mute" respectively and implement appropriate DAPM event
callbacks for "Speaker" and "Earpiece" DAPM widgets.
Fixes: 1137ceee76 ("ARM: OMAP1: ams-delta: Don't request unused GPIOs")
Signed-off-by: Janusz Krzysztofik <jmkrzyszt@gmail.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190907111650.15440-1-jmkrzyszt@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some tools use the snd_pcm_info_get_name() to try to identify PCMs or for
other purposes.
Currently it is left empty with the dmaengine-pcm, in this case copy the
pcm->id string as pcm->name.
For example IGT is using this to find the HDMI PCM for testing audio on it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reported-by: Arthur She <arthur.she@linaro.org>
Link: https://lore.kernel.org/r/20190906055524.7393-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>