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>
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.
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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
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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>
The pci express variant of the digigram lx6464es card has a different
device ID, but works without changes to the driver.
Thanks to Nikolas Slottke for reporting and testing.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Link: https://lore.kernel.org/r/20190906082119.40971-1-tim@klingt.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The destructor of AMDTP domain has WARN_ON() for the list of associated
AMDTP stream. Although this reminds a case that developers forget to
program consumer drivers to stop AMDTP domain, it hits when AMDTP domain
is not initialized yet. This occurs when initialization of sound card
fails as well and it's superfluous.
This commit removes the WARN_ON. Although the API to AMDTP domain does
nothing, it's left for future usage.
Fixes: 3ec3d7a3ff ("ALSA: firewire-lib: add AMDTP domain structure to handle several isoc contexts")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190906131414.15370-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FE-8 is the rest of model in TASCAM FireWire series. This device
has no functionality to process audio signal and MIDI messages. Instead,
it transfers control messages to host system corresponding to operations
for some faders, buttons and knobs on its surface.
Unlike the other devices in this series, the control messages are
transmitted by asynchronous transactions. Some registers of device are
used for registration of destination address for the transaction. The
transaction includes quadlet-aligned data up to 32 quadlets.
Userspace applications can receive the transaction and parse it for
control message via Linux FireWire subsystem, without any support by
ALSA firewire-tascam driver. Therefore the driver gives no support
for it.
This commit removes placeholder for FE-8 and add some comment for its
functionalities as notes.
$ python2 linux-firewire-utils/src/crpp < ~/git/am-config-rom/tascam/tascam-fe8.img
ROM header and bus information block
-----------------------------------------------------------------
400 040f4798 bus_info_length 4, crc_length 15, crc 18328 (should be 14256)
404 31333934 bus_name "1394"
408 20ff7002 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256)
40c 00022eff company_id 00022e |
410 a094dcb7 device_id ffa094dcb7 | EUI-64 00022effa094dcb7
root directory
-----------------------------------------------------------------
414 0004bccc directory_length 4, crc 48332
418 0300022e 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 0003eda4 directory_length 3, crc 60836
42c 1200022e specifier id
430 13800001 version
434 d4000004 --> dependent info directory at 444
eui-64 leaf at 438
-----------------------------------------------------------------
438 0002461e leaf_length 2, crc 17950
43c 00022eff company_id 00022e |
440 a094dcb7 device_id ffa094dcb7 | EUI-64 00022effa094dcb7
dependent info directory at 444
-----------------------------------------------------------------
444 0002ae47 directory_length 2, crc 44615
448 81000002 --> descriptor leaf at 450
44c 82000006 --> bus dependent info leaf at 464
descriptor leaf at 450
-----------------------------------------------------------------
450 0004a79e leaf_length 4, crc 42910
454 00000000 textual descriptor
458 00000000 minimal ASCII
45c 54415343 "TASC"
460 414d0000 "AM"
bus dependent info leaf at 464
-----------------------------------------------------------------
464 0004a7d8 leaf_length 4, crc 42968
468 00000000
46c 00000000
470 46452d38
474 00000000
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20190906125544.13800-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This notebook has 6 built in speakers for 5.1 surround support, however
only two got autodetected and have also not been assigned correctly.
This patch enables all speakers and also fixes muting when headphones are
plugged in.
The speaker layout is as follows:
pin 0x15 Front Left / Front Right
pin 0x18 Front Center / Subwoofer
pin 0x1b Rear Left / Rear Right (Surround)
The quirk will be enabled automatically on this hardware, but can also be
activated manually via the model=aspire-ethos module parameter.
Caveat: pin 0x1b is shared between headphones jack and surround speakers.
When headphones are plugged in, the surround speakers get muted
automatically by the hardware, however all other speakers remain
unmuted. Currently it's not possible to make use of the generic automute
function in the driver, because such shared pins are not supported.
If we would change the pin settings to identify the pin as headphones,
the surround channel and thus the ability to select 5.1 profiles would
get lost.
This quirk solves the above problem by monitoring jack state of 0x1b and
by connecting/disconnecting all remaining speaker pins when something
gets plugged in or unplugged from the headphones jack port.
Signed-off-by: Sergey Bostandzhyan <jin@mediatomb.cc>
Link: https://lore.kernel.org/r/20190906093343.GA7640@xn--80adja5bqm.su
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Variable pcm_idx is being initialized with a value that is never read
and is being re-assigned immediately afterwards. The assignment is
redundant and hence can be removed.
Addresses-Coverity: ("Unused value")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20190905154826.5916-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small HD-audio fixes:
- A regression fix for Realtek codecs due to the recent initialization
procedure change
- A fix for potential endless loop at the quirk table lookup
- Quirks for Lenovo, ASUS and HP machines
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Merge tag 'sound-5.3-rc8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small HD-audio fixes:
- A regression fix for Realtek codecs due to the recent
initialization procedure change
- A fix for potential endless loop at the quirk table lookup
- Quirks for Lenovo, ASUS and HP machines"
* tag 'sound-5.3-rc8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fix the problem of two front mics on a ThinkCentre
ALSA: hda/realtek - Enable internal speaker & headset mic of ASUS UX431FL
ALSA: hda/realtek - Add quirk for HP Pavilion 15
ALSA: hda/realtek - Fix overridden device-specific initialization
ALSA: hda - Fix potential endless loop at applying quirks
When do compile test, if SND_SOC_SOF_OF is not set, we get:
sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_request':
imx8.c:(.text+0xb0): undefined reference to `snd_sof_ipc_msgs_rx'
sound/soc/sof/imx/imx8.o: In function `imx8_ipc_msg_data':
imx8.c:(.text+0xf4): undefined reference to `sof_mailbox_read'
sound/soc/sof/imx/imx8.o: In function `imx8_dsp_handle_reply':
imx8.c:(.text+0x160): undefined reference to `sof_mailbox_read'
Make SND_SOC_SOF_IMX_TOPLEVEL always depends on SND_SOC_SOF_OF
Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: 202acc565a ("ASoC: SOF: imx: Add i.MX8 HW support")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190905064400.24800-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On the sm1, the TDMOUT number of input is extended and the
the gain enable bit moved to accommodate this extension
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20190905120120.31752-9-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
