The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage.
This patch introduces this functionality as to be seen in the mac
or windows scarlett control.
The correct address could already be found in the dump info, but was
never used. Without this patch inputs 3/4 are quite unusable else.
Signed-off-by: Jens Verwiebe <info@jensverwiebe.de>
Link: https://lore.kernel.org/r/384d65cd-5e87-91eb-9fc3-e57226f534c6@jensverwiebe.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Laptops like ASUS UX431FLC and UX431FL can share the same audio quirks.
But UX431FLC needs one more step to enable the internal speaker: Pull
the GPIO from CODEC to initialize the AMP.
Fixes: 60083f9e94 ("ALSA: hda/realtek - Enable internal speaker & headset mic of ASUS UX431FL")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191125093405.5702-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some more development work for v5.5. Highlights include:
- More cleanups from Morimoto-san.
- Trigger word detection for RT5677.
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Merge tag 'asoc-v5.5-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: More updates for v5.5
Some more development work for v5.5. Highlights include:
- More cleanups from Morimoto-san.
- Trigger word detection for RT5677.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An explicit Kconfig dependency is missing for the recent addition of
the timer support. CONFIG_SND_TIMER isn't always selected by SND_PCM.
Fixes: 26c53379f9 ("ALSA: aloop: Support selection of snd_timer instead of jiffies")
Reported-by: kbuild test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20191124083924.14049-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current code:
LENOVO-20QE000VMC-ThinkPadX1Carbon7th-20QE000VMC
With the patch:
LENOVO-20QE000VMC-ThinkPadX1Carbon7th
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20191120174435.30920-2-perex@perex.cz
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add append_dmi_string() function and make the code more readable.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191120174435.30920-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
The current driver only sets 0x76543210 and 0x67452301 for DALIGN.
This doesn’t work well for TDM split and ex-split mode for all SSIU.
This patch programs the DALIGN registers based on the SSIU number.
Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Cc: Jiada Wang <jiada_wang@mentor.com>
Cc: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Fixes: a914e44693 ("ASoC: rsnd: more clear rsnd_get_dalign() for DALIGN")
Signed-off-by: Nilkanth Ahirrao <anilkanth@jp.adit-jv.com>
Signed-off-by: Eugeniu Rosca <erosca@de.adit-jv.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20191121111023.10976-1-erosca@de.adit-jv.com
Signed-off-by: Mark Brown <broonie@kernel.org>
loopback_snd_timer_close_cable() function waits until all
scheduled tasklets are completed, but the timer is closed after that
and can generate more event callbacks, scheduling new tasklets,
that will not be synchronized with cable closing.
Move tasklet_kill() call to be executed after snd_timer_close()
call to avoid such case.
Fixes: 26c53379f9 ("ALSA: aloop: Support selection of snd_timer instead of jiffies")
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191122175218.17187-2-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
loopback_parse_timer_id() uses snd_card_ref(), that can lock on mutex,
also snd_timer_instance_new() uses non-atomic allocation, that can sleep.
So, both functions can not be called from loopback_snd_timer_open()
with cable->lock spinlock locked.
Moreover, most part of loopback_snd_timer_open() function body works
when the opposite stream of the same cable does not yet exist, and
the current stream is not yet completely open and can't be running,
so existing locking of loopback->cable_lock mutex is enough to protect
from conflicts with simultaneous opening or closing.
Locking of cable->lock spinlock is not needed in this case.
Fixes: 26c53379f9 ("ALSA: aloop: Support selection of snd_timer instead of jiffies")
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191122175218.17187-1-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of the new PCM sync_stop ops in ASoC component.
It's optional and can be NULL unless you need the sync operation.
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20191121190709.29121-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now PCM core accepts the NULL ioctl ops as default, and passing a proper
ioctl ops is no longer mandatory. Adjust soc_new_pcm() to allow also
the NULL for component ioctl ops, too.
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20191121190709.29121-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDMI codec may leave codec->relaxed_resume flag set even after
unbinding. Clear it unconditionally.
It's very unlikely that this actually matters in the real use case,
so just a fix for consistency.
Fixes: ade49db337 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI")
Link: https://lore.kernel.org/r/20191122132624.5482-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The old Nvidia chips have multiple HD-audio codecs on the same
HD-audio controller, and this doesn't work as expected with the current
audio component binding that is implemented under the one-codec-per-
controller assumption; at the probe time, the driver leads to several
kernel WARNING messages.
For the proper support, we may change the pin2port and port2pin to
traverse the codec list per the given pin number, but this needs more
development and testing.
As a quick workaround, instead, this patch drops the binding in the
audio side for these legacy chips since the audio component support in
nouveau graphics driver is still not merged (hence it's basically
unused).
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205625
Fixes: ade49db337 ("ALSA: hda/hdmi - Allow audio component for AMD/ATI and Nvidia HDMI")
Link: https://lore.kernel.org/r/20191122132000.4460-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return from pnp_irq is an unsigned integer type resource_size_t
and hence the error check for a positive non-error code is always
going to be true. A check for a non-failure return from pnp_irq
should in fact be for (resource_size_t)-1 rather than >= 0.
Addresses-Coverity: ("Unsigned compared against 0")
Fixes: a9824c868a ("[ALSA] Add CS4232 PnP BIOS support")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20191122131354.58042-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usb_mixer_controls_badd() that parses UAC3 BADD profiles misses a
NULL check for the given interfaces. When a malformed USB descriptor
is passed, this may lead to an Oops, as spotted by syzkaller.
Skip the iteration if the interface doesn't exist for avoiding the
crash.
Fixes: 17156f23e9 ("ALSA: usb: add UAC3 BADD profiles support")
Reported-by: syzbot+a36ab65c6653d7ccdd62@syzkaller.appspotmail.com
Suggested-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191122112840.24797-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The s6i6_gen2_info.ports[] array had the Mixer and PCM port type
entries in the wrong place. Use designators to explicitly specify the
array elements being set.
Fixes: 9e4d5c1be2 ("ALSA: usb-audio: Scarlett Gen 2 mixer interface")
Signed-off-by: Geoffrey D. Bennett <g@b4.vu>
Tested-by: Alex Fellows <alex.fellows@gmail.com>
Tested-by: Markus Schroetter <project.m.schroetter@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191110134356.GA31589@b4.vu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a new Dell machine which needs to apply the quirk
ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, try to use the fallback table
to fix it this time. And we could remove all pintbls of alc236
for applying DELL1_MIC_NO_PRESENCE on Dell machines.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20191121022644.8078-2-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a new Dell machine which needs to apply the quirk
ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, try to use the fallback table
to fix it this time. And we could remove all pintbls of alc256
for applying DELL1_MIC_NO_PRESENCE on Dell machines.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20191121022644.8078-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many PCI and other drivers performs snd_pcm_period_elapsed() simply in
its interrupt handler, so the sync_stop operation is just to call
synchronize_irq(). Instead of putting this call multiple times,
introduce the common card->sync_irq field. When this field is set,
PCM core performs synchronize_irq() for sync-stop operation. Each
driver just needs to copy its local IRQ number to card->sync_irq, and
that's all we need.
Link: https://lore.kernel.org/r/20191117085308.23915-8-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard programming model of a PCM sound driver is to process
snd_pcm_period_elapsed() from an interrupt handler. When a running
stream is stopped, PCM core calls the trigger-STOP PCM ops, sets the
stream state to SETUP, and moves on to the next step. This is
performed in an atomic manner -- this could be called from the interrupt
context, after all.
The problem is that, if the stream goes further and reaches to the
CLOSE state immediately, the stream might be still being processed in
snd_pcm_period_elapsed() in the interrupt context, and hits a NULL
dereference. Such a crash happens because of the atomic operation,
and we can't wait until the stream-stop finishes.
For addressing such a problem, this commit adds a new PCM ops,
sync_stop. This gets called at the appropriate places that need a
sync with the stream-stop, i.e. at hw_params, prepare and hw_free.
Some drivers already have a similar mechanism implemented locally, and
we'll refactor the code later.
Link: https://lore.kernel.org/r/20191117085308.23915-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently PCM ioctl ops is a mandatory field but almost all drivers
simply pass snd_pcm_lib_ioctl. For simplicity, allow to set NULL in
the field and call snd_pcm_lib_ioctl() as default.
Link: https://lore.kernel.org/r/20191117085308.23915-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update the documentation for the newly introduced managed buffer
allocation mode accordingly. The old preallocation is no longer
recommended.
Link: https://lore.kernel.org/r/20191117085308.23915-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the support for the feature to automatically allocate
and free PCM buffers, so called "managed buffer allocation" mode.
It's set up via new PCM helpers, snd_pcm_set_managed_buffer() and
snd_pcm_set_managed_buffer_all(), both of which correspond to the
existing preallocator helpers, snd_pcm_lib_preallocate_pages() and
snd_pcm_lib_preallocate_pages_for_all(). When the new helper is used,
it not only performs the pre-allocation of buffers, but also it
manages to call snd_pcm_lib_malloc_pages() before the PCM hw_params
ops and snd_lib_pcm_free() after the PCM hw_free ops inside PCM core,
respectively. This allows drivers to drop the explicit calls of the
memory allocation / release functions, and it will be a good amount of
code reduction in the end of this patch series.
When the PCM substream is set to the managed buffer allocation mode,
the managed_buffer_alloc flag is set in the substream object. Since
some drivers want to know when a buffer is newly allocated or
re-allocated at hw_params callback (e.g. want to set up the additional
stuff for the given buffer only at allocation time), now PCM core
turns on buffer_changed flag when the buffer has changed.
The standard conversions to use the new API will be straightforward:
- Replace snd_pcm_lib_preallocate*() calls with the corresponding
snd_pcm_set_managed_buffer*(); the arguments should be unchanged
- Drop superfluous snd_pcm_lib_malloc() and snd_pcm_lib_free() calls;
the check of snd_pcm_lib_malloc() returns should be replaced with
the check of runtime->buffer_changed flag.
- If hw_params or hw_free becomes empty, drop them from PCM ops
Link: https://lore.kernel.org/r/20191117085308.23915-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Show and change sound card timer source with read-write info
file in proc filesystem. Initial string can still be set as
module parameter.
The timer source string value can be changed at any time,
but it is latched by PCM substream open callback (the first one
for a particular cable). At this point it is actually used, that
is the string is parsed, and the timer is looked up and opened.
The timer source is set for a loopback card (the same as initial
setting by module parameter), but every cable uses the value,
current at the moment of open.
Setting the value to empty string switches the timer to jiffies.
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191120174955.6410-8-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
to do synchronous audio forwarding between hardware sound card and aloop
devices. Such an audio route could look like the following:
Sound card -> Loopback application -> ALSA loop device -> arecord
In this case the loopback device should use the sound timer of the sound
card. Without this patch the loopback application has to implement an
adaptive sample rate converter to align the different clocks of the
different ALSA devices.
The used timer can be selected by referring to a sound card, its device
and subdevice, when loading the module:
$ modprobe snd_aloop enable=1 timer_source=[<card>[.<dev>[.<subdev>]]]
<card> is the name (id) of the sound card or a card number.
<dev> and <subdev> are device and subdevice numbers (defaults are 0).
Empty string as a value of timer_source= parameter enables previous
functionality (using jiffies timer).
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191120174955.6410-7-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit does not change the behaviour. It only separates the jiffies
timer specific implementation from the generic part.
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191120174955.6410-5-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit only refactors the implementation. It does not change the
behaviour.
It is required to support other timers (e.g sound timer).
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191120174955.6410-4-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is required for additional timer implementations which could detect
errors and want to throw them.
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191120174955.6410-3-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Describe the unit of the variables used to calculate the hw pointer
depending on jiffies ticks.
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Link: https://lore.kernel.org/r/20191120174955.6410-2-andrew_gabbasov@mentor.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adjust indentation from spaces to tab (+optional two spaces) as in
coding style with command like:
$ sed -e 's/^ /\t/' -i */Kconfig
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191120133252.6365-1-krzk@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 957ce0c6b8 (ASoC: soc-pcm: check symmetry after
hw_params).
That commit cause soc_pcm_params_symmetry can't take effect.
cpu_dai->rate, cpu_dai->channels and cpu_dai->sample_bits
are updated in the middle of soc_pcm_hw_params, so move
soc_pcm_params_symmetry to the end of soc_pcm_hw_params is
not a good solution, for judgement of symmetry in the function
is always true.
FIXME:
According to the comments of that commit, I think the case
described in the commit should disable symmetric_rates
in Back-End, rather than changing the position of
soc_pcm_params_symmetry.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1573555602-5403-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The RST (reset-gpios) is low active so the driver must handle it
accordingly.
Add comments to explain clearly how the line is used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191120131753.6831-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the standard name for the gpion in DT: reset-gpios
Document that the RST line is low active and update the example
accordingly.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191120131753.6831-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This ASCII string can carry additional information about
soundcard components or configuration. Add the possibility
to set this string via the ASoC card.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20191119174933.25526-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Mic mute led does not work on HP ProBook 645 G4.
We can use CXT_FIXUP_MUTE_LED_GPIO fixup to support it.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191120082035.18937-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
soc-topology adds extra dai_link by using snd_soc_add_dai_link(),
and removes it by snd_soc_romove_dai_link().
This snd_soc_add/remove_dai_link() and/or its related
functions are unbalanced before, and now, these are balance-uped.
But, it finds the random operation issue, and it is reported by
Pierre-Louis.
When card was released, topology will call snd_soc_remove_dai_link()
via (A).
static void soc_cleanup_card_resources(struct snd_soc_card *card)
{
struct snd_soc_dai_link *link, *_link;
/* This should be called before snd_card_free() */
(A) soc_remove_link_components(card);
/* free the ALSA card at first; this syncs with pending operations */
if (card->snd_card) {
(B) snd_card_free(card->snd_card);
card->snd_card = NULL;
}
/* remove and free each DAI */
(X) soc_remove_link_dais(card);
for_each_card_links_safe(card, link, _link)
(C) snd_soc_remove_dai_link(card, link);
...
}
At (A), topology calls snd_soc_remove_dai_link().
Then topology rtd, and its related all data are freed.
Next, (B) is called, and then, pcm->private_free = soc_pcm_private_free()
is called.
static void soc_pcm_private_free(struct snd_pcm *pcm)
{
struct snd_soc_pcm_runtime *rtd = pcm->private_data;
/* need to sync the delayed work before releasing resources */
flush_delayed_work(&rtd->delayed_work);
snd_soc_pcm_component_free(rtd);
}
Here, it gets rtd via pcm->private_data.
But, topology related rtd are already freed at (A).
Normal sound card has no damage, becase it frees rtd at (C).
These are finalizing rtd related data.
Thus, these should be called when rtd was freed, not sound card
was freed. It is very natural and understandable.
In other words, pcm->private_free = soc_pcm_private_free()
is no longer needed.
Extra issue is that there is zero chance to call
soc_remove_dai() for topology related dai at (X).
Because (A) removes rtd connection from card too, and,
(X) is based on card connected rtd.
This means, (X) need to be called before (C) (= for normal sound)
and (A) (= for topology).
Now, I want to focus this patch which is the reason why
snd_card_free() = (B) is located there.
commit 4efda5f213
("ASoC: Fix use-after-free at card unregistration")
Original snd_card_free() was called last of this function.
But moved to top to avoid use-after-free issue.
The issue was happen at soc_pcm_free() which was pcm->private_free,
today it is updated/renamed to soc_pcm_private_free().
In other words, (B) need to be called before (C) (= for normal sound)
and (A) (= for topology), because it needs (not yet freed) rtd.
But, (A) need to be called before (B),
because it needs card->snd_card pointer.
If we call flush_delayed_work() and snd_soc_pcm_component_free()
(= same as soc_pcm_private_free()) when rtd was freed (= (C), (A)),
there is no reason to call snd_card_free() at top of this function.
It can be called end of this function, again.
But, in such case, it will likely break unbind again, as Takashi-san
reported. When unbind is performed in a busy state, the code may
release still-in-use resources.
At least we need to call snd_card_disconnect_sync() at the first place.
The final code will be...
static void soc_cleanup_card_resources(struct snd_soc_card *card)
{
struct snd_soc_dai_link *link, *_link;
if (card->snd_card)
(Z) snd_card_disconnect_sync(card->snd_card);
(X) soc_remove_link_dais(card);
(A) soc_remove_link_components(card);
for_each_card_links_safe(card, link, _link)
(C) snd_soc_remove_dai_link(card, link);
...
if (card->snd_card) {
(B) snd_card_free(card->snd_card);
card->snd_card = NULL;
}
}
To avoid release still-in-use resources,
call snd_card_disconnect_sync() at (Z).
(X) is needed for both non-topology and topology.
topology removes rtd via (A), and
non topology removes rtd via (C).
snd_card_free() is no longer related to use-after-free issue.
Thus, locating (B) is no problem.
Fixes: df95a16d2a ("ASoC: soc-core: fix RIP warning on card removal")
Fixes: bc7a9091e5 ("ASoC: soc-core: add soc_unbind_dai_link()")
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/87o8xax88g.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch uses rtd instead of pcm at snd_soc_pcm_component_new/free()
parameter.
This is prepare for dai_link remove bug fix on topology.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pnhqx89j.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When the Acer Switch 10 SW5-012 quirk was added we did not have
jack-detection support yet; and the builtin microphone selection of
the original quirk is wrong too.
Fix the microphone-input quirk and add jack-detection info so that the
internal-microphone and headphone/set jack on the Switch 10 work properly.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191119145138.59162-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds DP-MST support for GK104+ NVIDIA codecs.
GK104+ NVIDIA codecs support DP-MST audio. These codecs have 4
output converters and 4 pin widgets, with 4 device entries per pin
widget for a total of 16 device entries.
This patch moves the existing patch_nvhdmi() definition to
patch_nvhdmi_legacy(), used by pre-GK104 NVIDIA codecs. Redefine
patch_nvhdmi() to enable DP-MST support by setting codec->dp_mst and
spec->dyn_pcm_assign.
Introduce fresh logic for dynamic pcm assignment, making
sure that new pcm assignments are compatible with the legacy static
per_pin-pmc assignment that existed in the days before DP-MST.
Signed-off-by: Nikhil Mahale <nmahale@nvidia.com>
Reviewed-by: Aaron Plattner <aplattner@nvidia.com>
Link: https://lore.kernel.org/r/20191119084710.29267-5-nmahale@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch make it possible for non-acomp codecs to set
dyn_pcm_assign/dp_mst and get DP-MST audio support.
Document change notification HDA040-A for the Intel High Definition
Audio 1.0a specification introduces a Device Select verb for Digital
Display Pin Widgets that are multi-stream capable. This verb selects
a Device Entry that is used by subsequent Pin Widget verbs.
Once the Device Entry is selected, all subsequent Pin Widget verbs
controlling the sink device will be directed to the selected Device
Entry until the Device Select verb is updated with a new value.
These Pin Widget verbs include:
* Connection Select
* Get Connection List Entry
* Amplifier Gain/Mute
* Power State
* Pin Widget Control
* ELD Data
* DIP-Size
* DIP-Index
* DIP-Data
* DIP-XmitCtrl
* Content Protection Control
* ASP Channel Mapping
This patch adds calls to snd_hda_set_dev_select() to direct each of
these Pin Widget control verbs to the correct Device Entry.
snd_hda_get_connections() does not return per-device connection
list, therefore make use snd_hda_get_raw_connections() instead of
snd_hda_get_connections().
Signed-off-by: Nikhil Mahale <nmahale@nvidia.com>
Reviewed-by: Aaron Plattner <aplattner@nvidia.com>
Link: https://lore.kernel.org/r/20191119084710.29267-4-nmahale@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds DP-MST jack support which will be used on NVIDIA
platforms. Today, DP-MST audio is supported only if the codec has
acomp support. This patch makes it possible to add DP-MST support
for non-acomp codecs.
For the codecs supporting DP-MST audio, each pin can contain several
device entries. Each device entry is a virtual pin, described by
pin_nid and dev_id in struct hdmi_spec_per_pin. For monitor hotplug
event handling, non-acomp codecs enable and register jack-detection
for every hdmi_spec_per_pin.
This patch updates every relevant function in hda_jack.h and its
implementation in hda_jack.c, to consider dev_id along with pin_nid.
Changes to the HD Audio specification to support DP-MST audio are
described in the Intel Document Change Notification (DCN) number
HDA040-A.
From HDA040-A, "For the case of multi stream capable Digital Display
Pin Widget, [the Get Pin Sense verb] can be used to read a specific
Device Entry state as reported in Get Device List Entry verb." This
patch updates the read_pin_sense() function to take the dev_id as an
argument and pass it as a parameter to the Get Pin Sense verb.
Bits 15 through 20 from the Unsolicited Response for intrinsic
events contain the index of the Device Entry that generated the
event. This patch updates the Unsolicited Response event handlers to
extract the device entry index from the response and pass it to
snd_hda_jack_tbl_get_from_tag().
This patch updates snd_hda_jack_tbl_new() to take a dev_id argument
and store it in the jack structure, and to make sure not to generate
a different tag when called more than once for the same nid.
Signed-off-by: Nikhil Mahale <nmahale@nvidia.com>
Link: https://lore.kernel.org/r/20191119084710.29267-3-nmahale@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
s/snd_hda_pin_sense/snd_hda_jack_pin_sense/g
This aligns the snd_hda_pin_sense function name with the names of
other functions in hda_jack.h.
Signed-off-by: Nikhil Mahale <nmahale@nvidia.com>
Reviewed-by: Aaron Plattner <aplattner@nvidia.com>
Link: https://lore.kernel.org/r/20191119084710.29267-2-nmahale@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
